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Вылетает Asterisk при использвании перенаправления звонков.

Проблемы Asterisk без вэб-оболочек и их решения

Модераторы: april22, Zavr2008

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Vlad1983
Сообщения: 4251
Зарегистрирован: 09 авг 2011, 11:51

Re: Вылетает Asterisk при использвании перенаправления звонк

Сообщение Vlad1983 »

пробуйте сконвертить

Код: Выделить всё

asterisk -r
file convert /opt/var/lib/asterisk/moh/macroform-cold_day.ulaw /opt/var/lib/asterisk/moh/macroform-cold_day.alaw
file convert /opt/var/lib/asterisk/moh/macroform-cold_day.ulaw /opt/var/lib/asterisk/moh/macroform-cold_day.raw
file convert /opt/var/lib/asterisk/moh/macroform-cold_day.ulaw /opt/var/lib/asterisk/moh/macroform-cold_day.sln
file convert /opt/var/lib/asterisk/moh/macroform-robot_dity.ulaw /opt/var/lib/asterisk/moh/macroform-robot_dity.alaw
file convert /opt/var/lib/asterisk/moh/macroform-robot_dity.ulaw /opt/var/lib/asterisk/moh/macroform-robot_dity.raw
file convert /opt/var/lib/asterisk/moh/macroform-robot_dity.ulaw /opt/var/lib/asterisk/moh/macroform-robot_dity.sln
file convert /opt/var/lib/asterisk/moh/macroform-the_simplicity.ulaw /opt/var/lib/asterisk/moh/macroform-the_simplicity.alaw
file convert /opt/var/lib/asterisk/moh/macroform-the_simplicity.ulaw /opt/var/lib/asterisk/moh/macroform-the_simplicity.raw
file convert /opt/var/lib/asterisk/moh/macroform-the_simplicity.ulaw /opt/var/lib/asterisk/moh/macroform-the_simplicity.sln
file convert /opt/var/lib/asterisk/moh/manolo_camp-morning_coffee.ulaw /opt/var/lib/asterisk/moh/manolo_camp-morning_coffee.alaw
file convert /opt/var/lib/asterisk/moh/manolo_camp-morning_coffee.ulaw /opt/var/lib/asterisk/moh/manolo_camp-morning_coffee.raw
file convert /opt/var/lib/asterisk/moh/manolo_camp-morning_coffee.ulaw /opt/var/lib/asterisk/moh/manolo_camp-morning_coffee.sln
file convert /opt/var/lib/asterisk/moh/reno_project-system.ulaw /opt/var/lib/asterisk/moh/reno_project-system.alaw
file convert /opt/var/lib/asterisk/moh/reno_project-system.ulaw /opt/var/lib/asterisk/moh/reno_project-system.raw
file convert /opt/var/lib/asterisk/moh/reno_project-system.ulaw /opt/var/lib/asterisk/moh/reno_project-system.sln
ЛС: @rostel
Вернуться к началу
ccoll
Сообщения: 10
Зарегистрирован: 20 фев 2015, 22:43

Re: Вылетает Asterisk при использвании перенаправления звонк

Сообщение ccoll »

Сконвертировал файлы:

Код: Выделить всё

..........
..........
Converted /opt/var/lib/asterisk/moh/reno_project-system.ulaw to /opt/var/lib/asterisk/moh/reno_project-system.sln in 2859ms

Код: Выделить всё

CLI>
Class: default
        Mode: files
        Directory: /opt/var/lib/asterisk/moh
Class: default
        File: /opt/var/lib/asterisk/moh/macroform-cold_day
        File: /opt/var/lib/asterisk/moh/macroform-robot_dity
        File: /opt/var/lib/asterisk/moh/macroform-the_simplicity
        File: /opt/var/lib/asterisk/moh/manolo_camp-morning_coffee
        File: /opt/var/lib/asterisk/moh/reno_project-system
musiconhold.conf:

Код: Выделить всё

[default]
mode=files
directory=/opt/var/lib/asterisk/moh
Не помогло, вылетает Asterisk.
Вернуться к началу
Vlad1983
Сообщения: 4251
Зарегистрирован: 09 авг 2011, 11:51

Re: Вылетает Asterisk при использвании перенаправления звонк

Сообщение Vlad1983 »

попробуйте FreeSWITCH накатить
ЛС: @rostel
Вернуться к началу
Аватара пользователя
Zavr2008
Сообщения: 2215
Зарегистрирован: 27 янв 2011, 00:35
Контактная информация:
Контактная информация пользователя Zavr2008

Re: Вылетает Asterisk при использвании перенаправления звонк

Сообщение Zavr2008 »

телепаты в отпуске:

1. Опишите конкретно на какой номер трансферите этот SIP/users. если в город - там НЕТ tT в Dial. От этого и ануслинг.
2. canreinvite УСТАРЕЛ. используем directmedia=no
3. Наконец покажите SIP debug. Также включите set core verbose и debug побольше.
4. не вижу секцию general у sip.conf
5. callerid=users-hello у пира = записки школьника. УЧИМ матчасть.

P.S> Самое то главное пропустил:
"Asterisk 1.8.25.0" установлен на "ASUS WL-500G Premium V2"
отключаем все кодеки кроме alaw, ищем вообще Астеру памяти то хватает на зверьке, молимся тайпейскому гиганту чтоп хватило)
Российские E1 шлюзы Alvis. Модернизация УПАТС с E1,Подключение к ИС "Антифрод" E1 PRI/SS#7 УВР Телестор, Грифин и др..
Вернуться к началу
awsswa
Сообщения: 2390
Зарегистрирован: 09 июн 2012, 10:52
Откуда: Россия, Пермь skype: yarick_perm

Re: Вылетает Asterisk при использвании перенаправления звонк

Сообщение awsswa »

Vlad1983 писал(а):попробуйте FreeSWITCH накатить
фокус не прокатит в последнем релизе выпилен

честно я вообще не понимаю зачем вы скрутили дефолтный каталог /var/lib/asterisk/moh
все что надо было сделать это создать его - по умолчанию он отсуствует
платный суппорт по мере возможностей
Вернуться к началу
ccoll
Сообщения: 10
Зарегистрирован: 20 фев 2015, 22:43

Re: Вылетает Asterisk при использвании перенаправления звонк

Сообщение ccoll »

Zavr2008 писал(а):телепаты в отпуске:

1. Опишите конкретно на какой номер трансферите этот SIP/users. если в город - там НЕТ tT в Dial. От этого и ануслинг.
2. canreinvite УСТАРЕЛ. используем directmedia=no
3. Наконец покажите SIP debug. Также включите set core verbose и debug побольше.
4. не вижу секцию general у sip.conf
5. callerid=users-hello у пира = записки школьника. УЧИМ матчасть.

P.S> Самое то главное пропустил:
"Asterisk 1.8.25.0" установлен на "ASUS WL-500G Premium V2"
отключаем все кодеки кроме alaw, ищем вообще Астеру памяти то хватает на зверьке, молимся тайпейскому гиганту чтоп хватило)
1. Звоню пользователем andy - users. Подымаю трубку пользователем users и нажимаю клавишу # после чего asterisk вылетает. Хотел перевести звонок на пользователя 55555.
2. Заменил на directmedia=no.
3. Дебаг включил:

Код: Выделить всё

/opt/sbin/asterisk -rx "sip set debug on"

CLI> core set verbose 5
Set remote console verbosity to 5
CLI> core set debug 5
Core debug was 0 and is now 5
sip.conf:

Код: Выделить всё

[andy]
type=friend
secret=andy
nat=yes
videosupport=yes
directmedia=no
host=dynamic
defaultuser=andy
dtmfmode=rfc2833
allow=all
context=sip-dialout
qualify=yes
callgroup=1
pickupgroup=1

[users]
type=friend
secret=users
nat=yes
videosupport=yes
directmedia=no
host=dynamic
defaultuser=users
dtmfmode=rfc2833
allow=all
context=sip-dialout
qualify=yes
callgroup=1
pickupgroup=1

[55555]
type=friend
secret=1
dtmfmode=rfc2833
directmedia=no
context=sip-dialout
host=dynamic
nat=no
defaultuser=55555
qualify=yes
transport=udp
allow=all
callgroup=1
pickupgroup=1

[SPA31]
type=friend
secret=SPA
dtmfmode=rfc2833
directmedia=no
context=sip-dialout
host=dynamic
nat=no
defaultuser=SPA31
qualify=yes
transport=udp
allow=all
t38pt_udptl=yes,maxdatagram=400
callgroup=1
pickupgroup=1
extensions.conf:

Код: Выделить всё

[sip-dialout]
exten => andy,1,Dial(SIP/andy,,tT)
exten => 3,1,Dial(SIP/andy,,tT) ; короткий номер
exten => users,1,Dial(SIP/users,,tT)
exten => 1,1,Dial(SIP/users,,tT) ; короткий номер
exten => 55555,1,Dial(SIP/55555,,tT)
exten => 888,1,Dial(SIP/andy,,t&SIP/users,,t) ; входящий звонок c Linksys spa3102.
exten => _7777777!,1,Dial(SIP/SPA31,,D(${EXTEN})) ; звонок в город на определенный номер
4. не вижу секцию general у sip.conf
нет секции general.
Вернуться к началу
ccoll
Сообщения: 10
Зарегистрирован: 20 фев 2015, 22:43

Re: Вылетает Asterisk при использвании перенаправления звонк

Сообщение ccoll »

Лог1:

Код: Выделить всё

[Feb 25 00:24:03] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  0 [ 46]: OPTIONS sip:users@192.168.100.232:5060 SIP/2.0
[Feb 25 00:24:03] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  1 [ 65]: Via: SIP/2.0/UDP 192.168.100.19:5060;branch=z9hG4bK749f355a;rport
[Feb 25 00:24:03] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  2 [ 16]: Max-Forwards: 70
[Feb 25 00:24:03] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  3 [ 61]: From: "asterisk" <sip:asterisk@192.168.100.19>;tag=as63c1596e
[Feb 25 00:24:03] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  4 [ 36]: To: <sip:users@192.168.100.232:5060>
[Feb 25 00:24:03] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  5 [ 43]: Contact: <sip:asterisk@192.168.100.19:5060>
[Feb 25 00:24:03] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  6 [ 61]: Call-ID: 66b1a96203d7ac6f69ce58d94faa8128@192.168.100.19:5060
[Feb 25 00:24:03] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  7 [ 17]: CSeq: 102 OPTIONS
[Feb 25 00:24:03] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  8 [ 31]: User-Agent: Asterisk PBX 11.7.0
[Feb 25 00:24:03] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  9 [ 35]: Date: Wed, 25 Feb 2015 00:24:03 GMT
[Feb 25 00:24:03] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, P   UBLISH
[Feb 25 00:24:03] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header 11 [ 26]: Supported: replaces, timer
Reliably Transmitting (NAT) to 192.168.100.232:5060:
OPTIONS sip:users@192.168.100.232:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.19:5060;branch=z9hG4bK749f355a;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.100.19>;tag=as63c1596e
To: <sip:users@192.168.100.232:5060>
Contact: <sip:asterisk@192.168.100.19:5060>
Call-ID: 66b1a96203d7ac6f69ce58d94faa8128@192.168.100.19:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.7.0
Date: Wed, 25 Feb 2015 00:24:03 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[Feb 25 00:24:03] DEBUG[2154]: chan_sip.c:4335 __sip_reliable_xmit: *** SIP TIMER: Initializing retransmit timer on packet: Id  #56
[Feb 25 00:24:03] DEBUG[2154]: chan_sip.c:3875 __sip_xmit: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.100.232:5060

<--- SIP read from UDP:192.168.100.232:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.19:5060;branch=z9hG4bK749f355a;rport=5060
From: "asterisk" <sip:asterisk@192.168.100.19>;tag=as63c1596e
To: <sip:users@192.168.100.232:5060>;tag=007af7fabfbae41198937571a1da2b4c
Call-ID: 66b1a96203d7ac6f69ce58d94faa8128@192.168.100.19:5060
CSeq: 102 OPTIONS
Contact: <sip:users@192.168.100.232:5060>
Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE
Server: SIPPER for PhonerLite
Content-Length: 0

<------------->
[Feb 25 00:24:03] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  0 [ 14]: SIP/2.0 200 OK
[Feb 25 00:24:03] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  1 [ 70]: Via: SIP/2.0/UDP 192.168.100.19:5060;branch=z9hG4bK749f355a;rport=5060
[Feb 25 00:24:03] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  2 [ 61]: From: "asterisk" <sip:asterisk@192.168.100.19>;tag=as63c1596e
[Feb 25 00:24:03] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  3 [ 73]: To: <sip:users@192.168.100.232:5060>;tag=007af7fabfbae41198937571a1da2b4c
[Feb 25 00:24:03] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  4 [ 61]: Call-ID: 66b1a96203d7ac6f69ce58d94faa8128@192.168.100.19:5060
[Feb 25 00:24:03] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  5 [ 17]: CSeq: 102 OPTIONS
[Feb 25 00:24:03] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  6 [ 41]: Contact: <sip:users@192.168.100.232:5060>
[Feb 25 00:24:03] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  7 [ 71]: Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE
[Feb 25 00:24:03] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  8 [ 29]: Server: SIPPER for PhonerLite
[Feb 25 00:24:03] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  9 [ 17]: Content-Length: 0
--- (10 headers 0 lines) ---
[Feb 25 00:24:03] DEBUG[2154]: chan_sip.c:9167 find_call: = Looking for  Call ID: 66b1a96203d7ac6f69ce58d94faa8128@192.168.100.19:5060 (Checking To) --From    tag as63c1596e --To-tag 007af7fabfbae41198937571a1da2b4c
[Feb 25 00:24:03] DEBUG[2154]: chan_sip.c:4534 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #56
[Feb 25 00:24:03] DEBUG[2154]: chan_sip.c:4567 __sip_ack: Stopping retransmission on '66b1a96203d7ac6f69ce58d94faa8128@192.168.100.19:5060' of Request 102:    Match Found
[Feb 25 00:24:03] DEBUG[2154]: chan_sip.c:6828 sip_destroy: Destroying SIP dialog 66b1a96203d7ac6f69ce58d94faa8128@192.168.100.19:5060
Really destroying SIP dialog '66b1a96203d7ac6f69ce58d94faa8128@192.168.100.19:5060' Method: OPTIONS
[Feb 25 00:24:15] DEBUG[2154]: chan_sip.c:8770 sip_alloc: Allocating new SIP dialog for 19948d3468775d5d5b35b3b112587eb8@[c0a8:6413:a04a:ff7f::]:5060 - OPT   IONS (No RTP)
[Feb 25 00:24:15] DEBUG[2154]: acl.c:979 ast_ouraddrfor: For destination '192.168.10.101', our source address is '192.168.10.180'.
[Feb 25 00:24:15] DEBUG[2154]: chan_sip.c:4032 ast_sip_ouraddrfor: Setting SIP_TRANSPORT_UDP with address 192.168.10.180:5060
[Feb 25 00:24:15] DEBUG[2154]: chan_sip.c:8565 change_callid_pvt: SIP call-id changed from '19948d3468775d5d5b35b3b112587eb8@[c0a8:6413:a04a:ff7f::]:5060'    to '6d5ee43e21534d5e08f8e3a64e9fe2f6@192.168.10.180:5060'
[Feb 25 00:24:15] DEBUG[2154]: chan_sip.c:3518 initialize_initreq: Initializing initreq for method OPTIONS - callid 6d5ee43e21534d5e08f8e3a64e9fe2f6@192.16   8.10.180:5060
[Feb 25 00:24:15] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  0 [ 45]: OPTIONS sip:55555@192.168.10.101:5060 SIP/2.0
[Feb 25 00:24:15] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  1 [ 59]: Via: SIP/2.0/UDP 192.168.10.180:5060;branch=z9hG4bK3849d0e4
[Feb 25 00:24:15] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  2 [ 16]: Max-Forwards: 70
[Feb 25 00:24:15] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  3 [ 61]: From: "asterisk" <sip:asterisk@192.168.10.180>;tag=as1578bbab
[Feb 25 00:24:15] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  4 [ 35]: To: <sip:55555@192.168.10.101:5060>
[Feb 25 00:24:15] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  5 [ 43]: Contact: <sip:asterisk@192.168.10.180:5060>
[Feb 25 00:24:15] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  6 [ 61]: Call-ID: 6d5ee43e21534d5e08f8e3a64e9fe2f6@192.168.10.180:5060
[Feb 25 00:24:15] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  7 [ 17]: CSeq: 102 OPTIONS
[Feb 25 00:24:15] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  8 [ 31]: User-Agent: Asterisk PBX 11.7.0
[Feb 25 00:24:15] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  9 [ 35]: Date: Wed, 25 Feb 2015 00:24:15 GMT
[Feb 25 00:24:15] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, P   UBLISH
[Feb 25 00:24:15] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header 11 [ 26]: Supported: replaces, timer
Reliably Transmitting (no NAT) to 192.168.10.101:5060:
OPTIONS sip:55555@192.168.10.101:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.180:5060;branch=z9hG4bK3849d0e4
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.10.180>;tag=as1578bbab
To: <sip:55555@192.168.10.101:5060>
Contact: <sip:asterisk@192.168.10.180:5060>
Call-ID: 6d5ee43e21534d5e08f8e3a64e9fe2f6@192.168.10.180:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.7.0
Date: Wed, 25 Feb 2015 00:24:15 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[Feb 25 00:24:15] DEBUG[2154]: chan_sip.c:4335 __sip_reliable_xmit: *** SIP TIMER: Initializing retransmit timer on packet: Id  #59
[Feb 25 00:24:15] DEBUG[2154]: chan_sip.c:3875 __sip_xmit: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.10.101:5060

<--- SIP read from UDP:192.168.10.101:5060 --->
SIP/2.0 200 OK
To: <sip:55555@192.168.10.101:5060>;tag=61d3b64dfc71aaa3i0
From: "asterisk" <sip:asterisk@192.168.10.180>;tag=as1578bbab
Call-ID: 6d5ee43e21534d5e08f8e3a64e9fe2f6@192.168.10.180:5060
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 192.168.10.180:5060;branch=z9hG4bK3849d0e4
Server: Linksys/SPA3102-5.2.13(GW002)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces

<------------->
[Feb 25 00:24:15] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  0 [ 14]: SIP/2.0 200 OK
[Feb 25 00:24:15] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  1 [ 58]: To: <sip:55555@192.168.10.101:5060>;tag=61d3b64dfc71aaa3i0
[Feb 25 00:24:15] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  2 [ 61]: From: "asterisk" <sip:asterisk@192.168.10.180>;tag=as1578bbab
[Feb 25 00:24:15] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  3 [ 61]: Call-ID: 6d5ee43e21534d5e08f8e3a64e9fe2f6@192.168.10.180:5060
[Feb 25 00:24:15] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  4 [ 17]: CSeq: 102 OPTIONS
[Feb 25 00:24:15] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  5 [ 59]: Via: SIP/2.0/UDP 192.168.10.180:5060;branch=z9hG4bK3849d0e4
[Feb 25 00:24:15] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  6 [ 37]: Server: Linksys/SPA3102-5.2.13(GW002)
[Feb 25 00:24:15] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  7 [ 17]: Content-Length: 0
[Feb 25 00:24:15] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  8 [ 61]: Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
[Feb 25 00:24:15] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  9 [ 29]: Supported: x-sipura, replaces
--- (10 headers 0 lines) ---
[Feb 25 00:24:15] DEBUG[2154]: chan_sip.c:9167 find_call: = Looking for  Call ID: 6d5ee43e21534d5e08f8e3a64e9fe2f6@192.168.10.180:5060 (Checking To) --From    tag as1578bbab --To-tag 61d3b64dfc71aaa3i0
[Feb 25 00:24:15] DEBUG[2154]: chan_sip.c:4534 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #59
[Feb 25 00:24:15] DEBUG[2154]: chan_sip.c:4567 __sip_ack: Stopping retransmission on '6d5ee43e21534d5e08f8e3a64e9fe2f6@192.168.10.180:5060' of Request 102:    Match Found
[Feb 25 00:24:15] DEBUG[2154]: chan_sip.c:6828 sip_destroy: Destroying SIP dialog 6d5ee43e21534d5e08f8e3a64e9fe2f6@192.168.10.180:5060
Really destroying SIP dialog '6d5ee43e21534d5e08f8e3a64e9fe2f6@192.168.10.180:5060' Method: OPTIONS
[Feb 25 00:24:23] DEBUG[2154]: chan_sip.c:8770 sip_alloc: Allocating new SIP dialog for 1b9e5085018417457b4da6326520c87b@[c0a8:6413:a04a:ff7f::]:5060 - OPT   IONS (No RTP)
[Feb 25 00:24:23] DEBUG[2154]: acl.c:979 ast_ouraddrfor: For destination '192.168.10.43', our source address is '192.168.10.180'.
[Feb 25 00:24:23] DEBUG[2154]: chan_sip.c:4032 ast_sip_ouraddrfor: Setting SIP_TRANSPORT_UDP with address 192.168.10.180:5060
[Feb 25 00:24:23] DEBUG[2154]: chan_sip.c:8565 change_callid_pvt: SIP call-id changed from '1b9e5085018417457b4da6326520c87b@[c0a8:6413:a04a:ff7f::]:5060'    to '2e07f2f2060fcc045d15848746730671@192.168.10.180:5060'
[Feb 25 00:24:23] DEBUG[2154]: chan_sip.c:3518 initialize_initreq: Initializing initreq for method OPTIONS - callid 2e07f2f2060fcc045d15848746730671@192.16   8.10.180:5060
[Feb 25 00:24:23] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  0 [ 38]: OPTIONS sip:andy@192.168.10.43 SIP/2.0
[Feb 25 00:24:23] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  1 [ 65]: Via: SIP/2.0/UDP 192.168.10.180:5060;branch=z9hG4bK38983840;rport
[Feb 25 00:24:23] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  2 [ 16]: Max-Forwards: 70
[Feb 25 00:24:23] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  3 [ 61]: From: "asterisk" <sip:asterisk@192.168.10.180>;tag=as6603b223
[Feb 25 00:24:23] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  4 [ 28]: To: <sip:andy@192.168.10.43>
[Feb 25 00:24:23] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  5 [ 43]: Contact: <sip:asterisk@192.168.10.180:5060>
[Feb 25 00:24:23] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  6 [ 61]: Call-ID: 2e07f2f2060fcc045d15848746730671@192.168.10.180:5060
[Feb 25 00:24:23] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  7 [ 17]: CSeq: 102 OPTIONS
[Feb 25 00:24:23] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  8 [ 31]: User-Agent: Asterisk PBX 11.7.0
[Feb 25 00:24:23] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  9 [ 35]: Date: Wed, 25 Feb 2015 00:24:23 GMT
[Feb 25 00:24:23] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, P   UBLISH
[Feb 25 00:24:23] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header 11 [ 26]: Supported: replaces, timer
Reliably Transmitting (NAT) to 192.168.10.43:5060:
OPTIONS sip:andy@192.168.10.43 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.180:5060;branch=z9hG4bK38983840;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.10.180>;tag=as6603b223
To: <sip:andy@192.168.10.43>
Contact: <sip:asterisk@192.168.10.180:5060>
Call-ID: 2e07f2f2060fcc045d15848746730671@192.168.10.180:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.7.0
Date: Wed, 25 Feb 2015 00:24:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[Feb 25 00:24:23] DEBUG[2154]: chan_sip.c:4335 __sip_reliable_xmit: *** SIP TIMER: Initializing retransmit timer on packet: Id  #62
[Feb 25 00:24:23] DEBUG[2154]: chan_sip.c:3875 __sip_xmit: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.10.43:5060

<--- SIP read from UDP:192.168.10.43:5060 --->
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.10.180:5060;branch=z9hG4bK38983840;rport
From: "asterisk" <sip:asterisk@192.168.10.180>;tag=as6603b223
To: <sip:andy@192.168.10.43>;tag=M5Q-C
Call-ID: 2e07f2f2060fcc045d15848746730671@192.168.10.180:5060
CSeq: 102 OPTIONS

<------------->
[Feb 25 00:24:23] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  0 [ 14]: SIP/2.0 200 Ok
[Feb 25 00:24:23] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  1 [ 65]: Via: SIP/2.0/UDP 192.168.10.180:5060;branch=z9hG4bK38983840;rport
[Feb 25 00:24:23] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  2 [ 61]: From: "asterisk" <sip:asterisk@192.168.10.180>;tag=as6603b223
[Feb 25 00:24:23] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  3 [ 38]: To: <sip:andy@192.168.10.43>;tag=M5Q-C
[Feb 25 00:24:23] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  4 [ 61]: Call-ID: 2e07f2f2060fcc045d15848746730671@192.168.10.180:5060
[Feb 25 00:24:23] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  5 [ 17]: CSeq: 102 OPTIONS
--- (6 headers 0 lines) ---
[Feb 25 00:24:23] DEBUG[2154]: chan_sip.c:9167 find_call: = Looking for  Call ID: 2e07f2f2060fcc045d15848746730671@192.168.10.180:5060 (Checking To) --From    tag as6603b223 --To-tag M5Q-C
[Feb 25 00:24:23] DEBUG[2154]: chan_sip.c:4534 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #62
[Feb 25 00:24:23] DEBUG[2154]: chan_sip.c:4567 __sip_ack: Stopping retransmission on '2e07f2f2060fcc045d15848746730671@192.168.10.180:5060' of Request 102:    Match Found
[Feb 25 00:24:23] DEBUG[2154]: chan_sip.c:6828 sip_destroy: Destroying SIP dialog 2e07f2f2060fcc045d15848746730671@192.168.10.180:5060
Really destroying SIP dialog '2e07f2f2060fcc045d15848746730671@192.168.10.180:5060' Method: OPTIONS

<--- SIP read from UDP:192.168.10.43:5060 --->


<------------->
[Feb 25 00:24:26] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  0 [  0]:

<--- SIP read from UDP:192.168.10.43:5060 --->
INVITE sip:users@192.168.10.180 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.43:5060;branch=z9hG4bK.BnrBdc6Jp;rport
From: <sip:andy@192.168.10.180>;tag=1LEjyHs5q
To: sip:users@192.168.10.180
CSeq: 20 INVITE
Call-ID: QnoZcB0b4t
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Content-Length: 175
User-Agent: LinphoneAndroid/2.2.1.1 (belle-sip/1.2.4)
Contact: <sip:andy@192.168.10.43>;+sip.instance="<urn:uuid:af2e5b37-4eae-47ba-b166-af1e3b364580>"

v=0
o=andy 2300 2717 IN IP4 192.168.10.43
s=Talk
c=IN IP4 192.168.10.43
b=AS:380
t=0 0
m=audio 7076 RTP/AVP 9 3 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
<------------->
[Feb 25 00:24:38] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  0 [ 39]: INVITE sip:users@192.168.10.180 SIP/2.0
[Feb 25 00:24:38] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  1 [ 66]: Via: SIP/2.0/UDP 192.168.10.43:5060;branch=z9hG4bK.BnrBdc6Jp;rport
[Feb 25 00:24:38] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  2 [ 45]: From: <sip:andy@192.168.10.180>;tag=1LEjyHs5q
[Feb 25 00:24:38] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  3 [ 28]: To: sip:users@192.168.10.180
[Feb 25 00:24:38] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  4 [ 15]: CSeq: 20 INVITE
[Feb 25 00:24:38] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  5 [ 19]: Call-ID: QnoZcB0b4t
[Feb 25 00:24:38] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  6 [ 16]: Max-Forwards: 70
[Feb 25 00:24:38] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE   , INFO
[Feb 25 00:24:38] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  8 [ 29]: Content-Type: application/sdp
[Feb 25 00:24:38] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  9 [ 19]: Content-Length: 175
[Feb 25 00:24:38] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header 10 [ 53]: User-Agent: LinphoneAndroid/2.2.1.1 (belle-sip/1.2.4)
[Feb 25 00:24:38] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header 11 [ 97]: Contact: <sip:andy@192.168.10.43>;+sip.instance="<urn:uuid:af2e5b37-4eae-47   ba-b166-af1e3b364580>"
[Feb 25 00:24:38] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header 12 [  0]:
[Feb 25 00:24:38] DEBUG[2154]: chan_sip.c:9617 parse_request:    Body  0 [  3]: v=0
[Feb 25 00:24:38] DEBUG[2154]: chan_sip.c:9617 parse_request:    Body  1 [ 37]: o=andy 2300 2717 IN IP4 192.168.10.43
[Feb 25 00:24:38] DEBUG[2154]: chan_sip.c:9617 parse_request:    Body  2 [  6]: s=Talk
[Feb 25 00:24:38] DEBUG[2154]: chan_sip.c:9617 parse_request:    Body  3 [ 22]: c=IN IP4 192.168.10.43
[Feb 25 00:24:38] DEBUG[2154]: chan_sip.c:9617 parse_request:    Body  4 [  8]: b=AS:380
[Feb 25 00:24:38] DEBUG[2154]: chan_sip.c:9617 parse_request:    Body  5 [  5]: t=0 0
[Feb 25 00:24:38] DEBUG[2154]: chan_sip.c:9617 parse_request:    Body  6 [ 28]: m=audio 7076 RTP/AVP 9 3 101
[Feb 25 00:24:38] DEBUG[2154]: chan_sip.c:9617 parse_request:    Body  7 [ 33]: a=rtpmap:101 telephone-event/8000
[Feb 25 00:24:38] DEBUG[2154]: chan_sip.c:9654 parse_request:    Body  8 [ 15]: a=fmtp:101 0-11
--- (12 headers 9 lines) ---
[Feb 25 00:24:38] DEBUG[2154]: chan_sip.c:9167 find_call: = Looking for  Call ID: QnoZcB0b4t (Checking From) --From tag 1LEjyHs5q --To-tag
[Feb 25 00:24:38] DEBUG[2154]: acl.c:979 ast_ouraddrfor: For destination '192.168.10.43', our source address is '192.168.10.180'.
[Feb 25 00:24:38] DEBUG[2154]: chan_sip.c:4032 ast_sip_ouraddrfor: Setting SIP_TRANSPORT_UDP with address 192.168.10.180:5060
[Feb 25 00:24:38] DEBUG[2154]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '192.168.10.43:5060' into...
[Feb 25 00:24:38] DEBUG[2154]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '192.168.10.43' and port '5060'.
Sending to 192.168.10.43:5060 (no NAT)
[Feb 25 00:24:38] DEBUG[2154]: chan_sip.c:8770 sip_alloc: Allocating new SIP dialog for QnoZcB0b4t - INVITE (No RTP)
[Feb 25 00:24:38] DEBUG[2154][C-00000000]: chan_sip.c:28146 handle_incoming: **** Received INVITE (5) - Command in SIP INVITE
[Feb 25 00:24:38] DEBUG[2154][C-00000000]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '192.168.10.43:5060' into...
[Feb 25 00:24:38] DEBUG[2154][C-00000000]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '192.168.10.43' and port '5060'.
Sending to 192.168.10.43:5060 (no NAT)
[Feb 25 00:24:38] DEBUG[2154][C-00000000]: chan_sip.c:25270 handle_request_invite: Initializing initreq for method INVITE - callid QnoZcB0b4t
Using INVITE request as basis request - QnoZcB0b4t
[Feb 25 00:24:38] DEBUG[2154][C-00000000]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '192.168.10.180' into...
[Feb 25 00:24:38] DEBUG[2154][C-00000000]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '192.168.10.180' and port ''.
Found peer 'andy' for 'andy' from 192.168.10.43:5060

<--- Reliably Transmitting (NAT) to 192.168.10.43:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.10.43:5060;branch=z9hG4bK.BnrBdc6Jp;received=192.168.10.43;rport=5060
From: <sip:andy@192.168.10.180>;tag=1LEjyHs5q
To: sip:users@192.168.10.180;tag=as23e352d2
Call-ID: QnoZcB0b4t
CSeq: 20 INVITE
Server: Asterisk PBX 11.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="07eb8ddc"
Content-Length: 0


<------------>
[Feb 25 00:24:38] DEBUG[2154][C-00000000]: chan_sip.c:4335 __sip_reliable_xmit: *** SIP TIMER: Initializing retransmit timer on packet: Id  #65
[Feb 25 00:24:38] DEBUG[2154][C-00000000]: chan_sip.c:3875 __sip_xmit: Trying to put 'SIP/2.0 401' onto UDP socket destined for 192.168.10.43:5060
Scheduling destruction of SIP dialog 'QnoZcB0b4t' in 6400 ms (Method: INVITE)
[Feb 25 00:24:38] DEBUG[2154]: chan_sip.c:4119 retrans_pkt: SIP TIMER: Rescheduling retransmission #65 (1) SIP/2.0 - 1
[Feb 25 00:24:38] DEBUG[2154]: chan_sip.c:4139 retrans_pkt: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #65))
Retransmitting #1 (NAT) to 192.168.10.43:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.10.43:5060;branch=z9hG4bK.BnrBdc6Jp;received=192.168.10.43;rport=5060
From: <sip:andy@192.168.10.180>;tag=1LEjyHs5q
To: sip:users@192.168.10.180;tag=as23e352d2
Call-ID: QnoZcB0b4t
CSeq: 20 INVITE
Server: Asterisk PBX 11.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="07eb8ddc"
Content-Length: 0


---
[Feb 25 00:24:38] DEBUG[2154]: chan_sip.c:3875 __sip_xmit: Trying to put 'SIP/2.0 401' onto UDP socket destined for 192.168.10.43:5060
[Feb 25 00:24:38] DEBUG[2154]: chan_sip.c:4119 retrans_pkt: SIP TIMER: Rescheduling retransmission #65 (2) SIP/2.0 - 1
[Feb 25 00:24:38] DEBUG[2154]: chan_sip.c:4139 retrans_pkt: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #65))
Retransmitting #2 (NAT) to 192.168.10.43:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.10.43:5060;branch=z9hG4bK.BnrBdc6Jp;received=192.168.10.43;rport=5060
From: <sip:andy@192.168.10.180>;tag=1LEjyHs5q
To: sip:users@192.168.10.180;tag=as23e352d2
Call-ID: QnoZcB0b4t
CSeq: 20 INVITE
Server: Asterisk PBX 11.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="07eb8ddc"
Content-Length: 0


---
[Feb 25 00:24:38] DEBUG[2154]: chan_sip.c:3875 __sip_xmit: Trying to put 'SIP/2.0 401' onto UDP socket destined for 192.168.10.43:5060

<--- SIP read from UDP:192.168.10.43:5060 --->
ACK sip:users@192.168.10.180 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.43:5060;branch=z9hG4bK.BnrBdc6Jp;rport
Call-ID: QnoZcB0b4t
From: <sip:andy@192.168.10.180>;tag=1LEjyHs5q
To: <sip:users@192.168.10.180>;tag=as23e352d2
Contact: <sip:andy@192.168.10.43>;+sip.instance="<urn:uuid:af2e5b37-4eae-47ba-b166-af1e3b364580>"
Max-Forwards: 70
CSeq: 20 ACK

<------------->
[Feb 25 00:24:38] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  0 [ 36]: ACK sip:users@192.168.10.180 SIP/2.0
[Feb 25 00:24:38] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  1 [ 66]: Via: SIP/2.0/UDP 192.168.10.43:5060;branch=z9hG4bK.BnrBdc6Jp;rport
[Feb 25 00:24:38] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  2 [ 19]: Call-ID: QnoZcB0b4t
[Feb 25 00:24:38] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  3 [ 45]: From: <sip:andy@192.168.10.180>;tag=1LEjyHs5q
[Feb 25 00:24:38] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  4 [ 45]: To: <sip:users@192.168.10.180>;tag=as23e352d2
[Feb 25 00:24:38] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  5 [ 97]: Contact: <sip:andy@192.168.10.43>;+sip.instance="<urn:uuid:af2e5b37-4eae-47   ba-b166-af1e3b364580>"
[Feb 25 00:24:38] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  6 [ 16]: Max-Forwards: 70
[Feb 25 00:24:38] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  7 [ 12]: CSeq: 20 ACK
--- (8 headers 0 lines) ---
[Feb 25 00:24:38] DEBUG[2154]: chan_sip.c:9167 find_call: = Looking for  Call ID: QnoZcB0b4t (Checking From) --From tag 1LEjyHs5q --To-tag as23e352d2
[Feb 25 00:24:38] DEBUG[2154][C-00000000]: chan_sip.c:28146 handle_incoming: **** Received ACK (6) - Command in SIP ACK
[Feb 25 00:24:38] DEBUG[2154][C-00000000]: chan_sip.c:4534 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #65
[Feb 25 00:24:38] DEBUG[2154][C-00000000]: chan_sip.c:4567 __sip_ack: Stopping retransmission on 'QnoZcB0b4t' of Response 20: Match Found

<--- SIP read from UDP:192.168.10.43:5060 --->
INVITE sip:users@192.168.10.180 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.43:5060;branch=z9hG4bK.Vmq1qAKcq;rport
From: <sip:andy@192.168.10.180>;tag=1LEjyHs5q
To: sip:users@192.168.10.180
CSeq: 21 INVITE
Call-ID: QnoZcB0b4t
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Content-Length: 175
User-Agent: LinphoneAndroid/2.2.1.1 (belle-sip/1.2.4)
Contact: <sip:andy@192.168.10.43>;+sip.instance="<urn:uuid:af2e5b37-4eae-47ba-b166-af1e3b364580>"
Authorization: Digest realm="asterisk", nonce="07eb8ddc", username="andy", uri="sip:users@192.168.10.180", response="145dc6508356c27fa639be3f31aca31e"

v=0
o=andy 2300 2717 IN IP4 192.168.10.43
s=Talk
c=IN IP4 192.168.10.43
b=AS:380
t=0 0
m=audio 7076 RTP/AVP 9 3 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
<------------->
[Feb 25 00:24:38] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  0 [ 39]: INVITE sip:users@192.168.10.180 SIP/2.0
[Feb 25 00:24:38] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  1 [ 66]: Via: SIP/2.0/UDP 192.168.10.43:5060;branch=z9hG4bK.Vmq1qAKcq;rport
[Feb 25 00:24:38] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  2 [ 45]: From: <sip:andy@192.168.10.180>;tag=1LEjyHs5q
[Feb 25 00:24:38] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  3 [ 28]: To: sip:users@192.168.10.180
[Feb 25 00:24:38] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  4 [ 15]: CSeq: 21 INVITE
[Feb 25 00:24:38] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  5 [ 19]: Call-ID: QnoZcB0b4t
[Feb 25 00:24:38] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  6 [ 16]: Max-Forwards: 70
[Feb 25 00:24:38] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE   , INFO
[Feb 25 00:24:38] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  8 [ 29]: Content-Type: application/sdp
[Feb 25 00:24:38] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  9 [ 19]: Content-Length: 175
[Feb 25 00:24:38] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header 10 [ 53]: User-Agent: LinphoneAndroid/2.2.1.1 (belle-sip/1.2.4)
[Feb 25 00:24:38] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header 11 [ 97]: Contact: <sip:andy@192.168.10.43>;+sip.instance="<urn:uuid:af2e5b37-4eae-47   ba-b166-af1e3b364580>"
[Feb 25 00:24:38] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header 12 [150]: Authorization: Digest realm="asterisk", nonce="07eb8ddc", username="andy",    uri="sip:users@192.168.10.180", response="145dc6508356c27fa639be3f31aca31e"
[Feb 25 00:24:38] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header 13 [  0]:
[Feb 25 00:24:38] DEBUG[2154]: chan_sip.c:9617 parse_request:    Body  0 [  3]: v=0
[Feb 25 00:24:38] DEBUG[2154]: chan_sip.c:9617 parse_request:    Body  1 [ 37]: o=andy 2300 2717 IN IP4 192.168.10.43
[Feb 25 00:24:38] DEBUG[2154]: chan_sip.c:9617 parse_request:    Body  2 [  6]: s=Talk
[Feb 25 00:24:38] DEBUG[2154]: chan_sip.c:9617 parse_request:    Body  3 [ 22]: c=IN IP4 192.168.10.43
[Feb 25 00:24:38] DEBUG[2154]: chan_sip.c:9617 parse_request:    Body  4 [  8]: b=AS:380
[Feb 25 00:24:38] DEBUG[2154]: chan_sip.c:9617 parse_request:    Body  5 [  5]: t=0 0
[Feb 25 00:24:38] DEBUG[2154]: chan_sip.c:9617 parse_request:    Body  6 [ 28]: m=audio 7076 RTP/AVP 9 3 101
[Feb 25 00:24:38] DEBUG[2154]: chan_sip.c:9617 parse_request:    Body  7 [ 33]: a=rtpmap:101 telephone-event/8000
[Feb 25 00:24:38] DEBUG[2154]: chan_sip.c:9654 parse_request:    Body  8 [ 15]: a=fmtp:101 0-11
--- (13 headers 9 lines) ---
[Feb 25 00:24:38] DEBUG[2154]: chan_sip.c:9167 find_call: = Looking for  Call ID: QnoZcB0b4t (Checking From) --From tag 1LEjyHs5q --To-tag
[Feb 25 00:24:38] DEBUG[2154]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '192.168.10.180' into...
[Feb 25 00:24:38] DEBUG[2154]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '192.168.10.180' and port ''.
[Feb 25 00:24:38] DEBUG[2154]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '192.168.10.180' into...
[Feb 25 00:24:38] DEBUG[2154]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '192.168.10.180' and port ''.
[Feb 25 00:24:38] DEBUG[2154][C-00000000]: chan_sip.c:28146 handle_incoming: **** Received INVITE (5) - Command in SIP INVITE
[Feb 25 00:24:38] DEBUG[2154][C-00000000]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '192.168.10.43:5060' into...
[Feb 25 00:24:38] DEBUG[2154][C-00000000]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '192.168.10.43' and port '5060'.
Sending to 192.168.10.43:5060 (NAT)
[Feb 25 00:24:38] DEBUG[2154][C-00000000]: chan_sip.c:25270 handle_request_invite: Initializing initreq for method INVITE - callid QnoZcB0b4t
Using INVITE request as basis request - QnoZcB0b4t
[Feb 25 00:24:38] DEBUG[2154][C-00000000]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '192.168.10.180' into...
[Feb 25 00:24:38] DEBUG[2154][C-00000000]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '192.168.10.180' and port ''.
Found peer 'andy' for 'andy' from 192.168.10.43:5060
[Feb 25 00:24:38] DEBUG[2154][C-00000000]: rtp_engine.c:283 ast_rtp_instance_new: Using engine 'asterisk' for RTP instance '0x7ec83c'
[Feb 25 00:24:38] DEBUG[2154][C-00000000]: res_rtp_asterisk.c:1808 ast_rtp_new: Allocated port 18170 for RTP instance '0x7ec83c'
[Feb 25 00:24:38] DEBUG[2154][C-00000000]: rtp_engine.c:292 ast_rtp_instance_new: RTP instance '0x7ec83c' is setup and ready to go
[Feb 25 00:24:38] DEBUG[2154][C-00000000]: rtp_engine.c:283 ast_rtp_instance_new: Using engine 'asterisk' for RTP instance '0x7dfb44'
[Feb 25 00:24:38] DEBUG[2154][C-00000000]: res_rtp_asterisk.c:1808 ast_rtp_new: Allocated port 18952 for RTP instance '0x7dfb44'
[Feb 25 00:24:38] DEBUG[2154][C-00000000]: rtp_engine.c:292 ast_rtp_instance_new: RTP instance '0x7dfb44' is setup and ready to go
[Feb 25 00:24:38] DEBUG[2154][C-00000000]: res_rtp_asterisk.c:3947 ast_rtp_prop_set: Setup RTCP on RTP instance '0x7dfb44'
  == Using SIP VIDEO CoS mark 6
[Feb 25 00:24:38] DEBUG[2154][C-00000000]: res_rtp_asterisk.c:3947 ast_rtp_prop_set: Setup RTCP on RTP instance '0x7ec83c'
  == Using SIP RTP CoS mark 5
[Feb 25 00:24:38] DEBUG[2154][C-00000000]: chan_sip.c:5736 do_setnat: Setting NAT on RTP to On
[Feb 25 00:24:38] DEBUG[2154][C-00000000]: chan_sip.c:5740 do_setnat: Setting NAT on VRTP to On
[Feb 25 00:24:38] DEBUG[2154][C-00000000]: chan_sip.c:10049 process_sdp: Processing session-level SDP v=0... UNSUPPORTED OR FAILED.
[Feb 25 00:24:38] DEBUG[2154][C-00000000]: chan_sip.c:10049 process_sdp: Processing session-level SDP o=andy 2300 2717 IN IP4 192.168.10.43... OK.
[Feb 25 00:24:38] DEBUG[2154][C-00000000]: chan_sip.c:10049 process_sdp: Processing session-level SDP s=Talk... UNSUPPORTED OR FAILED.
[Feb 25 00:24:38] DEBUG[2154][C-00000000]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '192.168.10.43' into...
[Feb 25 00:24:38] DEBUG[2154][C-00000000]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '192.168.10.43' and port ''.
[Feb 25 00:24:38] DEBUG[2154][C-00000000]: chan_sip.c:10049 process_sdp: Processing session-level SDP c=IN IP4 192.168.10.43... OK.
[Feb 25 00:24:38] DEBUG[2154][C-00000000]: chan_sip.c:10049 process_sdp: Processing session-level SDP b=AS:380... UNSUPPORTED OR FAILED.
[Feb 25 00:24:38] DEBUG[2154][C-00000000]: chan_sip.c:10049 process_sdp: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED.
Found RTP audio format 9
[Feb 25 00:24:38] DEBUG[2154][C-00000000]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 9 based on m type on 0x7d9ec77c
Found RTP audio format 3
[Feb 25 00:24:38] DEBUG[2154][C-00000000]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 3 based on m type on 0x7d9ec77c
Found RTP audio format 101
[Feb 25 00:24:38] DEBUG[2154][C-00000000]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 101 based on m type on 0x7d9ec77c
Found audio description format telephone-event for ID 101
[Feb 25 00:24:38] DEBUG[2154][C-00000000]: chan_sip.c:10482 process_sdp: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK.
[Feb 25 00:24:38] DEBUG[2154][C-00000000]: chan_sip.c:10482 process_sdp: Processing media-level (audio) SDP a=fmtp:101 0-11... UNSUPPORTED OR FAILED.
Capabilities: us - (gsm|ulaw|alaw|h263|testlaw), peer - audio=(gsm|g722)/video=(nothing)/text=(nothing), combined - (gsm)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[Feb 25 00:24:38] DEBUG[2154][C-00000000]: res_rtp_asterisk.c:3994 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x7ec83c'
Peer audio RTP is at port 192.168.10.43:7076
[Feb 25 00:24:38] DEBUG[2154][C-00000000]: rtp_engine.c:515 ast_rtp_codecs_payloads_copy: Copying payload 3 from 0x7d9ec77c to 0x7ec9e8
[Feb 25 00:24:38] DEBUG[2154][C-00000000]: rtp_engine.c:515 ast_rtp_codecs_payloads_copy: Copying payload 9 from 0x7d9ec77c to 0x7ec9e8
[Feb 25 00:24:38] DEBUG[2154][C-00000000]: rtp_engine.c:515 ast_rtp_codecs_payloads_copy: Copying payload 101 from 0x7d9ec77c to 0x7ec9e8
[Feb 25 00:24:38] DEBUG[2154][C-00000000]: res_rtp_asterisk.c:3913 ast_rtp_prop_set: Ignoring duplicate RTCP property on RTP instance '0x7ec83c'
[Feb 25 00:24:38] DEBUG[2154][C-00000000]: res_rtp_asterisk.c:3994 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x7dfb44'
Peer doesn't provide video
[Feb 25 00:24:38] DEBUG[2154][C-00000000]: chan_sip.c:10753 process_sdp: We're settling with these formats: (gsm)
[Feb 25 00:24:38] DEBUG[2154][C-00000000]: chan_sip.c:25403 handle_request_invite: Checking SIP call limits for device andy
[Feb 25 00:24:38] DEBUG[2154][C-00000000]: chan_sip.c:6680 update_call_counter: Updating call counter for incoming call
[Feb 25 00:24:38] DEBUG[2154][C-00000000]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '192.168.10.180' into...
[Feb 25 00:24:38] DEBUG[2154][C-00000000]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '192.168.10.180' and port ''.
[Feb 25 00:24:38] DEBUG[2154][C-00000000]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '192.168.10.180' into...
[Feb 25 00:24:38] DEBUG[2154][C-00000000]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '192.168.10.180' and port ''.
Looking for users in sip-dialout (domain 192.168.10.180)
[Feb 25 00:24:38] DEBUG[2154][C-00000000]: format_pref.c:339 ast_codec_choose: Could not find preferred codec - Going for the best codec
[Feb 25 00:24:38] DEBUG[2154][C-00000000]: chan_sip.c:7951 sip_new: *** Our native formats are (gsm)
[Feb 25 00:24:38] DEBUG[2154][C-00000000]: chan_sip.c:7952 sip_new: *** Joint capabilities are (gsm)
[Feb 25 00:24:38] DEBUG[2154][C-00000000]: chan_sip.c:7953 sip_new: *** Our capabilities are (gsm|ulaw|alaw|h263|testlaw)
[Feb 25 00:24:38] DEBUG[2154][C-00000000]: chan_sip.c:7954 sip_new: *** AST_CODEC_CHOOSE formats are gsm
[Feb 25 00:24:38] DEBUG[2154][C-00000000]: chan_sip.c:7980 sip_new: This channel can handle video! HOLLYWOOD next!
[Feb 25 00:24:38] DEBUG[2154][C-00000000]: chan_sip.c:16241 build_route: build_route: Contact hop: <sip:andy@192.168.10.43>;+sip.instance="<urn:uuid:af2e5b   37-4eae-47ba-b166-af1e3b364580>"
list_route: hop: <sip:andy@192.168.10.43>
[Feb 25 00:24:38] DEBUG[2154][C-00000000]: chan_sip.c:25715 handle_request_invite: SIP/andy-00000000: New call is still down.... Trying...

<--- Transmitting (NAT) to 192.168.10.43:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.10.43:5060;branch=z9hG4bK.Vmq1qAKcq;received=192.168.10.43;rport=5060
From: <sip:andy@192.168.10.180>;tag=1LEjyHs5q
To: sip:users@192.168.10.180
Call-ID: QnoZcB0b4t
CSeq: 21 INVITE
Server: Asterisk PBX 11.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:users@192.168.10.180:5060>
Content-Length: 0


<------------>
[Feb 25 00:24:38] DEBUG[2154][C-00000000]: chan_sip.c:3875 __sip_xmit: Trying to put 'SIP/2.0 100' onto UDP socket destined for 192.168.10.43:5060
[Feb 25 00:24:38] DEBUG[2144]: devicestate.c:344 _ast_device_state: No provider found, checking channel drivers for SIP - andy
[Feb 25 00:24:38] DEBUG[2144]: chan_sip.c:29602 sip_devicestate: Checking device state for peer andy
[Feb 25 00:24:38] DEBUG[2144]: devicestate.c:467 do_state_change: Changing state for SIP/andy - state 1 (Not in use)
[Feb 25 00:24:38] DEBUG[2144]: devicestate.c:442 devstate_event: device 'SIP/andy' state '1'

<--- SIP read from UDP:192.168.10.43:5060 --->
ACK sip:users@192.168.10.180 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.43:5060;branch=z9hG4bK.BnrBdc6Jp;rport
Call-ID: QnoZcB0b4t
From: <sip:andy@192.168.10.180>;tag=1LEjyHs5q
To: <sip:users@192.168.10.180>;tag=as23e352d2
Contact: <sip:andy@192.168.10.43>;+sip.instance="<urn:uuid:af2e5b37-4eae-47ba-b166-af1e3b364580>"
Max-Forwards: 70
CSeq: 20 ACK

<------------->
[Feb 25 00:24:38] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  0 [ 36]: ACK sip:users@192.168.10.180 SIP/2.0
[Feb 25 00:24:38] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  1 [ 66]: Via: SIP/2.0/UDP 192.168.10.43:5060;branch=z9hG4bK.BnrBdc6Jp;rport
[Feb 25 00:24:38] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  2 [ 19]: Call-ID: QnoZcB0b4t
[Feb 25 00:24:38] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  3 [ 45]: From: <sip:andy@192.168.10.180>;tag=1LEjyHs5q
[Feb 25 00:24:38] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  4 [ 45]: To: <sip:users@192.168.10.180>;tag=as23e352d2
[Feb 25 00:24:38] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  5 [ 97]: Contact: <sip:andy@192.168.10.43>;+sip.instance="<urn:uuid:af2e5b37-4eae-47   ba-b166-af1e3b364580>"
[Feb 25 00:24:38] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  6 [ 16]: Max-Forwards: 70
[Feb 25 00:24:38] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  7 [ 12]: CSeq: 20 ACK
--- (8 headers 0 lines) ---
[Feb 25 00:24:38] DEBUG[2154]: chan_sip.c:9167 find_call: = Looking for  Call ID: QnoZcB0b4t (Checking From) --From tag 1LEjyHs5q --To-tag as23e352d2
[Feb 25 00:24:38] DEBUG[2154]: chan_sip.c:28422 handle_request_do: Invalid SIP message - rejected , no callid, len 344

<--- SIP read from UDP:192.168.10.43:5060 --->
ACK sip:users@192.168.10.180 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.43:5060;branch=z9hG4bK.BnrBdc6Jp;rport
Call-ID: QnoZcB0b4t
From: <sip:andy@192.168.10.180>;tag=1LEjyHs5q
To: <sip:users@192.168.10.180>;tag=as23e352d2
Contact: <sip:andy@192.168.10.43>;+sip.instance="<urn:uuid:af2e5b37-4eae-47ba-b166-af1e3b364580>"
Max-Forwards: 70
CSeq: 20 ACK

<------------->
[Feb 25 00:24:38] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  0 [ 36]: ACK sip:users@192.168.10.180 SIP/2.0
[Feb 25 00:24:38] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  1 [ 66]: Via: SIP/2.0/UDP 192.168.10.43:5060;branch=z9hG4bK.BnrBdc6Jp;rport
[Feb 25 00:24:38] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  2 [ 19]: Call-ID: QnoZcB0b4t
[Feb 25 00:24:38] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  3 [ 45]: From: <sip:andy@192.168.10.180>;tag=1LEjyHs5q
[Feb 25 00:24:38] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  4 [ 45]: To: <sip:users@192.168.10.180>;tag=as23e352d2
[Feb 25 00:24:38] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  5 [ 97]: Contact: <sip:andy@192.168.10.43>;+sip.instance="<urn:uuid:af2e5b37-4eae-47   ba-b166-af1e3b364580>"
[Feb 25 00:24:38] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  6 [ 16]: Max-Forwards: 70
[Feb 25 00:24:38] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  7 [ 12]: CSeq: 20 ACK
--- (8 headers 0 lines) ---
[Feb 25 00:24:38] DEBUG[2154]: chan_sip.c:9167 find_call: = Looking for  Call ID: QnoZcB0b4t (Checking From) --From tag 1LEjyHs5q --To-tag as23e352d2
[Feb 25 00:24:38] DEBUG[2154]: chan_sip.c:28422 handle_request_do: Invalid SIP message - rejected , no callid, len 344
[Feb 25 00:24:38] DEBUG[2181][C-00000000]: pbx.c:4890 pbx_extension_helper: Launching 'Dial'
    -- Executing [users@sip-dialout:1] Dial("SIP/andy-00000000", "SIP/users,,tT") in new stack
[Feb 25 00:24:38] DEBUG[2181][C-00000000]: chan_sip.c:29707 sip_request_call: Asked to create a SIP channel with formats: (gsm)
[Feb 25 00:24:38] DEBUG[2181][C-00000000]: chan_sip.c:8770 sip_alloc: Allocating new SIP dialog for 4b3db63057de03b1056906827e9f8dcd@[c0a8:6413:a04a:ff7f::   ]:5060 - INVITE (No RTP)
[Feb 25 00:24:38] DEBUG[2181][C-00000000]: rtp_engine.c:283 ast_rtp_instance_new: Using engine 'asterisk' for RTP instance '0x810404'
[Feb 25 00:24:38] DEBUG[2181][C-00000000]: res_rtp_asterisk.c:1808 ast_rtp_new: Allocated port 14582 for RTP instance '0x810404'
[Feb 25 00:24:38] DEBUG[2181][C-00000000]: rtp_engine.c:292 ast_rtp_instance_new: RTP instance '0x810404' is setup and ready to go
[Feb 25 00:24:38] DEBUG[2181][C-00000000]: rtp_engine.c:283 ast_rtp_instance_new: Using engine 'asterisk' for RTP instance '0x81bb34'
[Feb 25 00:24:38] DEBUG[2181][C-00000000]: res_rtp_asterisk.c:1808 ast_rtp_new: Allocated port 10946 for RTP instance '0x81bb34'
[Feb 25 00:24:38] DEBUG[2181][C-00000000]: rtp_engine.c:292 ast_rtp_instance_new: RTP instance '0x81bb34' is setup and ready to go
[Feb 25 00:24:38] DEBUG[2181][C-00000000]: res_rtp_asterisk.c:3947 ast_rtp_prop_set: Setup RTCP on RTP instance '0x81bb34'
  == Using SIP VIDEO CoS mark 6
[Feb 25 00:24:38] DEBUG[2181][C-00000000]: res_rtp_asterisk.c:3947 ast_rtp_prop_set: Setup RTCP on RTP instance '0x810404'
  == Using SIP RTP CoS mark 5
[Feb 25 00:24:38] DEBUG[2181][C-00000000]: chan_sip.c:5736 do_setnat: Setting NAT on RTP to On
[Feb 25 00:24:38] DEBUG[2181][C-00000000]: chan_sip.c:5740 do_setnat: Setting NAT on VRTP to On
[Feb 25 00:24:38] DEBUG[2181][C-00000000]: chan_sip.c:3642 obproxy_get: OBPROXY: Not applying OBproxy to this call
[Feb 25 00:24:38] DEBUG[2181][C-00000000]: acl.c:979 ast_ouraddrfor: For destination '192.168.100.232', our source address is '192.168.100.19'.
[Feb 25 00:24:38] DEBUG[2181][C-00000000]: chan_sip.c:4032 ast_sip_ouraddrfor: Setting SIP_TRANSPORT_UDP with address 192.168.100.19:5060
[Feb 25 00:24:38] DEBUG[2181][C-00000000]: chan_sip.c:5736 do_setnat: Setting NAT on RTP to On
[Feb 25 00:24:38] DEBUG[2181][C-00000000]: chan_sip.c:5740 do_setnat: Setting NAT on VRTP to On
[Feb 25 00:24:39] DEBUG[2181][C-00000000]: chan_sip.c:8565 change_callid_pvt: SIP call-id changed from '4b3db63057de03b1056906827e9f8dcd@[c0a8:6413:a04a:ff   7f::]:5060' to '45cb22644fd8103673bb63087ba6f0e4@192.168.100.19:5060'
[Feb 25 00:24:39] DEBUG[2181][C-00000000]: format_pref.c:339 ast_codec_choose: Could not find preferred codec - Going for the best codec
[Feb 25 00:24:39] DEBUG[2181][C-00000000]: chan_sip.c:7951 sip_new: *** Our native formats are (gsm)
[Feb 25 00:24:39] DEBUG[2181][C-00000000]: chan_sip.c:7952 sip_new: *** Joint capabilities are (gsm)
[Feb 25 00:24:39] DEBUG[2181][C-00000000]: chan_sip.c:7953 sip_new: *** Our capabilities are (gsm|ulaw|alaw|h263|testlaw)
[Feb 25 00:24:39] DEBUG[2181][C-00000000]: chan_sip.c:7954 sip_new: *** AST_CODEC_CHOOSE formats are gsm
[Feb 25 00:24:39] DEBUG[2181][C-00000000]: chan_sip.c:7956 sip_new: *** Our preferred formats from the incoming channel are (gsm)
[Feb 25 00:24:39] DEBUG[2181][C-00000000]: chan_sip.c:7980 sip_new: This channel can handle video! HOLLYWOOD next!
[Feb 25 00:24:39] DEBUG[2181][C-00000000]: channel_internal_api.c:882 ast_channel_callid_set: Channel Call ID changing from [C-00000000] to [C-00000000]
[Feb 25 00:24:39] DEBUG[2181][C-00000000]: channel.c:6507 ast_channel_inherit_variables: Not copying variable DIALEDTIME.
[Feb 25 00:24:39] DEBUG[2181][C-00000000]: channel.c:6507 ast_channel_inherit_variables: Not copying variable ANSWEREDTIME.
[Feb 25 00:24:39] DEBUG[2181][C-00000000]: channel.c:6507 ast_channel_inherit_variables: Not copying variable DIALEDPEERNAME.
[Feb 25 00:24:39] DEBUG[2181][C-00000000]: channel.c:6507 ast_channel_inherit_variables: Not copying variable DIALEDPEERNUMBER.
[Feb 25 00:24:39] DEBUG[2181][C-00000000]: channel.c:6507 ast_channel_inherit_variables: Not copying variable DIALSTATUS.
[Feb 25 00:24:39] DEBUG[2181][C-00000000]: channel.c:6507 ast_channel_inherit_variables: Not copying variable SIPCALLID.
[Feb 25 00:24:39] DEBUG[2181][C-00000000]: channel.c:6507 ast_channel_inherit_variables: Not copying variable SIPDOMAIN.
[Feb 25 00:24:39] DEBUG[2181][C-00000000]: channel.c:6507 ast_channel_inherit_variables: Not copying variable SIPURI.
[Feb 25 00:24:39] DEBUG[2181][C-00000000]: chan_sip.c:6356 sip_call: Outgoing Call for users
[Feb 25 00:24:39] DEBUG[2181][C-00000000]: chan_sip.c:6680 update_call_counter: Updating call counter for outgoing call
[Feb 25 00:24:39] DEBUG[2181][C-00000000]: chan_sip.c:13101 add_sdp: This call needs video offers!
[Feb 25 00:24:39] DEBUG[2181][C-00000000]: chan_sip.c:13151 add_sdp: ** Our capability: (gsm|ulaw|alaw|h263|testlaw) Video flag: False Text flag: False
[Feb 25 00:24:39] DEBUG[2181][C-00000000]: chan_sip.c:13152 add_sdp: ** Our prefcodec: (gsm)
Вернуться к началу
ccoll
Сообщения: 10
Зарегистрирован: 20 фев 2015, 22:43

Re: Вылетает Asterisk при использвании перенаправления звонк

Сообщение ccoll »

Продолжение, Лог2:

Код: Выделить всё

Audio is at 14582
Video is at 192.168.100.19:10946
Adding codec 100002 (gsm) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100003 (ulaw) to SDP
Adding codec 100017 (testlaw) to SDP
Adding video codec 200002 (h263) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
[Feb 25 00:24:39] DEBUG[2181][C-00000000]: chan_sip.c:13288 add_sdp: -- Done with adding codecs to SDP
[Feb 25 00:24:39] DEBUG[2181][C-00000000]: chan_sip.c:13482 add_sdp: Done building SDP. Settling with this capability: (gsm|ulaw|alaw|h263|testlaw)
[Feb 25 00:24:39] DEBUG[2181][C-00000000]: chan_sip.c:3518 initialize_initreq: Initializing initreq for method INVITE - callid 45cb22644fd8103673bb63087ba6   f0e4@192.168.100.19:5060
[Feb 25 00:24:39] DEBUG[2181][C-00000000]: chan_sip.c:9617 parse_request:  Header  0 [ 45]: INVITE sip:users@192.168.100.232:5060 SIP/2.0
[Feb 25 00:24:39] DEBUG[2181][C-00000000]: chan_sip.c:9617 parse_request:  Header  1 [ 65]: Via: SIP/2.0/UDP 192.168.100.19:5060;branch=z9hG4bK632b5235;rpo   rt
[Feb 25 00:24:39] DEBUG[2181][C-00000000]: chan_sip.c:9617 parse_request:  Header  2 [ 16]: Max-Forwards: 70
[Feb 25 00:24:39] DEBUG[2181][C-00000000]: chan_sip.c:9617 parse_request:  Header  3 [ 46]: From: <sip:andy@192.168.100.19>;tag=as5c9d91ca
[Feb 25 00:24:39] DEBUG[2181][C-00000000]: chan_sip.c:9617 parse_request:  Header  4 [ 36]: To: <sip:users@192.168.100.232:5060>
[Feb 25 00:24:39] DEBUG[2181][C-00000000]: chan_sip.c:9617 parse_request:  Header  5 [ 39]: Contact: <sip:andy@192.168.100.19:5060>
[Feb 25 00:24:39] DEBUG[2181][C-00000000]: chan_sip.c:9617 parse_request:  Header  6 [ 61]: Call-ID: 45cb22644fd8103673bb63087ba6f0e4@192.168.100.19:5060
[Feb 25 00:24:39] DEBUG[2181][C-00000000]: chan_sip.c:9617 parse_request:  Header  7 [ 16]: CSeq: 102 INVITE
[Feb 25 00:24:39] DEBUG[2181][C-00000000]: chan_sip.c:9617 parse_request:  Header  8 [ 31]: User-Agent: Asterisk PBX 11.7.0
[Feb 25 00:24:39] DEBUG[2181][C-00000000]: chan_sip.c:9617 parse_request:  Header  9 [ 35]: Date: Wed, 25 Feb 2015 00:24:39 GMT
[Feb 25 00:24:39] DEBUG[2181][C-00000000]: chan_sip.c:9617 parse_request:  Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOT   IFY, INFO, PUBLISH
[Feb 25 00:24:39] DEBUG[2181][C-00000000]: chan_sip.c:9617 parse_request:  Header 11 [ 26]: Supported: replaces, timer
[Feb 25 00:24:39] DEBUG[2181][C-00000000]: chan_sip.c:9617 parse_request:  Header 12 [ 29]: Content-Type: application/sdp
Reliably Transmitting (NAT) to 192.168.100.232:5060:
INVITE sip:users@192.168.100.232:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.19:5060;branch=z9hG4bK632b5235;rport
Max-Forwards: 70
From: <sip:andy@192.168.100.19>;tag=as5c9d91ca
To: <sip:users@192.168.100.232:5060>
Contact: <sip:andy@192.168.100.19:5060>
Call-ID: 45cb22644fd8103673bb63087ba6f0e4@192.168.100.19:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.7.0
Date: Wed, 25 Feb 2015 00:24:39 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 432

v=0
o=root 1314845379 1314845379 IN IP4 192.168.100.19
s=Asterisk PBX 11.7.0
c=IN IP4 192.168.100.19
b=CT:384
t=0 0
m=audio 14582 RTP/AVP 3 8 0 101
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=video 10946 RTP/AVP 34
a=rtpmap:34 H263/90000
a=fmtp:34 F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0
a=sendrecv

---
[Feb 25 00:24:39] DEBUG[2181][C-00000000]: chan_sip.c:4335 __sip_reliable_xmit: *** SIP TIMER: Initializing retransmit timer on packet: Id  #68
[Feb 25 00:24:39] DEBUG[2181][C-00000000]: chan_sip.c:3875 __sip_xmit: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.100.232:5060
    -- Called SIP/users

<--- SIP read from UDP:192.168.100.232:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.100.19:5060;branch=z9hG4bK632b5235;rport=5060
From: <sip:andy@192.168.100.19>;tag=as5c9d91ca
To: <sip:users@192.168.100.232:5060>
Call-ID: 45cb22644fd8103673bb63087ba6f0e4@192.168.100.19:5060
CSeq: 102 INVITE
Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE
Server: SIPPER for PhonerLite
Content-Length: 0

<------------->
[Feb 25 00:24:39] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  0 [ 18]: SIP/2.0 100 Trying
[Feb 25 00:24:39] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  1 [ 70]: Via: SIP/2.0/UDP 192.168.100.19:5060;branch=z9hG4bK632b5235;rport=5060
[Feb 25 00:24:39] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  2 [ 46]: From: <sip:andy@192.168.100.19>;tag=as5c9d91ca
[Feb 25 00:24:39] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  3 [ 36]: To: <sip:users@192.168.100.232:5060>
[Feb 25 00:24:39] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  4 [ 61]: Call-ID: 45cb22644fd8103673bb63087ba6f0e4@192.168.100.19:5060
[Feb 25 00:24:39] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  5 [ 16]: CSeq: 102 INVITE
[Feb 25 00:24:39] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  6 [ 71]: Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE
[Feb 25 00:24:39] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  7 [ 29]: Server: SIPPER for PhonerLite
[Feb 25 00:24:39] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  8 [ 17]: Content-Length: 0
--- (9 headers 0 lines) ---
[Feb 25 00:24:39] DEBUG[2154]: chan_sip.c:9167 find_call: = Looking for  Call ID: 45cb22644fd8103673bb63087ba6f0e4@192.168.100.19:5060 (Checking To) --From    tag as5c9d91ca --To-tag
[Feb 25 00:24:39] DEBUG[2154][C-00000000]: chan_sip.c:4601 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #68 - INVITE (got response)
[Feb 25 00:24:39] DEBUG[2154][C-00000000]: chan_sip.c:4608 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '45cb22644fd8103   673bb63087ba6f0e4@192.168.100.19:5060' Request 102: Found
[Feb 25 00:24:39] DEBUG[2154][C-00000000]: chan_sip.c:22601 handle_response_invite: SIP response 100 to standard invite

<--- SIP read from UDP:192.168.100.232:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.100.19:5060;branch=z9hG4bK632b5235;rport=5060
From: <sip:andy@192.168.100.19>;tag=as5c9d91ca
To: <sip:users@192.168.100.232:5060>;tag=00a46c10c0bae41198937571a1da2b4c
Call-ID: 45cb22644fd8103673bb63087ba6f0e4@192.168.100.19:5060
CSeq: 102 INVITE
Contact: <sip:users@192.168.100.232:5060>
Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE
Server: SIPPER for PhonerLite
Content-Length: 0

<------------->
[Feb 25 00:24:39] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  0 [ 19]: SIP/2.0 180 Ringing
[Feb 25 00:24:39] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  1 [ 70]: Via: SIP/2.0/UDP 192.168.100.19:5060;branch=z9hG4bK632b5235;rport=5060
[Feb 25 00:24:39] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  2 [ 46]: From: <sip:andy@192.168.100.19>;tag=as5c9d91ca
[Feb 25 00:24:39] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  3 [ 73]: To: <sip:users@192.168.100.232:5060>;tag=00a46c10c0bae41198937571a1da2b4c
[Feb 25 00:24:39] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  4 [ 61]: Call-ID: 45cb22644fd8103673bb63087ba6f0e4@192.168.100.19:5060
[Feb 25 00:24:39] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  5 [ 16]: CSeq: 102 INVITE
[Feb 25 00:24:39] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  6 [ 41]: Contact: <sip:users@192.168.100.232:5060>
[Feb 25 00:24:39] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  7 [ 71]: Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE
[Feb 25 00:24:39] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  8 [ 29]: Server: SIPPER for PhonerLite
[Feb 25 00:24:39] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  9 [ 17]: Content-Length: 0
--- (10 headers 0 lines) ---
[Feb 25 00:24:39] DEBUG[2154]: chan_sip.c:9167 find_call: = Looking for  Call ID: 45cb22644fd8103673bb63087ba6f0e4@192.168.100.19:5060 (Checking To) --From    tag as5c9d91ca --To-tag 00a46c10c0bae41198937571a1da2b4c
[Feb 25 00:24:39] DEBUG[2154][C-00000000]: chan_sip.c:4608 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '45cb22644fd8103   673bb63087ba6f0e4@192.168.100.19:5060' Request 102: Found
[Feb 25 00:24:39] DEBUG[2154][C-00000000]: chan_sip.c:22601 handle_response_invite: SIP response 180 to standard invite
[Feb 25 00:24:39] DEBUG[2154][C-00000000]: chan_sip.c:16241 build_route: build_route: Contact hop: <sip:users@192.168.100.232:5060>
list_route: hop: <sip:users@192.168.100.232:5060>
[Feb 25 00:24:39] DEBUG[2144]: devicestate.c:344 _ast_device_state: No provider found, checking channel drivers for SIP - users
[Feb 25 00:24:39] DEBUG[2144]: chan_sip.c:29602 sip_devicestate: Checking device state for peer users
[Feb 25 00:24:39] DEBUG[2144]: devicestate.c:467 do_state_change: Changing state for SIP/users - state 1 (Not in use)
[Feb 25 00:24:39] DEBUG[2144]: devicestate.c:442 devstate_event: device 'SIP/users' state '1'
    -- SIP/users-00000001 is ringing
[Feb 25 00:24:39] DEBUG[2181][C-00000000]: rtp_engine.c:1805 ast_rtp_instance_early_bridge: Setting early bridge SDP of 'SIP/andy-00000000' with that of 'S   IP/users-00000001'

<--- Transmitting (NAT) to 192.168.10.43:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.10.43:5060;branch=z9hG4bK.Vmq1qAKcq;received=192.168.10.43;rport=5060
From: <sip:andy@192.168.10.180>;tag=1LEjyHs5q
To: sip:users@192.168.10.180;tag=as6925596a
Call-ID: QnoZcB0b4t
CSeq: 21 INVITE
Server: Asterisk PBX 11.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:users@192.168.10.180:5060>
Content-Length: 0


<------------>
[Feb 25 00:24:39] DEBUG[2181][C-00000000]: chan_sip.c:3875 __sip_xmit: Trying to put 'SIP/2.0 180' onto UDP socket destined for 192.168.10.43:5060

<--- SIP read from UDP:192.168.100.232:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.19:5060;branch=z9hG4bK632b5235;rport=5060
From: <sip:andy@192.168.100.19>;tag=as5c9d91ca
To: <sip:users@192.168.100.232:5060>;tag=00a46c10c0bae41198937571a1da2b4c
Call-ID: 45cb22644fd8103673bb63087ba6f0e4@192.168.100.19:5060
CSeq: 102 INVITE
Contact: <sip:users@192.168.100.232:5060>
Content-Type: application/sdp
Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE
Supported: replaces, from-change
Server: SIPPER for PhonerLite
Content-Length: 305

v=0
o=- 2751694236 1 IN IP4 192.168.100.232
s=SIPPER for PhonerLite
c=IN IP4 192.168.100.232
t=0 0
m=audio 5062 RTP/AVP 3 8 0 101
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ssrc:554319825
a=sendrecv
m=video 0 RTP/AVP 34
<------------->
[Feb 25 00:24:43] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  0 [ 14]: SIP/2.0 200 OK
[Feb 25 00:24:43] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  1 [ 70]: Via: SIP/2.0/UDP 192.168.100.19:5060;branch=z9hG4bK632b5235;rport=5060
[Feb 25 00:24:43] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  2 [ 46]: From: <sip:andy@192.168.100.19>;tag=as5c9d91ca
[Feb 25 00:24:43] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  3 [ 73]: To: <sip:users@192.168.100.232:5060>;tag=00a46c10c0bae41198937571a1da2b4c
[Feb 25 00:24:43] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  4 [ 61]: Call-ID: 45cb22644fd8103673bb63087ba6f0e4@192.168.100.19:5060
[Feb 25 00:24:43] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  5 [ 16]: CSeq: 102 INVITE
[Feb 25 00:24:43] DEBUG[2181][C-00000000]: res_rtp_asterisk.c:3184 ast_rtcp_read: Got RTCP report of 60 bytes
[Feb 25 00:24:43] DEBUG[2181][C-00000000]: res_rtp_asterisk.c:3581 ast_rtp_read: 0x814f80 -- Probation learning mode pass with source address 192.168.100.2   32:5062
       > 0x814f80 -- Probation passed - setting RTP source address to 192.168.100.232:5062
[Feb 25 00:24:43] DEBUG[2181][C-00000000]: res_rtp_asterisk.c:3994 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x810404'
[Feb 25 00:24:43] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  6 [ 41]: Contact: <sip:users@192.168.100.232:5060>
[Feb 25 00:24:43] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  7 [ 29]: Content-Type: application/sdp
[Feb 25 00:24:43] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  8 [ 71]: Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE
[Feb 25 00:24:43] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  9 [ 32]: Supported: replaces, from-change
[Feb 25 00:24:43] DEBUG[2181][C-00000000]: res_rtp_asterisk.c:3581 ast_rtp_read: 0x814f80 -- Probation learning mode pass with source address 192.168.100.2   32:5062
       > 0x814f80 -- Probation passed - setting RTP source address to 192.168.100.232:5062
[Feb 25 00:24:43] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header 10 [ 29]: Server: SIPPER for PhonerLite
[Feb 25 00:24:43] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header 11 [ 19]: Content-Length: 305
[Feb 25 00:24:43] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header 12 [  0]:
[Feb 25 00:24:43] DEBUG[2154]: chan_sip.c:9617 parse_request:    Body  0 [  3]: v=0
[Feb 25 00:24:43] DEBUG[2154]: chan_sip.c:9617 parse_request:    Body  1 [ 39]: o=- 2751694236 1 IN IP4 192.168.100.232
[Feb 25 00:24:43] DEBUG[2154]: chan_sip.c:9617 parse_request:    Body  2 [ 23]: s=SIPPER for PhonerLite
[Feb 25 00:24:43] DEBUG[2154]: chan_sip.c:9617 parse_request:    Body  3 [ 24]: c=IN IP4 192.168.100.232
[Feb 25 00:24:43] DEBUG[2154]: chan_sip.c:9617 parse_request:    Body  4 [  5]: t=0 0
[Feb 25 00:24:43] DEBUG[2154]: chan_sip.c:9617 parse_request:    Body  5 [ 30]: m=audio 5062 RTP/AVP 3 8 0 101
[Feb 25 00:24:43] DEBUG[2154]: chan_sip.c:9617 parse_request:    Body  6 [ 19]: a=rtpmap:3 GSM/8000
[Feb 25 00:24:43] DEBUG[2154]: chan_sip.c:9617 parse_request:    Body  7 [ 20]: a=rtpmap:8 PCMA/8000
[Feb 25 00:24:43] DEBUG[2154]: chan_sip.c:9617 parse_request:    Body  8 [ 20]: a=rtpmap:0 PCMU/8000
[Feb 25 00:24:43] DEBUG[2154]: chan_sip.c:9617 parse_request:    Body  9 [ 33]: a=rtpmap:101 telephone-event/8000
[Feb 25 00:24:43] DEBUG[2154]: chan_sip.c:9617 parse_request:    Body 10 [ 15]: a=fmtp:101 0-16
[Feb 25 00:24:43] DEBUG[2154]: chan_sip.c:9617 parse_request:    Body 11 [ 16]: a=ssrc:554319825
[Feb 25 00:24:43] DEBUG[2154]: chan_sip.c:9617 parse_request:    Body 12 [ 10]: a=sendrecv
[Feb 25 00:24:43] DEBUG[2154]: chan_sip.c:9654 parse_request:    Body 13 [ 20]: m=video 0 RTP/AVP 34
--- (12 headers 14 lines) ---
[Feb 25 00:24:43] DEBUG[2154]: chan_sip.c:9167 find_call: = Looking for  Call ID: 45cb22644fd8103673bb63087ba6f0e4@192.168.100.19:5060 (Checking To) --From    tag as5c9d91ca --To-tag 00a46c10c0bae41198937571a1da2b4c
[Feb 25 00:24:43] DEBUG[2154][C-00000000]: chan_sip.c:4529 __sip_ack: Acked pending invite 102
[Feb 25 00:24:43] DEBUG[2154][C-00000000]: chan_sip.c:4567 __sip_ack: Stopping retransmission on '45cb22644fd8103673bb63087ba6f0e4@192.168.100.19:5060' of    Request 102: Match Found
[Feb 25 00:24:43] DEBUG[2154][C-00000000]: chan_sip.c:22601 handle_response_invite: SIP response 200 to standard invite
[Feb 25 00:24:43] DEBUG[2154][C-00000000]: chan_sip.c:10049 process_sdp: Processing session-level SDP v=0... UNSUPPORTED OR FAILED.
[Feb 25 00:24:43] DEBUG[2154][C-00000000]: chan_sip.c:10049 process_sdp: Processing session-level SDP o=- 2751694236 1 IN IP4 192.168.100.232... OK.
[Feb 25 00:24:43] DEBUG[2154][C-00000000]: chan_sip.c:10049 process_sdp: Processing session-level SDP s=SIPPER for PhonerLite... UNSUPPORTED OR FAILED.
[Feb 25 00:24:43] DEBUG[2154][C-00000000]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '192.168.100.232' into...
[Feb 25 00:24:43] DEBUG[2154][C-00000000]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '192.168.100.232' and port ''.
[Feb 25 00:24:43] DEBUG[2154][C-00000000]: chan_sip.c:10049 process_sdp: Processing session-level SDP c=IN IP4 192.168.100.232... OK.
[Feb 25 00:24:43] DEBUG[2154][C-00000000]: chan_sip.c:10049 process_sdp: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED.
Found RTP audio format 3
[Feb 25 00:24:43] DEBUG[2154][C-00000000]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 3 based on m type on 0x7d9ebe4c
Found RTP audio format 8
[Feb 25 00:24:43] DEBUG[2154][C-00000000]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 8 based on m type on 0x7d9ebe4c
Found RTP audio format 0
[Feb 25 00:24:43] DEBUG[2154][C-00000000]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 0 based on m type on 0x7d9ebe4c
Found RTP audio format 101
[Feb 25 00:24:43] DEBUG[2154][C-00000000]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 101 based on m type on 0x7d9ebe4c
Found audio description format GSM for ID 3
[Feb 25 00:24:43] DEBUG[2154][C-00000000]: chan_sip.c:10482 process_sdp: Processing media-level (audio) SDP a=rtpmap:3 GSM/8000... OK.
Found audio description format PCMA for ID 8
[Feb 25 00:24:43] DEBUG[2154][C-00000000]: chan_sip.c:10482 process_sdp: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK.
Found audio description format PCMU for ID 0
[Feb 25 00:24:43] DEBUG[2154][C-00000000]: chan_sip.c:10482 process_sdp: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK.
Found audio description format telephone-event for ID 101
[Feb 25 00:24:43] DEBUG[2154][C-00000000]: chan_sip.c:10482 process_sdp: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK.
[Feb 25 00:24:43] DEBUG[2154][C-00000000]: chan_sip.c:10482 process_sdp: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED OR FAILED.
[Feb 25 00:24:43] DEBUG[2154][C-00000000]: chan_sip.c:10482 process_sdp: Processing media-level (audio) SDP a=ssrc:554319825... UNSUPPORTED OR FAILED.
[Feb 25 00:24:43] DEBUG[2154][C-00000000]: chan_sip.c:10482 process_sdp: Processing media-level (audio) SDP a=sendrecv... OK.
[Feb 25 00:24:43] WARNING[2154][C-00000000]: chan_sip.c:10207 process_sdp: Ignoring video stream offer because port number is zero
[Feb 25 00:24:43] WARNING[2154][C-00000000]: chan_sip.c:10207 process_sdp: Ignoring video stream offer because port number is zero
Capabilities: us - (gsm|ulaw|alaw|h263|testlaw), peer - audio=(gsm|ulaw|alaw)/video=(nothing)/text=(nothing), combined - (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[Feb 25 00:24:43] DEBUG[2154][C-00000000]: res_rtp_asterisk.c:3994 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x810404'
Peer audio RTP is at port 192.168.100.232:5062
[Feb 25 00:24:43] DEBUG[2154][C-00000000]: rtp_engine.c:515 ast_rtp_codecs_payloads_copy: Copying payload 0 from 0x7d9ebe4c to 0x8105b0
[Feb 25 00:24:43] DEBUG[2154][C-00000000]: rtp_engine.c:515 ast_rtp_codecs_payloads_copy: Copying payload 3 from 0x7d9ebe4c to 0x8105b0
[Feb 25 00:24:43] DEBUG[2154][C-00000000]: rtp_engine.c:515 ast_rtp_codecs_payloads_copy: Copying payload 8 from 0x7d9ebe4c to 0x8105b0
[Feb 25 00:24:43] DEBUG[2154][C-00000000]: rtp_engine.c:515 ast_rtp_codecs_payloads_copy: Copying payload 101 from 0x7d9ebe4c to 0x8105b0
[Feb 25 00:24:43] DEBUG[2154][C-00000000]: res_rtp_asterisk.c:3913 ast_rtp_prop_set: Ignoring duplicate RTCP property on RTP instance '0x810404'
[Feb 25 00:24:43] DEBUG[2154][C-00000000]: res_rtp_asterisk.c:3994 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x81bb34'
Peer doesn't provide video
[Feb 25 00:24:43] DEBUG[2154][C-00000000]: chan_sip.c:10753 process_sdp: We're settling with these formats: (gsm|ulaw|alaw)
[Feb 25 00:24:43] DEBUG[2154][C-00000000]: chan_sip.c:10760 process_sdp: We have an owner, now see if we need to change this call
[Feb 25 00:24:43] DEBUG[2154][C-00000000]: chan_sip.c:10766 process_sdp: Setting native formats after processing SDP. peer joint formats (gsm|ulaw|alaw), o   ld nativeformats (gsm)
[Feb 25 00:24:43] DEBUG[2154][C-00000000]: channel.c:5361 set_format: Set channel SIP/users-00000001 to read format gsm
[Feb 25 00:24:43] DEBUG[2154][C-00000000]: channel.c:5361 set_format: Set channel SIP/users-00000001 to write format gsm
[Feb 25 00:24:43] DEBUG[2154][C-00000000]: chan_sip.c:6680 update_call_counter: Updating call counter for outgoing call
[Feb 25 00:24:43] DEBUG[2154][C-00000000]: chan_sip.c:16241 build_route: build_route: Contact hop: <sip:users@192.168.100.232:5060>
list_route: hop: <sip:users@192.168.100.232:5060>
[Feb 25 00:24:43] DEBUG[2154][C-00000000]: chan_sip.c:11898 reqprep: Strict routing enforced for session 45cb22644fd8103673bb63087ba6f0e4@192.168.100.19:50   60
set_destination: Parsing <sip:users@192.168.100.232:5060> for address/port to send to
[Feb 25 00:24:43] DEBUG[2154][C-00000000]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '192.168.100.232:5060' into...
[Feb 25 00:24:43] DEBUG[2154][C-00000000]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '192.168.100.232' and port '5060'.
set_destination: set destination to 192.168.100.232:5060
Transmitting (NAT) to 192.168.100.232:5060:
ACK sip:users@192.168.100.232:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.19:5060;branch=z9hG4bK3ce4a04b;rport
Max-Forwards: 70
From: <sip:andy@192.168.100.19>;tag=as5c9d91ca
To: <sip:users@192.168.100.232:5060>;tag=00a46c10c0bae41198937571a1da2b4c
Contact: <sip:andy@192.168.100.19:5060>
Call-ID: 45cb22644fd8103673bb63087ba6f0e4@192.168.100.19:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.7.0
Content-Length: 0


---
[Feb 25 00:24:43] DEBUG[2154][C-00000000]: chan_sip.c:3875 __sip_xmit: Trying to put 'ACK sip:use' onto UDP socket destined for 192.168.100.232:5060
[Feb 25 00:24:43] DEBUG[2144]: devicestate.c:344 _ast_device_state: No provider found, checking channel drivers for SIP - users
[Feb 25 00:24:43] DEBUG[2144]: chan_sip.c:29602 sip_devicestate: Checking device state for peer users
[Feb 25 00:24:43] DEBUG[2144]: devicestate.c:467 do_state_change: Changing state for SIP/users - state 1 (Not in use)
[Feb 25 00:24:43] DEBUG[2144]: devicestate.c:442 devstate_event: device 'SIP/users' state '1'
    -- SIP/users-00000001 answered SIP/andy-00000000
[Feb 25 00:24:43] DEBUG[2144]: devicestate.c:344 _ast_device_state: No provider found, checking channel drivers for SIP - andy
[Feb 25 00:24:43] DEBUG[2144]: chan_sip.c:29602 sip_devicestate: Checking device state for peer andy
[Feb 25 00:24:43] DEBUG[2144]: devicestate.c:467 do_state_change: Changing state for SIP/andy - state 1 (Not in use)
[Feb 25 00:24:43] DEBUG[2144]: devicestate.c:442 devstate_event: device 'SIP/andy' state '1'
[Feb 25 00:24:43] DEBUG[2181][C-00000000]: chan_sip.c:7277 sip_answer: SIP answering channel: SIP/andy-00000000
[Feb 25 00:24:43] DEBUG[2181][C-00000000]: res_rtp_asterisk.c:2175 ast_rtp_update_source: Setting the marker bit due to a source update
[Feb 25 00:24:43] DEBUG[2181][C-00000000]: chan_sip.c:13596 transmit_response_with_sdp: Setting framing from config on incoming call
[Feb 25 00:24:43] DEBUG[2181][C-00000000]: chan_sip.c:13151 add_sdp: ** Our capability: (gsm) Video flag: True Text flag: True
[Feb 25 00:24:43] DEBUG[2181][C-00000000]: chan_sip.c:13152 add_sdp: ** Our prefcodec: (nothing)
Audio is at 18170
Adding codec 100002 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
[Feb 25 00:24:43] DEBUG[2181][C-00000000]: chan_sip.c:13288 add_sdp: -- Done with adding codecs to SDP
[Feb 25 00:24:43] DEBUG[2181][C-00000000]: chan_sip.c:13482 add_sdp: Done building SDP. Settling with this capability: (gsm)

<--- Reliably Transmitting (NAT) to 192.168.10.43:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.10.43:5060;branch=z9hG4bK.Vmq1qAKcq;received=192.168.10.43;rport=5060
From: <sip:andy@192.168.10.180>;tag=1LEjyHs5q
To: sip:users@192.168.10.180;tag=as6925596a
Call-ID: QnoZcB0b4t
CSeq: 21 INVITE
Server: Asterisk PBX 11.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:users@192.168.10.180:5060>
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 1105373035 1105373035 IN IP4 192.168.10.180
s=Asterisk PBX 11.7.0
c=IN IP4 192.168.10.180
t=0 0
m=audio 18170 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
[Feb 25 00:24:43] DEBUG[2181][C-00000000]: chan_sip.c:4335 __sip_reliable_xmit: *** SIP TIMER: Initializing retransmit timer on packet: Id  #72
[Feb 25 00:24:43] DEBUG[2181][C-00000000]: chan_sip.c:3875 __sip_xmit: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.10.43:5060
[Feb 25 00:24:43] DEBUG[2181][C-00000000]: features.c:4429 ast_bridge_call: bridge answer set, chan answer set
[Feb 25 00:24:43] DEBUG[2181][C-00000000]: features.c:4250 clear_dialed_interfaces: Removing dialed interfaces datastore on SIP/users-00000001 since we're    bridging
[Feb 25 00:24:43] DEBUG[2181][C-00000000]: res_rtp_asterisk.c:2175 ast_rtp_update_source: Setting the marker bit due to a source update
[Feb 25 00:24:43] DEBUG[2181][C-00000000]: res_rtp_asterisk.c:2175 ast_rtp_update_source: Setting the marker bit due to a source update
[Feb 25 00:24:43] DEBUG[2181][C-00000000]: res_rtp_asterisk.c:3581 ast_rtp_read: 0x814f80 -- Probation learning mode pass with source address 192.168.100.2   32:5062
       > 0x814f80 -- Probation passed - setting RTP source address to 192.168.100.232:5062
[Feb 25 00:24:43] DEBUG[2181][C-00000000]: chan_sip.c:8392 sip_rtp_read: Oooh, format changed to gsm
[Feb 25 00:24:43] DEBUG[2181][C-00000000]: channel.c:5361 set_format: Set channel SIP/users-00000001 to read format gsm
[Feb 25 00:24:43] DEBUG[2181][C-00000000]: channel.c:5361 set_format: Set channel SIP/users-00000001 to write format gsm
[Feb 25 00:24:43] DEBUG[2181][C-00000000]: res_rtp_asterisk.c:2707 ast_rtp_write: Ooh, format changed from unknown to gsm
[Feb 25 00:24:43] DEBUG[2181][C-00000000]: res_rtp_asterisk.c:2742 ast_rtp_write: Created smoother: format: gsm ms: 20 len: 33
[Feb 25 00:24:43] DEBUG[2181][C-00000000]: res_rtp_asterisk.c:2600 ast_rtp_raw_write: Starting RTCP transmission on RTP instance '0x7ec83c'
[Feb 25 00:24:43] DEBUG[2154]: chan_sip.c:4119 retrans_pkt: SIP TIMER: Rescheduling retransmission #72 (1) SIP/2.0 - 1
[Feb 25 00:24:43] DEBUG[2154]: chan_sip.c:4139 retrans_pkt: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #72))
Retransmitting #1 (NAT) to 192.168.10.43:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.10.43:5060;branch=z9hG4bK.Vmq1qAKcq;received=192.168.10.43;rport=5060
From: <sip:andy@192.168.10.180>;tag=1LEjyHs5q
To: sip:users@192.168.10.180;tag=as6925596a
Call-ID: QnoZcB0b4t
CSeq: 21 INVITE
Server: Asterisk PBX 11.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:users@192.168.10.180:5060>
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 1105373035 1105373035 IN IP4 192.168.10.180
s=Asterisk PBX 11.7.0
c=IN IP4 192.168.10.180
t=0 0
m=audio 18170 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
[Feb 25 00:24:43] DEBUG[2154]: chan_sip.c:3875 __sip_xmit: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.10.43:5060
[Feb 25 00:24:43] DEBUG[2154]: chan_sip.c:4119 retrans_pkt: SIP TIMER: Rescheduling retransmission #72 (2) SIP/2.0 - 1
[Feb 25 00:24:44] DEBUG[2154]: chan_sip.c:4139 retrans_pkt: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #72))
Retransmitting #2 (NAT) to 192.168.10.43:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.10.43:5060;branch=z9hG4bK.Vmq1qAKcq;received=192.168.10.43;rport=5060
From: <sip:andy@192.168.10.180>;tag=1LEjyHs5q
To: sip:users@192.168.10.180;tag=as6925596a
Call-ID: QnoZcB0b4t
CSeq: 21 INVITE
Server: Asterisk PBX 11.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:users@192.168.10.180:5060>
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 1105373035 1105373035 IN IP4 192.168.10.180
s=Asterisk PBX 11.7.0
c=IN IP4 192.168.10.180
t=0 0
m=audio 18170 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
[Feb 25 00:24:44] DEBUG[2154]: chan_sip.c:3875 __sip_xmit: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.10.43:5060

<--- SIP read from UDP:192.168.10.43:5060 --->
ACK sip:users@192.168.10.180:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.43:5060;rport;branch=z9hG4bK.Hlx0r1wTj
From: <sip:andy@192.168.10.180>;tag=1LEjyHs5q
To: <sip:users@192.168.10.180>;tag=as6925596a
CSeq: 21 ACK
Call-ID: QnoZcB0b4t
Max-Forwards: 70
Authorization: Digest realm="asterisk", nonce="07eb8ddc", username="andy", uri="sip:users@192.168.10.180", response="145dc6508356c27fa639be3f31aca31e"

<------------->
[Feb 25 00:24:44] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  0 [ 41]: ACK sip:users@192.168.10.180:5060 SIP/2.0
[Feb 25 00:24:44] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  1 [ 66]: Via: SIP/2.0/UDP 192.168.10.43:5060;rport;branch=z9hG4bK.Hlx0r1wTj
[Feb 25 00:24:44] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  2 [ 45]: From: <sip:andy@192.168.10.180>;tag=1LEjyHs5q
[Feb 25 00:24:44] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  3 [ 45]: To: <sip:users@192.168.10.180>;tag=as6925596a
[Feb 25 00:24:44] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  4 [ 12]: CSeq: 21 ACK
[Feb 25 00:24:44] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  5 [ 19]: Call-ID: QnoZcB0b4t
[Feb 25 00:24:44] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  6 [ 16]: Max-Forwards: 70
[Feb 25 00:24:44] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  7 [150]: Authorization: Digest realm="asterisk", nonce="07eb8ddc", username="andy",    uri="sip:users@192.168.10.180", response="145dc6508356c27fa639be3f31aca31e"
--- (8 headers 0 lines) ---
[Feb 25 00:24:44] DEBUG[2154]: chan_sip.c:9167 find_call: = Looking for  Call ID: QnoZcB0b4t (Checking From) --From tag 1LEjyHs5q --To-tag as6925596a
[Feb 25 00:24:44] DEBUG[2154][C-00000000]: chan_sip.c:28146 handle_incoming: **** Received ACK (6) - Command in SIP ACK
[Feb 25 00:24:44] DEBUG[2154][C-00000000]: chan_sip.c:4534 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #72
[Feb 25 00:24:44] DEBUG[2154][C-00000000]: chan_sip.c:4567 __sip_ack: Stopping retransmission on 'QnoZcB0b4t' of Response 21: Match Found
[Feb 25 00:24:44] DEBUG[2181][C-00000000]: res_rtp_asterisk.c:3581 ast_rtp_read: 0x7fdfa0 -- Probation learning mode pass with source address 192.168.10.43   :7076
       > 0x7fdfa0 -- Probation passed - setting RTP source address to 192.168.10.43:7076
[Feb 25 00:24:44] DEBUG[2181][C-00000000]: res_rtp_asterisk.c:2707 ast_rtp_write: Ooh, format changed from unknown to gsm
[Feb 25 00:24:44] DEBUG[2181][C-00000000]: res_rtp_asterisk.c:2742 ast_rtp_write: Created smoother: format: gsm ms: 20 len: 33

<--- SIP read from UDP:192.168.10.43:5060 --->
ACK sip:users@192.168.10.180:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.43:5060;branch=z9hG4bK.Hlx0r1wTj;rport
From: <sip:andy@192.168.10.180>;tag=1LEjyHs5q
To: <sip:users@192.168.10.180>;tag=as6925596a
CSeq: 21 ACK
Call-ID: QnoZcB0b4t
Max-Forwards: 70
Authorization: Digest realm="asterisk", nonce="07eb8ddc", username="andy", uri="sip:users@192.168.10.180", response="145dc6508356c27fa639be3f31aca31e"

<------------->
[Feb 25 00:24:44] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  0 [ 41]: ACK sip:users@192.168.10.180:5060 SIP/2.0
[Feb 25 00:24:44] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  1 [ 66]: Via: SIP/2.0/UDP 192.168.10.43:5060;branch=z9hG4bK.Hlx0r1wTj;rport
[Feb 25 00:24:44] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  2 [ 45]: From: <sip:andy@192.168.10.180>;tag=1LEjyHs5q
[Feb 25 00:24:44] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  3 [ 45]: To: <sip:users@192.168.10.180>;tag=as6925596a
[Feb 25 00:24:44] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  4 [ 12]: CSeq: 21 ACK
[Feb 25 00:24:44] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  5 [ 19]: Call-ID: QnoZcB0b4t
[Feb 25 00:24:44] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  6 [ 16]: Max-Forwards: 70
[Feb 25 00:24:44] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  7 [150]: Authorization: Digest realm="asterisk", nonce="07eb8ddc", username="andy",    uri="sip:users@192.168.10.180", response="145dc6508356c27fa639be3f31aca31e"
--- (8 headers 0 lines) ---
[Feb 25 00:24:44] DEBUG[2154]: chan_sip.c:9167 find_call: = Looking for  Call ID: QnoZcB0b4t (Checking From) --From tag 1LEjyHs5q --To-tag as6925596a
[Feb 25 00:24:44] DEBUG[2154][C-00000000]: chan_sip.c:28146 handle_incoming: **** Received ACK (6) - Command in SIP ACK

<--- SIP read from UDP:192.168.10.43:5060 --->
ACK sip:users@192.168.10.180:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.43:5060;branch=z9hG4bK.Hlx0r1wTj;rport
From: <sip:andy@192.168.10.180>;tag=1LEjyHs5q
To: <sip:users@192.168.10.180>;tag=as6925596a
CSeq: 21 ACK
Call-ID: QnoZcB0b4t
Max-Forwards: 70
Authorization: Digest realm="asterisk", nonce="07eb8ddc", username="andy", uri="sip:users@192.168.10.180", response="145dc6508356c27fa639be3f31aca31e"

<------------->
[Feb 25 00:24:44] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  0 [ 41]: ACK sip:users@192.168.10.180:5060 SIP/2.0
[Feb 25 00:24:44] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  1 [ 66]: Via: SIP/2.0/UDP 192.168.10.43:5060;branch=z9hG4bK.Hlx0r1wTj;rport
[Feb 25 00:24:44] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  2 [ 45]: From: <sip:andy@192.168.10.180>;tag=1LEjyHs5q
[Feb 25 00:24:44] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  3 [ 45]: To: <sip:users@192.168.10.180>;tag=as6925596a
[Feb 25 00:24:44] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  4 [ 12]: CSeq: 21 ACK
[Feb 25 00:24:44] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  5 [ 19]: Call-ID: QnoZcB0b4t
[Feb 25 00:24:44] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  6 [ 16]: Max-Forwards: 70
[Feb 25 00:24:44] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  7 [150]: Authorization: Digest realm="asterisk", nonce="07eb8ddc", username="andy",    uri="sip:users@192.168.10.180", response="145dc6508356c27fa639be3f31aca31e"
--- (8 headers 0 lines) ---
[Feb 25 00:24:44] DEBUG[2154]: chan_sip.c:9167 find_call: = Looking for  Call ID: QnoZcB0b4t (Checking From) --From tag 1LEjyHs5q --To-tag as6925596a
[Feb 25 00:24:44] DEBUG[2154][C-00000000]: chan_sip.c:28146 handle_incoming: **** Received ACK (6) - Command in SIP ACK
[Feb 25 00:24:45] DEBUG[2154]: chan_sip.c:8770 sip_alloc: Allocating new SIP dialog for 7f53e8b771fdfd101b9dde2c5cd5bd7e@[c0a8:6413:a04a:ff7f::]:5060 - OPT   IONS (No RTP)
[Feb 25 00:24:45] DEBUG[2154]: acl.c:979 ast_ouraddrfor: For destination '192.168.10.101', our source address is '192.168.10.180'.
[Feb 25 00:24:45] DEBUG[2154]: chan_sip.c:4032 ast_sip_ouraddrfor: Setting SIP_TRANSPORT_UDP with address 192.168.10.180:5060
[Feb 25 00:24:45] DEBUG[2154]: chan_sip.c:8565 change_callid_pvt: SIP call-id changed from '7f53e8b771fdfd101b9dde2c5cd5bd7e@[c0a8:6413:a04a:ff7f::]:5060'    to '471fc8e37ccb18fe3f173c120cb987bb@192.168.10.180:5060'
[Feb 25 00:24:45] DEBUG[2154]: chan_sip.c:3518 initialize_initreq: Initializing initreq for method OPTIONS - callid 471fc8e37ccb18fe3f173c120cb987bb@192.16   8.10.180:5060
[Feb 25 00:24:45] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  0 [ 45]: OPTIONS sip:SPA31@192.168.10.101:5060 SIP/2.0
[Feb 25 00:24:45] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  1 [ 59]: Via: SIP/2.0/UDP 192.168.10.180:5060;branch=z9hG4bK42bb6616
[Feb 25 00:24:45] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  2 [ 16]: Max-Forwards: 70
[Feb 25 00:24:45] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  3 [ 61]: From: "asterisk" <sip:asterisk@192.168.10.180>;tag=as222b0efb
[Feb 25 00:24:45] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  4 [ 35]: To: <sip:SPA31@192.168.10.101:5060>
[Feb 25 00:24:45] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  5 [ 43]: Contact: <sip:asterisk@192.168.10.180:5060>
[Feb 25 00:24:45] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  6 [ 61]: Call-ID: 471fc8e37ccb18fe3f173c120cb987bb@192.168.10.180:5060
[Feb 25 00:24:45] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  7 [ 17]: CSeq: 102 OPTIONS
[Feb 25 00:24:45] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  8 [ 31]: User-Agent: Asterisk PBX 11.7.0
[Feb 25 00:24:45] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  9 [ 35]: Date: Wed, 25 Feb 2015 00:24:45 GMT
[Feb 25 00:24:45] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, P   UBLISH
[Feb 25 00:24:45] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header 11 [ 26]: Supported: replaces, timer
Reliably Transmitting (no NAT) to 192.168.10.101:5060:
OPTIONS sip:SPA31@192.168.10.101:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.180:5060;branch=z9hG4bK42bb6616
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.10.180>;tag=as222b0efb
To: <sip:SPA31@192.168.10.101:5060>
Contact: <sip:asterisk@192.168.10.180:5060>
Call-ID: 471fc8e37ccb18fe3f173c120cb987bb@192.168.10.180:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.7.0
Date: Wed, 25 Feb 2015 00:24:45 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[Feb 25 00:24:45] DEBUG[2154]: chan_sip.c:4335 __sip_reliable_xmit: *** SIP TIMER: Initializing retransmit timer on packet: Id  #74
[Feb 25 00:24:45] DEBUG[2154]: chan_sip.c:3875 __sip_xmit: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.10.101:5060

<--- SIP read from UDP:192.168.10.101:5060 --->
SIP/2.0 200 OK
To: <sip:SPA31@192.168.10.101:5060>;tag=e8a8c8fd324837b3i1
From: "asterisk" <sip:asterisk@192.168.10.180>;tag=as222b0efb
Call-ID: 471fc8e37ccb18fe3f173c120cb987bb@192.168.10.180:5060
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 192.168.10.180:5060;branch=z9hG4bK42bb6616
Server: Linksys/SPA3102-5.2.13(GW002)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces

<------------->
[Feb 25 00:24:45] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  0 [ 14]: SIP/2.0 200 OK
[Feb 25 00:24:45] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  1 [ 58]: To: <sip:SPA31@192.168.10.101:5060>;tag=e8a8c8fd324837b3i1
[Feb 25 00:24:45] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  2 [ 61]: From: "asterisk" <sip:asterisk@192.168.10.180>;tag=as222b0efb
[Feb 25 00:24:45] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  3 [ 61]: Call-ID: 471fc8e37ccb18fe3f173c120cb987bb@192.168.10.180:5060
[Feb 25 00:24:45] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  4 [ 17]: CSeq: 102 OPTIONS
[Feb 25 00:24:45] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  5 [ 59]: Via: SIP/2.0/UDP 192.168.10.180:5060;branch=z9hG4bK42bb6616
[Feb 25 00:24:45] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  6 [ 37]: Server: Linksys/SPA3102-5.2.13(GW002)
[Feb 25 00:24:45] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  7 [ 17]: Content-Length: 0
[Feb 25 00:24:45] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  8 [ 61]: Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
[Feb 25 00:24:45] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  9 [ 29]: Supported: x-sipura, replaces
--- (10 headers 0 lines) ---
[Feb 25 00:24:45] DEBUG[2154]: chan_sip.c:9167 find_call: = Looking for  Call ID: 471fc8e37ccb18fe3f173c120cb987bb@192.168.10.180:5060 (Checking To) --From    tag as222b0efb --To-tag e8a8c8fd324837b3i1
[Feb 25 00:24:45] DEBUG[2154]: chan_sip.c:4534 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #74
[Feb 25 00:24:45] DEBUG[2154]: chan_sip.c:4567 __sip_ack: Stopping retransmission on '471fc8e37ccb18fe3f173c120cb987bb@192.168.10.180:5060' of Request 102:    Match Found
[Feb 25 00:24:45] DEBUG[2154]: chan_sip.c:6828 sip_destroy: Destroying SIP dialog 471fc8e37ccb18fe3f173c120cb987bb@192.168.10.180:5060
Really destroying SIP dialog '471fc8e37ccb18fe3f173c120cb987bb@192.168.10.180:5060' Method: OPTIONS
[Feb 25 00:24:46] DEBUG[2181][C-00000000]: res_rtp_asterisk.c:3184 ast_rtcp_read: Got RTCP report of 124 bytes
[Feb 25 00:24:49] DEBUG[2181][C-00000000]: res_rtp_asterisk.c:3184 ast_rtcp_read: Got RTCP report of 124 bytes
[Feb 25 00:24:51] DEBUG[2181][C-00000000]: res_rtp_asterisk.c:3184 ast_rtcp_read: Got RTCP report of 124 bytes
[Feb 25 00:24:52] DEBUG[2181][C-00000000]: res_rtp_asterisk.c:3184 ast_rtcp_read: Got RTCP report of 84 bytes
[Feb 25 00:24:54] DEBUG[2181][C-00000000]: res_rtp_asterisk.c:3184 ast_rtcp_read: Got RTCP report of 124 bytes
[Feb 25 00:24:56] DEBUG[2181][C-00000000]: res_rtp_asterisk.c:3184 ast_rtcp_read: Got RTCP report of 124 bytes

<--- SIP read from UDP:192.168.10.43:5060 --->


<------------->
[Feb 25 00:24:56] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  0 [  0]:
[Feb 25 00:24:59] DEBUG[2181][C-00000000]: res_rtp_asterisk.c:3184 ast_rtcp_read: Got RTCP report of 124 bytes
[Feb 25 00:25:02] DEBUG[2181][C-00000000]: res_rtp_asterisk.c:3184 ast_rtcp_read: Got RTCP report of 84 bytes
[Feb 25 00:25:03] DEBUG[2154]: chan_sip.c:8770 sip_alloc: Allocating new SIP dialog for 5454d6473d7a2fb13656680d12f3f9d7@[c0a8:6413:a04a:ff7f::]:5060 - OPT   IONS (No RTP)
[Feb 25 00:25:03] DEBUG[2154]: acl.c:979 ast_ouraddrfor: For destination '192.168.100.232', our source address is '192.168.100.19'.
[Feb 25 00:25:03] DEBUG[2154]: chan_sip.c:4032 ast_sip_ouraddrfor: Setting SIP_TRANSPORT_UDP with address 192.168.100.19:5060
[Feb 25 00:25:03] DEBUG[2154]: chan_sip.c:8565 change_callid_pvt: SIP call-id changed from '5454d6473d7a2fb13656680d12f3f9d7@[c0a8:6413:a04a:ff7f::]:5060'    to '5e31b007360fb3b83b78ba3a3a184807@192.168.100.19:5060'
[Feb 25 00:25:03] DEBUG[2154]: chan_sip.c:3518 initialize_initreq: Initializing initreq for method OPTIONS - callid 5e31b007360fb3b83b78ba3a3a184807@192.16   8.100.19:5060
[Feb 25 00:25:03] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  0 [ 46]: OPTIONS sip:users@192.168.100.232:5060 SIP/2.0
[Feb 25 00:25:03] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  1 [ 65]: Via: SIP/2.0/UDP 192.168.100.19:5060;branch=z9hG4bK352ebb5f;rport
[Feb 25 00:25:03] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  2 [ 16]: Max-Forwards: 70
[Feb 25 00:25:03] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  3 [ 61]: From: "asterisk" <sip:asterisk@192.168.100.19>;tag=as4d14e09d
[Feb 25 00:25:03] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  4 [ 36]: To: <sip:users@192.168.100.232:5060>
[Feb 25 00:25:03] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  5 [ 43]: Contact: <sip:asterisk@192.168.100.19:5060>
[Feb 25 00:25:03] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  6 [ 61]: Call-ID: 5e31b007360fb3b83b78ba3a3a184807@192.168.100.19:5060
[Feb 25 00:25:03] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  7 [ 17]: CSeq: 102 OPTIONS
[Feb 25 00:25:03] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  8 [ 31]: User-Agent: Asterisk PBX 11.7.0
[Feb 25 00:25:03] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  9 [ 35]: Date: Wed, 25 Feb 2015 00:25:03 GMT
[Feb 25 00:25:03] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, P   UBLISH
[Feb 25 00:25:03] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header 11 [ 26]: Supported: replaces, timer
Reliably Transmitting (NAT) to 192.168.100.232:5060:
OPTIONS sip:users@192.168.100.232:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.19:5060;branch=z9hG4bK352ebb5f;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.100.19>;tag=as4d14e09d
To: <sip:users@192.168.100.232:5060>
Contact: <sip:asterisk@192.168.100.19:5060>
Call-ID: 5e31b007360fb3b83b78ba3a3a184807@192.168.100.19:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.7.0
Date: Wed, 25 Feb 2015 00:25:03 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[Feb 25 00:25:03] DEBUG[2154]: chan_sip.c:4335 __sip_reliable_xmit: *** SIP TIMER: Initializing retransmit timer on packet: Id  #77
[Feb 25 00:25:03] DEBUG[2154]: chan_sip.c:3875 __sip_xmit: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.100.232:5060

<--- SIP read from UDP:192.168.100.232:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.19:5060;branch=z9hG4bK352ebb5f;rport=5060
From: "asterisk" <sip:asterisk@192.168.100.19>;tag=as4d14e09d
To: <sip:users@192.168.100.232:5060>;tag=8056531fc0bae41198937571a1da2b4c
Call-ID: 5e31b007360fb3b83b78ba3a3a184807@192.168.100.19:5060
CSeq: 102 OPTIONS
Contact: <sip:users@192.168.100.232:5060>
Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE
Server: SIPPER for PhonerLite
Content-Length: 0

<------------->
[Feb 25 00:25:03] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  0 [ 14]: SIP/2.0 200 OK
[Feb 25 00:25:03] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  1 [ 70]: Via: SIP/2.0/UDP 192.168.100.19:5060;branch=z9hG4bK352ebb5f;rport=5060
[Feb 25 00:25:03] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  2 [ 61]: From: "asterisk" <sip:asterisk@192.168.100.19>;tag=as4d14e09d
[Feb 25 00:25:03] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  3 [ 73]: To: <sip:users@192.168.100.232:5060>;tag=8056531fc0bae41198937571a1da2b4c
[Feb 25 00:25:03] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  4 [ 61]: Call-ID: 5e31b007360fb3b83b78ba3a3a184807@192.168.100.19:5060
[Feb 25 00:25:03] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  5 [ 17]: CSeq: 102 OPTIONS
[Feb 25 00:25:03] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  6 [ 41]: Contact: <sip:users@192.168.100.232:5060>
[Feb 25 00:25:03] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  7 [ 71]: Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE
[Feb 25 00:25:03] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  8 [ 29]: Server: SIPPER for PhonerLite
[Feb 25 00:25:03] DEBUG[2154]: chan_sip.c:9617 parse_request:  Header  9 [ 17]: Content-Length: 0
--- (10 headers 0 lines) ---
[Feb 25 00:25:03] DEBUG[2154]: chan_sip.c:9167 find_call: = Looking for  Call ID: 5e31b007360fb3b83b78ba3a3a184807@192.168.100.19:5060 (Checking To) --From    tag as4d14e09d --To-tag 8056531fc0bae41198937571a1da2b4c
[Feb 25 00:25:03] DEBUG[2154]: chan_sip.c:4534 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #77
[Feb 25 00:25:03] DEBUG[2154]: chan_sip.c:4567 __sip_ack: Stopping retransmission on '5e31b007360fb3b83b78ba3a3a184807@192.168.100.19:5060' of Request 102:    Match Found
[Feb 25 00:25:03] DEBUG[2154]: chan_sip.c:6828 sip_destroy: Destroying SIP dialog 5e31b007360fb3b83b78ba3a3a184807@192.168.100.19:5060
Really destroying SIP dialog '5e31b007360fb3b83b78ba3a3a184807@192.168.100.19:5060' Method: OPTIONS
[Feb 25 00:25:04] DEBUG[2181][C-00000000]: res_rtp_asterisk.c:3184 ast_rtcp_read: Got RTCP report of 124 bytes
[Feb 25 00:25:04] DEBUG[2181][C-00000000]: res_rtp_asterisk.c:2852 create_dtmf_frame: Creating BEGIN DTMF Frame: 35 (#), at 192.168.100.232:5062
[Feb 25 00:25:04] DTMF[2181][C-00000000]: channel.c:4170 __ast_read: DTMF begin '#' received on SIP/users-00000001
[Feb 25 00:25:04] DTMF[2181][C-00000000]: channel.c:4181 __ast_read: DTMF begin passthrough '#' on SIP/users-00000001
[Feb 25 00:25:04] DEBUG[2181][C-00000000]: channel.c:7649 ast_generic_bridge: Got DTMF begin on channel (SIP/users-00000001)
[Feb 25 00:25:04] DEBUG[2181][C-00000000]: res_rtp_asterisk.c:2175 ast_rtp_update_source: Setting the marker bit due to a source update
[Feb 25 00:25:04] DEBUG[2181][C-00000000]: res_rtp_asterisk.c:2175 ast_rtp_update_source: Setting the marker bit due to a source update
[Feb 25 00:25:04] DEBUG[2181][C-00000000]: channel.c:8069 ast_channel_bridge: Bridge stops bridging channels SIP/andy-00000000 and SIP/users-00000001
[Feb 25 00:25:04] DEBUG[2181][C-00000000]: features.c:3611 feature_interpret_helper: Feature detected: fname=Blind Transfer sname=blindxfer exten=#
[Feb 25 00:25:05] DEBUG[2181][C-00000000]: features.c:4646 ast_bridge_call: Not passing DTMF through, since it may be a feature code
[Feb 25 00:25:05] DEBUG[2181][C-00000000]: res_rtp_asterisk.c:2175 ast_rtp_update_source: Setting the marker bit due to a source update
[Feb 25 00:25:05] DEBUG[2181][C-00000000]: res_rtp_asterisk.c:2175 ast_rtp_update_source: Setting the marker bit due to a source update
[Feb 25 00:25:05] DEBUG[2181][C-00000000]: res_rtp_asterisk.c:2852 create_dtmf_frame: Creating END DTMF Frame: 35 (#), at 192.168.100.232:5062
[Feb 25 00:25:05] DTMF[2181][C-00000000]: channel.c:4084 __ast_read: DTMF end '#' received on SIP/users-00000001, duration 80 ms
[Feb 25 00:25:05] DTMF[2181][C-00000000]: channel.c:4125 __ast_read: DTMF end accepted with begin '#' on SIP/users-00000001
[Feb 25 00:25:05] DTMF[2181][C-00000000]: channel.c:4140 __ast_read: DTMF end '#' detected to have actual duration 60 on the wire, emulation will be trigge   red on SIP/users-00000001
[Feb 25 00:25:05] DTMF[2181][C-00000000]: channel.c:4147 __ast_read: DTMF end '#' has duration 60 but want minimum 80, emulating on SIP/users-00000001
[Feb 25 00:25:05] DTMF[2181][C-00000000]: channel.c:4204 __ast_read: DTMF end emulation of '#' queued on SIP/users-00000001
[Feb 25 00:25:05] DEBUG[2181][C-00000000]: channel.c:7649 ast_generic_bridge: Got DTMF end on channel (SIP/users-00000001)
[Feb 25 00:25:05] DEBUG[2181][C-00000000]: res_rtp_asterisk.c:2175 ast_rtp_update_source: Setting the marker bit due to a source update
[Feb 25 00:25:05] DEBUG[2181][C-00000000]: res_rtp_asterisk.c:2175 ast_rtp_update_source: Setting the marker bit due to a source update
[Feb 25 00:25:05] DEBUG[2181][C-00000000]: channel.c:8069 ast_channel_bridge: Bridge stops bridging channels SIP/andy-00000000 and SIP/users-00000001
[Feb 25 00:25:05] DEBUG[2181][C-00000000]: features.c:3740 feature_interpret: Feature interpret: chan=SIP/andy-00000000, peer=SIP/users-00000001, code=#, s   ense=2, features=2, dynamic=#
[Feb 25 00:25:05] DEBUG[2181][C-00000000]: features.c:3611 feature_interpret_helper: Feature detected: fname=Blind Transfer sname=blindxfer exten=#
[Feb 25 00:25:05] DEBUG[2181][C-00000000]: features.c:2521 builtin_blindtransfer: Executing Blind Transfer SIP/andy-00000000, SIP/users-00000001 (sense=2)
[Feb 25 00:25:05] DEBUG[2181][C-00000000]: res_rtp_asterisk.c:2175 ast_rtp_update_source: Setting the marker bit due to a source update
    -- Started music on hold, class 'default', on SIP/andy-00000000
[Feb 25 00:25:05] DEBUG[2181][C-00000000]: channel.c:3577 ast_settimeout: Scheduling timer at (50 requested / 50 actual) timer ticks per second
CLI>
Disconnected from Asterisk server
Asterisk cleanly ending (0).
Executing last minute cleanups
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ded
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Re: Вылетает Asterisk при использвании перенаправления звонк

Сообщение ded »

ded писал(а):Ваш Asus не может проиграть музыку, которую туда ему засунули.
Для начала уберите музыку в ожидании качестве КПВ - гудков вызова.
Потом убедитесь, что вы подсунули годный файл музыки. МР3 не нужно, нужен некомпрессированный wav, и не стерео, а моно, РСМб 8Кгц.
Вы пробовали так? Проигнорировали. У вас такая палитра кодеков в скромном по ресурсам устройстве, зачем Вы усложняете ему задачу?
У вас предпочтительный выбор идёт кодек gsm
[Feb 25 00:24:39] DEBUG[2181][C-00000000]: chan_sip.c:13151 add_sdp: ** Our capability: (gsm|ulaw|alaw|h263|testlaw) Video flag: False Text flag: False
[Feb 25 00:24:39] DEBUG[2181][C-00000000]: chan_sip.c:13152 add_sdp: ** Our prefcodec: (gsm)
Не хотите попробовать capability: (ulaw|alaw) ??
да ещё и звоните с софтфона с видео поддержкой!
[Feb 25 00:24:39] DEBUG[2181][C-00000000]: chan_sip.c:13101 add_sdp: This call needs video offers!
Может сначала упростите опыт?
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Zavr2008
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Зарегистрирован: 27 янв 2011, 00:35
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Re: Вылетает Asterisk при использвании перенаправления звонк

Сообщение Zavr2008 »

Dial(SIP/andy,,t&SIP/users,,t)
Это Вы где такое подсмотрели?
В УЧЕБНИК однозначно..

учимся писАть правильно:

Код: Выделить всё

Dial(SIP/andy&SIP/users,,tT)
И учимся указывать верные данные в секции [general] в sip*.conf. Также и читаем до просветления про формат поля callerid у пиров sip.

Музыку также можно попробовать потестить отдельно:

Код: Выделить всё

exten => 6000,1,Answer
exten => 6000,2,MusicOnHold()
Российские E1 шлюзы Alvis. Модернизация УПАТС с E1,Подключение к ИС "Антифрод" E1 PRI/SS#7 УВР Телестор, Грифин и др..
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