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roher
СообщениеДобавлено: Пн май 12, 2014 11:38 
Не в сети

Зарегистрирован: Чт апр 03, 2014 11:42
Сообщений: 4
Добрый день!
У меня дома настроен Asterisk 11.8.1 на Debian-сервере в связке с D-Link DVG-7111S для выхода в город.
И есть очень большая проблема: при любом звонке через FXO-порт D-Link'а Asterisk вообще перестаёт обрабатывать звонки и заваливает лог сообщениями вида:
[Apr 19 08:39:52] WARNING[3400] chan_sip.c: Autodestruct on dialog 'BD21-1143-4100062266E8A23356586-022@SipHost' with owner SIP/1000-00000002 in place (Method: BYE). Rescheduling destruction for 10000 ms
[Apr 19 08:39:52] WARNING[3400] chan_sip.c: Autodestruct on dialog '20853266711a687119364bfb485b4a96@172.16.0.10:5060' with owner SIP/2000-00000003 in place (Method: BYE). Rescheduling destruction for 10000 ms
Помогает только ребут Астериска, а после следующего городского звонка (входящего или исходящего) история повторяется.
Расскажу подробнее о сетапе (все ip-адреса, порты и номера отличаются от реальных в целях безопасности):
IP-адрес Астериска 172.16.0.10, сидит на порту 5060, IP-адрес DLink'а 172.16.0.20
В Астериске заведено два юзера для FXO и FXS портов железки: 1000 и 2000. Т.е. 1000 отвечает за внутреннюю трубку, подключённую к DLink'у, а 2000 - за внешнюю линию.
Соответственно эти номера прописаны в настойках DLink'а.

Предположим, мы звоним с внутреннего номера (1000) на городской номер 100 (в Москве МГТС это бесплатный робот точного времени).
Соответственно в диалплане эта строка (пусть в контексте context) выглядит так: Dial(SIP/2000/100)

Вот SIP-лог звонка:
Скрытый текст: показать
[Apr 19 08:39:13] VERBOSE[3400] chan_sip.c:
<--- SIP read from UDP:172.16.0.20:5060 --->
INVITE sip:100@172.16.0.10:5060;user=phone SIP/2.0
Allow:INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,REFER,SUBSCRIBE,NOTIFY,UPDATE
Via: SIP/2.0/UDP 172.16.0.20:5060;branch=z9hG4bKf73a3d86b3a377ab
From: "1000" <sip:1000@172.16.0.10>;tag=7ec3f92d-16226
To: <sip:100@172.16.0.10:5060;user=phone>
Call-ID: BD21-1143-4100062266E8A23356586-022@SipHost
CSeq:1141 INVITE
Contact: <sip:1000@172.16.0.20:5060>
Expires:90
Max-Forwards:70
Min-SE:90
Session-Expires: 1800
Supported: replaces,timer
User-Agent:dlink
Content-Type: application/sdp
Content-Length: 333

v=0
o=1000 2123421310 2123421310 IN IP4 172.16.0.20
s=Session SDP
c=IN IP4 172.16.0.20
t=0 0
m=audio 9000 RTP/AVP 8 4 18 2 0 101
a=rtpmap:8 PCMA/8000/1
a=rtpmap:4 G723/8000/1
a=rtpmap:18 G729/8000/1
a=fmtp:18 annexb=no
a=rtpmap:2 G726-32/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-16
<------------->
[Apr 19 08:39:13] VERBOSE[3400] chan_sip.c: --- (16 headers 14 lines) ---
[Apr 19 08:39:13] VERBOSE[3400] chan_sip.c: Sending to 172.16.0.20:5060 (no NAT)
[Apr 19 08:39:13] VERBOSE[3400][C-00000001] chan_sip.c: Sending to 172.16.0.20:5060 (no NAT)
[Apr 19 08:39:13] VERBOSE[3400][C-00000001] chan_sip.c: Using INVITE request as basis request - BD21-1143-4100062266E8A23356586-022@SipHost
[Apr 19 08:39:13] VERBOSE[3400][C-00000001] chan_sip.c: Found peer '1000' for '1000' from 172.16.0.20:5060
[Apr 19 08:39:13] VERBOSE[3400][C-00000001] chan_sip.c:
<--- Reliably Transmitting (no NAT) to 172.16.0.20:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.16.0.20:5060;branch=z9hG4bKf73a3d86b3a377ab;received=172.16.0.20
From: "1000" <sip:1000@172.16.0.10>;tag=7ec3f92d-16226
To: <sip:100@172.16.0.10:5060;user=phone>;tag=as174558a4
Call-ID: BD21-1143-4100062266E8A23356586-022@SipHost
CSeq: 1141 INVITE
Server: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1c734ae6"
Content-Length: 0


<------------>
[Apr 19 08:39:13] VERBOSE[3400][C-00000001] chan_sip.c: Scheduling destruction of SIP dialog 'BD21-1143-4100062266E8A23356586-022@SipHost' in 6400 ms (Method: INVITE)
[Apr 19 08:39:13] VERBOSE[3400] chan_sip.c:
<--- SIP read from UDP:172.16.0.20:5060 --->
ACK sip:100@172.16.0.10:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 172.16.0.20:5060;branch=z9hG4bKf73a3d86b3a377ab
From: "1000" <sip:1000@172.16.0.10>;tag=7ec3f92d-16226
To: <sip:100@172.16.0.10:5060;user=phone>;tag=as174558a4
Call-ID: BD21-1143-4100062266E8A23356586-022@SipHost
CSeq:1141 ACK
Max-Forwards:70
Content-Length: 0

<------------->
[Apr 19 08:39:13] VERBOSE[3400] chan_sip.c: --- (8 headers 0 lines) ---
[Apr 19 08:39:13] VERBOSE[3400] chan_sip.c:
<--- SIP read from UDP:172.16.0.20:5060 --->
INVITE sip:100@172.16.0.10:5060;user=phone SIP/2.0
Allow:INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,REFER,SUBSCRIBE,NOTIFY,UPDATE
Via: SIP/2.0/UDP 172.16.0.20:5060;branch=z9hG4bK2330025bbd617c5d
From: "1000" <sip:1000@172.16.0.10>;tag=7ec3f92d-16226
To: <sip:100@172.16.0.10:5060;user=phone>
Call-ID: BD21-1143-4100062266E8A23356586-022@SipHost
CSeq:1142 INVITE
Contact: <sip:1000@172.16.0.20:5060>
Expires:90
Max-Forwards:70
Min-SE:90
Authorization:Digest username="1000",realm="asterisk",nonce="1c734ae6",uri="sip:100@172.16.0.10:5060;user=phone",response="3b02d882988d13ba6558854920fffcf6",algorithm=MD5
Session-Expires: 1800
Supported: replaces,timer
User-Agent:dlink
Content-Type: application/sdp
Content-Length: 333

v=0
o=1000 2123421310 2123421310 IN IP4 172.16.0.20
s=Session SDP
c=IN IP4 172.16.0.20
t=0 0
m=audio 9000 RTP/AVP 8 4 18 2 0 101
a=rtpmap:8 PCMA/8000/1
a=rtpmap:4 G723/8000/1
a=rtpmap:18 G729/8000/1
a=fmtp:18 annexb=no
a=rtpmap:2 G726-32/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-16
<------------->
[Apr 19 08:39:13] VERBOSE[3400] chan_sip.c: --- (17 headers 14 lines) ---
[Apr 19 08:39:13] VERBOSE[3400][C-00000001] chan_sip.c: Sending to 172.16.0.20:5060 (no NAT)
[Apr 19 08:39:13] VERBOSE[3400][C-00000001] chan_sip.c: Using INVITE request as basis request - BD21-1143-4100062266E8A23356586-022@SipHost
[Apr 19 08:39:13] VERBOSE[3400][C-00000001] chan_sip.c: Found peer '1000' for '1000' from 172.16.0.20:5060
[Apr 19 08:39:13] VERBOSE[3400][C-00000001] netsock2.c: == Using SIP RTP CoS mark 5
[Apr 19 08:39:13] VERBOSE[3400][C-00000001] chan_sip.c: Found RTP audio format 8
[Apr 19 08:39:13] VERBOSE[3400][C-00000001] chan_sip.c: Found RTP audio format 4
[Apr 19 08:39:13] VERBOSE[3400][C-00000001] chan_sip.c: Found RTP audio format 18
[Apr 19 08:39:13] VERBOSE[3400][C-00000001] chan_sip.c: Found RTP audio format 2
[Apr 19 08:39:13] VERBOSE[3400][C-00000001] chan_sip.c: Found RTP audio format 0
[Apr 19 08:39:13] VERBOSE[3400][C-00000001] chan_sip.c: Found RTP audio format 101
[Apr 19 08:39:13] VERBOSE[3400][C-00000001] chan_sip.c: Found audio description format PCMA for ID 8
[Apr 19 08:39:13] VERBOSE[3400][C-00000001] chan_sip.c: Found audio description format G723 for ID 4
[Apr 19 08:39:13] VERBOSE[3400][C-00000001] chan_sip.c: Found audio description format G729 for ID 18
[Apr 19 08:39:13] VERBOSE[3400][C-00000001] chan_sip.c: Found audio description format G726-32 for ID 2
[Apr 19 08:39:13] VERBOSE[3400][C-00000001] chan_sip.c: Found audio description format PCMU for ID 0
[Apr 19 08:39:13] VERBOSE[3400][C-00000001] chan_sip.c: Found audio description format telephone-event for ID 101
[Apr 19 08:39:13] VERBOSE[3400][C-00000001] chan_sip.c: Capabilities: us - (gsm|ulaw|alaw), peer - audio=(g723|ulaw|alaw|g726|g729)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
[Apr 19 08:39:13] VERBOSE[3400][C-00000001] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[Apr 19 08:39:13] VERBOSE[3400][C-00000001] chan_sip.c: Peer audio RTP is at port 172.16.0.20:9000
[Apr 19 08:39:13] VERBOSE[3400][C-00000001] chan_sip.c: Looking for 100 in context (domain 172.16.0.10)
[Apr 19 08:39:13] VERBOSE[3400][C-00000001] chan_sip.c: list_route: hop: <sip:1000@172.16.0.20:5060>
[Apr 19 08:39:13] VERBOSE[3400][C-00000001] chan_sip.c:
<--- Transmitting (no NAT) to 172.16.0.20:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.16.0.20:5060;branch=z9hG4bK2330025bbd617c5d;received=172.16.0.20
From: "1000" <sip:1000@172.16.0.10>;tag=7ec3f92d-16226
To: <sip:100@172.16.0.10:5060;user=phone>
Call-ID: BD21-1143-4100062266E8A23356586-022@SipHost
CSeq: 1142 INVITE
Server: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:100@172.16.0.10:5060>
Content-Length: 0


<------------>
[Apr 19 08:39:13] VERBOSE[14401][C-00000001] pbx.c: -- Executing [100@context:1] Dial("SIP/1000-00000002", "SIP/2000/100,120,mXWKT") in new stack
[Apr 19 08:39:13] VERBOSE[14401][C-00000001] netsock2.c: == Using SIP RTP CoS mark 5
[Apr 19 08:39:13] VERBOSE[14401][C-00000001] chan_sip.c: Audio is at 13716
[Apr 19 08:39:13] VERBOSE[14401][C-00000001] chan_sip.c: Adding codec 100004 (alaw) to SDP
[Apr 19 08:39:13] VERBOSE[14401][C-00000001] chan_sip.c: Adding codec 100003 (ulaw) to SDP
[Apr 19 08:39:13] VERBOSE[14401][C-00000001] chan_sip.c: Adding codec 100002 (gsm) to SDP
[Apr 19 08:39:13] VERBOSE[14401][C-00000001] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Apr 19 08:39:13] VERBOSE[14401][C-00000001] chan_sip.c: Reliably Transmitting (no NAT) to 172.16.0.20:5060:
INVITE sip:100@172.16.0.20:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.0.10:5060;branch=z9hG4bK16e17a80
Max-Forwards: 70
From: "PSTN Phone" <sip:1000@172.16.0.10:5060>;tag=as145b939c
To: <sip:100@172.16.0.20:5060>
Contact: <sip:1000@172.16.0.10:5060>
Call-ID: 20853266711a687119364bfb485b4a96@172.16.0.10:5060
CSeq: 102 INVITE
User-Agent: Asterisk
Date: Sat, 19 Apr 2014 04:39:13 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 529

v=0
o=root 387132854 387132854 IN IP4 172.16.0.10
s=Asterisk PBX 11.8.1
c=IN IP4 172.16.0.10
t=0 0
m=audio 13716 RTP/AVP 8 0 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=ice-ufrag:2a32b5623206171467a6506743ae0475
a=ice-pwd:1a5be7d976a9c3a84e7dd3943a1b5150
a=candidate:Hac100062 1 UDP 2130706431 172.16.0.10 13716 typ host
a=candidate:Hac100062 2 UDP 2130706430 172.16.0.10 13717 typ host
a=sendrecv

---
[Apr 19 08:39:13] VERBOSE[14401][C-00000001] app_dial.c: -- Called SIP/2000/100
[Apr 19 08:39:13] VERBOSE[14401][C-00000001] res_musiconhold.c: -- Started music on hold, class 'default', on SIP/1000-00000002
[Apr 19 08:39:13] VERBOSE[14401][C-00000001] chan_sip.c: Audio is at 10282
[Apr 19 08:39:13] VERBOSE[14401][C-00000001] chan_sip.c: Adding codec 100004 (alaw) to SDP
[Apr 19 08:39:13] VERBOSE[14401][C-00000001] chan_sip.c: Adding codec 100003 (ulaw) to SDP
[Apr 19 08:39:13] VERBOSE[14401][C-00000001] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Apr 19 08:39:13] VERBOSE[14401][C-00000001] chan_sip.c:
<--- Transmitting (no NAT) to 172.16.0.20:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.16.0.20:5060;branch=z9hG4bK2330025bbd617c5d;received=172.16.0.20
From: "1000" <sip:1000@172.16.0.10>;tag=7ec3f92d-16226
To: <sip:100@172.16.0.10:5060;user=phone>;tag=as3b3d3d2c
Call-ID: BD21-1143-4100062266E8A23356586-022@SipHost
CSeq: 1142 INVITE
Server: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:100@172.16.0.10:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 506

v=0
o=root 887912993 887912993 IN IP4 172.16.0.10
s=Asterisk PBX 11.8.1
c=IN IP4 172.16.0.10
t=0 0
m=audio 10282 RTP/AVP 8 0 101
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=ice-ufrag:53afa21f5bc7d61b2ceaee3132cfb05f
a=ice-pwd:25f4c76f163a074a6fa7421231f75cd8
a=candidate:Hac100062 1 UDP 2130706431 172.16.0.10 10282 typ host
a=candidate:Hac100062 2 UDP 2130706430 172.16.0.10 10283 typ host
a=sendrecv

<------------>
[Apr 19 08:39:13] WARNING[14401][C-00000001] mp3/interface.c: Junk at the beginning of frame 49443303
[Apr 19 08:39:13] VERBOSE[3400] chan_sip.c:
<--- SIP read from UDP:172.16.0.20:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.16.0.10:5060;branch=z9hG4bK16e17a80
From: "PSTN Phone" <sip:1000@172.16.0.10:5060>;tag=as145b939c
To: <sip:100@172.16.0.20:5060>
Call-ID: 20853266711a687119364bfb485b4a96@172.16.0.10:5060
CSeq:102 INVITE
Content-Type: application/sdp
Content-Length: 0

<------------->
[Apr 19 08:39:13] VERBOSE[3400] chan_sip.c: --- (8 headers 0 lines) ---
[Apr 19 08:39:13] VERBOSE[14401][C-00000001] res_rtp_asterisk.c: > 0x964c468 -- Probation passed - setting RTP source address to 172.16.0.20:9000
[Apr 19 08:39:15] VERBOSE[3400] chan_sip.c:
<--- SIP read from UDP:172.16.0.20:5060 --->
SIP/2.0 183 Session in progress
Via: SIP/2.0/UDP 172.16.0.10:5060;branch=z9hG4bK16e17a80
From: "PSTN Phone" <sip:1000@172.16.0.10:5060>;tag=as145b939c
To: <sip:100@172.16.0.20:5060>;tag=33e5ff91-16228
Call-ID: 20853266711a687119364bfb485b4a96@172.16.0.10:5060
CSeq:102 INVITE
Contact: <sip:172.16.0.20:5060>
User-Agent:dlink
Content-Type: application/sdp
Content-Length: 217

v=0
o=2000 2123423630 2123423630 IN IP4 172.16.0.20
s=Session SDP
c=IN IP4 172.16.0.20
t=0 0
m=audio 9002 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-16
a=sendonly
<------------->
[Apr 19 08:39:15] VERBOSE[3400] chan_sip.c: --- (10 headers 10 lines) ---
[Apr 19 08:39:15] VERBOSE[3400][C-00000001] chan_sip.c: list_route: hop: <sip:172.16.0.20:5060>
[Apr 19 08:39:15] VERBOSE[3400][C-00000001] chan_sip.c: Found RTP audio format 8
[Apr 19 08:39:15] VERBOSE[3400][C-00000001] chan_sip.c: Found RTP audio format 101
[Apr 19 08:39:15] VERBOSE[3400][C-00000001] chan_sip.c: Found audio description format PCMA for ID 8
[Apr 19 08:39:15] VERBOSE[3400][C-00000001] chan_sip.c: Found audio description format telephone-event for ID 101
[Apr 19 08:39:15] VERBOSE[3400][C-00000001] chan_sip.c: Capabilities: us - (gsm|ulaw|alaw), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
[Apr 19 08:39:15] VERBOSE[3400][C-00000001] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[Apr 19 08:39:15] VERBOSE[3400][C-00000001] chan_sip.c: Peer audio RTP is at port 172.16.0.20:9002
[Apr 19 08:39:15] VERBOSE[14401][C-00000001] app_dial.c: -- Call on SIP/2000-00000003 placed on hold
[Apr 19 08:39:15] VERBOSE[14401][C-00000001] res_musiconhold.c: -- Stopped music on hold on SIP/1000-00000002
[Apr 19 08:39:15] VERBOSE[14401][C-00000001] res_musiconhold.c: -- Started music on hold, class 'default', on SIP/1000-00000002
[Apr 19 08:39:15] VERBOSE[14401][C-00000001] app_dial.c: -- SIP/2000-00000003 is making progress passing it to SIP/1000-00000002
[Apr 19 08:39:15] WARNING[14401][C-00000001] mp3/interface.c: Junk at the beginning of frame 49443303
[Apr 19 08:39:15] VERBOSE[14401][C-00000001] res_rtp_asterisk.c: > 0x94fe578 -- Probation passed - setting RTP source address to 172.16.0.20:9002
[Apr 19 08:39:23] VERBOSE[3400] chan_sip.c:
<--- SIP read from UDP:172.16.0.20:5060 --->
SIP/2.0 200 OK
Allow:INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,REFER,SUBSCRIBE,NOTIFY,UPDATE
Via: SIP/2.0/UDP 172.16.0.10:5060;branch=z9hG4bK16e17a80
From: "PSTN Phone" <sip:1000@172.16.0.10:5060>;tag=as145b939c
To: <sip:100@172.16.0.20:5060>;tag=33e5ff91-16228
Call-ID: 20853266711a687119364bfb485b4a96@172.16.0.10:5060
CSeq:102 INVITE
Contact: <sip:172.16.0.20:5060>
User-Agent:dlink
Content-Type: application/sdp
Content-Length: 205

v=0
o=2000 2123431110 2123431110 IN IP4 172.16.0.20
s=Session SDP
c=IN IP4 172.16.0.20
t=0 0
m=audio 9002 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-16
<------------->
[Apr 19 08:39:23] VERBOSE[3400] chan_sip.c: --- (11 headers 9 lines) ---
[Apr 19 08:39:23] VERBOSE[3400][C-00000001] chan_sip.c: Found RTP audio format 8
[Apr 19 08:39:23] VERBOSE[3400][C-00000001] chan_sip.c: Found RTP audio format 101
[Apr 19 08:39:23] VERBOSE[3400][C-00000001] chan_sip.c: Found audio description format PCMA for ID 8
[Apr 19 08:39:23] VERBOSE[3400][C-00000001] chan_sip.c: Found audio description format telephone-event for ID 101
[Apr 19 08:39:23] VERBOSE[3400][C-00000001] chan_sip.c: Capabilities: us - (gsm|ulaw|alaw), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
[Apr 19 08:39:23] VERBOSE[3400][C-00000001] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[Apr 19 08:39:23] VERBOSE[3400][C-00000001] chan_sip.c: Peer audio RTP is at port 172.16.0.20:9002
[Apr 19 08:39:23] VERBOSE[3400][C-00000001] chan_sip.c: list_route: hop: <sip:172.16.0.20:5060>
[Apr 19 08:39:23] VERBOSE[3400][C-00000001] chan_sip.c: set_destination: Parsing <sip:172.16.0.20:5060> for address/port to send to
[Apr 19 08:39:23] VERBOSE[3400][C-00000001] chan_sip.c: set_destination: set destination to 172.16.0.20:5060
[Apr 19 08:39:23] VERBOSE[3400][C-00000001] chan_sip.c: Transmitting (no NAT) to 172.16.0.20:5060:
ACK sip:172.16.0.20:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.0.10:5060;branch=z9hG4bK6a8cdf26
Max-Forwards: 70
From: "PSTN Phone" <sip:1000@172.16.0.10:5060>;tag=as145b939c
To: <sip:100@172.16.0.20:5060>;tag=33e5ff91-16228
Contact: <sip:1000@172.16.0.10:5060>
Call-ID: 20853266711a687119364bfb485b4a96@172.16.0.10:5060
CSeq: 102 ACK
User-Agent: Asterisk
Content-Length: 0


---
[Apr 19 08:39:23] VERBOSE[14401][C-00000001] app_dial.c: -- Call on SIP/2000-00000003 left from hold
[Apr 19 08:39:23] VERBOSE[14401][C-00000001] res_musiconhold.c: -- Stopped music on hold on SIP/1000-00000002
[Apr 19 08:39:23] VERBOSE[14401][C-00000001] app_dial.c: -- SIP/2000-00000003 answered SIP/1000-00000002
[Apr 19 08:39:23] VERBOSE[14401][C-00000001] chan_sip.c: Audio is at 10282
[Apr 19 08:39:23] VERBOSE[14401][C-00000001] chan_sip.c: Adding codec 100004 (alaw) to SDP
[Apr 19 08:39:23] VERBOSE[14401][C-00000001] chan_sip.c: Adding codec 100003 (ulaw) to SDP
[Apr 19 08:39:23] VERBOSE[14401][C-00000001] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Apr 19 08:39:23] VERBOSE[14401][C-00000001] chan_sip.c:
<--- Reliably Transmitting (no NAT) to 172.16.0.20:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.0.20:5060;branch=z9hG4bK2330025bbd617c5d;received=172.16.0.20
From: "1000" <sip:1000@172.16.0.10>;tag=7ec3f92d-16226
To: <sip:100@172.16.0.10:5060;user=phone>;tag=as3b3d3d2c
Call-ID: BD21-1143-4100062266E8A23356586-022@SipHost
CSeq: 1142 INVITE
Server: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:100@172.16.0.10:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 506

v=0
o=root 887912993 887912993 IN IP4 172.16.0.10
s=Asterisk PBX 11.8.1
c=IN IP4 172.16.0.10
t=0 0
m=audio 10282 RTP/AVP 8 0 101
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=ice-ufrag:53afa21f5bc7d61b2ceaee3132cfb05f
a=ice-pwd:25f4c76f163a074a6fa7421231f75cd8
a=candidate:Hac100062 1 UDP 2130706431 172.16.0.10 10282 typ host
a=candidate:Hac100062 2 UDP 2130706430 172.16.0.10 10283 typ host
a=sendrecv

<------------>
[Apr 19 08:39:23] VERBOSE[3400] chan_sip.c: Retransmitting #1 (no NAT) to 172.16.0.20:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.0.20:5060;branch=z9hG4bK2330025bbd617c5d;received=172.16.0.20
From: "1000" <sip:1000@172.16.0.10>;tag=7ec3f92d-16226
To: <sip:100@172.16.0.10:5060;user=phone>;tag=as3b3d3d2c
Call-ID: BD21-1143-4100062266E8A23356586-022@SipHost
CSeq: 1142 INVITE
Server: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:100@172.16.0.10:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 506

v=0
o=root 887912993 887912993 IN IP4 172.16.0.10
s=Asterisk PBX 11.8.1
c=IN IP4 172.16.0.10
t=0 0
m=audio 10282 RTP/AVP 8 0 101
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=ice-ufrag:53afa21f5bc7d61b2ceaee3132cfb05f
a=ice-pwd:25f4c76f163a074a6fa7421231f75cd8
a=candidate:Hac100062 1 UDP 2130706431 172.16.0.10 10282 typ host
a=candidate:Hac100062 2 UDP 2130706430 172.16.0.10 10283 typ host
a=sendrecv

---
[Apr 19 08:39:23] VERBOSE[14401][C-00000001] res_rtp_asterisk.c: > 0x94fe578 -- Probation passed - setting RTP source address to 172.16.0.20:9002
[Apr 19 08:39:23] VERBOSE[3400] chan_sip.c:
<--- SIP read from UDP:172.16.0.20:5060 --->
ACK sip:100@172.16.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.0.20:5060;branch=z9hG4bKea4e119ec5abf0ed
From: "1000" <sip:1000@172.16.0.10>;tag=7ec3f92d-16226
To: <sip:100@172.16.0.10:5060;user=phone>;tag=as3b3d3d2c
Call-ID: BD21-1143-4100062266E8A23356586-022@SipHost
CSeq:1142 ACK
Max-Forwards:70
Authorization:Digest username="1000",realm="asterisk",nonce="1c734ae6",uri="sip:100@172.16.0.10:5060;user=phone",response="3b02d882988d13ba6558854920fffcf6",algorithm=MD5
User-Agent:dlink
Content-Length: 0

<------------->
[Apr 19 08:39:23] VERBOSE[3400] chan_sip.c: --- (10 headers 0 lines) ---
[Apr 19 08:39:23] VERBOSE[3400] chan_sip.c:
<--- SIP read from UDP:172.16.0.20:5060 --->
ACK sip:100@172.16.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.0.20:5060;branch=z9hG4bKea4e119ec5abf0ed
From: "1000" <sip:1000@172.16.0.10>;tag=7ec3f92d-16226
To: <sip:100@172.16.0.10:5060;user=phone>;tag=as3b3d3d2c
Call-ID: BD21-1143-4100062266E8A23356586-022@SipHost
CSeq:1142 ACK
Max-Forwards:70
Authorization:Digest username="1000",realm="asterisk",nonce="1c734ae6",uri="sip:100@172.16.0.10:5060;user=phone",response="3b02d882988d13ba6558854920fffcf6",algorithm=MD5
User-Agent:dlink
Content-Length: 0

<------------->
[Apr 19 08:39:23] VERBOSE[3400] chan_sip.c: --- (10 headers 0 lines) ---
[Apr 19 08:39:26] VERBOSE[3400] chan_sip.c:
<--- SIP read from UDP:172.16.0.20:5060 --->
BYE sip:100@172.16.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.0.20:5060;branch=z9hG4bK45eff9d052d63fb4
From: "1000" <sip:1000@172.16.0.10>;tag=7ec3f92d-16226
To: <sip:100@172.16.0.10:5060;user=phone>;tag=as3b3d3d2c
Call-ID: BD21-1143-4100062266E8A23356586-022@SipHost
CSeq:1143 BYE
Max-Forwards:70
User-Agent:dlink
Content-Length: 0

<------------->
[Apr 19 08:39:26] VERBOSE[3400] chan_sip.c: --- (9 headers 0 lines) ---
[Apr 19 08:39:26] VERBOSE[3400][C-00000001] chan_sip.c: Sending to 172.16.0.20:5060 (no NAT)
[Apr 19 08:39:26] VERBOSE[3400][C-00000001] chan_sip.c: Scheduling destruction of SIP dialog 'BD21-1143-4100062266E8A23356586-022@SipHost' in 6400 ms (Method: BYE)
[Apr 19 08:39:26] VERBOSE[3400][C-00000001] chan_sip.c:
<--- Transmitting (no NAT) to 172.16.0.20:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.0.20:5060;branch=z9hG4bK45eff9d052d63fb4;received=172.16.0.20
From: "1000" <sip:1000@172.16.0.10>;tag=7ec3f92d-16226
To: <sip:100@172.16.0.10:5060;user=phone>;tag=as3b3d3d2c
Call-ID: BD21-1143-4100062266E8A23356586-022@SipHost
CSeq: 1143 BYE
Server: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
[Apr 19 08:39:26] VERBOSE[14401][C-00000001] pbx.c: -- Executing [h@context:9996] NoOp("SIP/1000-00000002", "") in new stack
[Apr 19 08:39:26] WARNING[14401][C-00000001] res_odbc.c: SetConnectAttr (Txn isolation) returned an error: HY000: [MySQL][ODBC 5.1 Driver]MySQL server has gone away
[Apr 19 08:39:33] WARNING[3400] chan_sip.c: Autodestruct on dialog 'BD21-1143-4100062266E8A23356586-022@SipHost' with owner SIP/1000-00000002 in place (Method: BYE). Rescheduling destruction for 10000 ms
[Apr 19 08:39:33] VERBOSE[3400] chan_sip.c:
<--- SIP read from UDP:172.16.0.20:5060 --->
BYE sip:1000@172.16.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.0.20:5060;branch=z9hG4bK9438a22e89e0c201
From: <sip:100@172.16.0.20:5060>;tag=33e5ff91-16228
To: "PSTN Phone" <sip:1000@172.16.0.10:5060>;tag=as145b939c
Call-ID: 20853266711a687119364bfb485b4a96@172.16.0.10:5060
CSeq:2 BYE
Contact: <sip:172.16.0.20:5060>
Max-Forwards:70
User-Agent:dlink
Content-Length: 0

<------------->
[Apr 19 08:39:33] VERBOSE[3400] chan_sip.c: --- (10 headers 0 lines) ---
[Apr 19 08:39:33] VERBOSE[3400][C-00000001] chan_sip.c: Sending to 172.16.0.20:5060 (no NAT)
[Apr 19 08:39:33] VERBOSE[3400][C-00000001] chan_sip.c: Scheduling destruction of SIP dialog '20853266711a687119364bfb485b4a96@172.16.0.10:5060' in 6400 ms (Method: BYE)
[Apr 19 08:39:33] VERBOSE[3400][C-00000001] chan_sip.c:
<--- Transmitting (no NAT) to 172.16.0.20:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.0.20:5060;branch=z9hG4bK9438a22e89e0c201;received=172.16.0.20
From: <sip:100@172.16.0.20:5060>;tag=33e5ff91-16228
To: "PSTN Phone" <sip:1000@172.16.0.10:5060>;tag=as145b939c
Call-ID: 20853266711a687119364bfb485b4a96@172.16.0.10:5060
CSeq: 2 BYE
Server: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
[Apr 19 08:39:39] WARNING[3400] chan_sip.c: Autodestruct on dialog 'BD21-1143-4100062266E8A23356586-022@SipHost' with owner SIP/1000-00000002 in place (Method: BYE). Rescheduling destruction for 10000 ms
[Apr 19 08:39:39] WARNING[3400] chan_sip.c: Autodestruct on dialog '20853266711a687119364bfb485b4a96@172.16.0.10:5060' with owner SIP/2000-00000003 in place (Method: BYE). Rescheduling destruction for 10000 ms
[Apr 19 08:39:43] VERBOSE[3400] chan_sip.c:
<--- SIP read from UDP:172.16.0.21:5060 --->

<------------->
[Apr 19 08:39:45] VERBOSE[3400] chan_sip.c: Reliably Transmitting (no NAT) to 172.16.0.20:5060:
OPTIONS sip:1000@172.16.0.20:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.0.10:5060;branch=z9hG4bK1d4d5f19
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.16.0.10:5060>;tag=as05bf74dd
To: <sip:1000@172.16.0.20:5060>
Contact: <sip:asterisk@172.16.0.10:5060>
Call-ID: 1dc734840c3aa32a689796ea59af2ce0@172.16.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk
Date: Sat, 19 Apr 2014 04:39:45 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[Apr 19 08:39:45] VERBOSE[3400] chan_sip.c:
<--- SIP read from UDP:172.16.0.20:5060 --->
SIP/2.0 200 OK
Allow:INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,REFER,SUBSCRIBE,NOTIFY,UPDATE
Via: SIP/2.0/UDP 172.16.0.10:5060;branch=z9hG4bK1d4d5f19
From: "asterisk" <sip:asterisk@172.16.0.10:5060>;tag=as05bf74dd
To: <sip:1000@172.16.0.20:5060>;tag=c8bcb663-16258
Call-ID: 1dc734840c3aa32a689796ea59af2ce0@172.16.0.10:5060
CSeq:102 OPTIONS
Contact: <sip:1000@172.16.0.20:5060>
User-Agent:dlink
Content-Length: 0

<------------->
[Apr 19 08:39:45] VERBOSE[3400] chan_sip.c: --- (10 headers 0 lines) ---
[Apr 19 08:39:45] VERBOSE[3400] chan_sip.c: Really destroying SIP dialog '1dc734840c3aa32a689796ea59af2ce0@172.16.0.10:5060' Method: OPTIONS
[Apr 19 08:39:45] WARNING[3400] chan_sip.c: Autodestruct on dialog 'BD21-1143-4100062266E8A23356586-022@SipHost' with owner SIP/1000-00000002 in place (Method: BYE). Rescheduling destruction for 10000 ms
[Apr 19 08:39:46] VERBOSE[3400] chan_sip.c: Reliably Transmitting (no NAT) to 172.16.0.20:5060:
OPTIONS sip:2000@172.16.0.20:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.0.10:5060;branch=z9hG4bK2ccbad6b
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.16.0.10:5060>;tag=as0d4a0493
To: <sip:2000@172.16.0.20:5060>
Contact: <sip:asterisk@172.16.0.10:5060>
Call-ID: 49f2eada73ed0bf4420190e276e6b815@172.16.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk
Date: Sat, 19 Apr 2014 04:39:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[Apr 19 08:39:46] WARNING[3400] chan_sip.c: Autodestruct on dialog '20853266711a687119364bfb485b4a96@172.16.0.10:5060' with owner SIP/2000-00000003 in place (Method: BYE). Rescheduling destruction for 10000 ms
[Apr 19 08:39:46] VERBOSE[3400] chan_sip.c:
<--- SIP read from UDP:172.16.0.20:5060 --->
SIP/2.0 200 OK
Allow:INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,REFER,SUBSCRIBE,NOTIFY,UPDATE
Via: SIP/2.0/UDP 172.16.0.10:5060;branch=z9hG4bK2ccbad6b
From: "asterisk" <sip:asterisk@172.16.0.10:5060>;tag=as0d4a0493
To: <sip:2000@172.16.0.20:5060>;tag=ebeebdae-16259
Call-ID: 49f2eada73ed0bf4420190e276e6b815@172.16.0.10:5060
CSeq:102 OPTIONS
Contact: <sip:2000@172.16.0.20:5060>
User-Agent:dlink
Content-Length: 0

<------------->
[Apr 19 08:39:46] VERBOSE[3400] chan_sip.c: --- (10 headers 0 lines) ---
[Apr 19 08:39:46] VERBOSE[3400] chan_sip.c: Really destroying SIP dialog '49f2eada73ed0bf4420190e276e6b815@172.16.0.10:5060' Method: OPTIONS
[Apr 19 08:39:52] WARNING[3400] chan_sip.c: Autodestruct on dialog 'BD21-1143-4100062266E8A23356586-022@SipHost' with owner SIP/1000-00000002 in place (Method: BYE). Rescheduling destruction for 10000 ms
[Apr 19 08:39:52] WARNING[3400] chan_sip.c: Autodestruct on dialog '20853266711a687119364bfb485b4a96@172.16.0.10:5060' with owner SIP/2000-00000003 in place (Method: BYE). Rescheduling destruction for 10000 ms
[Apr 19 08:39:58] WARNING[3400] chan_sip.c: Autodestruct on dialog 'BD21-1143-4100062266E8A23356586-022@SipHost' with owner SIP/1000-00000002 in place (Method: BYE). Rescheduling destruction for 10000 ms
[Apr 19 08:39:58] WARNING[3400] chan_sip.c: Autodestruct on dialog '20853266711a687119364bfb485b4a96@172.16.0.10:5060' with owner SIP/2000-00000003 in place (Method: BYE). Rescheduling destruction for 10000 ms
[Apr 19 08:40:05] WARNING[3400] chan_sip.c: Autodestruct on dialog 'BD21-1143-4100062266E8A23356586-022@SipHost' with owner SIP/1000-00000002 in place (Method: BYE). Rescheduling destruction for 10000 ms
[Apr 19 08:40:05] WARNING[3400] chan_sip.c: Autodestruct on dialog '20853266711a687119364bfb485b4a96@172.16.0.10:5060' with owner SIP/2000-00000003 in place (Method: BYE). Rescheduling destruction for 10000 ms

Также прикладываю excel-файл с красиво выписанным SIP-логом этого звонка.

А вот вывод некоторых инфо-команд, я думаю он может быть полезен (звонок правда другой, но ситуация абсолютно аналогична):
Скрытый текст: показать
CLI> sip show channels
Peer User/ANR Call ID Format Hold Last Message Expiry Peer
172.16.0.20 100 059d7f0a2e79cea (alaw) No Rx: BYE 2000
172.16.0.20 1000 BD21-1143-46800 (alaw) No Rx: BYE 1000
3 active SIP dialogs

CLI> sip show channel 059d7f0a2e79cea

* SIP Call
Curr. trans. direction: Outgoing
Call-ID: 059d7f0a2e79cea66d4d38b947216f40@172.16.0.10:5060
Owner channel ID: SIP/2000-00000019
Our Codec Capability: (gsm|ulaw|alaw)
Non-Codec Capability (DTMF): 1
Their Codec Capability: (alaw)
Joint Codec Capability: (alaw)
Format: (alaw)
T.38 support No
Video support No
MaxCallBR: 384 kbps
Theoretical Address: 172.16.0.20:5060
Received Address: 172.16.0.20:5060
SIP Transfer mode: open
Force rport: No
Audio IP: 172.16.0.10 (local)
Our Tag: as5ee32903
Their Tag: ec65f16f-800289
SIP User agent: dlink
Username: 100
Peername: 2000
Original uri: sip:172.16.0.20:5060
Need Destroy: No
Last Message: Rx: BYE
Promiscuous Redir: No
Route: <sip:172.16.0.20:5060>
DTMF Mode: rfc2833
SIP Options: (none)
Session-Timer: Inactive

CLI> sip show history 059d7f0a2e79cea

* SIP Call
1. NewChan Channel SIP/2000-00000019 - from 059d7f0a2e79cea66d4d38b947216
2. TxReqRel INVITE / 102 INVITE - INVITE
3. ReTx 200 INVITE sip:100@172.16.0.20:5060 SIP/2.0
4. Rx SIP/2.0 / 102 INVITE / 100 Trying
5. Rx SIP/2.0 / 102 INVITE / 100 Trying
6. Rx SIP/2.0 / 102 INVITE / 183 Session in progress
7. Hold SIP/2.0
8. Rx SIP/2.0 / 102 INVITE / 200 OK
9. Unhold SIP/2.0
10. TxReq ACK / 102 ACK - ACK
11. Rx BYE / 12 BYE / sip:1000@172.16.0.10:5060
12. RTCPaudio Quality:ssrc=1980451042;themssrc=2232333582;lp=0;rxjitter=0.003
13. RTCPaudioJitter Quality:minrxjitter=0.000000;maxrxjitter=0.000000;avgrxjitter=0
14. RTCPaudioLoss Quality:minrxlost=0.000000;maxrxlost=0.000000;avgrxlost=0.00000
15. RTCPaudioRTT Quality:minrtt=0.000000;maxrtt=0.000000;avgrtt=0.000000;stdevrt
16. SchedDestroy 6400 ms
17. TxResp SIP/2.0 / 12 BYE - 200 OK

CLI> sip show channel BD21-1143-46800

* SIP Call
Curr. trans. direction: Incoming
Call-ID: BD21-1143-468002877881A024BDFB-032@SipHost
Owner channel ID: SIP/1000-00000018
Our Codec Capability: (gsm|ulaw|alaw)
Non-Codec Capability (DTMF): 1
Their Codec Capability: (g723|ulaw|alaw|g726|g729)
Joint Codec Capability: (ulaw|alaw)
Format: (alaw)
T.38 support No
Video support No
MaxCallBR: 384 kbps
Theoretical Address: 172.16.0.20:5060
Received Address: 172.16.0.20:5060
SIP Transfer mode: open
Force rport: No
Audio IP: 172.16.0.10 (local)
Our Tag: as2cdb5dbd
Their Tag: ef763358-800287
SIP User agent: dlink
Username: 1000
Peername: 1000
Original uri: sip:1000@172.16.0.20:5060
Caller-ID: 1000
Need Destroy: No
Last Message: Rx: BYE
Promiscuous Redir: No
Route: <sip:1000@172.16.0.20:5060>
DTMF Mode: rfc2833
SIP Options: replaces replace timer
Session-Timer: Inactive

CLI> sip show history BD21-1143-46800

* SIP Call
1. Rx INVITE / 1408 INVITE / sip:100@172.16.0.10:5060;user=phone
2. AuthChal Auth challenge sent for - nc 0
3. TxRespRel SIP/2.0 / 1408 INVITE - 401 Unauthorized
4. SchedDestroy 6400 ms
5. ReTx 200 SIP/2.0 401 Unauthorized
6. Rx ACK / 1408 ACK / sip:100@172.16.0.10:5060;user=phone
7. Rx INVITE / 1409 INVITE / sip:100@172.16.0.10:5060;user=phone
8. CancelDestroy
9. Invite New call: BD21-1143-468002877881A024BDFB-032@SipHost
10. AuthOK Auth challenge successful for 1000
11. NewChan Channel SIP/1000-00000018 - from BD21-1143-468002877881A024BDFB-
12. TxResp SIP/2.0 / 1409 INVITE - 100 Trying
13. TxResp SIP/2.0 / 1409 INVITE - 183 Session Progress
14. TxRespRel SIP/2.0 / 1409 INVITE - 200 OK
15. ReTx 200 SIP/2.0 200 OK
16. Rx ACK / 1409 ACK / sip:100@172.16.0.10:5060
17. Rx ACK / 1409 ACK / sip:100@172.16.0.10:5060
18. Rx BYE / 1410 BYE / sip:100@172.16.0.10:5060
19. RTCPaudio Quality:ssrc=1937276587;themssrc=3832984186;lp=0;rxjitter=0.000
20. RTCPaudioJitter Quality:minrxjitter=0.000000;maxrxjitter=0.000000;avgrxjitter=0
21. RTCPaudioLoss Quality:minrxlost=0.000000;maxrxlost=0.000000;avgrxlost=0.00000
22. RTCPaudioRTT Quality:minrtt=0.000000;maxrtt=0.000000;avgrtt=0.000000;stdevrt
23. SchedDestroy 6400 ms
24. TxResp SIP/2.0 / 1410 BYE - 200 OK

CLI> core show channels
Channel Location State Application(Data)
SIP/1000-00000018 100@context Up Dial(SIP/2000/100,120,mXWKT)
SIP/2000-00000019 (None) Up AppDial((Outgoing Line))
2 active channels
1 active call
13 calls processed

CLI> core show channel SIP/1000-00000018
-- General --
Name: SIP/1000-00000018
Type: SIP
UniqueID: 1398193286.24
LinkedID: 1398193286.24
Caller ID: 1000
Caller ID Name: PSTN Phone
Connected Line ID: (N/A)
Connected Line ID Name: (N/A)
Eff. Connected Line ID: (N/A)
Eff. Connected Line ID Name: (N/A)
DNID Digits: 100
Language: ru
State: Up (6)
Rings: 0
NativeFormats: (alaw)
WriteFormat: alaw
ReadFormat: alaw
WriteTranscode: No
ReadTranscode: No
1st File Descriptor: 21
Frames in: 839
Frames out: 851
Time to Hangup: 0
Elapsed Time: 0h5m8s
Direct Bridge: <none>
Indirect Bridge: <none>
-- PBX --
Context: context
Extension: 100
Priority: 16
Call Group: 2
Pickup Group: 0
Application: Dial
Data: SIP/2000/100,120,mXWKT
Blocking in: (Not Blocking)
Call Identifer: [C-0000000e]
Variables:
DIALEDTIME=17
ANSWEREDTIME=7
RTPAUDIOQOSRTTBRIDGED=minrtt=0.000000;maxrtt=0.000000;avgrtt=0.000000;stdevrtt=0.000000;
RTPAUDIOQOSLOSSBRIDGED=minrxlost=0.000000;maxrxlost=0.000000;avgrxlost=0.000000;stdevrxlost=0.000000;reported_minlost=0.000000;reported_maxlost=0.000000;reported_avglost=0.000000;reported_stdevlost=0.000000;
RTPAUDIOQOSJITTERBRIDGED=minrxjitter=0.000000;maxrxjitter=0.000000;avgrxjitter=0.000000;stdevrxjitter=0.000000;reported_minjitter=0.000000;reported_maxjitter=0.000000;reported_avgjitter=0.000000;reported_stdevjitter=0.000000;
RTPAUDIOQOSBRIDGED=ssrc=1980451042;themssrc=2232333582;lp=0;rxjitter=0.003224;rxcount=326;txjitter=0.000000;txcount=324;rlp=0;rtt=0.000000
RTPAUDIOQOSRTT=minrtt=0.000000;maxrtt=0.000000;avgrtt=0.000000;stdevrtt=0.000000;
RTPAUDIOQOSLOSS=minrxlost=0.000000;maxrxlost=0.000000;avgrxlost=0.000000;stdevrxlost=0.000000;reported_minlost=0.000000;reported_maxlost=0.000000;reported_avglost=0.000000;reported_stdevlost=0.000000;
RTPAUDIOQOSJITTER=minrxjitter=0.000000;maxrxjitter=0.000000;avgrxjitter=0.000000;stdevrxjitter=0.000000;reported_minjitter=0.000000;reported_maxjitter=0.000000;reported_avgjitter=0.000000;reported_stdevjitter=0.000000;
RTPAUDIOQOS=ssrc=1937276587;themssrc=3832984186;lp=0;rxjitter=0.000216;rxcount=319;txjitter=0.000000;txcount=849;rlp=0;rtt=0.000000
BRIDGEPVTCALLID=059d7f0a2e79cea66d4d38b ... .0.10:5060
BRIDGEPEER=SIP/2000-00000019
DIALEDPEERNUMBER=2000/100
DIALEDPEERNAME=SIP/2000-00000019
DIALSTATUS=ANSWER
DYNAMIC_FEATURES=
check=
options=mXWKT
waitTime=120
dialString=SIP/2000/100
phoneNumber=100
ARGC=4
ARG4=mXWKT
ARG3=120
ARG2=SIP/2000/100
ARG1=100
phoneNumber=100
ARGC=1
ARG1=100
SIPCALLID=BD21-1143-468002877881A024BDFB-032@SipHost
SIPDOMAIN=172.16.0.10
SIPURI=sip:1000@172.16.0.20:5060

CDR Variables:
level 1: dnid=100
level 1: clid="PSTN Phone" <1000>
level 1: src=1000
level 1: dst=100
level 1: dcontext=context
level 1: channel=SIP/1000-00000018
level 1: dstchannel=SIP/2000-00000019
level 1: lastapp=Dial
level 1: lastdata=SIP/2000/100,120,mXWKT
level 1: start=2014-04-22 23:01:26
level 1: answer=2014-04-22 23:01:36
level 1: duration=307
level 1: billsec=297
level 1: disposition=ANSWERED
level 1: amaflags=DOCUMENTATION
level 1: uniqueid=1398193286.24
level 1: linkedid=1398193286.24
level 1: sequence=36

CLI> core show channel SIP/2000-00000019
-- General --
Name: SIP/2000-00000019
Type: SIP
UniqueID: 1398193286.25
LinkedID: 1398193286.24
Caller ID: 100
Caller ID Name: (N/A)
Connected Line ID: 1000
Connected Line ID Name: PSTN Phone
Eff. Connected Line ID: 1000
Eff. Connected Line ID Name: PSTN Phone
DNID Digits: (N/A)
Language: ru
State: Up (6)
Rings: 0
NativeFormats: (alaw)
WriteFormat: alaw
ReadFormat: alaw
WriteTranscode: No
ReadTranscode: No
1st File Descriptor: 26
Frames in: 750
Frames out: 324
Time to Hangup: 0
Elapsed Time: 0h6m12s
Direct Bridge: <none>
Indirect Bridge: <none>
-- PBX --
Context: extcontext
Extension:
Priority: 1
Call Group: 2
Pickup Group: 0
Application: AppDial
Data: (Outgoing Line)
Blocking in: (Not Blocking)
Call Identifer: [C-0000000e]
Variables:
RTPAUDIOQOSRTT=minrtt=0.000000;maxrtt=0.000000;avgrtt=0.000000;stdevrtt=0.000000;
RTPAUDIOQOSLOSS=minrxlost=0.000000;maxrxlost=0.000000;avgrxlost=0.000000;stdevrxlost=0.000000;reported_minlost=0.000000;reported_maxlost=0.000000;reported_avglost=0.000000;reported_stdevlost=0.000000;
RTPAUDIOQOSJITTER=minrxjitter=0.000000;maxrxjitter=0.000000;avgrxjitter=0.000000;stdevrxjitter=0.000000;reported_minjitter=0.000000;reported_maxjitter=0.000000;reported_avgjitter=0.000000;reported_stdevjitter=0.000000;
RTPAUDIOQOS=ssrc=1980451042;themssrc=2232333582;lp=0;rxjitter=0.003224;rxcount=326;txjitter=0.000000;txcount=324;rlp=0;rtt=0.000000
RTPAUDIOQOSRTTBRIDGED=minrtt=0.000000;maxrtt=0.000000;avgrtt=0.000000;stdevrtt=0.000000;
RTPAUDIOQOSLOSSBRIDGED=minrxlost=0.000000;maxrxlost=0.000000;avgrxlost=0.000000;stdevrxlost=0.000000;reported_minlost=0.000000;reported_maxlost=0.000000;reported_avglost=0.000000;reported_stdevlost=0.000000;
RTPAUDIOQOSJITTERBRIDGED=minrxjitter=0.000000;maxrxjitter=0.000000;avgrxjitter=0.000000;stdevrxjitter=0.000000;reported_minjitter=0.000000;reported_maxjitter=0.000000;reported_avgjitter=0.000000;reported_stdevjitter=0.000000;
RTPAUDIOQOSBRIDGED=ssrc=1937276587;themssrc=3832984186;lp=0;rxjitter=0.000216;rxcount=319;txjitter=0.000000;txcount=849;rlp=0;rtt=0.000000
BRIDGEPVTCALLID=BD21-1143-468002877881A024BDFB-032@SipHost
BRIDGEPEER=SIP/1000-00000018
DIALEDPEERNUMBER=2000/100
SIPCALLID=059d7f0a2e79cea66d4d38b947216 ... .0.10:5060

CDR Variables:
level 1: dnid=
level 1: clid=100
level 1: src=100
level 1: dst=s
level 1: dcontext=extcontext
level 1: channel=SIP/2000-00000019
level 1: start=2014-04-22 23:01:26
level 1: answer=2014-04-22 23:01:36
level 1: duration=371
level 1: billsec=361
level 1: disposition=ANSWERED
level 1: amaflags=DOCUMENTATION
level 1: uniqueid=1398193286.25
level 1: linkedid=1398193286.24
level 1: sequence=35

Помогите пожалуйста это победить, уже перерыл кучу форумов, гуглил, но никакого толкового решения этой проблемы не увидел.


Вложения:
call_log.xls [49.5 KiB]
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Alexey Motkov
СообщениеДобавлено: Пн май 19, 2014 10:28 
Не в сети
Сотрудник D-LINK
Сотрудник D-LINK

Зарегистрирован: Пн янв 11, 2010 09:40
Сообщений: 4400
Вопрос скорее к компани DIGIUM, но межете попробовать запретить для этих линий кодек GSM, также покажите лог со шлюза во время данного звонка.
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