asterisk.ru

Знаю что проблема очень избитая, но для моего случая почему-то решения не нашел, может кто нибудь наставит на путь истинный.
Версия Asterisk 13 Сборка FreePBX-64bit-6.12.65
1. Имеем телефон 1, телефон 2, телефон 3 (территориально находится в другом районе), и сотовый телефон.
2. Телефон 1 и телефон 2 заведен на астериск.
Настройки

PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:

Код: Выделить всё

host=10.72.0.9
port=5060
username=+375232232302@ims.beltel.by
secret=12345
type=friend
fromdomain=ims.beltel.by
dtmfmode=rfc2833
insecure=invite,port
disallow=all
allow=alaw
nat=yes
canreinvite=no
context=from-trunk

Строка регистрации

Код: Выделить всё

+375232232302@ims.beltel.by:123456:+375232232302@ims.beltel.by@10.72.0.9:5060/+375232232302

Настройка второго транка аналогично первому,
Asterisk поднят на Hyper-V 2012, с двумя сетевухами, настройки ниже

PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:

eth0 static
ip 192.168.1.240
mask 255.255.255.0
eth1 dhcp
10.223.222.135
mask 255.255.128.0
gateway 10.223.128.1
в eth0 воткнута локальная сеть
в eth1 воткнут dsl-модем в режиме бриджа

Проблема заключается в том что при звонке с телефона 3, либо с сотового телефона не слышно ни в одну ни в другую сторону, что характерно при звонке с сотового при поднятии трубки 1, слышны небольшие потрескивания, хотя отсчет времени не начинается, а на телефоне 1 отсчет времени начинается. Так же при звонке с телефона 1 на телефон 2 всё прекрасно слышится, так же всё прекрасно слышится если звонить с телефона 1 либо 2 на телефон 3 либо на сотовый телефон, не знаю имеет ли это значение, но при звонке с сотового телефона, номер сотового определяется как 8441234567 хотя на самом деле номер либо 80441234567 либо +375441234567. Так же если подключить прекрасный китайский модем который выдает наш оператор, H208N то входящие прекрасным образом начинают работать. Очень прошу помощи, потому что уже не знаю что проверять...
sip show peers в общем то стандартно

PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:

Код: Выделить всё

Name/username             Host                                    Dyn Forcerport Comedia    ACL Port     Status      Description                      
+375232232302/+3752322323 10.72.0.9                                   Yes        Yes            5060     OK (35 ms)                                   
+375232232305/+3752322323 10.72.0.9                                   Yes        Yes            5060     OK (34 ms)                                   
101/101                   192.168.1.10                             D  Yes        Yes         A  5061     OK (155 ms)                                  
102/102                   192.168.1.19                             D  Yes        Yes         A  5061     OK (157 ms)

sip set debug ip 10.72.0.9
core set verbose 0

PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:

Код: Выделить всё

[2015-06-17 17:33:04] VERBOSE[1899] chan_sip.c:
<--- SIP read from UDP:10.72.0.9:5060 --->
INVITE sip:+375232232302@10.223.222.135:5061 SIP/2.0
Via: SIP/2.0/UDP 10.72.0.9:5060;branch=z9hG4bK99092371o349ayrz2x7z29wb3T39988
Call-ID: asbc1434562393558226028285@10.1.0.97
From: <tel:8441234567;phone-context=+375>;tag=sbc0405339597509
To: <sip:+375232232302@10.72.0.9:5060;transport=udp;user=phone>
CSeq: 1 INVITE
Allow: INVITE,ACK,BYE,CANCEL,UPDATE,INFO,PRACK,NOTIFY,REFER,SUBSCRIBE,OPTIONS,MESSAGE
Contact: <sip:10.72.0.9:5060;did=1412039382258>
Max-Forwards: 69
Supported: 100rel,timer
P-Asserted-Identity: <tel:8441234567;phone-context=+375>
P-Called-Party-ID: <tel:+375232232302>
Session-Expires: 600
Min-SE: 600
Content-Length: 224
Content-Type: application/sdp

v=0
o=- 253539630 253539630 IN IP4 10.72.0.10
s=SBC call
c=IN IP4 10.72.0.10
t=0 0
m=audio 28020 RTP/AVP 8 0 4 18 116
a=fmtp:4 annexa=no
a=fmtp:18 annexb=no
a=rtpmap:116 telephone-event/8000
a=ptime:20
a=sendrecv
<------------->
[2015-06-17 17:33:04] VERBOSE[1899] chan_sip.c: --- (16 headers 11 lines) ---
[2015-06-17 17:33:04] VERBOSE[1899] chan_sip.c: Sending to 10.72.0.9:5060 (NAT)
[2015-06-17 17:33:04] VERBOSE[1899][C-0000000a] chan_sip.c: Sending to 10.72.0.9:5060 (NAT)
[2015-06-17 17:33:04] VERBOSE[1899][C-0000000a] chan_sip.c: Using INVITE request as basis request - asbc1434562393558226028285@10.1.0.97
[2015-06-17 17:33:04] VERBOSE[1899][C-0000000a] chan_sip.c: Found peer '+375232232302' for '8441234567' from 10.72.0.9:5060
[2015-06-17 17:33:04] VERBOSE[1899][C-0000000a] chan_sip.c: Found RTP audio format 8
[2015-06-17 17:33:04] VERBOSE[1899][C-0000000a] chan_sip.c: Found RTP audio format 0
[2015-06-17 17:33:04] VERBOSE[1899][C-0000000a] chan_sip.c: Found RTP audio format 4
[2015-06-17 17:33:04] VERBOSE[1899][C-0000000a] chan_sip.c: Found RTP audio format 18
[2015-06-17 17:33:04] VERBOSE[1899][C-0000000a] chan_sip.c: Found RTP audio format 116
[2015-06-17 17:33:04] VERBOSE[1899][C-0000000a] chan_sip.c: Found audio description format telephone-event for ID 116
[2015-06-17 17:33:04] VERBOSE[1899][C-0000000a] chan_sip.c: Capabilities: us - (alaw), peer - audio=(ulaw|g723|alaw|g729)/video=(nothing)/text=(nothing), combined - (alaw)
[2015-06-17 17:33:04] VERBOSE[1899][C-0000000a] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[2015-06-17 17:33:04] VERBOSE[1899][C-0000000a] chan_sip.c: Peer audio RTP is at port 10.72.0.10:28020
[2015-06-17 17:33:04] VERBOSE[1899][C-0000000a] chan_sip.c: Looking for +375232232302 in from-trunk (domain 10.223.222.135)
[2015-06-17 17:33:04] VERBOSE[1899][C-0000000a] sip/route.c: sip_route_dump: route/path hop: <sip:10.72.0.9:5060;did=1412039382258>
[2015-06-17 17:33:04] VERBOSE[1899][C-0000000a] chan_sip.c:
<--- Transmitting (NAT) to 10.72.0.9:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.72.0.9:5060;branch=z9hG4bK99092371o349ayrz2x7z29wb3T39988;received=10.72.0.9;rport=5060
From: <tel:8441234567;phone-context=+375>;tag=sbc0405339597509
To: <sip:+375232232302@10.72.0.9:5060;transport=udp;user=phone>
Call-ID: asbc1434562393558226028285@10.1.0.97
CSeq: 1 INVITE
Server: FPBX-12.0.68(13.3.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 600;refresher=uas
Contact: <sip:+375232232302@10.223.222.135:5061>
Content-Length: 0


<------------>
[2015-06-17 17:33:04] WARNING[56604][C-0000000a] func_channel.c: Unknown or unavailable item requested: 'reversecharge'
[2015-06-17 17:33:04] VERBOSE[56604][C-0000000a] chan_sip.c:
<--- Transmitting (NAT) to 10.72.0.9:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.72.0.9:5060;branch=z9hG4bK99092371o349ayrz2x7z29wb3T39988;received=10.72.0.9;rport=5060
From: <tel:8441234567;phone-context=+375>;tag=sbc0405339597509
To: <sip:+375232232302@10.72.0.9:5060;transport=udp;user=phone>;tag=as429999e8
Call-ID: asbc1434562393558226028285@10.1.0.97
CSeq: 1 INVITE
Server: FPBX-12.0.68(13.3.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 600;refresher=uas
Contact: <sip:+375232232302@10.223.222.135:5061>
Content-Length: 0


<------------>
[2015-06-17 17:33:04] VERBOSE[56604][C-0000000a] chan_sip.c:
<--- Transmitting (NAT) to 10.72.0.9:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.72.0.9:5060;branch=z9hG4bK99092371o349ayrz2x7z29wb3T39988;received=10.72.0.9;rport=5060
From: <tel:8441234567;phone-context=+375>;tag=sbc0405339597509
To: <sip:+375232232302@10.72.0.9:5060;transport=udp;user=phone>;tag=as429999e8
Call-ID: asbc1434562393558226028285@10.1.0.97
CSeq: 1 INVITE
Server: FPBX-12.0.68(13.3.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 600;refresher=uas
Contact: <sip:+375232232302@10.223.222.135:5061>
Content-Length: 0


<------------>
[2015-06-17 17:33:06] VERBOSE[56604][C-0000000a] chan_sip.c: Audio is at 13326
[2015-06-17 17:33:06] VERBOSE[56604][C-0000000a] chan_sip.c: Adding codec alaw to SDP
[2015-06-17 17:33:06] VERBOSE[56604][C-0000000a] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[2015-06-17 17:33:06] VERBOSE[56604][C-0000000a] chan_sip.c:
<--- Reliably Transmitting (NAT) to 10.72.0.9:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.72.0.9:5060;branch=z9hG4bK99092371o349ayrz2x7z29wb3T39988;received=10.72.0.9;rport=5060
From: <tel:8441234567;phone-context=+375>;tag=sbc0405339597509
To: <sip:+375232232302@10.72.0.9:5060;transport=udp;user=phone>;tag=as429999e8
Call-ID: asbc1434562393558226028285@10.1.0.97
CSeq: 1 INVITE
Server: FPBX-12.0.68(13.3.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 600;refresher=uas
Contact: <sip:+375232232302@10.223.222.135:5061>
Content-Type: application/sdp
Require: timer
Content-Length: 253

v=0
o=root 497129578 497129578 IN IP4 10.223.222.135
s=Asterisk PBX 13.3.2
c=IN IP4 10.223.222.135
t=0 0
m=audio 13326 RTP/AVP 8 116
a=rtpmap:8 PCMA/8000
a=rtpmap:116 telephone-event/8000
a=fmtp:116 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<------------>
[2015-06-17 17:33:07] VERBOSE[1899] chan_sip.c: Retransmitting #1 (NAT) to 10.72.0.9:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.72.0.9:5060;branch=z9hG4bK99092371o349ayrz2x7z29wb3T39988;received=10.72.0.9;rport=5060
From: <tel:8441234567;phone-context=+375>;tag=sbc0405339597509
To: <sip:+375232232302@10.72.0.9:5060;transport=udp;user=phone>;tag=as429999e8
Call-ID: asbc1434562393558226028285@10.1.0.97
CSeq: 1 INVITE
Server: FPBX-12.0.68(13.3.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 600;refresher=uas
Contact: <sip:+375232232302@10.223.222.135:5061>
Content-Type: application/sdp
Require: timer
Content-Length: 253

v=0
o=root 497129578 497129578 IN IP4 10.223.222.135
s=Asterisk PBX 13.3.2
c=IN IP4 10.223.222.135
t=0 0
m=audio 13326 RTP/AVP 8 116
a=rtpmap:8 PCMA/8000
a=rtpmap:116 telephone-event/8000
a=fmtp:116 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

---
[2015-06-17 17:33:07] VERBOSE[1899] chan_sip.c:
<--- SIP read from UDP:10.72.0.9:5060 --->
ACK sip:+375232232302@10.223.222.135:5061 SIP/2.0
Via: SIP/2.0/UDP 10.72.0.9:5060;branch=z9hG4bK01aiwiyw9y9xbwyry07449y41T40825
Call-ID: asbc1434562393558226028285@10.1.0.97
From: <tel:8441234567;phone-context=+375>;tag=sbc0405339597509
To: <sip:+375232232302@10.72.0.9:5060;transport=udp;user=phone>;tag=as429999e8
CSeq: 1 ACK
Max-Forwards: 69
Content-Length: 0

<------------->
[2015-06-17 17:33:07] VERBOSE[1899] chan_sip.c: --- (8 headers 0 lines) ---
[2015-06-17 17:33:07] VERBOSE[1899] chan_sip.c:
<--- SIP read from UDP:10.72.0.9:5060 --->
ACK sip:+375232232302@10.223.222.135:5061 SIP/2.0
Via: SIP/2.0/UDP 10.72.0.9:5060;branch=z9hG4bK01aiwiyw9y9xbwyry07449y41T40825
Call-ID: asbc1434562393558226028285@10.1.0.97
From: <tel:8441234567;phone-context=+375>;tag=sbc0405339597509
To: <sip:+375232232302@10.72.0.9:5060;transport=udp;user=phone>;tag=as429999e8
CSeq: 1 ACK
Max-Forwards: 69
Content-Length: 0

<------------->
[2015-06-17 17:33:07] VERBOSE[1899] chan_sip.c: --- (8 headers 0 lines) ---
[2015-06-17 17:33:07] VERBOSE[1899] chan_sip.c:
<--- SIP read from UDP:10.72.0.9:5060 --->
INVITE sip:+375232232302@10.223.222.135:5061 SIP/2.0
Via: SIP/2.0/UDP 10.72.0.9:5060;branch=z9hG4bK32043212xix3799brxyx34xo2T40875
Call-ID: asbc1434562393558226028285@10.1.0.97
From: <tel:8441234567;phone-context=+375>;tag=sbc0405339597509
To: <sip:+375232232302@10.72.0.9:5060;transport=udp;user=phone>;tag=as429999e8
CSeq: 2 INVITE
Contact: <sip:10.72.0.9:5060;did=1412039382258>
Max-Forwards: 69
Supported: timer
Session-Expires: 600;refresher=uas
Min-SE: 600
Content-Length: 0

<------------->
[2015-06-17 17:33:07] VERBOSE[1899] chan_sip.c: --- (12 headers 0 lines) ---
[2015-06-17 17:33:07] VERBOSE[1899][C-0000000a] chan_sip.c: Sending to 10.72.0.9:5060 (NAT)
[2015-06-17 17:33:07] VERBOSE[1899][C-0000000a] chan_sip.c:
<--- Transmitting (NAT) to 10.72.0.9:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.72.0.9:5060;branch=z9hG4bK32043212xix3799brxyx34xo2T40875;received=10.72.0.9;rport=5060
From: <tel:8441234567;phone-context=+375>;tag=sbc0405339597509
To: <sip:+375232232302@10.72.0.9:5060;transport=udp;user=phone>;tag=as429999e8
Call-ID: asbc1434562393558226028285@10.1.0.97
CSeq: 2 INVITE
Server: FPBX-12.0.68(13.3.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 600;refresher=uas
Contact: <sip:+375232232302@10.223.222.135:5061>
Content-Length: 0


<------------>
[2015-06-17 17:33:07] VERBOSE[1899][C-0000000a] chan_sip.c: Audio is at 13326
[2015-06-17 17:33:07] VERBOSE[1899][C-0000000a] chan_sip.c: Adding codec alaw to SDP
[2015-06-17 17:33:07] VERBOSE[1899][C-0000000a] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[2015-06-17 17:33:07] VERBOSE[1899][C-0000000a] chan_sip.c:
<--- Reliably Transmitting (NAT) to 10.72.0.9:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.72.0.9:5060;branch=z9hG4bK32043212xix3799brxyx34xo2T40875;received=10.72.0.9;rport=5060
From: <tel:8441234567;phone-context=+375>;tag=sbc0405339597509
To: <sip:+375232232302@10.72.0.9:5060;transport=udp;user=phone>;tag=as429999e8
Call-ID: asbc1434562393558226028285@10.1.0.97
CSeq: 2 INVITE
Server: FPBX-12.0.68(13.3.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 600;refresher=uas
Contact: <sip:+375232232302@10.223.222.135:5061>
Content-Type: application/sdp
Require: timer
Content-Length: 253

v=0
o=root 497129578 497129579 IN IP4 10.223.222.135
s=Asterisk PBX 13.3.2
c=IN IP4 10.223.222.135
t=0 0
m=audio 13326 RTP/AVP 8 116
a=rtpmap:8 PCMA/8000
a=rtpmap:116 telephone-event/8000
a=fmtp:116 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<------------>
[2015-06-17 17:33:07] VERBOSE[1899] chan_sip.c: Retransmitting #1 (NAT) to 10.72.0.9:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.72.0.9:5060;branch=z9hG4bK32043212xix3799brxyx34xo2T40875;received=10.72.0.9;rport=5060
From: <tel:8441234567;phone-context=+375>;tag=sbc0405339597509
To: <sip:+375232232302@10.72.0.9:5060;transport=udp;user=phone>;tag=as429999e8
Call-ID: asbc1434562393558226028285@10.1.0.97
CSeq: 2 INVITE
Server: FPBX-12.0.68(13.3.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 600;refresher=uas
Contact: <sip:+375232232302@10.223.222.135:5061>
Content-Type: application/sdp
Require: timer
Content-Length: 253

v=0
o=root 497129578 497129579 IN IP4 10.223.222.135
s=Asterisk PBX 13.3.2
c=IN IP4 10.223.222.135
t=0 0
m=audio 13326 RTP/AVP 8 116
a=rtpmap:8 PCMA/8000
a=rtpmap:116 telephone-event/8000
a=fmtp:116 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

---
[2015-06-17 17:33:07] VERBOSE[1899] chan_sip.c: Retransmitting #2 (NAT) to 10.72.0.9:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.72.0.9:5060;branch=z9hG4bK32043212xix3799brxyx34xo2T40875;received=10.72.0.9;rport=5060
From: <tel:8441234567;phone-context=+375>;tag=sbc0405339597509
To: <sip:+375232232302@10.72.0.9:5060;transport=udp;user=phone>;tag=as429999e8
Call-ID: asbc1434562393558226028285@10.1.0.97
CSeq: 2 INVITE
Server: FPBX-12.0.68(13.3.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 600;refresher=uas
Contact: <sip:+375232232302@10.223.222.135:5061>
Content-Type: application/sdp
Require: timer
Content-Length: 253

v=0
o=root 497129578 497129579 IN IP4 10.223.222.135
s=Asterisk PBX 13.3.2
c=IN IP4 10.223.222.135
t=0 0
m=audio 13326 RTP/AVP 8 116
a=rtpmap:8 PCMA/8000
a=rtpmap:116 telephone-event/8000
a=fmtp:116 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

---
[2015-06-17 17:33:07] VERBOSE[1899] chan_sip.c: Retransmitting #3 (NAT) to 10.72.0.9:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.72.0.9:5060;branch=z9hG4bK32043212xix3799brxyx34xo2T40875;received=10.72.0.9;rport=5060
From: <tel:8441234567;phone-context=+375>;tag=sbc0405339597509
To: <sip:+375232232302@10.72.0.9:5060;transport=udp;user=phone>;tag=as429999e8
Call-ID: asbc1434562393558226028285@10.1.0.97
CSeq: 2 INVITE
Server: FPBX-12.0.68(13.3.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 600;refresher=uas
Contact: <sip:+375232232302@10.223.222.135:5061>
Content-Type: application/sdp
Require: timer
Content-Length: 253

v=0
o=root 497129578 497129579 IN IP4 10.223.222.135
s=Asterisk PBX 13.3.2
c=IN IP4 10.223.222.135
t=0 0
m=audio 13326 RTP/AVP 8 116
a=rtpmap:8 PCMA/8000
a=rtpmap:116 telephone-event/8000
a=fmtp:116 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

---
[2015-06-17 17:33:08] VERBOSE[1899] chan_sip.c: Retransmitting #4 (NAT) to 10.72.0.9:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.72.0.9:5060;branch=z9hG4bK32043212xix3799brxyx34xo2T40875;received=10.72.0.9;rport=5060
From: <tel:8441234567;phone-context=+375>;tag=sbc0405339597509
To: <sip:+375232232302@10.72.0.9:5060;transport=udp;user=phone>;tag=as429999e8
Call-ID: asbc1434562393558226028285@10.1.0.97
CSeq: 2 INVITE
Server: FPBX-12.0.68(13.3.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 600;refresher=uas
Contact: <sip:+375232232302@10.223.222.135:5061>
Content-Type: application/sdp
Require: timer
Content-Length: 253

v=0
o=root 497129578 497129579 IN IP4 10.223.222.135
s=Asterisk PBX 13.3.2
c=IN IP4 10.223.222.135
t=0 0
m=audio 13326 RTP/AVP 8 116
a=rtpmap:8 PCMA/8000
a=rtpmap:116 telephone-event/8000
a=fmtp:116 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

---
[2015-06-17 17:33:10] VERBOSE[1899] chan_sip.c: Retransmitting #5 (NAT) to 10.72.0.9:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.72.0.9:5060;branch=z9hG4bK32043212xix3799brxyx34xo2T40875;received=10.72.0.9;rport=5060
From: <tel:8441234567;phone-context=+375>;tag=sbc0405339597509
To: <sip:+375232232302@10.72.0.9:5060;transport=udp;user=phone>;tag=as429999e8
Call-ID: asbc1434562393558226028285@10.1.0.97
CSeq: 2 INVITE
Server: FPBX-12.0.68(13.3.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 600;refresher=uas
Contact: <sip:+375232232302@10.223.222.135:5061>
Content-Type: application/sdp
Require: timer
Content-Length: 253

v=0
o=root 497129578 497129579 IN IP4 10.223.222.135
s=Asterisk PBX 13.3.2
c=IN IP4 10.223.222.135
t=0 0
m=audio 13326 RTP/AVP 8 116
a=rtpmap:8 PCMA/8000
a=rtpmap:116 telephone-event/8000
a=fmtp:116 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

---
[2015-06-17 17:33:13] VERBOSE[1899] chan_sip.c: Retransmitting #6 (NAT) to 10.72.0.9:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.72.0.9:5060;branch=z9hG4bK32043212xix3799brxyx34xo2T40875;received=10.72.0.9;rport=5060
From: <tel:8441234567;phone-context=+375>;tag=sbc0405339597509
To: <sip:+375232232302@10.72.0.9:5060;transport=udp;user=phone>;tag=as429999e8
Call-ID: asbc1434562393558226028285@10.1.0.97
CSeq: 2 INVITE
Server: FPBX-12.0.68(13.3.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 600;refresher=uas
Contact: <sip:+375232232302@10.223.222.135:5061>
Content-Type: application/sdp
Require: timer
Content-Length: 253

v=0
o=root 497129578 497129579 IN IP4 10.223.222.135
s=Asterisk PBX 13.3.2
c=IN IP4 10.223.222.135
t=0 0
m=audio 13326 RTP/AVP 8 116
a=rtpmap:8 PCMA/8000
a=rtpmap:116 telephone-event/8000
a=fmtp:116 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

---
[2015-06-17 17:33:13] WARNING[1899] chan_sip.c: Retransmission timeout reached on transmission asbc1434562393558226028285@10.1.0.97 for seqno 2 (Non-critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6398ms with no response
[2015-06-17 17:33:13] WARNING[1899] chan_sip.c: Timeout on asbc1434562393558226028285@10.1.0.97 on non-critical invite transaction.
[2015-06-17 17:33:19] NOTICE[1899] chan_sip.c: -- Re-registration for +375232232305@10.72.0.9
[2015-06-17 17:33:19] VERBOSE[1899] chan_sip.c: REGISTER 12 headers, 0 lines
[2015-06-17 17:33:19] VERBOSE[1899] chan_sip.c: Reliably Transmitting (NAT) to 10.72.0.9:5060:
REGISTER sip:ims.beltel.by SIP/2.0
Via: SIP/2.0/UDP 10.223.222.135:5061;branch=z9hG4bK7f892d01;rport
Max-Forwards: 70
From: <sip:+375232232305@ims.beltel.by>;tag=as56a24fec
To: <sip:+375232232305@ims.beltel.by>
Call-ID: 27d2b51d29e25bd00b3408fc1ce6c695@[::1]
CSeq: 106 REGISTER
Supported: replaces, timer
User-Agent: FPBX-12.0.68(13.3.2)
Authorization: Digest username="+375232232305@ims.beltel.by", realm="ims.beltel.by", algorithm=MD5, uri="sip:ims.beltel.by", nonce="Qx8hngc7jo8YmNCPgH/qKA==", response="48cebb54f2000593fa5df9b0dca48e45"
Expires: 120
Contact: <sip:+375232232305@10.223.222.135:5061>
Content-Length: 0


---
[2015-06-17 17:33:19] VERBOSE[1899] chan_sip.c:
<--- SIP read from UDP:10.72.0.9:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.223.222.135:5061;branch=z9hG4bK7f892d01;received=10.223.222.135;rport=5061
Call-ID: 27d2b51d29e25bd00b3408fc1ce6c695@[::1]
From: <sip:+375232232305@ims.beltel.by>;tag=as56a24fec
To: <sip:+375232232305@ims.beltel.by>;tag=slaql9rn
CSeq: 106 REGISTER
WWW-Authenticate: Digest realm="ims.beltel.by",nonce="AulTX9WBWqwi3XA6ZUvytA==",stale=true,algorithm=MD5
Content-Length: 0

<------------->
[2015-06-17 17:33:19] VERBOSE[1899] chan_sip.c: --- (8 headers 0 lines) ---
[2015-06-17 17:33:19] VERBOSE[1899] chan_sip.c: Responding to challenge, registration to domain/host name 10.72.0.9
[2015-06-17 17:33:19] VERBOSE[1899] chan_sip.c: REGISTER 12 headers, 0 lines
[2015-06-17 17:33:19] VERBOSE[1899] chan_sip.c: Reliably Transmitting (NAT) to 10.72.0.9:5060:
REGISTER sip:ims.beltel.by SIP/2.0
Via: SIP/2.0/UDP 10.223.222.135:5061;branch=z9hG4bK31ffeaca;rport
Max-Forwards: 70
From: <sip:+375232232305@ims.beltel.by>;tag=as56a24fec
To: <sip:+375232232305@ims.beltel.by>
Call-ID: 27d2b51d29e25bd00b3408fc1ce6c695@[::1]
CSeq: 107 REGISTER
Supported: replaces, timer
User-Agent: FPBX-12.0.68(13.3.2)
Authorization: Digest username="+375232232305@ims.beltel.by", realm="ims.beltel.by", algorithm=MD5, uri="sip:ims.beltel.by", nonce="AulTX9WBWqwi3XA6ZUvytA==", response="bce0f0c0b07b804555181c65ab42147c"
Expires: 120
Contact: <sip:+375232232305@10.223.222.135:5061>
Content-Length: 0


---
[2015-06-17 17:33:19] NOTICE[1899] chan_sip.c: -- Re-registration for +375232232302@10.72.0.9
[2015-06-17 17:33:19] VERBOSE[1899] chan_sip.c: REGISTER 12 headers, 0 lines
[2015-06-17 17:33:19] VERBOSE[1899] chan_sip.c: Reliably Transmitting (NAT) to 10.72.0.9:5060:
REGISTER sip:ims.beltel.by SIP/2.0
Via: SIP/2.0/UDP 10.223.222.135:5061;branch=z9hG4bK0ac36689;rport
Max-Forwards: 70
From: <sip:+375232232302@ims.beltel.by>;tag=as1257a698
To: <sip:+375232232302@ims.beltel.by>
Call-ID: 56caf4840180e6ba0c425bca32fc2059@[::1]
CSeq: 106 REGISTER
Supported: replaces, timer
User-Agent: FPBX-12.0.68(13.3.2)
Authorization: Digest username="+375232232302@ims.beltel.by", realm="ims.beltel.by", algorithm=MD5, uri="sip:ims.beltel.by", nonce="kLOddWRJ97TtW5+9qV4U6w==", response="632b76dfc06fc60e3b66e6e3baae221b"
Expires: 120
Contact: <sip:+375232232302@10.223.222.135:5061>
Content-Length: 0


---
[2015-06-17 17:33:19] VERBOSE[1899] chan_sip.c:
<--- SIP read from UDP:10.72.0.9:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.223.222.135:5061;branch=z9hG4bK0ac36689;received=10.223.222.135;rport=5061
Call-ID: 56caf4840180e6ba0c425bca32fc2059@[::1]
From: <sip:+375232232302@ims.beltel.by>;tag=as1257a698
To: <sip:+375232232302@ims.beltel.by>;tag=zr3y4riz
CSeq: 106 REGISTER
WWW-Authenticate: Digest realm="ims.beltel.by",nonce="AG7hdCRYnN4K3vDM65YDqQ==",stale=true,algorithm=MD5
Content-Length: 0

<------------->
[2015-06-17 17:33:19] VERBOSE[1899] chan_sip.c: --- (8 headers 0 lines) ---
[2015-06-17 17:33:19] VERBOSE[1899] chan_sip.c: Responding to challenge, registration to domain/host name 10.72.0.9
[2015-06-17 17:33:19] VERBOSE[1899] chan_sip.c: REGISTER 12 headers, 0 lines
[2015-06-17 17:33:19] VERBOSE[1899] chan_sip.c: Reliably Transmitting (NAT) to 10.72.0.9:5060:
REGISTER sip:ims.beltel.by SIP/2.0
Via: SIP/2.0/UDP 10.223.222.135:5061;branch=z9hG4bK00a0aba7;rport
Max-Forwards: 70
From: <sip:+375232232302@ims.beltel.by>;tag=as1257a698
To: <sip:+375232232302@ims.beltel.by>
Call-ID: 56caf4840180e6ba0c425bca32fc2059@[::1]
CSeq: 107 REGISTER
Supported: replaces, timer
User-Agent: FPBX-12.0.68(13.3.2)
Authorization: Digest username="+375232232302@ims.beltel.by", realm="ims.beltel.by", algorithm=MD5, uri="sip:ims.beltel.by", nonce="AG7hdCRYnN4K3vDM65YDqQ==", response="5e8764e69bc3d5d6f92b0d0cdf30b257"
Expires: 120
Contact: <sip:+375232232302@10.223.222.135:5061>
Content-Length: 0


---
[2015-06-17 17:33:19] VERBOSE[1899] chan_sip.c:
<--- SIP read from UDP:10.72.0.9:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.223.222.135:5061;branch=z9hG4bK31ffeaca;received=10.223.222.135;rport=5061
Call-ID: 27d2b51d29e25bd00b3408fc1ce6c695@[::1]
From: <sip:+375232232305@ims.beltel.by>;tag=as56a24fec
To: <sip:+375232232305@ims.beltel.by>;tag=tq9oqotr
CSeq: 107 REGISTER
Service-Route: <sip:orig@scscf01.ims.beltel.by;lr;Dpt=7c04_d7512246;ca=8f26>
P-Associated-URI: <sip:+375232232305@ims.beltel.by>,<sip:+375232232305@ims.beltel.by;user=phone>
Path: <sip:term@pcscf01.ims.beltel.by;lr;ssn;TYPE=V4;IP=10.65.0.26;PORT=59858;Dpt=7c02_86>
Accept-Resource-Priority: wps.4
Contact: <sip:+375232232305@10.223.222.135:5061>;q=1;expires=120
Content-Length: 0

<------------->
[2015-06-17 17:33:19] VERBOSE[1899] chan_sip.c: --- (12 headers 0 lines) ---
[2015-06-17 17:33:19] NOTICE[1899] chan_sip.c: Outbound Registration: Expiry for 10.72.0.9 is 120 sec (Scheduling reregistration in 105 s)
[2015-06-17 17:33:19] VERBOSE[1899] chan_sip.c: Really destroying SIP dialog '27d2b51d29e25bd00b3408fc1ce6c695@[::1]' Method: REGISTER
[2015-06-17 17:33:19] NOTICE[1899] chan_sip.c: -- Re-registration for +375232232304@10.72.0.9
[2015-06-17 17:33:19] VERBOSE[1899] chan_sip.c: REGISTER 12 headers, 0 lines
[2015-06-17 17:33:19] VERBOSE[1899] chan_sip.c: Reliably Transmitting (NAT) to 10.72.0.9:5060:
REGISTER sip:ims.beltel.by SIP/2.0
Via: SIP/2.0/UDP 10.223.222.135:5061;branch=z9hG4bK61f9e14a;rport
Max-Forwards: 70
From: <sip:+375232232304@ims.beltel.by>;tag=as2185dcf4
To: <sip:+375232232304@ims.beltel.by>
Call-ID: 5196a3b37d7d8fe11b587663737e4a21@[::1]
CSeq: 106 REGISTER
Supported: replaces, timer
User-Agent: FPBX-12.0.68(13.3.2)
Authorization: Digest username="+375232232304@ims.beltel.by", realm="ims.beltel.by", algorithm=MD5, uri="sip:ims.beltel.by", nonce="mc+iU2GsjlAHqdQSky/gLw==", response="7dc8fac95756b0bb35069be1202060a9"
Expires: 120
Contact: <sip:+375232232304@10.223.222.135:5061>
Content-Length: 0


---
[2015-06-17 17:33:19] VERBOSE[1899] chan_sip.c:
<--- SIP read from UDP:10.72.0.9:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.223.222.135:5061;branch=z9hG4bK00a0aba7;received=10.223.222.135;rport=5061
Call-ID: 56caf4840180e6ba0c425bca32fc2059@[::1]
From: <sip:+375232232302@ims.beltel.by>;tag=as1257a698
To: <sip:+375232232302@ims.beltel.by>;tag=0ii92wo4
CSeq: 107 REGISTER
Service-Route: <sip:orig@scscf01.ims.beltel.by;lr;Dpt=7b94_b3576246;ca=98af>
P-Associated-URI: <sip:+375232232302@ims.beltel.by>,<sip:+375232232302@ims.beltel.by;user=phone>
Path: <sip:term@pcscf01.ims.beltel.by;lr;ssn;TYPE=V4;IP=10.65.0.25;PORT=50919;Dpt=7b92_86>
Accept-Resource-Priority: wps.4
Contact: <sip:+375232232302@10.223.222.135:5061>;q=1;expires=120
Content-Length: 0

<------------->
[2015-06-17 17:33:19] VERBOSE[1899] chan_sip.c: --- (12 headers 0 lines) ---
[2015-06-17 17:33:19] NOTICE[1899] chan_sip.c: Outbound Registration: Expiry for 10.72.0.9 is 120 sec (Scheduling reregistration in 105 s)
[2015-06-17 17:33:19] VERBOSE[1899] chan_sip.c: Really destroying SIP dialog '56caf4840180e6ba0c425bca32fc2059@[::1]' Method: REGISTER
[2015-06-17 17:33:19] VERBOSE[1899] chan_sip.c:
<--- SIP read from UDP:10.72.0.9:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.223.222.135:5061;branch=z9hG4bK61f9e14a;received=10.223.222.135;rport=5061
Call-ID: 5196a3b37d7d8fe11b587663737e4a21@[::1]
From: <sip:+375232232304@ims.beltel.by>;tag=as2185dcf4
To: <sip:+375232232304@ims.beltel.by>;tag=hgilokrr
CSeq: 106 REGISTER
WWW-Authenticate: Digest realm="ims.beltel.by",nonce="R2phMPKhgg6rTNEjcmLMYQ==",stale=true,algorithm=MD5
Content-Length: 0

<------------->
[2015-06-17 17:33:19] VERBOSE[1899] chan_sip.c: --- (8 headers 0 lines) ---
[2015-06-17 17:33:19] VERBOSE[1899] chan_sip.c: Responding to challenge, registration to domain/host name 10.72.0.9
[2015-06-17 17:33:19] VERBOSE[1899] chan_sip.c: REGISTER 12 headers, 0 lines
[2015-06-17 17:33:19] VERBOSE[1899] chan_sip.c: Reliably Transmitting (NAT) to 10.72.0.9:5060:
REGISTER sip:ims.beltel.by SIP/2.0
Via: SIP/2.0/UDP 10.223.222.135:5061;branch=z9hG4bK3fccf66a;rport
Max-Forwards: 70
From: <sip:+375232232304@ims.beltel.by>;tag=as2185dcf4
To: <sip:+375232232304@ims.beltel.by>
Call-ID: 5196a3b37d7d8fe11b587663737e4a21@[::1]
CSeq: 107 REGISTER
Supported: replaces, timer
User-Agent: FPBX-12.0.68(13.3.2)
Authorization: Digest username="+375232232304@ims.beltel.by", realm="ims.beltel.by", algorithm=MD5, uri="sip:ims.beltel.by", nonce="R2phMPKhgg6rTNEjcmLMYQ==", response="c61360ef46c33d8df193bd79b6eb334e"
Expires: 120
Contact: <sip:+375232232304@10.223.222.135:5061>
Content-Length: 0


---
[2015-06-17 17:33:19] VERBOSE[1899] chan_sip.c:
<--- SIP read from UDP:10.72.0.9:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.223.222.135:5061;branch=z9hG4bK3fccf66a;received=10.223.222.135;rport=5061
Call-ID: 5196a3b37d7d8fe11b587663737e4a21@[::1]
From: <sip:+375232232304@ims.beltel.by>;tag=as2185dcf4
To: <sip:+375232232304@ims.beltel.by>;tag=viijjvrx
CSeq: 107 REGISTER
Service-Route: <sip:orig@scscf01.ims.beltel.by;lr;Dpt=7be4_20534246;ca=869f>
P-Associated-URI: <sip:+375232232304@ims.beltel.by>,<sip:+375232232304@ims.beltel.by;user=phone>
Path: <sip:term@pcscf01.ims.beltel.by;lr;ssn;TYPE=V4;IP=10.65.0.26;PORT=41310;Dpt=7be2_86>
Accept-Resource-Priority: wps.4
Contact: <sip:+375232232304@10.223.222.135:5061>;q=1;expires=120
Content-Length: 0

<------------->
[2015-06-17 17:33:19] VERBOSE[1899] chan_sip.c: --- (12 headers 0 lines) ---
[2015-06-17 17:33:19] NOTICE[1899] chan_sip.c: Outbound Registration: Expiry for 10.72.0.9 is 120 sec (Scheduling reregistration in 105 s)
[2015-06-17 17:33:19] VERBOSE[1899] chan_sip.c: Really destroying SIP dialog '5196a3b37d7d8fe11b587663737e4a21@[::1]' Method: REGISTER
[2015-06-17 17:33:38] NOTICE[1899] chan_sip.c: Disconnecting call 'SIP/+375232232302-00000014' for lack of RTP activity in 31 seconds
[2015-06-17 17:33:38] VERBOSE[56604][C-0000000a] chan_sip.c: Scheduling destruction of SIP dialog 'asbc1434562393558226028285@10.1.0.97' in 6400 ms (Method: INVITE)
[2015-06-17 17:33:38] VERBOSE[56604][C-0000000a] chan_sip.c: Reliably Transmitting (NAT) to 10.72.0.9:5060:
BYE sip:10.72.0.9:5060;did=1412039382258 SIP/2.0
Via: SIP/2.0/UDP 10.223.222.135:5061;branch=z9hG4bK2ad10373;rport
Max-Forwards: 70
From: <sip:+375232232302@10.72.0.9:5060;transport=udp;user=phone>;tag=as429999e8
To: <tel:8441234567;phone-context=+375>;tag=sbc0405339597509
Call-ID: asbc1434562393558226028285@10.1.0.97
CSeq: 102 BYE
User-Agent: FPBX-12.0.68(13.3.2)
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
[2015-06-17 17:33:38] VERBOSE[1899] chan_sip.c: Retransmitting #1 (NAT) to 10.72.0.9:5060:
BYE sip:10.72.0.9:5060;did=1412039382258 SIP/2.0
Via: SIP/2.0/UDP 10.223.222.135:5061;branch=z9hG4bK2ad10373;rport
Max-Forwards: 70
From: <sip:+375232232302@10.72.0.9:5060;transport=udp;user=phone>;tag=as429999e8
To: <tel:8441234567;phone-context=+375>;tag=sbc0405339597509
Call-ID: asbc1434562393558226028285@10.1.0.97
CSeq: 102 BYE
User-Agent: FPBX-12.0.68(13.3.2)
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
[2015-06-17 17:33:38] VERBOSE[1899] chan_sip.c:
<--- SIP read from UDP:10.72.0.9:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.223.222.135:5061;branch=z9hG4bK2ad10373;rport=5061
Call-ID: asbc1434562393558226028285@10.1.0.97
From: <sip:+375232232302@10.72.0.9:5060;transport=udp;user=phone>;tag=as429999e8
To: <tel:8441234567;phone-context=+375>;tag=sbc0405339597509
CSeq: 102 BYE
Content-Length: 0

<------------->
[2015-06-17 17:33:38] VERBOSE[1899] chan_sip.c: --- (7 headers 0 lines) ---
[2015-06-17 17:33:38] VERBOSE[1899][C-0000000a] chan_sip.c: SIP Response message for INCOMING dialog BYE arrived
[2015-06-17 17:33:38] VERBOSE[1899] chan_sip.c: Really destroying SIP dialog 'asbc1434562393558226028285@10.1.0.97' Method: INVITE
[2015-06-17 17:33:38] VERBOSE[1899] chan_sip.c:
<--- SIP read from UDP:10.72.0.9:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.223.222.135:5061;branch=z9hG4bK2ad10373;rport=5061
Call-ID: asbc1434562393558226028285@10.1.0.97
From: <sip:+375232232302@10.72.0.9:5060;transport=udp;user=phone>;tag=as429999e8
To: <tel:8441234567;phone-context=+375>;tag=sbc0405339597509
CSeq: 102 BYE
Content-Length: 0

<------------->
[2015-06-17 17:33:38] VERBOSE[1899] chan_sip.c: --- (7 headers 0 lines) ---
[2015-06-17 17:33:48] VERBOSE[1899] chan_sip.c: Reliably Transmitting (NAT) to 10.72.0.9:5060:
OPTIONS sip:10.72.0.9 SIP/2.0
Via: SIP/2.0/UDP 10.223.222.135:5061;branch=z9hG4bK7607028c;rport
Max-Forwards: 70
From: "Unknown" <sip:+375232232305@10.223.222.135:5061>;tag=as38a273e1
To: <sip:10.72.0.9>
Contact: <sip:+375232232305@10.223.222.135:5061>
Call-ID: 02636cca29c4b32f5e79720d459a03a7@10.223.222.135:5061
CSeq: 102 OPTIONS
User-Agent: FPBX-12.0.68(13.3.2)
Date: Wed, 17 Jun 2015 14:33:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---
[2015-06-17 17:33:48] VERBOSE[1899] chan_sip.c:
<--- SIP read from UDP:10.72.0.9:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.223.222.135:5061;branch=z9hG4bK7607028c;rport
Call-ID: 02636cca29c4b32f5e79720d459a03a7@10.223.222.135:5061
From: "Unknown"<sip:+375232232305@10.223.222.135:5061>;tag=as38a273e1
To: <sip:10.72.0.9>;tag=sbc04051dndd8e2
CSeq: 102 OPTIONS
Content-Length: 0

<------------->
[2015-06-17 17:33:48] VERBOSE[1899] chan_sip.c: --- (7 headers 0 lines) ---
[2015-06-17 17:33:48] VERBOSE[1899] chan_sip.c: Really destroying SIP dialog '02636cca29c4b32f5e79720d459a03a7@10.223.222.135:5061' Method: OPTIONS
[2015-06-17 17:33:48] VERBOSE[1899] chan_sip.c: Reliably Transmitting (NAT) to 10.72.0.9:5060:
OPTIONS sip:10.72.0.9 SIP/2.0
Via: SIP/2.0/UDP 10.223.222.135:5061;branch=z9hG4bK5d87c8b2;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@10.223.222.135:5061>;tag=as28a1c89d
To: <sip:10.72.0.9>
Contact: <sip:Unknown@10.223.222.135:5061>
Call-ID: 6bd6853b7dcce3750a55482218788397@10.223.222.135:5061
CSeq: 102 OPTIONS
User-Agent: FPBX-12.0.68(13.3.2)
Date: Wed, 17 Jun 2015 14:33:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---
[2015-06-17 17:33:48] VERBOSE[1899] chan_sip.c:
<--- SIP read from UDP:10.72.0.9:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.223.222.135:5061;branch=z9hG4bK5d87c8b2;rport
Call-ID: 6bd6853b7dcce3750a55482218788397@10.223.222.135:5061
From: "Unknown"<sip:Unknown@10.223.222.135:5061>;tag=as28a1c89d
To: <sip:10.72.0.9>;tag=sbc04057k3dkdl0
CSeq: 102 OPTIONS
Content-Length: 0

<------------->
[2015-06-17 17:33:48] VERBOSE[1899] chan_sip.c: --- (7 headers 0 lines) ---
[2015-06-17 17:33:48] VERBOSE[1899] chan_sip.c: Really destroying SIP dialog '6bd6853b7dcce3750a55482218788397@10.223.222.135:5061' Method: OPTIONS

rtp set debug on

PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:

Код: Выделить всё

 [2015-06-17 17:43:30] WARNING[56973][C-0000000c] func_channel.c: Unknown or unavailable item requested: 'reversecharge'
[2015-06-17 17:43:32] VERBOSE[56975][C-0000000c] res_rtp_asterisk.c: Got RTP packet from 192.168.1.10:16304 (type 08, seq 007197, ts 763470928, len 000160)
[2015-06-17 17:43:32] VERBOSE[56973][C-0000000c] res_rtp_asterisk.c: Sent RTP packet to 10.72.0.10:35470 (type 08, seq 054691, ts 763470928, len 000160)
[2015-06-17 17:43:32] VERBOSE[56975][C-0000000c] res_rtp_asterisk.c: Got RTP packet from 192.168.1.10:16304 (type 08, seq 007198, ts 763471088, len 000160)
[2015-06-17 17:43:32] VERBOSE[56973][C-0000000c] res_rtp_asterisk.c: Sent RTP packet to 10.72.0.10:35470 (type 08, seq 054692, ts 763471088, len 000160)
[2015-06-17 17:43:32] VERBOSE[56975][C-0000000c] res_rtp_asterisk.c: Got RTP packet from 192.168.1.10:16304 (type 08, seq 007199, ts 763471248, len 000160)
[2015-06-17 17:43:32] VERBOSE[56973][C-0000000c] res_rtp_asterisk.c: Sent RTP packet to 10.72.0.10:35470 (type 08, seq 054693, ts 763471248, len 000160)
[2015-06-17 17:43:32] VERBOSE[56975][C-0000000c] res_rtp_asterisk.c: Got RTP packet from 192.168.1.10:16304 (type 08, seq 007200, ts 763471408, len 000160)
[2015-06-17 17:43:32] VERBOSE[56973][C-0000000c] res_rtp_asterisk.c: Sent RTP packet to 10.72.0.10:35470 (type 08, seq 054694, ts 763471408, len 000160)
[2015-06-17 17:43:32] VERBOSE[56975][C-0000000c] res_rtp_asterisk.c: Got RTP packet from 192.168.1.10:16304 (type 08, seq 007201, ts 763471568, len 000160)
[2015-06-17 17:43:32] VERBOSE[56973][C-0000000c] res_rtp_asterisk.c: Sent RTP packet to 10.72.0.10:35470 (type 08, seq 054695, ts 763471568, len 000160)
[2015-06-17 17:43:32] VERBOSE[56975][C-0000000c] res_rtp_asterisk.c: Got RTP packet from 192.168.1.10:16304 (type 08, seq 007202, ts 763471728, len 000160)
[2015-06-17 17:43:32] VERBOSE[56973][C-0000000c] res_rtp_asterisk.c: Sent RTP packet to 10.72.0.10:35470 (type 08, seq 054696, ts 763471728, len 000160)
[2015-06-17 17:43:32] VERBOSE[56975][C-0000000c] res_rtp_asterisk.c: Got RTP packet from 192.168.1.10:16304 (type 08, seq 007203, ts 763471888, len 000160)
[2015-06-17 17:43:32] VERBOSE[56973][C-0000000c] res_rtp_asterisk.c: Sent RTP packet to 10.72.0.10:35470 (type 08, seq 054697, ts 763471888, len 000160)
[2015-06-17 17:43:32] VERBOSE[56975][C-0000000c] res_rtp_asterisk.c: Got RTP packet from 192.168.1.10:16304 (type 08, seq 007204, ts 763472048, len 000160)
[2015-06-17 17:43:32] VERBOSE[56973][C-0000000c] res_rtp_asterisk.c: Sent RTP packet to 10.72.0.10:35470 (type 08, seq 054698, ts 763472048, len 000160)
[2015-06-17 17:43:32] VERBOSE[56975][C-0000000c] res_rtp_asterisk.c: Got RTP packet from 192.168.1.10:16304 (type 08, seq 007205, ts 763472208, len 
.................
[2015-06-17 17:43:32] VERBOSE[56973][C-0000000c] res_rtp_asterisk.c: Sent RTP packet to 10.72.0.10:35470 (type 08, seq 054712, ts 763474288, len 000160)
[2015-06-17 17:43:32] VERBOSE[56975][C-0000000c] res_rtp_asterisk.c: Got RTP packet from 192.168.1.10:16304 (type 08, seq 007219, ts 763474448, len 
[2015-06-17 17:43:33] VERBOSE[56973][C-0000000c] res_rtp_asterisk.c: Sent RTP packet to 10.72.0.10:35470 (type 08, seq 054737, ts 763478288, len 000160)
.............
[2015-06-17 17:44:03] NOTICE[1899] chan_sip.c: Disconnecting call 'SIP/+375232232302-00000018' for lack of RTP activity in 31 seconds

Насколько я понимаю не приходят ко мне RTP пакеты, но не могу понять почему... Может кто сталкивался с такой проблемой? Спасибо за внимание

С такой проблемой сталкивались и сталкиваются умельцы, вчера поставившие себе Астериск на виртуалку (чай не боги горшки обжигают!), несчастные клиенты beltel.by.
Правильный посыл в вашем случае - "обращайтесь в службу техподдержки Белтел" ибо это ваш оператор, вы ему платите, пускай раскуривает. Они освоили 2-3 модели шлюзиков, типа Hybertone, и всем их ставят, и на этом что - всё что ли? Их высокопрофильную работу бесплатно делать не будут, разве что уж совсем от нечего делать. Пусть ставят тестовую версию Астериска себе и пишут рекомендации своим клиентам по подключению.
http://forum.asterisk.ru/viewtopic.php?f=3&t=4104

У Вас, уважаемый ТС, катастрофически не хватает а) теоретических знаний по предмету, б) практической чуйки и в) терпения просматривать ваши же анализы мочи и кала - логи. В которых всё, в общем то, видно. Что сигнальный провайдерский ИП 10.72.0.9:5060, а в SDP почему-то присутствует 10.72.0.10 (с чего бы ВДРУГ?), и для него Вы уверено указали nat=yes (кто это подсказал?), ну и Астериск лупит туда - раз
Transmitting (NAT) to 10.72.0.9:5060
потом повторяет
chan_sip.c: Retransmitting #1 (NAT) to 10.72.0.9:5060: - ответа то нету! И ещё
chan_sip.c: Retransmitting #2 (NAT) to 10.72.0.9:5060:
.....
chan_sip.c: Retransmitting #5 (NAT) to 10.72.0.9:5060: вот так несколько раз безответно.
Почему Вы это не разбираете самостоятельно?
Я Вас сразу направлю в платный суппорт, вам так проще будет, хорошо? Эту штангу Вам не поднять. Вопросов слишком много.

Про NAT, кончено, уместное замечание.

но столько пафоса ИМХО не стоило вываливать.

У меня аналогичная проблема решилась отключением услуги "Музыкальный марафон", по дефолту включаемой любимым БТК.

Везёт вам!
С такой удачей надо в казино ходить, а не транки с Белтелекомом настраивать.
За пафос извиняйте.

вчера поставившие себе Астериск

Дело не в астере - дело в понимании маршрутизация-(хренов роутер) и т.д.Увы, ded но ... нежелании ИСКАТЬ траблы и ВИДЕТЬ. Бел-Эстон-Укр - не важно, везде берут НОВЫХ подаванов, которые киску на курсах "учили" и понимания просто нет. Ну можно прикрыться "вчера поставил" - никто не пробовал ЭТУ фразу сказать ЖЕНЩИНЕ в первую ... ночь

А, вообще-то, всех с праздником!!!

ded писал(а): С такой удачей надо в казино ходить, а не транки с Белтелекомом настраивать.

никакой удачи. Просто транки бтк без гудка работали, а транки с гудком - нет. Через один и тот же сервер, с одними и теми же параметрами. Снифали, логи вычитывали - тупо трафик с БТК не идет.

Гудок как-то связан с услугой "Музыкальный марафон"?

ну, имелось в виду не гудок, а "музыкальный марафон".

Включаем "марафон" обратно, перестают работать.

ps: у мтса это называется прсото «Персональный ГУДОК»

Про гудки.
В традиционной телефонии гудки гудит станция к которой подключен вызываемый абонент. Т.е. получив вызов станция смотрит БД (ограничения, услуги там всякие) и абоненту посылает вызов, а в обратную сторону КПВ. Если это мобильный мутатор, тут сценарии делятся на 2 штуки-абонет припейд или котракт. И далее по сценарию для своего типа, если есть какие-нить услуги, то срабатывает триггер, который запускает обработку события, типа вместо гудков в обратную сторону проиграть музыку. Управляет всем этим тоже компутер, использующий свой протокол.
Во всей этой лабуде есть одно скрытое от пользователей явление INTERNETWORKING между разными системами и протоколами.
Вот он родимый и вносит всякие каверзы в функционирование. Он родимый определяется стандартами и производители и програмеры должны эти стандарты соблюдать.

ded писал(а):С такой проблемой сталкивались и сталкиваются умельцы, вчера поставившие себе Астериск на виртуалку (чай не боги горшки обжигают!)

у меня астер как раз на виртуалке .
уже не раз пожалел, что не на железе (периодически возникает эхо на аналоговых каналах или вообще отваливается аналоговый шлюз)

asterisk.ru

Входящие звонки - тишина

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Re: Входящие звонки - тишина

Re: Входящие звонки - тишина

Re: Входящие звонки - тишина

Re: Входящие звонки - тишина

Re: Входящие звонки - тишина

Re: Входящие звонки - тишина

Re: Входящие звонки - тишина

Re: Входящие звонки - тишина