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плавающая проблема со звуком

https://forum.asterisk.ru/viewtopic.php?t=7640

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плавающая проблема со звуком

Добавлено: 03 июл 2016, 23:36
alexkab
Добрый день.
Решили перевести пользователей на рарус софтфон (использовать его как сип клиент) и временами (может быть 1 звонок из 10) при входящем звонке от сип провайдера Telphin, не идет звук от вызывающего абонента (мы его не слышим, он нас слышит). При использовании 3cxphone и железного телефона, такой проблемы не было.С другими sip провайдером рарус софтфон вроде бы работает стабильно, проблем пока не замечено. Помогите отдебажить по чьей вине это происходит и можно ли исправить?
PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER: Звук есть
Изображение
PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER: Нет звука
Изображение
213.170.92.166 телфин
192.168.200.9 астериск 11.13
192.168.220.98 софтфон рарус
95.80.121.253 телфин войс?


PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER: CLI когда нет звука
<--- SIP read from UDP:213.170.92.166:5060 --->
INVITE sip:000127483@192.168.200.9:5060 SIP/2.0
Record-Route: <sip:7474YXYYZZ4@213.170.92.166;lr=on;ftag=1c100376760;ldi=17.cc23;tfs=AAAAABICCQ0ADwkGCglzAXVdUUAARVddXlpcXS5jb206NTA2MA-->
Record-Route: <sip:213.170.100.150;lr;ftag=1c100376760;did=17.93a153f6>
Via: SIP/2.0/UDP 213.170.92.166;branch=z9hG4bK8a08.70038ea22ce9c7bcfa3ae9d464137946.0
Via: SIP/2.0/UDP 213.170.100.150:5060;branch=z9hG4bK8a08.7b05cfd5.0
Via: SIP/2.0/UDP 95.80.121.253;branch=z9hG4bKac100383652
Max-Forwards: 69
From: +79508019234 <sip:+79508019234@sip.telphin.com:5060>;tag=1c100376760
To: <sip:7474YXYYZZZ@213.170.100.150;user=phone>
P-Preferred-Identity: <sip:+79508019234@sip.telphin.com:5060>
Call-ID: 10037573114420028289@95.80.121.253
CSeq: 1 INVITE
Contact: <sip:9508019234@95.80.121.253:5060>
Supported: em,100rel,timer,replaces,path,early-session,resource-priority,sdp-anat
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 304

v=0
o=AudiocodesGW 100370112 100369777 IN IP4 95.80.121.253
s=Phone-Call
c=IN IP4 95.80.121.253
t=0 0
m=audio 6940 RTP/AVP 8 0 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
<------------->
--- (17 headers 14 lines) ---
Sending to 213.170.92.166:5060 (no NAT)
Sending to 213.170.92.166:5060 (no NAT)
Using INVITE request as basis request - 10037573114420028289@95.80.121.253
Found peer 'telphin' for '+79508019234' from 213.170.92.166:5060
== Using SIP RTP CoS mark 5
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw|g729), peer - audio=(ulaw|alaw|g729)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 95.80.121.253:6940
Looking for 000127483 in ctx-department-high (domain 192.168.200.9)
list_route: hop: <sip:7474YXYYZZ4@213.170.92.166;lr=on;ftag=1c100376760;ldi=17.cc23;tfs=AAAAABICCQ0ADwkGCglzAXVdUUAARVddXlpcXS5jb206NTA2MA-->
list_route: hop: <sip:213.170.100.150;lr;ftag=1c100376760;did=17.93a153f6>

<--- Transmitting (NAT) to 213.170.92.166:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 213.170.92.166;branch=z9hG4bK8a08.70038ea22ce9c7bcfa3ae9d464137946.0;received=213.170.92.166;rport=5060
Via: SIP/2.0/UDP 213.170.100.150:5060;branch=z9hG4bK8a08.7b05cfd5.0
Via: SIP/2.0/UDP 95.80.121.253;branch=z9hG4bKac100383652
Record-Route: <sip:7474YXYYZZ4@213.170.92.166;lr=on;ftag=1c100376760;ldi=17.cc23;tfs=AAAAABICCQ0ADwkGCglzAXVdUUAARVddXlpcXS5jb206NTA2MA-->
Record-Route: <sip:213.170.100.150;lr;ftag=1c100376760;did=17.93a153f6>
From: +79508019234 <sip:+79508019234@sip.telphin.com:5060>;tag=1c100376760
To: <sip:7474YXYYZZZ@213.170.100.150;user=phone>
Call-ID: 10037573114420028289@95.80.121.253
CSeq: 1 INVITE
Server: Asterisk PBX 11.13.1~dfsg-2+b1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:000127483@178.234.22.219:5060>
Content-Length: 0


<------------>
-- Executing [000127483@ctx-department-high:1] Dial("SIP/telphin-000008d3", "SIP/111&SIP/112&SIP/113&SIP/114&SIP/115&SIP/116&SIP/117&SIP/118&SIP/119&SIP/121&SIP/122&SIP/123&SIP/124&SIP/125&SIP/126&SIP/127&SIP/128&SIP/129&SIP/501&SIP/502,40,gtTM(record^000127483)") in new stack

-- SIP/502-000008d9 is ringing

<--- Transmitting (NAT) to 213.170.92.166:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 213.170.92.166;branch=z9hG4bK8a08.70038ea22ce9c7bcfa3ae9d464137946.0;received=213.170.92.166;rport=5060
Via: SIP/2.0/UDP 213.170.100.150:5060;branch=z9hG4bK8a08.7b05cfd5.0
Via: SIP/2.0/UDP 95.80.121.253;branch=z9hG4bKac100383652
Record-Route: <sip:7474YXYYZZ4@213.170.92.166;lr=on;ftag=1c100376760;ldi=17.cc23;tfs=AAAAABICCQ0ADwkGCglzAXVdUUAARVddXlpcXS5jb206NTA2MA-->
Record-Route: <sip:213.170.100.150;lr;ftag=1c100376760;did=17.93a153f6>
From: +79508019234 <sip:+79508019234@sip.telphin.com:5060>;tag=1c100376760
To: <sip:7474YXYYZZZ@213.170.100.150;user=phone>;tag=as7fa48790
Call-ID: 10037573114420028289@95.80.121.253
CSeq: 1 INVITE
Server: Asterisk PBX 11.13.1~dfsg-2+b1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:000127483@178.234.22.219:5060>
Content-Length: 0


<------------>
-- SIP/501-000008d8 is ringing

Reliably Transmitting (NAT) to 213.170.92.166:5060:
OPTIONS sip:sip.telphin.com SIP/2.0
Via: SIP/2.0/UDP 178.234.22.219:5060;branch=z9hG4bK501864eb;rport
Max-Forwards: 70
From: "asterisk" <sip:000127483@178.234.22.219>;tag=as04709ce7
To: <sip:sip.telphin.com>
Contact: <sip:000127483@178.234.22.219:5060>
Call-ID: 3ac9264f44d1bb750dab3f40460327c1@178.234.22.219:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.13.1~dfsg-2+b1
Date: Sun, 03 Jul 2016 19:00:30 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---
Reliably Transmitting (NAT) to 213.170.92.166:5060:
OPTIONS sip:sip.telphin.com SIP/2.0
Via: SIP/2.0/UDP 178.234.22.219:5060;branch=z9hG4bK52e179fd;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@178.234.22.219>;tag=as442d2718
To: <sip:sip.telphin.com>
Contact: <sip:asterisk@178.234.22.219:5060>
Call-ID: 7823a2be641008ed51d4b367736f77cf@178.234.22.219:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.13.1~dfsg-2+b1
Date: Sun, 03 Jul 2016 19:00:30 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:213.170.92.166:5060 --->
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.200.9:5060;branch=z9hG4bK501864eb;rport=5060;received=192.168.200.9
From: "asterisk" <sip:000127483@192.168.200.9:5060>;tag=as04709ce7
To: <sip:sip.telphin.com>;tag=599861ab72d4f5a03e7b922e1a9ee85b.7734
Call-ID: 3ac9264f44d1bb750dab3f40460327c1@178.234.22.219:5060
CSeq: 102 OPTIONS
Server: Telphin SoftSwitch
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '3ac9264f44d1bb750dab3f40460327c1@178.234.22.219:5060' Method: OPTIONS

<--- SIP read from UDP:213.170.92.166:5060 --->
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.200.9:5060;branch=z9hG4bK52e179fd;rport=5060;received=192.168.200.9
From: "asterisk" <sip:asterisk@192.168.200.9:5060>;tag=as442d2718
To: <sip:sip.telphin.com>;tag=599861ab72d4f5a03e7b922e1a9ee85b.ff9d
Call-ID: 7823a2be641008ed51d4b367736f77cf@178.234.22.219:5060
CSeq: 102 OPTIONS
Server: Telphin SoftSwitch
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '7823a2be641008ed51d4b367736f77cf@178.234.22.219:5060' Method: OPTIONS
-- SIP/501-000008d8 connected line has changed. Saving it until answer for SIP/telphin-000008d3
-- SIP/501-000008d8 answered SIP/telphin-000008d3
-- Executing [s@macro-record:1] Set("SIP/501-000008d8", "filename=/var/records/20160703220031") in new stack
-- Executing [s@macro-record:2] Set("SIP/501-000008d8", "AUDIOHOOK_INHERIT(MixMonitor)=yes") in new stack
-- Executing [s@macro-record:3] Set("SIP/501-000008d8", "CDR(userfield)=20160703220031") in new stack
-- Executing [s@macro-record:4] MixMonitor("SIP/501-000008d8", "/var/records/20160703220031.wav,b,/usr/bin/lame -S -V2 /var/records/20160703220031.wav /var/records/20160703220031.mp3 && rm -f /var/records/20160703220031.wav && chmod 766 /var/records/20160703220031.mp3") in new stack
== Begin MixMonitor Recording SIP/501-000008d8
-- Executing [s@macro-record:5] Set("SIP/501-000008d8", "AUDIOHOOK_INHERIT(MixMonitor)=yes") in new stack
-- Executing [s@macro-record:6] MacroExit("SIP/501-000008d8", "") in new stack
Audio is at 13938
Adding codec 100008 (g729) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100003 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 213.170.92.166:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 213.170.92.166;branch=z9hG4bK8a08.70038ea22ce9c7bcfa3ae9d464137946.0;received=213.170.92.166;rport=5060
Via: SIP/2.0/UDP 213.170.100.150:5060;branch=z9hG4bK8a08.7b05cfd5.0
Via: SIP/2.0/UDP 95.80.121.253;branch=z9hG4bKac100383652
Record-Route: <sip:7474YXYYZZ4@213.170.92.166;lr=on;ftag=1c100376760;ldi=17.cc23;tfs=AAAAABICCQ0ADwkGCglzAXVdUUAARVddXlpcXS5jb206NTA2MA-->
Record-Route: <sip:213.170.100.150;lr;ftag=1c100376760;did=17.93a153f6>
From: +79508019234 <sip:+79508019234@sip.telphin.com:5060>;tag=1c100376760
To: <sip:7474YXYYZZZ@213.170.100.150;user=phone>;tag=as7fa48790
Call-ID: 10037573114420028289@95.80.121.253
CSeq: 1 INVITE
Server: Asterisk PBX 11.13.1~dfsg-2+b1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:000127483@178.234.22.219:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 317

v=0
o=root 54023434 54023434 IN IP4 178.234.22.219
s=Asterisk PBX 11.13.1~dfsg-2+b1
c=IN IP4 178.234.22.219
t=0 0
m=audio 13938 RTP/AVP 18 8 0 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
== Begin MixMonitor Recording SIP/telphin-000008d3
== Begin MixMonitor Recording SIP/501-000008d8

<--- SIP read from UDP:213.170.92.166:5060 --->
ACK sip:000127483@192.168.200.9:5060 SIP/2.0
Record-Route: <sip:000127483@213.170.92.166;lr=on;ftag=1c100376760>
Record-Route: <sip:213.170.100.150;lr;ftag=1c100376760>
Via: SIP/2.0/UDP 213.170.92.166;branch=z9hG4bK8a08.036ceba8e515a9dc06f08894043f49c5.0
Via: SIP/2.0/UDP 213.170.100.150:5060;branch=z9hG4bK8a08.7b05cfd5.2
Via: SIP/2.0/UDP 95.80.121.253;rport=5060;received=95.80.121.253;branch=z9hG4bKac130509415
Max-Forwards: 69
From: <sip:+79508019234@sip.telphin.com:5060>;tag=1c100376760
To: <sip:7474YXYYZZZ@213.170.100.150;user=phone>;tag=as7fa48790
Call-ID: 10037573114420028289@95.80.121.253
CSeq: 1 ACK
Contact: <sip:9508019234@95.80.121.253:5060>
Supported: em,timer,replaces,path,early-session,resource-priority
Content-Length: 0

<------------->
--- (14 headers 0 lines) ---
== MixMonitor close filestream (mixed)
== MixMonitor close filestream (mixed)
== End MixMonitor Recording SIP/501-000008d8
== Executing [/usr/bin/lame -S -V2 /var/records/20160703220031.wav /var/records/20160703220031.mp3 && rm -f /var/records/20160703220031.wav && chmod 766 /var/records/20160703220031.mp3]
-- Executing [000127483@ctx-department-high:2] Hangup("SIP/telphin-000008d3", "") in new stack
== Spawn extension (ctx-department-high, 000127483, 2) exited non-zero on 'SIP/telphin-000008d3'
Scheduling destruction of SIP dialog '10037573114420028289@95.80.121.253' in 6400 ms (Method: ACK)
set_destination: Parsing <sip:7474YXYYZZ4@213.170.92.166;lr=on;ftag=1c100376760;ldi=17.cc23;tfs=AAAAABICCQ0ADwkGCglzAXVdUUAARVddXlpcXS5jb206NTA2MA--> for address/port to send to
set_destination: set destination to 213.170.92.166:5060
Reliably Transmitting (NAT) to 213.170.92.166:5060:
BYE sip:9508019234@95.80.121.253:5060 SIP/2.0
Via: SIP/2.0/UDP 178.234.22.219:5060;branch=z9hG4bK7d0a13ec;rport
Route: <sip:7474YXYYZZ4@213.170.92.166;lr=on;ftag=1c100376760;ldi=17.cc23;tfs=AAAAABICCQ0ADwkGCglzAXVdUUAARVddXlpcXS5jb206NTA2MA-->,<sip:213.170.100.150;lr;ftag=1c100376760;did=17.93a153f6>
Max-Forwards: 70
From: <sip:7474YXYYZZZ@213.170.100.150;user=phone>;tag=as7fa48790
To: +79508019234 <sip:+79508019234@sip.telphin.com:5060>;tag=1c100376760
Call-ID: 10037573114420028289@95.80.121.253
CSeq: 102 BYE
User-Agent: Asterisk PBX 11.13.1~dfsg-2+b1
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
== MixMonitor close filestream (mixed)
== End MixMonitor Recording SIP/telphin-000008d3
== End MixMonitor Recording SIP/501-000008d8
Retransmitting #1 (NAT) to 213.170.92.166:5060:
BYE sip:9508019234@95.80.121.253:5060 SIP/2.0
Via: SIP/2.0/UDP 178.234.22.219:5060;branch=z9hG4bK7d0a13ec;rport
Route: <sip:7474YXYYZZ4@213.170.92.166;lr=on;ftag=1c100376760;ldi=17.cc23;tfs=AAAAABICCQ0ADwkGCglzAXVdUUAARVddXlpcXS5jb206NTA2MA-->,<sip:213.170.100.150;lr;ftag=1c100376760;did=17.93a153f6>
Max-Forwards: 70
From: <sip:7474YXYYZZZ@213.170.100.150;user=phone>;tag=as7fa48790
To: +79508019234 <sip:+79508019234@sip.telphin.com:5060>;tag=1c100376760
Call-ID: 10037573114420028289@95.80.121.253
CSeq: 102 BYE
User-Agent: Asterisk PBX 11.13.1~dfsg-2+b1
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---

<--- SIP read from UDP:213.170.92.166:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.200.9:5060;received=192.168.200.9;branch=z9hG4bK7d0a13ec;rport=5060
From: <sip:7474YXYYZZZ@213.170.100.150;user=phone>;tag=as7fa48790
To: +79508019234 <sip:+79508019234@sip.telphin.com:5060>;tag=1c100376760
Call-ID: 10037573114420028289@95.80.121.253
CSeq: 102 BYE
Contact: <sip:9508019234@95.80.121.253:5060>
Record-Route: <sip:213.170.100.150;lr;ftag=1c100376760;did=17.93a153f6>,<sip:7474YXYYZZ4@213.170.92.166;lr=on;ftag=1c100376760;ldi=17.cc23;tfs=AAAAABICCQ0ADwkGCglzAXVdUUAARVddXlpcXS5jb206NTA2MA-->
Supported: em,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-Mediant 1000/v.6.00A.032.003
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '10037573114420028289@95.80.121.253' Method: ACK

<--- SIP read from UDP:213.170.92.166:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.200.9:5060;received=192.168.200.9;branch=z9hG4bK7d0a13ec;rport=5060
From: <sip:7474YXYYZZZ@213.170.100.150;user=phone>;tag=as7fa48790
To: +79508019234 <sip:+79508019234@sip.telphin.com:5060>;tag=1c100376760
Call-ID: 10037573114420028289@95.80.121.253
CSeq: 102 BYE
Contact: <sip:9508019234@95.80.121.253:5060>
Record-Route: <sip:213.170.100.150;lr;ftag=1c100376760;did=17.93a153f6>,<sip:7474YXYYZZ4@213.170.92.166;lr=on;ftag=1c100376760;ldi=17.cc23;tfs=AAAAABICCQ0ADwkGCglzAXVdUUAARVddXlpcXS5jb206NTA2MA-->
Supported: em,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-Mediant 1000/v.6.00A.032.003
Content-Length: 0


REGISTER 11 headers, 0 lines
Reliably Transmitting (no NAT) to 213.170.92.166:5060:
REGISTER sip:sip.telphin.com SIP/2.0
Via: SIP/2.0/UDP 178.234.22.219:5060;branch=z9hG4bK0e73663e
Max-Forwards: 70
From: <sip:000127483@sip.telphin.com>;tag=as02409708
To: <sip:000127483@sip.telphin.com>
Call-ID: 1c243f487fb7d3624fd2bacb1ecb7378@192.168.220.9
CSeq: 4676 REGISTER
User-Agent: Asterisk PBX 11.13.1~dfsg-2+b1
Authorization: Digest username="000127483", realm="sip.telphin.com", algorithm=MD5, uri="sip:sip.telphin.com", nonce="V3lhzFd5YKBhAl9alJTyLB6joq4X8RQE", response="a4a0a3ac8b3471e18b433fc26f70ffc3"
Expires: 120
Contact: <sip:000127483@178.234.22.219:5060>
Content-Length: 0


---

<--- SIP read from UDP:213.170.92.166:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.200.9:5060;branch=z9hG4bK0e73663e
From: <sip:000127483@sip.telphin.com>;tag=as02409708
To: <sip:000127483@sip.telphin.com>;tag=599861ab72d4f5a03e7b922e1a9ee85b.c4f4
Call-ID: 1c243f487fb7d3624fd2bacb1ecb7378@192.168.220.9
CSeq: 4676 REGISTER
Contact: <sip:000127483@192.168.200.9:5060>;expires=1800
Server: Telphin SoftSwitch
Content-Length: 0


Reliably Transmitting (NAT) to 213.170.92.166:5060:
OPTIONS sip:sip.telphin.com SIP/2.0
Via: SIP/2.0/UDP 178.234.22.219:5060;branch=z9hG4bK3aafc77a;rport
Max-Forwards: 70
From: "asterisk" <sip:000127483@178.234.22.219>;tag=as73e85553
To: <sip:sip.telphin.com>
Contact: <sip:000127483@178.234.22.219:5060>
Call-ID: 7173e76057964f312d9e29d34e06f3f5@178.234.22.219:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.13.1~dfsg-2+b1
Date: Sun, 03 Jul 2016 19:01:30 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---
Reliably Transmitting (NAT) to 213.170.92.166:5060:
OPTIONS sip:sip.telphin.com SIP/2.0
Via: SIP/2.0/UDP 178.234.22.219:5060;branch=z9hG4bK602c0460;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@178.234.22.219>;tag=as2aab1ded
To: <sip:sip.telphin.com>
Contact: <sip:asterisk@178.234.22.219:5060>
Call-ID: 2c6404cd5bacd2053d9537e01ee09c87@178.234.22.219:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.13.1~dfsg-2+b1
Date: Sun, 03 Jul 2016 19:01:30 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:213.170.92.166:5060 --->
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.200.9:5060;branch=z9hG4bK3aafc77a;rport=5060;received=192.168.200.9
From: "asterisk" <sip:000127483@192.168.200.9:5060>;tag=as73e85553
To: <sip:sip.telphin.com>;tag=599861ab72d4f5a03e7b922e1a9ee85b.e916
Call-ID: 7173e76057964f312d9e29d34e06f3f5@178.234.22.219:5060
CSeq: 102 OPTIONS
Server: Telphin SoftSwitch
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '7173e76057964f312d9e29d34e06f3f5@178.234.22.219:5060' Method: OPTIONS

<--- SIP read from UDP:213.170.92.166:5060 --->
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.200.9:5060;branch=z9hG4bK602c0460;rport=5060;received=192.168.200.9
From: "asterisk" <sip:asterisk@192.168.200.9:5060>;tag=as2aab1ded
To: <sip:sip.telphin.com>;tag=599861ab72d4f5a03e7b922e1a9ee85b.6c43
Call-ID: 2c6404cd5bacd2053d9537e01ee09c87@178.234.22.219:5060
CSeq: 102 OPTIONS
Server: Telphin SoftSwitch
Content-Length: 0

Re: плавающая проблема со звуком

Добавлено: 04 июл 2016, 09:25
ded
1) Надо обращаться к разработчиками Рарус, что Вас подвигло сюда то писать, если явно косяк там?

2) Судя по логу - вызов из телфина, с мобильного +7950ХХХХХХХ падает на кучу локальных телефонов
Dial("SIP/telphin-000008d3", "SIP/111&SIP/112&SIP/113&SIP/114&SIP/115&SIP/116&SIP/117&SIP/118&SIP/119&SIP/121&SIP/122&SIP/123&SIP/124&SIP/125&SIP/126&SIP/127&SIP/128&SIP/129&SIP/501&SIP/502
из которых 501 видать и есть тот самый рарус-софтфон.
При этом Телфин и ваш Астериск устанавливают три кодека для этой сессии:

Код: Выделить всё

Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw|g729), peer - audio=(ulaw|alaw|g729)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|g729)
и я могу предположить, что Телфин открывает голос в g729, а в рарусе его нет.

3) Много бессмысленных для этого разбора пакетов OPTIONS, лишний раз говорит о том, что логи не понимаете, или не хотите читать, Вам - в раздел Бизнес, и там искать платную поддержку.

Re: плавающая проблема со звуком

Добавлено: 04 июл 2016, 11:51
alexkab
1. Сначала подумал обратиться на форумы астериска, может подскажут, потом буду писать в рарус.
2. В 9 случаях из 10 звук есть, из за этого я подумал что дело не в наличии кодеков.
Еще раз вставлю картинки wireshark'а, потому что в первом сообщении не закрепились.
PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER: Звук есть
voice.jpeg
voice.jpeg (79.05 КБ) 8941 просмотр
PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER: Звука нет
no_voice.jpeg
no_voice.jpeg (84.3 КБ) 8941 просмотр

Re: плавающая проблема со звуком

Добавлено: 04 июл 2016, 11:59
april22
явно видно - что тут проблема в кодеках .
выключите везде 729 , и пронаблюдайте .

Re: плавающая проблема со звуком

Добавлено: 04 июл 2016, 23:18
alexkab
Не включаю G729, проблема осталась, звука иногда нет.
PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER: Звук есть
voice_without729.jpeg
voice_without729.jpeg (85.75 КБ) 8895 просмотров
PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER: Звука нет
no_voice_without729.jpeg
no_voice_without729.jpeg (76.16 КБ) 8895 просмотров
PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER: CLI звук есть
<--- SIP read from UDP:213.170.92.166:5060 --->
INVITE sip:000127483@192.168.200.9:5060 SIP/2.0
Record-Route: <sip:7474XXXXXXX@213.170.92.166;lr=on;ftag=1c1303544129;ldi=6b73.4744;tfs=AAAAABICCQ0ADwkGCglzAXVdUUAARVddXlpcXS5jb206NTA2MA-->
Record-Route: <sip:213.170.100.150;lr;ftag=1c1303544129;did=6b7.c2f1f0e2>
Via: SIP/2.0/UDP 213.170.92.166;branch=z9hG4bKde7f.078734efca238c1b852ba1b4f13ed1d4.0
Via: SIP/2.0/UDP 213.170.100.150:5060;branch=z9hG4bKde7f.534c8df3.0
Via: SIP/2.0/UDP 95.80.121.253;branch=z9hG4bKac1303550993
Max-Forwards: 69
From: +7950XXXXXXX <sip:+7950XXXXXXX@sip.telphin.com:5060>;tag=1c1303544129
To: <sip:474XXXXXXX@213.170.100.150;user=phone>
P-Preferred-Identity: <sip:+7950XXXXXXX@sip.telphin.com:5060>
Call-ID: 1303543094154200291054@95.80.121.253
CSeq: 1 INVITE
Contact: <sip:950XXXXXXX@95.80.121.253:5060>
Supported: em,100rel,timer,replaces,path,early-session,resource-priority,sdp-anat
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 306

v=0
o=AudiocodesGW 1303537507 1303537171 IN IP4 95.80.121.253
s=Phone-Call
c=IN IP4 95.80.121.253
t=0 0
m=audio 6040 RTP/AVP 8 0 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
<------------->
--- (17 headers 14 lines) ---
Sending to 213.170.92.166:5060 (no NAT)
Sending to 213.170.92.166:5060 (no NAT)
Using INVITE request as basis request - 1303543094154200291054@95.80.121.253
Found peer 'telphin' for '+7950XXXXXXX' from 213.170.92.166:5060
== Using SIP RTP CoS mark 5
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw), peer - audio=(ulaw|alaw|g729)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 95.80.121.253:6040
Looking for 000127483 in ctx-department-high (domain 192.168.200.9)
list_route: hop: <sip:7474XXXXXXX@213.170.92.166;lr=on;ftag=1c1303544129;ldi=6b73.4744;tfs=AAAAABICCQ0ADwkGCglzAXVdUUAARVddXlpcXS5jb206NTA2MA-->
list_route: hop: <sip:213.170.100.150;lr;ftag=1c1303544129;did=6b7.c2f1f0e2>

<--- Transmitting (NAT) to 213.170.92.166:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 213.170.92.166;branch=z9hG4bKde7f.078734efca238c1b852ba1b4f13ed1d4.0;received=213.170.92.166;rport=5060
Via: SIP/2.0/UDP 213.170.100.150:5060;branch=z9hG4bKde7f.534c8df3.0
Via: SIP/2.0/UDP 95.80.121.253;branch=z9hG4bKac1303550993
Record-Route: <sip:7474XXXXXXX@213.170.92.166;lr=on;ftag=1c1303544129;ldi=6b73.4744;tfs=AAAAABICCQ0ADwkGCglzAXVdUUAARVddXlpcXS5jb206NTA2MA-->
Record-Route: <sip:213.170.100.150;lr;ftag=1c1303544129;did=6b7.c2f1f0e2>
From: +7950XXXXXXX <sip:+7950XXXXXXX@sip.telphin.com:5060>;tag=1c1303544129
To: <sip:474XXXXXXX@213.170.100.150;user=phone>
Call-ID: 1303543094154200291054@95.80.121.253
CSeq: 1 INVITE
Server: Asterisk PBX 11.13.1~dfsg-2+b1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:000127483@XXX.XXX.XX.XXX:5060>
Content-Length: 0

<------------>
-- Executing [000127483@ctx-department-high:1] Dial("SIP/telphin-00000cc0", "SIP/111&SIP/112&SIP/113&SIP/114&SIP/115&SIP/116&SIP/117&SIP/118&SIP/119&SIP/121&SIP/122&SIP/123&SIP/124&SIP/125&SIP/126&SIP/127&SIP/128&SIP/129&SIP/501&SIP/502,40,gtTM(record^000127483)") in new stack
-- Called SIP/502
-- SIP/502-00000cc8 is ringing

<--- Transmitting (NAT) to 213.170.92.166:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 213.170.92.166;branch=z9hG4bKde7f.078734efca238c1b852ba1b4f13ed1d4.0;received=213.170.92.166;rport=5060
Via: SIP/2.0/UDP 213.170.100.150:5060;branch=z9hG4bKde7f.534c8df3.0
Via: SIP/2.0/UDP 95.80.121.253;branch=z9hG4bKac1303550993
Record-Route: <sip:7474XXXXXXX@213.170.92.166;lr=on;ftag=1c1303544129;ldi=6b73.4744;tfs=AAAAABICCQ0ADwkGCglzAXVdUUAARVddXlpcXS5jb206NTA2MA-->
Record-Route: <sip:213.170.100.150;lr;ftag=1c1303544129;did=6b7.c2f1f0e2>
From: +7950XXXXXXX <sip:+7950XXXXXXX@sip.telphin.com:5060>;tag=1c1303544129
To: <sip:474XXXXXXX@213.170.100.150;user=phone>;tag=as679ce487
Call-ID: 1303543094154200291054@95.80.121.253
CSeq: 1 INVITE
Server: Asterisk PBX 11.13.1~dfsg-2+b1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:000127483@XXX.XXX.XX.XXX:5060>
Content-Length: 0

<------------>
-- SIP/502-00000cc8 connected line has changed. Saving it until answer for SIP/telphin-00000cc0
-- SIP/502-00000cc8 answered SIP/telphin-00000cc0
-- Executing [s@macro-record:1] Set("SIP/502-00000cc8", "filename=/var/records/20160704224317") in new stack
-- Executing [s@macro-record:2] Set("SIP/502-00000cc8", "AUDIOHOOK_INHERIT(MixMonitor)=yes") in new stack
-- Executing [s@macro-record:3] Set("SIP/502-00000cc8", "CDR(userfield)=20160704224317") in new stack
-- Executing [s@macro-record:4] MixMonitor("SIP/502-00000cc8", "/var/records/20160704224317.wav,b,/usr/bin/lame -S -V2 /var/records/20160704224317.wav /var/records/20160704224317.mp3 && rm -f /var/records/20160704224317.wav && chmod 766 /var/records/20160704224317.mp3") in new stack
== Begin MixMonitor Recording SIP/502-00000cc8
-- Executing [s@macro-record:5] Set("SIP/502-00000cc8", "AUDIOHOOK_INHERIT(MixMonitor)=yes") in new stack
-- Executing [s@macro-record:6] MacroExit("SIP/502-00000cc8", "") in new stack
Audio is at 16248
Adding codec 100004 (alaw) to SDP
Adding codec 100003 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 213.170.92.166:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 213.170.92.166;branch=z9hG4bKde7f.078734efca238c1b852ba1b4f13ed1d4.0;received=213.170.92.166;rport=5060
Via: SIP/2.0/UDP 213.170.100.150:5060;branch=z9hG4bKde7f.534c8df3.0
Via: SIP/2.0/UDP 95.80.121.253;branch=z9hG4bKac1303550993
Record-Route: <sip:7474XXXXXXX@213.170.92.166;lr=on;ftag=1c1303544129;ldi=6b73.4744;tfs=AAAAABICCQ0ADwkGCglzAXVdUUAARVddXlpcXS5jb206NTA2MA-->
Record-Route: <sip:213.170.100.150;lr;ftag=1c1303544129;did=6b7.c2f1f0e2>
From: +7950XXXXXXX <sip:+7950XXXXXXX@sip.telphin.com:5060>;tag=1c1303544129
To: <sip:474XXXXXXX@213.170.100.150;user=phone>;tag=as679ce487
Call-ID: 1303543094154200291054@95.80.121.253
CSeq: 1 INVITE
Server: Asterisk PBX 11.13.1~dfsg-2+b1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:000127483@XXX.XXX.XX.XXX:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 274

v=0
o=root 1444871650 1444871650 IN IP4 XXX.XXX.XX.XXX
s=Asterisk PBX 11.13.1~dfsg-2+b1
c=IN IP4 XXX.XXX.XX.XXX
t=0 0
m=audio 16248 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=ptime:20
a=sendrecv

<------------>
Retransmitting #1 (NAT) to 213.170.92.166:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 213.170.92.166;branch=z9hG4bKde7f.078734efca238c1b852ba1b4f13ed1d4.0;received=213.170.92.166;rport=5060
Via: SIP/2.0/UDP 213.170.100.150:5060;branch=z9hG4bKde7f.534c8df3.0
Via: SIP/2.0/UDP 95.80.121.253;branch=z9hG4bKac1303550993
Record-Route: <sip:7474XXXXXXX@213.170.92.166;lr=on;ftag=1c1303544129;ldi=6b73.4744;tfs=AAAAABICCQ0ADwkGCglzAXVdUUAARVddXlpcXS5jb206NTA2MA-->
Record-Route: <sip:213.170.100.150;lr;ftag=1c1303544129;did=6b7.c2f1f0e2>
From: +7950XXXXXXX <sip:+7950XXXXXXX@sip.telphin.com:5060>;tag=1c1303544129
To: <sip:474XXXXXXX@213.170.100.150;user=phone>;tag=as679ce487
Call-ID: 1303543094154200291054@95.80.121.253
CSeq: 1 INVITE
Server: Asterisk PBX 11.13.1~dfsg-2+b1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:000127483@XXX.XXX.XX.XXX:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 274

v=0
o=root 1444871650 1444871650 IN IP4 XXX.XXX.XX.XXX
s=Asterisk PBX 11.13.1~dfsg-2+b1
c=IN IP4 XXX.XXX.XX.XXX
t=0 0
m=audio 16248 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=ptime:20
a=sendrecv

---

<--- SIP read from UDP:213.170.92.166:5060 --->
ACK sip:000127483@192.168.200.9:5060 SIP/2.0
Record-Route: <sip:000127483@213.170.92.166;lr=on;ftag=1c1303544129>
Record-Route: <sip:213.170.100.150;lr;ftag=1c1303544129>
Via: SIP/2.0/UDP 213.170.92.166;branch=z9hG4bKde7f.6a94b77b5bf6115e3bde3fae286f60e1.0
Via: SIP/2.0/UDP 213.170.100.150:5060;branch=z9hG4bKde7f.534c8df3.2
Via: SIP/2.0/UDP 95.80.121.253;rport=5060;received=95.80.121.253;branch=z9hG4bKac1324580769
Max-Forwards: 69
From: <sip:+7950XXXXXXX@sip.telphin.com:5060>;tag=1c1303544129
To: <sip:474XXXXXXX@213.170.100.150;user=phone>;tag=as679ce487
Call-ID: 1303543094154200291054@95.80.121.253
CSeq: 1 ACK
Contact: <sip:950XXXXXXX@95.80.121.253:5060>
Supported: em,timer,replaces,path,early-session,resource-priority
Content-Length: 0

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

<--- SIP read from UDP:213.170.92.166:5060 --->
ACK sip:000127483@192.168.200.9:5060 SIP/2.0
Record-Route: <sip:000127483@213.170.92.166;lr=on;ftag=1c1303544129>
Record-Route: <sip:213.170.100.150;lr;ftag=1c1303544129>
Via: SIP/2.0/UDP 213.170.92.166;branch=z9hG4bKde7f.6a94b77b5bf6115e3bde3fae286f60e1.0
Via: SIP/2.0/UDP 213.170.100.150:5060;branch=z9hG4bKde7f.534c8df3.2
Via: SIP/2.0/UDP 95.80.121.253;rport=5060;received=95.80.121.253;branch=z9hG4bKac1324580769
Max-Forwards: 69
From: <sip:+7950XXXXXXX@sip.telphin.com:5060>;tag=1c1303544129
To: <sip:474XXXXXXX@213.170.100.150;user=phone>;tag=as679ce487
Call-ID: 1303543094154200291054@95.80.121.253
CSeq: 1 ACK
Contact: <sip:950XXXXXXX@95.80.121.253:5060>
Supported: em,timer,replaces,path,early-session,resource-priority
Content-Length: 0

<------------->
Scheduling destruction of SIP dialog '1303543094154200291054@95.80.121.253' in 6400 ms (Method: ACK)
set_destination: Parsing <sip:7474XXXXXXX@213.170.92.166;lr=on;ftag=1c1303544129;ldi=6b73.4744;tfs=AAAAABICCQ0ADwkGCglzAXVdUUAARVddXlpcXS5jb206NTA2MA--> for address/port to send to
set_destination: set destination to 213.170.92.166:5060
Reliably Transmitting (NAT) to 213.170.92.166:5060:
BYE sip:950XXXXXXX@95.80.121.253:5060 SIP/2.0
Via: SIP/2.0/UDP XXX.XXX.XX.XXX:5060;branch=z9hG4bK1ddebf58;rport
Route: <sip:7474XXXXXXX@213.170.92.166;lr=on;ftag=1c1303544129;ldi=6b73.4744;tfs=AAAAABICCQ0ADwkGCglzAXVdUUAARVddXlpcXS5jb206NTA2MA-->,<sip:213.170.100.150;lr;ftag=1c1303544129;did=6b7.c2f1f0e2>
Max-Forwards: 70
From: <sip:474XXXXXXX@213.170.100.150;user=phone>;tag=as679ce487
To: +7950XXXXXXX <sip:+7950XXXXXXX@sip.telphin.com:5060>;tag=1c1303544129
Call-ID: 1303543094154200291054@95.80.121.253
CSeq: 102 BYE
User-Agent: Asterisk PBX 11.13.1~dfsg-2+b1
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER: CLI звука нет
<--- SIP read from UDP:213.170.92.166:5060 --->
INVITE sip:000127483@192.168.200.9:5060 SIP/2.0
Record-Route: <sip:7474XXXXXXX@213.170.92.166;lr=on;ftag=1c1343176331;ldi=b211.751;tfs=AAAAABICCQ0ADwkGCglzAXVdUUAARVddXlpcXS5jb206NTA2MA-->
Record-Route: <sip:213.170.100.150;lr;ftag=1c1343176331;did=b21.1f5fa7a3>
Via: SIP/2.0/UDP 213.170.92.166;branch=z9hG4bK261d.2ad58747da90635f67c71249e67ff499.0
Via: SIP/2.0/UDP 213.170.100.150:5060;branch=z9hG4bK261d.41588592.0
Via: SIP/2.0/UDP 95.80.121.253;branch=z9hG4bKac1343183205
Max-Forwards: 69
From: +7950XXXXXXX <sip:+7950XXXXXXX@sip.telphin.com:5060>;tag=1c1343176331
To: <sip:474XXXXXXX@213.170.100.150;user=phone>
P-Preferred-Identity: <sip:+7950XXXXXXX@sip.telphin.com:5060>
Call-ID: 1343175276154200292357@95.80.121.253
CSeq: 1 INVITE
Contact: <sip:950XXXXXXX@95.80.121.253:5060>
Supported: em,100rel,timer,replaces,path,early-session,resource-priority,sdp-anat
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 306

v=0
o=AudiocodesGW 1343169676 1343169346 IN IP4 95.80.121.253
s=Phone-Call
c=IN IP4 95.80.121.253
t=0 0
m=audio 7110 RTP/AVP 8 0 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
<------------->
--- (17 headers 14 lines) ---
Sending to 213.170.92.166:5060 (no NAT)
Sending to 213.170.92.166:5060 (no NAT)
Using INVITE request as basis request - 1343175276154200292357@95.80.121.253
Found peer 'telphin' for '+7950XXXXXXX' from 213.170.92.166:5060
== Using SIP RTP CoS mark 5
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw), peer - audio=(ulaw|alaw|g729)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 95.80.121.253:7110
Looking for 000127483 in ctx-department-high (domain 192.168.200.9)
list_route: hop: <sip:7474XXXXXXX@213.170.92.166;lr=on;ftag=1c1343176331;ldi=b211.751;tfs=AAAAABICCQ0ADwkGCglzAXVdUUAARVddXlpcXS5jb206NTA2MA-->
list_route: hop: <sip:213.170.100.150;lr;ftag=1c1343176331;did=b21.1f5fa7a3>

<--- Transmitting (NAT) to 213.170.92.166:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 213.170.92.166;branch=z9hG4bK261d.2ad58747da90635f67c71249e67ff499.0;received=213.170.92.166;rport=5060
Via: SIP/2.0/UDP 213.170.100.150:5060;branch=z9hG4bK261d.41588592.0
Via: SIP/2.0/UDP 95.80.121.253;branch=z9hG4bKac1343183205
Record-Route: <sip:7474XXXXXXX@213.170.92.166;lr=on;ftag=1c1343176331;ldi=b211.751;tfs=AAAAABICCQ0ADwkGCglzAXVdUUAARVddXlpcXS5jb206NTA2MA-->
Record-Route: <sip:213.170.100.150;lr;ftag=1c1343176331;did=b21.1f5fa7a3>
From: +7950XXXXXXX <sip:+7950XXXXXXX@sip.telphin.com:5060>;tag=1c1343176331
To: <sip:474XXXXXXX@213.170.100.150;user=phone>
Call-ID: 1343175276154200292357@95.80.121.253
CSeq: 1 INVITE
Server: Asterisk PBX 11.13.1~dfsg-2+b1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:000127483@XXX.XXX.XX.XXX:5060>
Content-Length: 0

<------------>
-- Executing [000127483@ctx-department-high:1] Dial("SIP/telphin-00000d62", "SIP/111&SIP/112&SIP/113&SIP/114&SIP/115&SIP/116&SIP/117&SIP/118&SIP/119&SIP/121&SIP/122&SIP/123&SIP/124&SIP/125&SIP/126&SIP/127&SIP/128&SIP/129&SIP/501&SIP/502,40,gtTM(record^000127483)") in new stack
-- Called SIP/502
-- SIP/502-00000d6a is ringing

<--- Transmitting (NAT) to 213.170.92.166:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 213.170.92.166;branch=z9hG4bK261d.2ad58747da90635f67c71249e67ff499.0;received=213.170.92.166;rport=5060
Via: SIP/2.0/UDP 213.170.100.150:5060;branch=z9hG4bK261d.41588592.0
Via: SIP/2.0/UDP 95.80.121.253;branch=z9hG4bKac1343183205
Record-Route: <sip:7474XXXXXXX@213.170.92.166;lr=on;ftag=1c1343176331;ldi=b211.751;tfs=AAAAABICCQ0ADwkGCglzAXVdUUAARVddXlpcXS5jb206NTA2MA-->
Record-Route: <sip:213.170.100.150;lr;ftag=1c1343176331;did=b21.1f5fa7a3>
From: +7950XXXXXXX <sip:+7950XXXXXXX@sip.telphin.com:5060>;tag=1c1343176331
To: <sip:474XXXXXXX@213.170.100.150;user=phone>;tag=as1285b3ce
Call-ID: 1343175276154200292357@95.80.121.253
CSeq: 1 INVITE
Server: Asterisk PBX 11.13.1~dfsg-2+b1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:000127483@XXX.XXX.XX.XXX:5060>
Content-Length: 0

<------------>
-- SIP/502-00000d6a connected line has changed. Saving it until answer for SIP/telphin-00000d62
-- SIP/502-00000d6a answered SIP/telphin-00000d62
-- Executing [s@macro-record:1] Set("SIP/502-00000d6a", "filename=/var/records/20160704225620") in new stack
-- Executing [s@macro-record:2] Set("SIP/502-00000d6a", "AUDIOHOOK_INHERIT(MixMonitor)=yes") in new stack
-- Executing [s@macro-record:3] Set("SIP/502-00000d6a", "CDR(userfield)=20160704225620") in new stack
-- Executing [s@macro-record:4] MixMonitor("SIP/502-00000d6a", "/var/records/20160704225620.wav,b,/usr/bin/lame -S -V2 /var/records/20160704225620.wav /var/records/20160704225620.mp3 && rm -f /var/records/20160704225620.wav && chmod 766 /var/records/20160704225620.mp3") in new stack
== Begin MixMonitor Recording SIP/502-00000d6a
-- Executing [s@macro-record:5] Set("SIP/502-00000d6a", "AUDIOHOOK_INHERIT(MixMonitor)=yes") in new stack
-- Executing [s@macro-record:6] MacroExit("SIP/502-00000d6a", "") in new stack
Audio is at 14898
Adding codec 100004 (alaw) to SDP
Adding codec 100003 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 213.170.92.166:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 213.170.92.166;branch=z9hG4bK261d.2ad58747da90635f67c71249e67ff499.0;received=213.170.92.166;rport=5060
Via: SIP/2.0/UDP 213.170.100.150:5060;branch=z9hG4bK261d.41588592.0
Via: SIP/2.0/UDP 95.80.121.253;branch=z9hG4bKac1343183205
Record-Route: <sip:7474XXXXXXX@213.170.92.166;lr=on;ftag=1c1343176331;ldi=b211.751;tfs=AAAAABICCQ0ADwkGCglzAXVdUUAARVddXlpcXS5jb206NTA2MA-->
Record-Route: <sip:213.170.100.150;lr;ftag=1c1343176331;did=b21.1f5fa7a3>
From: +7950XXXXXXX <sip:+7950XXXXXXX@sip.telphin.com:5060>;tag=1c1343176331
To: <sip:474XXXXXXX@213.170.100.150;user=phone>;tag=as1285b3ce
Call-ID: 1343175276154200292357@95.80.121.253
CSeq: 1 INVITE
Server: Asterisk PBX 11.13.1~dfsg-2+b1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:000127483@XXX.XXX.XX.XXX:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 274

v=0
o=root 1706338222 1706338222 IN IP4 XXX.XXX.XX.XXX
s=Asterisk PBX 11.13.1~dfsg-2+b1
c=IN IP4 XXX.XXX.XX.XXX
t=0 0
m=audio 14898 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=ptime:20
a=sendrecv

<------------>
<--- SIP read from UDP:213.170.92.166:5060 --->
ACK sip:000127483@192.168.200.9:5060 SIP/2.0
Record-Route: <sip:000127483@213.170.92.166;lr=on;ftag=1c1343176331>
Record-Route: <sip:213.170.100.150;lr;ftag=1c1343176331>
Via: SIP/2.0/UDP 213.170.92.166;branch=z9hG4bK261d.a1371b7c9870a085a3f90cc57541128c.0
Via: SIP/2.0/UDP 213.170.100.150:5060;branch=z9hG4bK261d.41588592.2
Via: SIP/2.0/UDP 95.80.121.253;rport=5060;received=95.80.121.253;branch=z9hG4bKac1362168175
Max-Forwards: 69
From: <sip:+7950XXXXXXX@sip.telphin.com:5060>;tag=1c1343176331
To: <sip:474XXXXXXX@213.170.100.150;user=phone>;tag=as1285b3ce
Call-ID: 1343175276154200292357@95.80.121.253
CSeq: 1 ACK
Contact: <sip:950XXXXXXX@95.80.121.253:5060>
Supported: em,timer,replaces,path,early-session,resource-priority
Content-Length: 0

<------------->
<--- SIP read from UDP:213.170.92.166:5060 --->
BYE sip:000127483@192.168.200.9:5060 SIP/2.0
Record-Route: <sip:000127483@213.170.92.166;lr=on;ftag=1c1343176331>
Record-Route: <sip:213.170.100.150;lr;ftag=1c1343176331>
Via: SIP/2.0/UDP 213.170.92.166;branch=z9hG4bKf51d.e0261dccd5ff05e9a0fc28cb878606b9.0
Via: SIP/2.0/UDP 213.170.100.150:5060;branch=z9hG4bKf51d.65332993.0
Via: SIP/2.0/UDP 95.80.121.253;rport=5060;received=95.80.121.253;branch=z9hG4bKac1585705916
Max-Forwards: 69
From: <sip:+7950XXXXXXX@sip.telphin.com:5060>;tag=1c1343176331
To: <sip:474XXXXXXX@213.170.100.150;user=phone>;tag=as1285b3ce
Call-ID: 1343175276154200292357@95.80.121.253
CSeq: 2 BYE
Supported: em,timer,replaces,path,early-session,resource-priority
Reason: Q.850 ;cause=16
Content-Length: 0

Re: плавающая проблема со звуком

Добавлено: 05 июл 2016, 09:58
april22
Peer audio RTP is at port 95.80.121.253:7110
мож сюда поглядеть ?

Re: плавающая проблема со звуком

Добавлено: 06 июл 2016, 16:06
alexkab
мож сюда поглядеть ?
Не понял Вас. Что Вы имеете ввиду ??
У меня на астериске в rtp.conf разрешены 10000-20000. В дебаге где звук есть - открывается порт 6040
Peer audio RTP is at port 95.80.121.253:6040
. На маршрутизаторе полный нат, никаких ограничений нет. Кстати дебаг из консоли и вышерасположенные скрины - это разные вызовы.

Re: плавающая проблема со звуком

Добавлено: 06 июл 2016, 16:23
ded
ded писал(а):1) Надо обращаться к разработчиками Рарус.

Re: плавающая проблема со звуком

Добавлено: 06 июл 2016, 16:36
alexkab
Уважаемый ded, я извиняюсь за свою назойливость, ибо хочется немножко понимания. Почему нужно обращаться именно в рарус, а не в телфин ? (как вы это определили и что в таком случае им говорить), ведь из вышеуказанного видно rtp не бегут от 95.80.121.253. Поясните плиз.

Re: плавающая проблема со звуком

Добавлено: 06 июл 2016, 16:39
ded
alexkab писал(а):При использовании 3cxphone и железного телефона, такой проблемы не было.
Часовой пояс: UTC+03:00
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