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Условный перевод без голоса

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

Модераторы: april22, Zavr2008

Re: Условный перевод без голоса

Сообщение carassin » 21 мар 2017, 11:01

fecyt писал(а):Почему вызывается эта мелодия, может где-то в контексте это происходит и т.д. и т.п.

И еще одна правка, вполне возможно, что это реакция на 180 Ringing от Вашего телефона, т.к. там нет sdp, то Астериск вызывает мелодию и тем самым эмулирует гудки от аппарата. А т.к. для А абонента это одна медиа сессия, он слушает эти гудки :)


Я так полагаю что мелодия вызывается из features.conf благодаря опции xfersound, которая закоментирована. Сейчас beep не вызывается (xfersound = ), но по прежнему слышу хрипые гудки.

А нельзя ли как-то попросить Asterisk по другому реагировать на 180 Ringing без sdp ? Или он по другому не может ?
carassin
 
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Re: Условный перевод без голоса

Сообщение ded » 21 мар 2017, 11:17

Google: asterisk sip trying 180 ringing 183 session progress
ded
 
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Re: Условный перевод без голоса

Сообщение fecyt » 21 мар 2017, 15:13

carassin писал(а):А нельзя ли как-то попросить Asterisk по другому реагировать на 180 Ringing без sdp ? Или он по другому не может ?

Это было предположение, и я могу быть не прав. По факту, у Вас создается два rtp потока, один с гудками, другой с Вашим голосом от телефона, все накладывается и получается искажение.
Реакция ли это на 180 Ringing или просто вызов мелодии, при переводе, нужно, полагаю, смотреть в дебаге астериска. Выяснив, что это, проще будет найти решение.
fecyt
 
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Re: Условный перевод без голоса

Сообщение fecyt » 21 мар 2017, 15:29

Пожажите, пожалуйста, на текущий момент вывод в консоль проблемного звонка.
fecyt
 
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Re: Условный перевод без голоса

Сообщение fecyt » 21 мар 2017, 16:01

Потестировал сейчас на виртуалке, жаль мирокофона нет. И да, мелодия вызывается из-за xfersound = beep, ну вот Вы ее отключили, и интересно, создается второй rtp поток и с чем он идет от астера. Кидайте заодно и pcap.
fecyt
 
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Re: Условный перевод без голоса

Сообщение ded » 21 мар 2017, 16:05

fecyt писал(а):Потестировал сейчас на виртуалке, жаль мирокофона нет.
Микрофон то у Вас должен быть тоже виртуальный!
ded
 
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Re: Условный перевод без голоса

Сообщение carassin » 21 мар 2017, 16:34

fecyt, спасибо что помогаете

[Показать] Спойлер: sip set debug on
<--- SIP read from UDP:192.168.5.222:5060 --->
INVITE sip:22@192.168.9.22;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.5.222:5060;branch=z9hG4bK324563163311103692
From: 11 <sip:11@192.168.9.22:5060>;tag=2595919630
To: "22" <sip:22@192.168.9.22;user=phone>
Call-ID: 3971215423117-26893168827707@192.168.5.222
CSeq: 1 INVITE
Contact: <sip:11@192.168.5.222:5060>
Max-Forwards: 70
Supported: replaces, join, path
User-Agent: DLINK DPH-150S FRU2.2.162.67
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE
Content-Type: application/sdp
Content-Length: 214

v=0
o=2000 2769618127 2034412529 IN IP4 192.168.5.222
s=A conversation
c=IN IP4 192.168.5.222
t=0 0
m=audio 10160 RTP/AVP 0 8 18
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=sendrecv
<------------->
--- (13 headers 10 lines) ---
Sending to 192.168.5.222:5060 (no NAT)
Sending to 192.168.5.222:5060 (no NAT)
Using INVITE request as basis request - 3971215423117-26893168827707@192.168.5.222
Found peer '11' for '11' from 192.168.5.222:5060
== Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Capabilities: us - (ulaw|alaw|ilbc), peer - audio=(ulaw|alaw|g729)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 192.168.5.222:10160
Looking for 22 in krs-main (domain 192.168.9.22)
list_route: hop: <sip:11@192.168.5.222:5060>

<--- Transmitting (no NAT) to 192.168.5.222:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.5.222:5060;branch=z9hG4bK324563163311103692;received=192.168.5.222
From: 11 <sip:11@192.168.9.22:5060>;tag=2595919630
To: "22" <sip:22@192.168.9.22;user=phone>
Call-ID: 3971215423117-26893168827707@192.168.5.222
CSeq: 1 INVITE
Server: Asterisk PBX 11.13.1~dfsg-2+deb8u2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:22@192.168.9.22:5060>
Content-Length: 0


<------------>
-- Executing [22@krs-main:1] Dial("SIP/11-00000016", "SIP/22,10,Tt") in new stack
== Using SIP RTP CoS mark 5
Audio is at 10730
Adding codec 100004 (alaw) to SDP
Adding codec 100003 (ulaw) to SDP
Adding codec 100010 (ilbc) to SDP
Reliably Transmitting (no NAT) to 192.168.8.160:5060:
INVITE sip:22@192.168.8.160:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.9.22:5060;branch=z9hG4bK296dfbc9
Max-Forwards: 70
From: "11" <sip:11@192.168.9.22>;tag=as5e717be8
To: <sip:22@192.168.8.160:5060>
Contact: <sip:11@192.168.9.22:5060>
Call-ID: 67713f763adac84a5fcc3a3a455e68fe@192.168.9.22:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.13.1~dfsg-2+deb8u2
Date: Tue, 21 Mar 2017 16:25:30 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 263

v=0
o=root 1011778363 1011778363 IN IP4 192.168.9.22
s=Asterisk PBX 11.13.1~dfsg-2+deb8u2
c=IN IP4 192.168.9.22
t=0 0
m=audio 10730 RTP/AVP 8 0 97
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=ptime:20
a=sendrecv

---
-- Called SIP/22

<--- SIP read from UDP:192.168.8.160:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.9.22:5060;branch=z9hG4bK296dfbc9
From: "11" <sip:11@192.168.9.22>;tag=as5e717be8
To: <sip:22@192.168.8.160:5060>
Call-ID: 67713f763adac84a5fcc3a3a455e68fe@192.168.9.22:5060
CSeq: 102 INVITE
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE
Content-Length: 0

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

<--- SIP read from UDP:192.168.8.160:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.9.22:5060;branch=z9hG4bK296dfbc9
From: "11" <sip:11@192.168.9.22>;tag=as5e717be8
To: <sip:22@192.168.8.160:5060>;tag=175419759
Call-ID: 67713f763adac84a5fcc3a3a455e68fe@192.168.9.22:5060
CSeq: 102 INVITE
Contact: <sip:22@192.168.8.160:5060>
User-Agent: DLINK DPH-150S FRU2.2.162.67
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
list_route: hop: <sip:22@192.168.8.160:5060>
-- SIP/22-00000017 is ringing

<--- Transmitting (no NAT) to 192.168.5.222:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.5.222:5060;branch=z9hG4bK324563163311103692;received=192.168.5.222
From: 11 <sip:11@192.168.9.22:5060>;tag=2595919630
To: "22" <sip:22@192.168.9.22;user=phone>;tag=as5bd01027
Call-ID: 3971215423117-26893168827707@192.168.5.222
CSeq: 1 INVITE
Server: Asterisk PBX 11.13.1~dfsg-2+deb8u2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:22@192.168.9.22:5060>
Content-Length: 0


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

<--- SIP read from UDP:192.168.8.160:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.9.22:5060;branch=z9hG4bK296dfbc9
From: "11" <sip:11@192.168.9.22>;tag=as5e717be8
To: <sip:22@192.168.8.160:5060>;tag=175419759
Call-ID: 67713f763adac84a5fcc3a3a455e68fe@192.168.9.22:5060
CSeq: 102 INVITE
Contact: <sip:22@192.168.8.160:5060>
Supported: 100rel, replaces, timer
User-Agent: DLINK DPH-150S FRU2.2.162.67
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE
Content-Type: application/sdp
Content-Length: 185

v=0
o=22 1625720524 900112426 IN IP4 192.168.8.160
s=A conversation
c=IN IP4 192.168.8.160
t=0 0
m=audio 10112 RTP/AVP 8 0
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=sendrecv
<------------->
--- (12 headers 9 lines) ---
Found RTP audio format 8
Found RTP audio format 0
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Capabilities: us - (ulaw|alaw|ilbc), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 192.168.8.160:10112
list_route: hop: <sip:22@192.168.8.160:5060>
set_destination: Parsing <sip:22@192.168.8.160:5060> for address/port to send to
set_destination: set destination to 192.168.8.160:5060
Transmitting (no NAT) to 192.168.8.160:5060:
ACK sip:22@192.168.8.160:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.9.22:5060;branch=z9hG4bK1b15c00c
Max-Forwards: 70
From: "11" <sip:11@192.168.9.22>;tag=as5e717be8
To: <sip:22@192.168.8.160:5060>;tag=175419759
Contact: <sip:11@192.168.9.22:5060>
Call-ID: 67713f763adac84a5fcc3a3a455e68fe@192.168.9.22:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.13.1~dfsg-2+deb8u2
Content-Length: 0


---
-- SIP/22-00000017 answered SIP/11-00000016
Audio is at 14032
Adding codec 100004 (alaw) to SDP
Adding codec 100003 (ulaw) to SDP

<--- Reliably Transmitting (no NAT) to 192.168.5.222:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.5.222:5060;branch=z9hG4bK324563163311103692;received=192.168.5.222
From: 11 <sip:11@192.168.9.22:5060>;tag=2595919630
To: "22" <sip:22@192.168.9.22;user=phone>;tag=as5bd01027
Call-ID: 3971215423117-26893168827707@192.168.5.222
CSeq: 1 INVITE
Server: Asterisk PBX 11.13.1~dfsg-2+deb8u2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:22@192.168.9.22:5060>
Content-Type: application/sdp
Content-Length: 216

v=0
o=root 484034046 484034046 IN IP4 192.168.9.22
s=Asterisk PBX 11.13.1~dfsg-2+deb8u2
c=IN IP4 192.168.9.22
t=0 0
m=audio 14032 RTP/AVP 8 0
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=ptime:20
a=sendrecv

<------------>
-- Locally bridging SIP/11-00000016 and SIP/22-00000017

<--- SIP read from UDP:192.168.5.222:5060 --->
ACK sip:22@192.168.9.22:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.222:5060;branch=z9hG4bK3223039002477226014
From: 11 <sip:11@192.168.9.22:5060>;tag=2595919630
To: "22" <sip:22@192.168.9.22;user=phone>;tag=as5bd01027
Call-ID: 3971215423117-26893168827707@192.168.5.222
CSeq: 1 ACK
Contact: <sip:11@192.168.5.222:5060>
Max-Forwards: 70
User-Agent: DLINK DPH-150S FRU2.2.162.67
Content-Length: 0

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

<--- SIP read from UDP:192.168.8.160:5060 --->
INFO sip:11@192.168.9.22:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.8.160:5060;branch=z9hG4bK1634484642342132707
From: <sip:22@192.168.8.160:5060>;tag=175419759
To: "11" <sip:11@192.168.9.22>;tag=as5e717be8
Call-ID: 67713f763adac84a5fcc3a3a455e68fe@192.168.9.22:5060
CSeq: 1 INFO
Max-Forwards: 70
User-Agent: DLINK DPH-150S FRU2.2.162.67
Content-Type: application/dtmf-relay
Content-Length: 25

Signal=10
Duration=160
<------------->
--- (10 headers 2 lines) ---
Receiving INFO!
* DTMF-relay event received: *

<--- Transmitting (no NAT) to 192.168.8.160:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.8.160:5060;branch=z9hG4bK1634484642342132707;received=192.168.8.160
From: <sip:22@192.168.8.160:5060>;tag=175419759
To: "11" <sip:11@192.168.9.22>;tag=as5e717be8
Call-ID: 67713f763adac84a5fcc3a3a455e68fe@192.168.9.22:5060
CSeq: 1 INFO
Server: Asterisk PBX 11.13.1~dfsg-2+deb8u2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
-- Locally bridging SIP/11-00000016 and SIP/22-00000017

<--- SIP read from UDP:192.168.8.160:5060 --->
INFO sip:11@192.168.9.22:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.8.160:5060;branch=z9hG4bK32647130832637723941
From: <sip:22@192.168.8.160:5060>;tag=175419759
To: "11" <sip:11@192.168.9.22>;tag=as5e717be8
Call-ID: 67713f763adac84a5fcc3a3a455e68fe@192.168.9.22:5060
CSeq: 2 INFO
Max-Forwards: 70
User-Agent: DLINK DPH-150S FRU2.2.162.67
Content-Type: application/dtmf-relay
Content-Length: 24

Signal=2
Duration=160
<------------->
--- (10 headers 2 lines) ---
Receiving INFO!
* DTMF-relay event received: 2

<--- Transmitting (no NAT) to 192.168.8.160:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.8.160:5060;branch=z9hG4bK32647130832637723941;received=192.168.8.160
From: <sip:22@192.168.8.160:5060>;tag=175419759
To: "11" <sip:11@192.168.9.22>;tag=as5e717be8
Call-ID: 67713f763adac84a5fcc3a3a455e68fe@192.168.9.22:5060
CSeq: 2 INFO
Server: Asterisk PBX 11.13.1~dfsg-2+deb8u2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
-- Started music on hold, class 'default', on SIP/11-00000016
-- <SIP/22-00000017> Playing 'pbx-transfer.gsm' (language 'ru')

<--- SIP read from UDP:192.168.8.160:5060 --->
INFO sip:11@192.168.9.22:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.8.160:5060;branch=z9hG4bK162164696126134358
From: <sip:22@192.168.8.160:5060>;tag=175419759
To: "11" <sip:11@192.168.9.22>;tag=as5e717be8
Call-ID: 67713f763adac84a5fcc3a3a455e68fe@192.168.9.22:5060
CSeq: 3 INFO
Max-Forwards: 70
User-Agent: DLINK DPH-150S FRU2.2.162.67
Content-Type: application/dtmf-relay
Content-Length: 24

Signal=3
Duration=160
<------------->
--- (10 headers 2 lines) ---
Receiving INFO!
* DTMF-relay event received: 3

<--- Transmitting (no NAT) to 192.168.8.160:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.8.160:5060;branch=z9hG4bK162164696126134358;received=192.168.8.160
From: <sip:22@192.168.8.160:5060>;tag=175419759
To: "11" <sip:11@192.168.9.22>;tag=as5e717be8
Call-ID: 67713f763adac84a5fcc3a3a455e68fe@192.168.9.22:5060
CSeq: 3 INFO
Server: Asterisk PBX 11.13.1~dfsg-2+deb8u2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


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

<--- SIP read from UDP:192.168.8.160:5060 --->
INFO sip:11@192.168.9.22:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.8.160:5060;branch=z9hG4bK251824994190571893
From: <sip:22@192.168.8.160:5060>;tag=175419759
To: "11" <sip:11@192.168.9.22>;tag=as5e717be8
Call-ID: 67713f763adac84a5fcc3a3a455e68fe@192.168.9.22:5060
CSeq: 4 INFO
Max-Forwards: 70
User-Agent: DLINK DPH-150S FRU2.2.162.67
Content-Type: application/dtmf-relay
Content-Length: 24

Signal=3
Duration=160
<------------->
--- (10 headers 2 lines) ---
Receiving INFO!
* DTMF-relay event received: 3

<--- Transmitting (no NAT) to 192.168.8.160:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.8.160:5060;branch=z9hG4bK251824994190571893;received=192.168.8.160
From: <sip:22@192.168.8.160:5060>;tag=175419759
To: "11" <sip:11@192.168.9.22>;tag=as5e717be8
Call-ID: 67713f763adac84a5fcc3a3a455e68fe@192.168.9.22:5060
CSeq: 4 INFO
Server: Asterisk PBX 11.13.1~dfsg-2+deb8u2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
-- Executing [33@krs-main:1] Dial("Local/33@krs-main-00000004;2", "SIP/33,10,Tt") in new stack
== Using SIP RTP CoS mark 5
Audio is at 15142
Adding codec 100004 (alaw) to SDP
Adding codec 100003 (ulaw) to SDP
Adding codec 100010 (ilbc) to SDP
Reliably Transmitting (no NAT) to 192.168.5.22:5060:
INVITE sip:33@192.168.5.22;line=d2f6e1318e0e0d9 SIP/2.0
Via: SIP/2.0/UDP 192.168.9.22:5060;branch=z9hG4bK2cf76607
Max-Forwards: 70
From: <sip:22@192.168.9.22>;tag=as59f1fac3
To: <sip:33@192.168.5.22;line=d2f6e1318e0e0d9>
Contact: <sip:22@192.168.9.22:5060>
Call-ID: 2cfeb8950e016e2f0800096121174741@192.168.9.22:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.13.1~dfsg-2+deb8u2
Date: Tue, 21 Mar 2017 16:25:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 261

v=0
o=root 313745517 313745517 IN IP4 192.168.9.22
s=Asterisk PBX 11.13.1~dfsg-2+deb8u2
c=IN IP4 192.168.9.22
t=0 0
m=audio 15142 RTP/AVP 8 0 97
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=ptime:20
a=sendrecv

---
-- Called SIP/33

<--- SIP read from UDP:192.168.5.22:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.9.22:5060;branch=z9hG4bK2cf76607
From: <sip:22@192.168.9.22>;tag=as59f1fac3
To: <sip:33@192.168.5.22;line=d2f6e1318e0e0d9>
Call-ID: 2cfeb8950e016e2f0800096121174741@192.168.9.22:5060
CSeq: 102 INVITE
User-Agent: Linphone/3.6.1 (eXosip2/4.1.0)
Content-Length: 0

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

<--- SIP read from UDP:192.168.5.22:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.9.22:5060;branch=z9hG4bK2cf76607
From: <sip:22@192.168.9.22>;tag=as59f1fac3
To: <sip:33@192.168.5.22;line=d2f6e1318e0e0d9>;tag=520096285
Call-ID: 2cfeb8950e016e2f0800096121174741@192.168.9.22:5060
CSeq: 102 INVITE
Contact: <sip:33@192.168.5.22:5060>
User-Agent: Linphone/3.6.1 (eXosip2/4.1.0)
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
list_route: hop: <sip:33@192.168.5.22:5060>
-- SIP/33-00000018 is ringing
-- Local/33@krs-main-00000004;1 is ringing

<--- SIP read from UDP:192.168.5.222:5060 --->
OPTIONS sip:192.168.9.22:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.222:5060;branch=z9hG4bK29904310133066314995
From: 11 <sip:11@192.168.9.22:5060>;tag=858928046
To: <sip:192.168.9.22:5060>
Call-ID: 120241465010541-2008854223503@192.168.5.222
CSeq: 1 OPTIONS
Max-Forwards: 70
User-Agent: DLINK DPH-150S FRU2.2.162.67
Accept: application/sdp
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Sending to 192.168.5.222:5060 (no NAT)
Looking for s in krs-main (domain 192.168.9.22)

<--- Transmitting (no NAT) to 192.168.5.222:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.5.222:5060;branch=z9hG4bK29904310133066314995;received=192.168.5.222
From: 11 <sip:11@192.168.9.22:5060>;tag=858928046
To: <sip:192.168.9.22:5060>;tag=as7e603c42
Call-ID: 120241465010541-2008854223503@192.168.5.222
CSeq: 1 OPTIONS
Server: Asterisk PBX 11.13.1~dfsg-2+deb8u2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '120241465010541-2008854223503@192.168.5.222' in 32000 ms (Method: OPTIONS)

<--- SIP read from UDP:192.168.8.160:5060 --->
BYE sip:11@192.168.9.22:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.8.160:5060;branch=z9hG4bK218101932412491599
From: <sip:22@192.168.8.160:5060>;tag=175419759
To: "11" <sip:11@192.168.9.22>;tag=as5e717be8
Call-ID: 67713f763adac84a5fcc3a3a455e68fe@192.168.9.22:5060
CSeq: 5 BYE
Contact: <sip:22@192.168.8.160:5060>
Max-Forwards: 70
User-Agent: DLINK DPH-150S FRU2.2.162.67
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Sending to 192.168.8.160:5060 (no NAT)
Scheduling destruction of SIP dialog '67713f763adac84a5fcc3a3a455e68fe@192.168.9.22:5060' in 32000 ms (Method: BYE)

<--- Transmitting (no NAT) to 192.168.8.160:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.8.160:5060;branch=z9hG4bK218101932412491599;received=192.168.8.160
From: <sip:22@192.168.8.160:5060>;tag=175419759
To: "11" <sip:11@192.168.9.22>;tag=as5e717be8
Call-ID: 67713f763adac84a5fcc3a3a455e68fe@192.168.9.22:5060
CSeq: 5 BYE
Server: Asterisk PBX 11.13.1~dfsg-2+deb8u2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
-- Nobody picked up in 10000 ms
Scheduling destruction of SIP dialog '2cfeb8950e016e2f0800096121174741@192.168.9.22:5060' in 32000 ms (Method: INVITE)
Reliably Transmitting (no NAT) to 192.168.5.22:5060:
CANCEL sip:33@192.168.5.22;line=d2f6e1318e0e0d9 SIP/2.0
Via: SIP/2.0/UDP 192.168.9.22:5060;branch=z9hG4bK2cf76607
Max-Forwards: 70
From: <sip:22@192.168.9.22>;tag=as59f1fac3
To: <sip:33@192.168.5.22;line=d2f6e1318e0e0d9>
Call-ID: 2cfeb8950e016e2f0800096121174741@192.168.9.22:5060
CSeq: 102 CANCEL
User-Agent: Asterisk PBX 11.13.1~dfsg-2+deb8u2
Content-Length: 0


---
Scheduling destruction of SIP dialog '2cfeb8950e016e2f0800096121174741@192.168.9.22:5060' in 32000 ms (Method: INVITE)
-- Executing [33@krs-main:2] Hangup("Local/33@krs-main-00000004;2", "") in new stack
== Spawn extension (krs-main, 33, 2) exited non-zero on 'Local/33@krs-main-00000004;2'

<--- SIP read from UDP:192.168.5.22:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.9.22:5060;branch=z9hG4bK2cf76607
From: <sip:22@192.168.9.22>;tag=as59f1fac3
To: <sip:33@192.168.5.22;line=d2f6e1318e0e0d9>;tag=520096285
Call-ID: 2cfeb8950e016e2f0800096121174741@192.168.9.22:5060
CSeq: 102 CANCEL
User-Agent: Linphone/3.6.1 (eXosip2/4.1.0)
Content-Length: 0

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

<--- SIP read from UDP:192.168.5.22:5060 --->
SIP/2.0 487 Request Cancelled
Via: SIP/2.0/UDP 192.168.9.22:5060;branch=z9hG4bK2cf76607
From: <sip:22@192.168.9.22>;tag=as59f1fac3
To: <sip:33@192.168.5.22;line=d2f6e1318e0e0d9>;tag=520096285
Call-ID: 2cfeb8950e016e2f0800096121174741@192.168.9.22:5060
CSeq: 102 INVITE
User-Agent: Linphone/3.6.1 (eXosip2/4.1.0)
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Transmitting (no NAT) to 192.168.5.22:5060:
ACK sip:33@192.168.5.22:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.9.22:5060;branch=z9hG4bK2cf76607
Max-Forwards: 70
From: <sip:22@192.168.9.22>;tag=as59f1fac3
To: <sip:33@192.168.5.22;line=d2f6e1318e0e0d9>;tag=520096285
Contact: <sip:22@192.168.9.22:5060>
Call-ID: 2cfeb8950e016e2f0800096121174741@192.168.9.22:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.13.1~dfsg-2+deb8u2
Content-Length: 0


---
Scheduling destruction of SIP dialog '2cfeb8950e016e2f0800096121174741@192.168.9.22:5060' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:192.168.5.22:5060 --->
jaK
<------------->
== Using SIP RTP CoS mark 5
Audio is at 15820
Adding codec 100004 (alaw) to SDP
Adding codec 100003 (ulaw) to SDP
Adding codec 100010 (ilbc) to SDP
Reliably Transmitting (no NAT) to 192.168.8.160:5060:
INVITE sip:22@192.168.8.160:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.9.22:5060;branch=z9hG4bK75f5b75b
Max-Forwards: 70
From: "11" <sip:11@192.168.9.22>;tag=as6900fc43
To: <sip:22@192.168.8.160:5060>
Contact: <sip:11@192.168.9.22:5060>
Call-ID: 7bd45fb35e39e51f5c3473c142e6029a@192.168.9.22:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.13.1~dfsg-2+deb8u2
Date: Tue, 21 Mar 2017 16:25:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 263

v=0
o=root 1190521306 1190521306 IN IP4 192.168.9.22
s=Asterisk PBX 11.13.1~dfsg-2+deb8u2
c=IN IP4 192.168.9.22
t=0 0
m=audio 15820 RTP/AVP 8 0 97
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:192.168.8.160:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.9.22:5060;branch=z9hG4bK75f5b75b
From: "11" <sip:11@192.168.9.22>;tag=as6900fc43
To: <sip:22@192.168.8.160:5060>
Call-ID: 7bd45fb35e39e51f5c3473c142e6029a@192.168.9.22:5060
CSeq: 102 INVITE
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE
Content-Length: 0

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

<--- SIP read from UDP:192.168.8.160:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.9.22:5060;branch=z9hG4bK75f5b75b
From: "11" <sip:11@192.168.9.22>;tag=as6900fc43
To: <sip:22@192.168.8.160:5060>;tag=3092313993
Call-ID: 7bd45fb35e39e51f5c3473c142e6029a@192.168.9.22:5060
CSeq: 102 INVITE
Contact: <sip:22@192.168.8.160:5060>
User-Agent: DLINK DPH-150S FRU2.2.162.67
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
list_route: hop: <sip:22@192.168.8.160:5060>
-- SIP/22-00000019 is ringing

<--- SIP read from UDP:192.168.8.160:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.9.22:5060;branch=z9hG4bK75f5b75b
From: "11" <sip:11@192.168.9.22>;tag=as6900fc43
To: <sip:22@192.168.8.160:5060>;tag=3092313993
Call-ID: 7bd45fb35e39e51f5c3473c142e6029a@192.168.9.22:5060
CSeq: 102 INVITE
Contact: <sip:22@192.168.8.160:5060>
Supported: 100rel, replaces, timer
User-Agent: DLINK DPH-150S FRU2.2.162.67
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE
Content-Type: application/sdp
Content-Length: 185

v=0
o=22 254456053 2854410380 IN IP4 192.168.8.160
s=A conversation
c=IN IP4 192.168.8.160
t=0 0
m=audio 10114 RTP/AVP 8 0
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=sendrecv
<------------->
--- (12 headers 9 lines) ---
Found RTP audio format 8
Found RTP audio format 0
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Capabilities: us - (ulaw|alaw|ilbc), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 192.168.8.160:10114
list_route: hop: <sip:22@192.168.8.160:5060>
set_destination: Parsing <sip:22@192.168.8.160:5060> for address/port to send to
set_destination: set destination to 192.168.8.160:5060
Transmitting (no NAT) to 192.168.8.160:5060:
ACK sip:22@192.168.8.160:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.9.22:5060;branch=z9hG4bK686a3c69
Max-Forwards: 70
From: "11" <sip:11@192.168.9.22>;tag=as6900fc43
To: <sip:22@192.168.8.160:5060>;tag=3092313993
Contact: <sip:11@192.168.9.22:5060>
Call-ID: 7bd45fb35e39e51f5c3473c142e6029a@192.168.9.22:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.13.1~dfsg-2+deb8u2
Content-Length: 0


---
-- Stopped music on hold on SIP/11-00000016
-- Locally bridging SIP/11-00000016 and SIP/22-00000019
== Spawn extension (krs-main, 22, 1) exited non-zero on 'Transfered/SIP/11-00000016<ZOMBIE>'
Really destroying SIP dialog '40231059923169-20423240515183@192.168.8.160' Method: OPTIONS

<--- SIP read from UDP:192.168.5.222:5060 --->
BYE sip:22@192.168.9.22:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.222:5060;branch=z9hG4bK3763163241299491
From: 11 <sip:11@192.168.9.22:5060>;tag=2595919630
To: "22" <sip:22@192.168.9.22;user=phone>;tag=as5bd01027
Call-ID: 3971215423117-26893168827707@192.168.5.222
CSeq: 2 BYE
Contact: <sip:11@192.168.5.222:5060>
Max-Forwards: 70
User-Agent: DLINK DPH-150S FRU2.2.162.67
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Sending to 192.168.5.222:5060 (no NAT)
Scheduling destruction of SIP dialog '3971215423117-26893168827707@192.168.5.222' in 32000 ms (Method: BYE)

<--- Transmitting (no NAT) to 192.168.5.222:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.5.222:5060;branch=z9hG4bK3763163241299491;received=192.168.5.222
From: 11 <sip:11@192.168.9.22:5060>;tag=2595919630
To: "22" <sip:22@192.168.9.22;user=phone>;tag=as5bd01027
Call-ID: 3971215423117-26893168827707@192.168.5.222
CSeq: 2 BYE
Server: Asterisk PBX 11.13.1~dfsg-2+deb8u2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '7bd45fb35e39e51f5c3473c142e6029a@192.168.9.22:5060' in 32000 ms (Method: INVITE)
set_destination: Parsing <sip:22@192.168.8.160:5060> for address/port to send to
set_destination: set destination to 192.168.8.160:5060
Reliably Transmitting (no NAT) to 192.168.8.160:5060:
BYE sip:22@192.168.8.160:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.9.22:5060;branch=z9hG4bK68a8e73a
Max-Forwards: 70
From: "11" <sip:11@192.168.9.22>;tag=as6900fc43
To: <sip:22@192.168.8.160:5060>;tag=3092313993
Call-ID: 7bd45fb35e39e51f5c3473c142e6029a@192.168.9.22:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX 11.13.1~dfsg-2+deb8u2
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---

<--- SIP read from UDP:192.168.8.160:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.9.22:5060;branch=z9hG4bK68a8e73a
From: "11" <sip:11@192.168.9.22>;tag=as6900fc43
To: <sip:22@192.168.8.160:5060>;tag=3092313993
Call-ID: 7bd45fb35e39e51f5c3473c142e6029a@192.168.9.22:5060
CSeq: 103 BYE
User-Agent: DLINK DPH-150S FRU2.2.162.67
Content-Length: 0


https://cloud.mail.ru/public/Pz6s/kHZXTAiy1
carassin
 
Сообщений: 11
Зарегистрирован: 20 июл 2015, 11:46

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