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Проблема с входящими звонками

Новичком считается только что прочитавший «Астериск - будущее телефонии»
http://asterisk.ru/knowledgebase/books
и пытающийся сделать большее

Модераторы: april22, Zavr2008

Ответить
musho5755
Сообщения: 37
Зарегистрирован: 07 ноя 2015, 16:41

Проблема с входящими звонками

Сообщение musho5755 »

Подключился к провайдеру без аутентификации, Исходящие звонки работают нормально. Входящие идут, устанавливается сип соединение есть invite, получает ОК от Астериска, но не посылает АСК в ответ. После поднятия трубки астериск не знает куда отправить RTP пакеты. Rtp debug так показывает

Код: Выделить всё

Sent RTP P2P packet to 88.115.XXX.XXX:7116 (type 08, len 000160)
Sent RTP P2P packet to 88.115.XXX.XXX:7116 (type 08, len 000160)
Sent RTP P2P packet to 88.115.XXX.XXX:7116 (type 08, len 000160)
Sent RTP P2P packet to 88.115.XXX.XXX:7116 (type 08, len 000160)
Sent RTP P2P packet to 88.115.XXX.XXX:7116 (type 08, len 000160)
Sent RTP P2P packet to 88.115.XXX.XXX:7116 (type 08, len 000160)
Sent RTP P2P packet to 88.115.XXX.XXX:7116 (type 08, len 000160)
Sent RTP P2P packet to 88.115.XXX.XXX:7116 (type 08, len 000160)
Sent RTP P2P packet to 88.115.XXX.XXX:7116 (type 08, len 000160)
Sent RTP P2P packet to 88.115.XXX.XXX:7116 (type 08, len 000160)
Sent RTP P2P packet to 88.115.XXX.XXX:7116 (type 08, len 000160)
Sent RTP P2P packet to 88.115.XXX.XXX:7116 (type 08, len 000160)
Sent RTP P2P packet to 88.115.XXX.XXX:7116 (type 08, len 000160)
Sent RTP P2P packet to 88.115.XXX.XXX:7116 (type 08, len 000160)
Sent RTP P2P packet to 88.115.XXX.XXX:7116 (type 08, len 000160)
Sent RTP P2P packet to 88.115.XXX.XXX:7116 (type 08, len 000160)
Sent RTP P2P packet to 88.115.XXX.XXX:7116 (type 08, len 000160)
[2016-03-26 12:33:11] WARNING[1370][C-0000012d]: chan_sip.c:9867 process_sdp: Insufficient information for SDP (m= not found)
Проблема мне кажется с NAT-ом. Но что конкретно не правильно делаю, не понимаю. Не подскажете?
sip.conf

Код: Выделить всё

[general]
externip=196.88.XX.XX 
externaddr=196.88.XX.XX
localnet=192.168.230.0/255.255.0.0
alwaysauthreject=yes
allowguest=no; 
bindaddr = 192.168.230.250
bindport=5060;
;textsupport=yes
;videosupport=yes
rtupdate = no
rtcachefriends = no
rtsavesysname = no
directmedia = no

[orange]
type=friend
context=from-provider
host=88.118.XX.XX
fromdomain=196.88.XX.XX
outboundproxy=88.115.XX.XX
media_address=88.115.XX.XX
ignoresdpversion=yes
insecure=port,invite
canreinvite=no
qualify=yes
nat=force_rport,comedia instead
directmedia=yes
disallow=all
allow=g729
allow=alaw
allow=ulaw
dtmfmode = rfc2833
Вернуться к началу
Аватара пользователя
zzuz
Сообщения: 1658
Зарегистрирован: 21 сен 2010, 13:33
Контактная информация:
Контактная информация пользователя zzuz

Re: Проблема с входящими звонками

Сообщение zzuz »

Insufficient information for SDP (m= not found)

К провайдеру.
Линия24 - Системы Массового Телефонного Обслуживания
Вернуться к началу
musho5755
Сообщения: 37
Зарегистрирован: 07 ноя 2015, 16:41

Re: Проблема с входящими звонками

Сообщение musho5755 »

То есть у меня все правильно настроено?
Вернуться к началу
virus_net
Сообщения: 2337
Зарегистрирован: 05 июн 2013, 08:12
Откуда: Москва

Re: Проблема с входящими звонками

Сообщение virus_net »

Т.е. берем tcpdump или включаем sip debug и смотрим SIP пакет INVITE от провайдера и зрим в секцию с SDP.
мой SIP URI sip:virus_net@asterisk.ru
bitname.ru - Домены .bit (namecoin) .emc .coin .lib .bazar (emercoin)
ENUMER - звони бесплатно и напрямую.
Вернуться к началу
musho5755
Сообщения: 37
Зарегистрирован: 07 ноя 2015, 16:41

Re: Проблема с входящими звонками

Сообщение musho5755 »

Вроде бы все нормально показывает. НЕ понимаю где проблема (((
PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:
<--- SIP read from UDP:88.115.XX.XX:5060 --->
INVITE sip:60658980@196.88.XX.XX;user=phone SIP/2.0
Content-Length:247
From:<sip:96352471@SIP.ORANGE;user=phone>;tag=AA48W2ciXf0fA8U8
To:<sip:60658980@196.88.XX.XX;user=phone>
Via:SIP/2.0/UDP SIP.ORANGE:5060;branch=z9hG4bK5XcU74cdA6ZaV8jX
Call-ID:6968FD724DAB329DE2470F03@4e42ffffffff
CSeq:1 INVITE
Max-Forwards:70
Route:<sip:196.88.XX.XX:5060;lr>
Record-Route:<sip:AAQAATjpCAABZAAAA1QAAfMYK@88.115.XX.XX:5060;lr>
Request-Disposition:no-fork
P-Asserted-Identity:<sip:96352471@SIP.ORANGE;user=phone;cpc=ordinary>,<tel:96352471;cpc=ordinary;phone-context=+374>
Session-Expires:1800;refresher=uac
Contact:sip:88.115.XX.XX:5060
Supported:100rel,precondition,timer,norefersub
Allow:ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,UPDATE
P-Charging-Vector:icid-value=c5SVCAEnqmJ6AAK3p~TMBU5CABw-;icid-generated-at=88.115.XX.XX;orig-ioi=SIP.ORANGE
P-Early-Media:supported
Content-Type:application/sdp
Content-Disposition:session;handling=required

v=0
o=- 0 0 IN IP4 11.3.XX.XX
s=-
c=IN IP4 88.115.XX.XX
t=0 0
m=audio 4938 RTP/AVP 18 8 96
b=AS:64
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:96 telephone-event/8000
a=ptime:20
a=maxptime:20
a=3gOoBTC
<------------->
--- (20 headers 14 lines) ---
Sending to 88.115.XX.XX:5060 (no NAT)
Sending to 88.115.XX.XX:5060 (no NAT)
Using INVITE request as basis request - 6968FD724DAB329DE2470F03@4e42ffffffff
Found peer 'orange' for '96352471' from 88.115.XX.XX:5060
Found RTP audio format 18
Found RTP audio format 8
Found RTP audio format 96
Found audio description format G729 for ID 18
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 96
Capabilities: us - (ulaw|alaw|g729), peer - audio=(alaw|g729)/video=(nothing)/text=(nothing), combined - (alaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 88.115.XX.XX:4938
Looking for 60658980 in default (domain 196.88.XX.XX)
list_route: hop: <sip:AAQAATjpCAABZAAAA1QAAfMYK@88.115.XX.XX:5060;lr>

<--- Transmitting (no NAT) to 88.115.XX.XX:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP SIP.ORANGE:5060;branch=z9hG4bK5XcU74cdA6ZaV8jX;received=88.115.XX.XX
Record-Route: <sip:AAQAATjpCAABZAAAA1QAAfMYK@88.115.XX.XX:5060;lr>
From: <sip:96352471@SIP.ORANGE;user=phone>;tag=AA48W2ciXf0fA8U8
To: <sip:60658980@196.88.XX.XX;user=phone>
Call-ID: 6968FD724DAB329DE2470F03@4e42ffffffff
CSeq: 1 INVITE
Server: Asterisk PBX 11.20.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uac
Contact: <sip:60658980@196.88.XX.XX:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 88.115.XX.XX:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP SIP.ORANGE:5060;branch=z9hG4bK5XcU74cdA6ZaV8jX;received=88.115.XX.XX
Record-Route: <sip:AAQAATjpCAABZAAAA1QAAfMYK@88.115.XX.XX:5060;lr>
From: <sip:96352471@SIP.ORANGE;user=phone>;tag=AA48W2ciXf0fA8U8
To: <sip:60658980@196.88.XX.XX;user=phone>;tag=as61207fc4
Call-ID: 6968FD724DAB329DE2470F03@4e42ffffffff
CSeq: 1 INVITE
Server: Asterisk PBX 11.20.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uac
Contact: <sip:60658980@196.88.XX.XX:5060>
Content-Length: 0


Audio is at 50200
Adding codec 100008 (g729) to SDP
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 88.115.XX.XX:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP SIP.ORANGE:5060;branch=z9hG4bK5XcU74cdA6ZaV8jX;received=88.115.XX.XX
Record-Route: <sip:AAQAATjpCAABZAAAA1QAAfMYK@88.115.XX.XX:5060;lr>
From: <sip:96352471@SIP.ORANGE;user=phone>;tag=AA48W2ciXf0fA8U8
To: <sip:60658980@196.88.XX.XX;user=phone>;tag=as61207fc4
Call-ID: 6968FD724DAB329DE2470F03@4e42ffffffff
CSeq: 1 INVITE
Server: Asterisk PBX 11.20.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uac
Contact: <sip:60658980@196.88.XX.XX:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 280

v=0
o=root 389331348 389331348 IN IP4 196.88.XX.XX
s=Asterisk PBX 11.20.0
c=IN IP4 196.88.XX.XX
t=0 0
m=audio 50200 RTP/AVP 18 8 96
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:88.115.XX.XX:5060 --->
ACK sip:60658980@196.88.XX.XX:5060 SIP/2.0
Content-Length:0
From:<sip:96352471@SIP.ORANGE;user=phone>;tag=AA48W2ciXf0fA8U8
To:<sip:60658980@196.88.XX.XX;user=phone>;tag=as61207fc4
Via:SIP/2.0/UDP SIP.ORANGE:5060;branch=z9hG4bKabD6aefD41dU03Wc
Call-ID:6968FD724DAB329DE2470F03@4e42ffffffff
CSeq:1 ACK
Max-Forwards:70
Request-Disposition:no-fork

<------------->
--- (9 headers 0 lines) ---
set_destination: Parsing <sip:AAQAATjpCAABZAAAA1QAAfMYK@88.115.XX.XX:5060;lr> for address/port to send to
set_destination: set destination to 88.115.XX.XX:5060
Audio is at 50200
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 88.115.XX.XX:5060:
INVITE sip:88.115.XX.XX:5060 SIP/2.0
Via: SIP/2.0/UDP 196.88.XX.XX:5060;branch=z9hG4bK00b0bf77
Route: <sip:AAQAATjpCAABZAAAA1QAAfMYK@88.115.XX.XX:5060;lr>
Max-Forwards: 70
From: <sip:60658980@196.88.XX.XX;user=phone>;tag=as61207fc4
To: <sip:96352471@SIP.ORANGE;user=phone>;tag=AA48W2ciXf0fA8U8
Contact: <sip:60658980@196.88.XX.XX:5060>
Call-ID: 6968FD724DAB329DE2470F03@4e42ffffffff
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.20.0
Session-Expires: 1800;refresher=uas
Min-SE: 90
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 235

v=0
o=root 389331348 389331349 IN IP4 192.168.250.80
s=Asterisk PBX 11.20.0
c=IN IP4 192.168.250.80
t=0 0
m=audio 53182 RTP/AVP 8 96
a=rtpmap:8 PCMA/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:88.115.XX.XX:5060 --->
INVITE sip:60658980@196.88.XX.XX:5060 SIP/2.0
Content-Length:211
From:<sip:96352471@SIP.ORANGE;user=phone>;tag=AA48W2ciXf0fA8U8
To:<sip:60658980@196.88.XX.XX;user=phone>;tag=as61207fc4
Via:SIP/2.0/UDP SIP.ORANGE:5060;branch=z9hG4bKDXVZg75XhZ40Zj4j
Call-ID:6968FD724DAB329DE2470F03@4e42ffffffff
CSeq:2 INVITE
Max-Forwards:70
Record-Route:<sip:AAQAATjpCAABZAAAA1QAAfMYK@88.115.XX.XX:5060;lr>
Request-Disposition:no-fork
Session-Expires:1800;refresher=uac
Contact:sip:88.115.XX.XX:5060
Supported:timer
Allow:ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,UPDATE
P-Charging-Vector:icid-value=c5SVCAEnqmJ6AAK3p~TMBU5CABw-;icid-generated-at=88.115.XX.XX;orig-ioi=SIP.ORANGE
Content-Type:application/sdp
Content-Disposition:session;handling=required

v=0
o=- 0 1 IN IP4 11.3.XX.XX
s=-
c=IN IP4 88.115.XX.XX
t=0 0
m=audio 4938 RTP/AVP 18 96
b=AS:8
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:96 telephone-event/8000
a=ptime:20
a=maxptime:20
<------------->
--- (17 headers 12 lines) ---

<--- Reliably Transmitting (no NAT) to 88.115.XX.XX:5060 --->
SIP/2.0 491 Request Pending
Via: SIP/2.0/UDP SIP.ORANGE:5060;branch=z9hG4bKDXVZg75XhZ40Zj4j;received=88.115.XX.XX
From: <sip:96352471@SIP.ORANGE;user=phone>;tag=AA48W2ciXf0fA8U8
To: <sip:60658980@196.88.XX.XX;user=phone>;tag=as61207fc4
Call-ID: 6968FD724DAB329DE2470F03@4e42ffffffff
CSeq: 2 INVITE
Server: Asterisk PBX 11.20.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uac
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


<------------>
Sending to 88.115.XX.XX:5060 (no NAT)

<--- SIP read from UDP:88.115.XX.XX:5060 --->
ACK sip:60658980@196.88.XX.XX:5060 SIP/2.0
Content-Length:0
From:<sip:96352471@SIP.ORANGE;user=phone>;tag=AA48W2ciXf0fA8U8
To:<sip:60658980@196.88.XX.XX;user=phone>;tag=as61207fc4
Via:SIP/2.0/UDP SIP.ORANGE:5060;branch=z9hG4bKDXVZg75XhZ40Zj4j
Call-ID:6968FD724DAB329DE2470F03@4e42ffffffff
CSeq:2 ACK
Max-Forwards:70

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:88.115.XX.XX:5060 --->
SIP/2.0 491 Request Pending
Content-Length:0
From:<sip:60658980@196.88.XX.XX;user=phone>;tag=as61207fc4
To:<sip:96352471@SIP.ORANGE;user=phone>;tag=AA48W2ciXf0fA8U8
Via:SIP/2.0/UDP 196.88.XX.XX:5060;branch=z9hG4bK00b0bf77
Call-ID:6968FD724DAB329DE2470F03@4e42ffffffff
CSeq:102 INVITE
Retry-After:8

<------------->
--- (8 headers 0 lines) ---
set_destination: Parsing <sip:AAQAATjpCAABZAAAA1QAAfMYK@88.115.XX.XX:5060;lr> for address/port to send to
set_destination: set destination to 88.115.XX.XX:5060
Transmitting (no NAT) to 88.115.XX.XX:5060:
ACK sip:88.115.XX.XX:5060 SIP/2.0
Via: SIP/2.0/UDP 196.88.XX.XX:5060;branch=z9hG4bK00b0bf77
Route: <sip:AAQAATjpCAABZAAAA1QAAfMYK@88.115.XX.XX:5060;lr>
Max-Forwards: 70
From: <sip:60658980@196.88.XX.XX;user=phone>;tag=as61207fc4
To: <sip:96352471@SIP.ORANGE;user=phone>;tag=AA48W2ciXf0fA8U8
Contact: <sip:60658980@196.88.XX.XX:5060>
Call-ID: 6968FD724DAB329DE2470F03@4e42ffffffff
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.20.0
Content-Length: 0


---
set_destination: Parsing <sip:AAQAATjpCAABZAAAA1QAAfMYK@88.115.XX.XX:5060;lr> for address/port to send to
set_destination: set destination to 88.115.XX.XX:5060
Audio is at 50200
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 88.115.XX.XX:5060:
INVITE sip:88.115.XX.XX:5060 SIP/2.0
Via: SIP/2.0/UDP 196.88.XX.XX:5060;branch=z9hG4bK3953dd5b
Route: <sip:AAQAATjpCAABZAAAA1QAAfMYK@88.115.XX.XX:5060;lr>
Max-Forwards: 70
From: <sip:60658980@196.88.XX.XX;user=phone>;tag=as61207fc4
To: <sip:96352471@SIP.ORANGE;user=phone>;tag=AA48W2ciXf0fA8U8
Contact: <sip:60658980@196.88.XX.XX:5060>
Call-ID: 6968FD724DAB329DE2470F03@4e42ffffffff
CSeq: 103 INVITE
User-Agent: Asterisk PBX 11.20.0
Session-Expires: 1800;refresher=uas
Min-SE: 90
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 235

v=0
o=root 389331348 389331350 IN IP4 192.168.250.80
s=Asterisk PBX 11.20.0
c=IN IP4 192.168.250.80
t=0 0
m=audio 53182 RTP/AVP 8 96
a=rtpmap:8 PCMA/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:88.115.XX.XX:5060 --->
SIP/2.0 200 OK
Content-Length:201
From:<sip:60658980@196.88.XX.XX;user=phone>;tag=as61207fc4
To:<sip:96352471@SIP.ORANGE;user=phone>;tag=AA48W2ciXf0fA8U8
Via:SIP/2.0/UDP 196.88.XX.XX:5060;branch=z9hG4bK3953dd5b
Call-ID:6968FD724DAB329DE2470F03@4e42ffffffff
CSeq:103 INVITE
Record-Route:<sip:AAQAATjpCAABZAAAA1QAAfMYK@88.115.XX.XX:5060;lr>
Session-Expires:1800;refresher=uas
Contact:sip:88.115.XX.XX:5060
Require:timer
Allow:ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,UPDATE
P-Charging-Vector:icid-value=c5SVCAEnqmJ6AAK3p~TMBU5CABw-;icid-generated-at=88.115.XX.XX;orig-ioi=SIP.ORANGE;term-ioi=SIP.ORANGE
Content-Type:application/sdp
Content-Disposition:session;handling=required

v=0
o=- 0 2 IN IP4 11.3.XX.XX
s=-
c=IN IP4 88.115.XX.XX
t=0 0
m=audio 4938 RTP/AVP 8 96
b=AS:64
a=sendrecv
a=rtpmap:8 PCMA/8000
a=rtpmap:96 telephone-event/8000
a=ptime:20
a=maxptime:20
<------------->
--- (15 headers 12 lines) ---
Found RTP audio format 8
Found RTP audio format 96
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 96
Capabilities: us - (ulaw|alaw|g729), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 88.115.XX.XX:4938
set_destination: Parsing <sip:AAQAATjpCAABZAAAA1QAAfMYK@88.115.XX.XX:5060;lr> for address/port to send to
set_destination: set destination to 88.115.XX.XX:5060
Transmitting (no NAT) to 88.115.XX.XX:5060:
ACK sip:88.115.XX.XX:5060 SIP/2.0
Via: SIP/2.0/UDP 196.88.XX.XX:5060;branch=z9hG4bK0eefbe48
Route: <sip:AAQAATjpCAABZAAAA1QAAfMYK@88.115.XX.XX:5060;lr>
Max-Forwards: 70
From: <sip:60658980@196.88.XX.XX;user=phone>;tag=as61207fc4
To: <sip:96352471@SIP.ORANGE;user=phone>;tag=AA48W2ciXf0fA8U8
Contact: <sip:60658980@196.88.XX.XX:5060>
Call-ID: 6968FD724DAB329DE2470F03@4e42ffffffff
CSeq: 103 ACK
User-Agent: Asterisk PBX 11.20.0
Content-Length: 0


---
set_destination: Parsing <sip:AAQAATjpCAABZAAAA1QAAfMYK@88.115.XX.XX:5060;lr> for address/port to send to
set_destination: set destination to 88.115.XX.XX:5060
Audio is at 50200
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 88.115.XX.XX:5060:
INVITE sip:88.115.XX.XX:5060 SIP/2.0
Via: SIP/2.0/UDP 196.88.XX.XX:5060;branch=z9hG4bK5ff89ca7
Route: <sip:AAQAATjpCAABZAAAA1QAAfMYK@88.115.XX.XX:5060;lr>
Max-Forwards: 70
From: <sip:60658980@196.88.XX.XX;user=phone>;tag=as61207fc4
To: <sip:96352471@SIP.ORANGE;user=phone>;tag=AA48W2ciXf0fA8U8
Contact: <sip:60658980@196.88.XX.XX:5060>
Call-ID: 6968FD724DAB329DE2470F03@4e42ffffffff
CSeq: 104 INVITE
User-Agent: Asterisk PBX 11.20.0
Session-Expires: 1800;refresher=uas
Min-SE: 90
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 235

v=0
o=root 389331348 389331351 IN IP4 192.168.250.80
s=Asterisk PBX 11.20.0
c=IN IP4 192.168.250.80
t=0 0
m=audio 53182 RTP/AVP 8 96
a=rtpmap:8 PCMA/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:88.115.XX.XX:5060 --->
SIP/2.0 200 OK
Content-Length:201
From:<sip:60658980@196.88.XX.XX;user=phone>;tag=as61207fc4
To:<sip:96352471@SIP.ORANGE;user=phone>;tag=AA48W2ciXf0fA8U8
Via:SIP/2.0/UDP 196.88.XX.XX:5060;branch=z9hG4bK5ff89ca7
Call-ID:6968FD724DAB329DE2470F03@4e42ffffffff
CSeq:104 INVITE
Record-Route:<sip:AAQAATjpCAABZAAAA1QAAfMYK@88.115.XX.XX:5060;lr>
Session-Expires:1800;refresher=uas
Contact:sip:88.115.XX.XX:5060
Require:timer
Allow:ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,UPDATE
P-Charging-Vector:icid-value=c5SVCAEnqmJ6AAK3p~TMBU5CABw-;icid-generated-at=88.115.XX.XX;orig-ioi=SIP.ORANGE;term-ioi=SIP.ORANGE
Content-Type:application/sdp
Content-Disposition:session;handling=required

v=0
o=- 0 3 IN IP4 11.3.XX.XX
s=-
c=IN IP4 88.115.XX.XX
t=0 0
m=audio 4938 RTP/AVP 8 96
b=AS:64
a=sendrecv
a=rtpmap:8 PCMA/8000
a=rtpmap:96 telephone-event/8000
a=ptime:20
a=maxptime:20
<------------->
--- (15 headers 12 lines) ---
Found RTP audio format 8
Found RTP audio format 96
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 96
Capabilities: us - (ulaw|alaw|g729), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 88.115.XX.XX:4938
set_destination: Parsing <sip:AAQAATjpCAABZAAAA1QAAfMYK@88.115.XX.XX:5060;lr> for address/port to send to
set_destination: set destination to 88.115.XX.XX:5060
Transmitting (no NAT) to 88.115.XX.XX:5060:
ACK sip:88.115.XX.XX:5060 SIP/2.0
Via: SIP/2.0/UDP 196.88.XX.XX:5060;branch=z9hG4bK297d17af
Route: <sip:AAQAATjpCAABZAAAA1QAAfMYK@88.115.XX.XX:5060;lr>
Max-Forwards: 70
From: <sip:60658980@196.88.XX.XX;user=phone>;tag=as61207fc4
To: <sip:96352471@SIP.ORANGE;user=phone>;tag=AA48W2ciXf0fA8U8
Contact: <sip:60658980@196.88.XX.XX:5060>
Call-ID: 6968FD724DAB329DE2470F03@4e42ffffffff
CSeq: 104 ACK
User-Agent: Asterisk PBX 11.20.0
Content-Length: 0


<--- SIP read from UDP:88.115.XX.XX:5060 --->
INVITE sip:60658980@196.88.XX.XX:5060 SIP/2.0
Content-Length:211
From:<sip:96352471@SIP.ORANGE;user=phone>;tag=AA48W2ciXf0fA8U8
To:<sip:60658980@196.88.XX.XX;user=phone>;tag=as61207fc4
Via:SIP/2.0/UDP SIP.ORANGE:5060;branch=z9hG4bKj251..0h2aYC0391
Call-ID:6968FD724DAB329DE2470F03@4e42ffffffff
CSeq:3 INVITE
Max-Forwards:70
Record-Route:<sip:AAQAATjpCAABZAAAA1QAAfMYK@88.115.XX.XX:5060;lr>
Request-Disposition:no-fork
Session-Expires:1800;refresher=uac
Contact:sip:88.115.XX.XX:5060
Supported:timer
Allow:ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,UPDATE
P-Charging-Vector:icid-value=c5SVCAEnqmJ6AAK3p~TMBU5CABw-;icid-generated-at=88.115.XX.XX;orig-ioi=SIP.ORANGE
Content-Type:application/sdp
Content-Disposition:session;handling=required

v=0
o=- 0 4 IN IP4 11.3.XX.XX
s=-
c=IN IP4 88.115.XX.XX
t=0 0
m=audio 4938 RTP/AVP 18 96
b=AS:8
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:96 telephone-event/8000
a=ptime:20
a=maxptime:20
<------------->
--- (17 headers 12 lines) ---
Sending to 88.115.XX.XX:5060 (no NAT)
Found RTP audio format 18
Found RTP audio format 96
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 96
Capabilities: us - (ulaw|alaw|g729), peer - audio=(g729)/video=(nothing)/text=(nothing), combined - (g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 88.115.XX.XX:4938

<--- Transmitting (no NAT) to 88.115.XX.XX:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP SIP.ORANGE:5060;branch=z9hG4bKj251..0h2aYC0391;received=88.115.XX.XX
Record-Route: <sip:AAQAATjpCAABZAAAA1QAAfMYK@88.115.XX.XX:5060;lr>
From: <sip:96352471@SIP.ORANGE;user=phone>;tag=AA48W2ciXf0fA8U8
To: <sip:60658980@196.88.XX.XX;user=phone>;tag=as61207fc4
Call-ID: 6968FD724DAB329DE2470F03@4e42ffffffff
CSeq: 3 INVITE
Server: Asterisk PBX 11.20.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uac
Contact: <sip:60658980@196.88.XX.XX:5060>
Content-Length: 0


<------------>
Audio is at 50200
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 88.115.XX.XX:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP SIP.ORANGE:5060;branch=z9hG4bKj251..0h2aYC0391;received=88.115.XX.XX
Record-Route: <sip:AAQAATjpCAABZAAAA1QAAfMYK@88.115.XX.XX:5060;lr>
From: <sip:96352471@SIP.ORANGE;user=phone>;tag=AA48W2ciXf0fA8U8
To: <sip:60658980@196.88.XX.XX;user=phone>;tag=as61207fc4
Call-ID: 6968FD724DAB329DE2470F03@4e42ffffffff
CSeq: 3 INVITE
Server: Asterisk PBX 11.20.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uac
Contact: <sip:60658980@196.88.XX.XX:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 136

v=0
o=root 389331348 389331352 IN IP4 192.168.250.80
s=Asterisk PBX 11.20.0
c=IN IP4 192.168.250.80
t=0 0
m=audio 0 RTP/AVP 18 96

<------------>

<--- SIP read from UDP:88.115.XX.XX:5060 --->
ACK sip:60658980@196.88.XX.XX:5060 SIP/2.0
Content-Length:0
From:<sip:96352471@SIP.ORANGE;user=phone>;tag=AA48W2ciXf0fA8U8
To:<sip:60658980@196.88.XX.XX;user=phone>;tag=as61207fc4
Via:SIP/2.0/UDP SIP.ORANGE:5060;branch=z9hG4bKa0ZU5D5gg6_f_Cje
Call-ID:6968FD724DAB329DE2470F03@4e42ffffffff
CSeq:3 ACK
Max-Forwards:70
Request-Disposition:no-fork

<------------->
--- (9 headers 0 lines) ---
[2016-03-26 12:21:43] WARNING[1370][C-00000115]: chan_sip.c:9867 process_sdp: Insufficient information for SDP (m= not found)
set_destination: Parsing <sip:AAQAATjpCAABZAAAA1QAAfMYK@88.115.XX.XX:5060;lr> for address/port to send to
set_destination: set destination to 88.115.XX.XX:5060
Audio is at 50200
Adding codec 100008 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 88.115.XX.XX:5060:
INVITE sip:88.115.XX.XX:5060 SIP/2.0
Via: SIP/2.0/UDP 196.88.XX.XX:5060;branch=z9hG4bK085e0054
Route: <sip:AAQAATjpCAABZAAAA1QAAfMYK@88.115.XX.XX:5060;lr>
Max-Forwards: 70
From: <sip:60658980@196.88.XX.XX;user=phone>;tag=as61207fc4
To: <sip:96352471@SIP.ORANGE;user=phone>;tag=AA48W2ciXf0fA8U8
Contact: <sip:60658980@196.88.XX.XX:5060>
Call-ID: 6968FD724DAB329DE2470F03@4e42ffffffff
CSeq: 105 INVITE
User-Agent: Asterisk PBX 11.20.0
Session-Expires: 1800;refresher=uas
Min-SE: 90
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 256

v=0
o=root 389331348 389331353 IN IP4 196.88.XX.XX
s=Asterisk PBX 11.20.0
c=IN IP4 196.88.XX.XX
t=0 0
m=audio 50200 RTP/AVP 18 96
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:88.115.XX.XX:5060 --->
SIP/2.0 200 OK
Content-Length:223
From:<sip:60658980@196.88.XX.XX;user=phone>;tag=as61207fc4
To:<sip:96352471@SIP.ORANGE;user=phone>;tag=AA48W2ciXf0fA8U8
Via:SIP/2.0/UDP 196.88.XX.XX:5060;branch=z9hG4bK085e0054
Call-ID:6968FD724DAB329DE2470F03@4e42ffffffff
CSeq:105 INVITE
Record-Route:<sip:AAQAATjpCAABZAAAA1QAAfMYK@88.115.XX.XX:5060;lr>
Session-Expires:1800;refresher=uas
Contact:sip:88.115.XX.XX:5060
Require:timer
Allow:ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,UPDATE
P-Charging-Vector:icid-value=c5SVCAEnqmJ6AAK3p~TMBU5CABw-;icid-generated-at=88.115.XX.XX;orig-ioi=SIP.ORANGE;term-ioi=SIP.ORANGE
Content-Type:application/sdp
Content-Disposition:session;handling=required

v=0
o=- 0 5 IN IP4 11.3.XX.XX
s=-
c=IN IP4 88.115.XX.XX
t=0 0
m=audio 4938 RTP/AVP 18 96
b=AS:8
a=sendrecv
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:96 telephone-event/8000
a=ptime:20
a=maxptime:20
<------------->
--- (15 headers 13 lines) ---
Found RTP audio format 18
Found RTP audio format 96
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 96
Capabilities: us - (ulaw|alaw|g729), peer - audio=(g729)/video=(nothing)/text=(nothing), combined - (g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 88.115.XX.XX:4938
set_destination: Parsing <sip:AAQAATjpCAABZAAAA1QAAfMYK@88.115.XX.XX:5060;lr> for address/port to send to
set_destination: set destination to 88.115.XX.XX:5060
Transmitting (no NAT) to 88.115.XX.XX:5060:
ACK sip:88.115.XX.XX:5060 SIP/2.0
Via: SIP/2.0/UDP 196.88.XX.XX:5060;branch=z9hG4bK6bd36a2f
Route: <sip:AAQAATjpCAABZAAAA1QAAfMYK@88.115.XX.XX:5060;lr>
Max-Forwards: 70
From: <sip:60658980@196.88.XX.XX;user=phone>;tag=as61207fc4
To: <sip:96352471@SIP.ORANGE;user=phone>;tag=AA48W2ciXf0fA8U8
Contact: <sip:60658980@196.88.XX.XX:5060>
Call-ID: 6968FD724DAB329DE2470F03@4e42ffffffff
CSeq: 105 ACK
User-Agent: Asterisk PBX 11.20.0
Content-Length: 0


---
Scheduling destruction of SIP dialog '6968FD724DAB329DE2470F03@4e42ffffffff' in 6400 ms (Method: ACK)
set_destination: Parsing <sip:AAQAATjpCAABZAAAA1QAAfMYK@88.115.XX.XX:5060;lr> for address/port to send to
set_destination: set destination to 88.115.XX.XX:5060
Reliably Transmitting (no NAT) to 88.115.XX.XX:5060:
BYE sip:88.115.XX.XX:5060 SIP/2.0
Via: SIP/2.0/UDP 196.88.XX.XX:5060;branch=z9hG4bK20508e71
Route: <sip:AAQAATjpCAABZAAAA1QAAfMYK@88.115.XX.XX:5060;lr>
Max-Forwards: 70
From: <sip:60658980@196.88.XX.XX;user=phone>;tag=as61207fc4
To: <sip:96352471@SIP.ORANGE;user=phone>;tag=AA48W2ciXf0fA8U8
Call-ID: 6968FD724DAB329DE2470F03@4e42ffffffff
CSeq: 106 BYE
User-Agent: Asterisk PBX 11.20.0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---

<--- SIP read from UDP:88.115.XX.XX:5060 --->
SIP/2.0 200 OK
Content-Length:0
From:<sip:60658980@196.88.XX.XX;user=phone>;tag=as61207fc4
To:<sip:96352471@SIP.ORANGE;user=phone>;tag=AA48W2ciXf0fA8U8
Via:SIP/2.0/UDP 196.88.XX.XX:5060;branch=z9hG4bK20508e71
Call-ID:6968FD724DAB329DE2470F03@4e42ffffffff
CSeq:106 BYE
Allow:ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,UPDATE

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '6968FD724DAB329DE2470F03@4e42ffffffff' Method: ACK
Вернуться к началу
musho5755
Сообщения: 37
Зарегистрирован: 07 ноя 2015, 16:41

Re: Проблема с входящими звонками

Сообщение musho5755 »

Просто указал canreinvite=yes а надо было поставить no
Вернуться к началу
Ответить

Вернуться в «Вопросы новичков»

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