как использовать провайдера телефонии с другого астериска
Re: как использовать провайдера телефонии с другого астериск
если получится то завтра попробую смоделировать ситуацию с freepbx- distro
Своими вопросами , вы загоняете меня в ГУГЛЬ.
Re: как использовать провайдера телефонии с другого астериск
Связь настроил. звонки проходят. дозваниваюсь. все ок. возникает следующая проблема:
каким образом могу передавать определенные параметры: у меня происходит разногласие следующее: звоню с экстеншена 3333 на номер 79046528473, проходит вызов. 2012-09-27 19:21:24 3052 79046528473 SIP/3052-00002f0c SIP/outAlterfone-00002f0d BUSY 0s выдает такое сообщение. все ок. 3052 - эьтто регистрационный аккаунт на elastixe..........потом пью чай. прихожу звоню и у меня 2012-09-27 19:16:22 3333 s SIP/outtest-00002ef4 ANSWERED 4s т.е. присваивается 3333 номер-экстеншен. как мне сделать так, чтобы оставался 3052? исполдьзовать сам этот экстеншен 3052 не вариант в связи с большим количеством таких номеров.
Как у меня решается сейчас: регистрация проходит по 1 аккунту например 3052. Звоню с экстеншена 3333 ( это первый астериск). На втором заведен провайдер и входящее правило с номера 3052 направлять в транк провайдера. но мне приходят разные виды номеров, которые указал выше. ту проблему я решил путем введения аналогиного экстеншена (аккаунта) на первом астериске. и все звонки стали проходить. но если таких экстеншенов будет куча? как можно организовать обие правила в этом случае?
каким образом могу передавать определенные параметры: у меня происходит разногласие следующее: звоню с экстеншена 3333 на номер 79046528473, проходит вызов. 2012-09-27 19:21:24 3052 79046528473 SIP/3052-00002f0c SIP/outAlterfone-00002f0d BUSY 0s выдает такое сообщение. все ок. 3052 - эьтто регистрационный аккаунт на elastixe..........потом пью чай. прихожу звоню и у меня 2012-09-27 19:16:22 3333 s SIP/outtest-00002ef4 ANSWERED 4s т.е. присваивается 3333 номер-экстеншен. как мне сделать так, чтобы оставался 3052? исполдьзовать сам этот экстеншен 3052 не вариант в связи с большим количеством таких номеров.
Как у меня решается сейчас: регистрация проходит по 1 аккунту например 3052. Звоню с экстеншена 3333 ( это первый астериск). На втором заведен провайдер и входящее правило с номера 3052 направлять в транк провайдера. но мне приходят разные виды номеров, которые указал выше. ту проблему я решил путем введения аналогиного экстеншена (аккаунта) на первом астериске. и все звонки стали проходить. но если таких экстеншенов будет куча? как можно организовать обие правила в этом случае?
Re: как использовать провайдера телефонии с другого астериск
Ага, я сделал так, как написал April22. есть ньюанс один. его я описал чуть выше. проблема аналогичная
Ниже дамп между двумя астерисками:
\\\\\\\\\\\\\\\\\\\\\\дамп астериска, с провайдером\\\\\\\\\\\\\\\\\\\
<--- SIP read from UDP:172.24.26.40:5060 --->
INVITE sip:79046528473@172.24.26.48:5060 SIP/2.0
Via: SIP/2.0/UDP 172.24.26.40:5060;branch=z9hG4bK5bd2d834;rport
Max-Forwards: 70
From: "123456" <sip:123456@172.24.26.40>;tag=as1fb4a99f
To: <sip:79046528473@172.24.26.48:5060>
Contact: <sip:123456@172.24.26.40:5060>
Call-ID: 5f129c263cb7a99719b046bb5549e391@172.24.26.40:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.8.1(1.8.11.0)
Date: Fri, 28 Sep 2012 08:09:56 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 282
v=0
o=root 829674207 829674207 IN IP4 172.24.26.40
s=Asterisk PBX 1.8.11.0
c=IN IP4 172.24.26.40
t=0 0
m=audio 17318 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.24.26.40:5060 (NAT)
Using INVITE request as basis request - 5f129c263cb7a99719b046bb5549e391@172.24. 26.40:5060
Found peer 'out_test' for '123456' from 172.24.26.40:5060
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format telephone-event for ID 101
Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0xe (gsm|ulaw|alaw)/ video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephon e-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.24.26.40:17318
Looking for 79046528473 in from-trunk-sip-out_test (domain 172.24.26.48)
list_route: hop: <sip:123456@172.24.26.40:5060>
<--- Transmitting (NAT) to 172.24.26.40:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.24.26.40:5060;branch=z9hG4bK5bd2d834;received=172.24.26.40; rport=5060
From: "123456" <sip:123456@172.24.26.40>;tag=as1fb4a99f
To: <sip:79046528473@172.24.26.48:5060>
Call-ID: 5f129c263cb7a99719b046bb5549e391@172.24.26.40:5060
CSeq: 102 INVITE
Server: FPBX-2.8.1(1.8.11.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H
Supported: replaces, timer
Contact: <sip:79046528473@172.24.26.48:5060>
Content-Length: 0
<--- Reliably Transmitting (NAT) to 172.24.26.40:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.24.26.40:5060;branch=z9hG4bK5bd2d834;received=172.24.26.40; rport=5060
From: "123456" <sip:123456@172.24.26.40>;tag=as1fb4a99f
To: <sip:79046528473@172.24.26.48:5060>;tag=as4fe5084a
Call-ID: 5f129c263cb7a99719b046bb5549e391@172.24.26.40:5060
CSeq: 102 INVITE
Server: FPBX-2.8.1(1.8.11.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H
Supported: replaces, timer
Contact: <sip:79046528473@172.24.26.48:5060>
Content-Type: application/sdp
Content-Length: 282
v=0
o=root 284048546 284048546 IN IP4 172.24.26.48
s=Asterisk PBX 1.8.11.0
c=IN IP4 172.24.26.48
t=0 0
m=audio 15688 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/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:172.24.26.40:5060 --->
ACK sip:79046528473@172.24.26.48:5060 SIP/2.0
Via: SIP/2.0/UDP 172.24.26.40:5060;branch=z9hG4bK1c2f8dfd;rport
Max-Forwards: 70
From: "123456" <sip:123456@172.24.26.40>;tag=as1fb4a99f
To: <sip:79046528473@172.24.26.48:5060>;tag=as4fe5084a
Contact: <sip:123456@172.24.26.40:5060>
Call-ID: 5f129c263cb7a99719b046bb5549e391@172.24.26.40:5060
CSeq: 102 ACK
User-Agent: FPBX-2.8.1(1.8.11.0)
Content-Length: 0
<--- SIP read from UDP:172.24.26.40:5060 --->
BYE sip:79046528473@172.24.26.48:5060 SIP/2.0
Via: SIP/2.0/UDP 172.24.26.40:5060;branch=z9hG4bK3f226292;rport
Max-Forwards: 70
From: "123456" <sip:123456@172.24.26.40>;tag=as1fb4a99f
To: <sip:79046528473@172.24.26.48:5060>;tag=as4fe5084a
Call-ID: 5f129c263cb7a99719b046bb5549e391@172.24.26.40:5060
CSeq: 103 BYE
User-Agent: FPBX-2.8.1(1.8.11.0)
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
Sending to 172.24.26.40:5060 (NAT)
Scheduling destruction of SIP dialog '5f129c263cb7a99719b046bb5549e391@172.24.26 .40:5060' in 32000 ms (Method: BYE)
<--- Transmitting (NAT) to 172.24.26.40:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.24.26.40:5060;branch=z9hG4bK3f226292;received=172.24.26.40; rport=5060
From: "123456" <sip:123456@172.24.26.40>;tag=as1fb4a99f
To: <sip:79046528473@172.24.26.48:5060>;tag=as4fe5084a
Call-ID: 5f129c263cb7a99719b046bb5549e391@172.24.26.40:5060
CSeq: 103 BYE
Server: FPBX-2.8.1(1.8.11.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H
Supported: replaces, timer
Content-Length: 0
\\\\\\\\\\дамп с астериска, который с екстеншенами(астер1)\\\\\\\\\\\\\\\\\\\\\\
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Audio is at 17318
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 172.24.26.48:5060:
INVITE sip:79046528473@172.24.26.48:5060 SIP/2.0
Via: SIP/2.0/UDP 172.24.26.40:5060;branch=z9hG4bK5bd2d834;rport
Max-Forwards: 70
From: "123456" <sip:123456@172.24.26.40>;tag=as1fb4a99f
To: <sip:79046528473@172.24.26.48:5060>
Contact: <sip:123456@172.24.26.40:5060>
Call-ID: 5f129c263cb7a99719b046bb5549e391@172.24.26.40:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.8.1(1.8.11.0)
Date: Fri, 28 Sep 2012 08:09:56 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 282
v=0
o=root 829674207 829674207 IN IP4 172.24.26.40
s=Asterisk PBX 1.8.11.0
c=IN IP4 172.24.26.40
t=0 0
m=audio 17318 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
---
-- Called SIP/out_test2/79046528473
<--- SIP read from UDP:172.24.26.48:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.24.26.40:5060;branch=z9hG4bK5bd2d834;received=172.24.26.40;rport=5060
From: "123456" <sip:123456@172.24.26.40>;tag=as1fb4a99f
To: <sip:79046528473@172.24.26.48:5060>
Call-ID: 5f129c263cb7a99719b046bb5549e391@172.24.26.40:5060
CSeq: 102 INVITE
Server: FPBX-2.8.1(1.8.11.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
ontact: <sip:79046528473@172.24.26.48:5060>
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
<--- SIP read from UDP:172.24.26.48:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.24.26.40:5060;branch=z9hG4bK5bd2d834;received=172.24.26.40;rport=5060
From: "123456" <sip:123456@172.24.26.40>;tag=as1fb4a99f
To: <sip:79046528473@172.24.26.48:5060>;tag=as4fe5084a
Call-ID: 5f129c263cb7a99719b046bb5549e391@172.24.26.40:5060
CSeq: 102 INVITE
Server: FPBX-2.8.1(1.8.11.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:79046528473@172.24.26.48:5060>
Content-Type: application/sdp
Content-Length: 282
v=0
o=root 284048546 284048546 IN IP4 172.24.26.48
s=Asterisk PBX 1.8.11.0
c=IN IP4 172.24.26.48
t=0 0
m=audio 15688 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (12 headers 13 lines) ---
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.24.26.48:15688
list_route: hop: <sip:79046528473@172.24.26.48:5060>
set_destination: Parsing <sip:79046528473@172.24.26.48:5060> for address/port to send to
set_destination: set destination to 172.24.26.48:5060
Transmitting (NAT) to 172.24.26.48:5060:
ACK sip:79046528473@172.24.26.48:5060 SIP/2.0
Via: SIP/2.0/UDP 172.24.26.40:5060;branch=z9hG4bK1c2f8dfd;rport
Max-Forwards: 70
From: "123456" <sip:123456@172.24.26.40>;tag=as1fb4a99f
To: <sip:79046528473@172.24.26.48:5060>;tag=as4fe5084a
Contact: <sip:123456@172.24.26.40:5060>
Call-ID: 5f129c263cb7a99719b046bb5549e391@172.24.26.40:5060
CSeq: 102 ACK
User-Agent: FPBX-2.8.1(1.8.11.0)
Content-Length: 0
---
-- SIP/out_test2-000000d3 answered SIP/3333-000000d2
-- Locally bridging SIP/3333-000000d2 and SIP/out_test2-000000d3
-- Executing [h@macro-dialout-trunk:1] Macro("SIP/3333-000000d2", "hangupcall,") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("SIP/3333-000000d2", "1?endmixmoncheck") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] NoOp("SIP/3333-000000d2", "End of MIXMON check") in new stack
-- Executing [s@macro-hangupcall:10] GotoIf("SIP/3333-000000d2", "1?nomeetmemon") in new stack
-- Goto (macro-hangupcall,s,15)
-- Executing [s@macro-hangupcall:15] NoOp("SIP/3333-000000d2", "MEETME_RECORDINGFILE=") in new stack
-- Executing [s@macro-hangupcall:16] GotoIf("SIP/3333-000000d2", "1?noautomon") in new stack
-- Goto (macro-hangupcall,s,18)
-- Executing [s@macro-hangupcall:18] NoOp("SIP/3333-000000d2", "TOUCH_MONITOR_OUTPUT=") in new stack
-- Executing [s@macro-hangupcall:19] GotoIf("SIP/3333-000000d2", "1?noautomon2") in new stack
-- Goto (macro-hangupcall,s,25)
-- Executing [s@macro-hangupcall:25] NoOp("SIP/3333-000000d2", "MONITOR_FILENAME=") in new stack
-- Executing [s@macro-hangupcall:26] GotoIf("SIP/3333-000000d2", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,29)
-- Executing [s@macro-hangupcall:29] GotoIf("SIP/3333-000000d2", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,32)
-- Executing [s@macro-hangupcall:32] GotoIf("SIP/3333-000000d2", "1?theend") in new stack
-- Goto (macro-hangupcall,s,34)
-- Executing [s@macro-hangupcall:34] Hangup("SIP/3333-000000d2", "") in new stack
== Spawn extension (macro-hangupcall, s, 34) exited non-zero on 'SIP/3333-000000d2' in macro 'hangupcall'
== Spawn extension (macro-dialout-trunk, h, 1) exited non-zero on 'SIP/3333-000000d2'
Scheduling destruction of SIP dialog '5f129c263cb7a99719b046bb5549e391@172.24.26.40:5060' in 32000 ms (Method: INVITE)
set_destination: Parsing <sip:79046528473@172.24.26.48:5060> for address/port to send to
set_destination: set destination to 172.24.26.48:5060
Reliably Transmitting (NAT) to 172.24.26.48:5060:
BYE sip:79046528473@172.24.26.48:5060 SIP/2.0
Via: SIP/2.0/UDP 172.24.26.40:5060;branch=z9hG4bK3f226292;rport
Max-Forwards: 70
From: "123456" <sip:123456@172.24.26.40>;tag=as1fb4a99f
To: <sip:79046528473@172.24.26.48:5060>;tag=as4fe5084a
Call-ID: 5f129c263cb7a99719b046bb5549e391@172.24.26.40:5060
CSeq: 103 BYE
User-Agent: FPBX-2.8.1(1.8.11.0)
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
---
== Spawn extension (macro-dialout-trunk, s, 19) exited non-zero on 'SIP/3333-000000d2' in macro 'dialout-trunk'
== Spawn extension (from-internal, 79046528473, 7) exited non-zero on 'SIP/3333-000000d2'
<--- SIP read from UDP:172.24.26.48:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.24.26.40:5060;branch=z9hG4bK3f226292;received=172.24.26.40;rport=5060
From: "123456" <sip:123456@172.24.26.40>;tag=as1fb4a99f
To: <sip:79046528473@172.24.26.48:5060>;tag=as4fe5084a
Call-ID: 5f129c263cb7a99719b046bb5549e391@172.24.26.40:5060
CSeq: 103 BYE
Server: FPBX-2.8.1(1.8.11.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '5f129c263cb7a99719b046bb5549e391@172.24.26.40:5060' Method: INVITE
-- Remote UNIX connection
-- Remote UNIX connection disconnected
\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\
внешне ситуаия выглядит так: звонок уходит с первого астера, приходит во второй, и уже второй его принимает, но не может как я понял номер передать, пришедший с первого. 2012-09-28 10:12:31 123456 s SIP/out_test-000032c3 ANSWERED 4s пустой канал приемник.
Ниже дамп между двумя астерисками:
\\\\\\\\\\\\\\\\\\\\\\дамп астериска, с провайдером\\\\\\\\\\\\\\\\\\\
<--- SIP read from UDP:172.24.26.40:5060 --->
INVITE sip:79046528473@172.24.26.48:5060 SIP/2.0
Via: SIP/2.0/UDP 172.24.26.40:5060;branch=z9hG4bK5bd2d834;rport
Max-Forwards: 70
From: "123456" <sip:123456@172.24.26.40>;tag=as1fb4a99f
To: <sip:79046528473@172.24.26.48:5060>
Contact: <sip:123456@172.24.26.40:5060>
Call-ID: 5f129c263cb7a99719b046bb5549e391@172.24.26.40:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.8.1(1.8.11.0)
Date: Fri, 28 Sep 2012 08:09:56 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 282
v=0
o=root 829674207 829674207 IN IP4 172.24.26.40
s=Asterisk PBX 1.8.11.0
c=IN IP4 172.24.26.40
t=0 0
m=audio 17318 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.24.26.40:5060 (NAT)
Using INVITE request as basis request - 5f129c263cb7a99719b046bb5549e391@172.24. 26.40:5060
Found peer 'out_test' for '123456' from 172.24.26.40:5060
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format telephone-event for ID 101
Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0xe (gsm|ulaw|alaw)/ video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephon e-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.24.26.40:17318
Looking for 79046528473 in from-trunk-sip-out_test (domain 172.24.26.48)
list_route: hop: <sip:123456@172.24.26.40:5060>
<--- Transmitting (NAT) to 172.24.26.40:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.24.26.40:5060;branch=z9hG4bK5bd2d834;received=172.24.26.40; rport=5060
From: "123456" <sip:123456@172.24.26.40>;tag=as1fb4a99f
To: <sip:79046528473@172.24.26.48:5060>
Call-ID: 5f129c263cb7a99719b046bb5549e391@172.24.26.40:5060
CSeq: 102 INVITE
Server: FPBX-2.8.1(1.8.11.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H
Supported: replaces, timer
Contact: <sip:79046528473@172.24.26.48:5060>
Content-Length: 0
<--- Reliably Transmitting (NAT) to 172.24.26.40:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.24.26.40:5060;branch=z9hG4bK5bd2d834;received=172.24.26.40; rport=5060
From: "123456" <sip:123456@172.24.26.40>;tag=as1fb4a99f
To: <sip:79046528473@172.24.26.48:5060>;tag=as4fe5084a
Call-ID: 5f129c263cb7a99719b046bb5549e391@172.24.26.40:5060
CSeq: 102 INVITE
Server: FPBX-2.8.1(1.8.11.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H
Supported: replaces, timer
Contact: <sip:79046528473@172.24.26.48:5060>
Content-Type: application/sdp
Content-Length: 282
v=0
o=root 284048546 284048546 IN IP4 172.24.26.48
s=Asterisk PBX 1.8.11.0
c=IN IP4 172.24.26.48
t=0 0
m=audio 15688 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/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:172.24.26.40:5060 --->
ACK sip:79046528473@172.24.26.48:5060 SIP/2.0
Via: SIP/2.0/UDP 172.24.26.40:5060;branch=z9hG4bK1c2f8dfd;rport
Max-Forwards: 70
From: "123456" <sip:123456@172.24.26.40>;tag=as1fb4a99f
To: <sip:79046528473@172.24.26.48:5060>;tag=as4fe5084a
Contact: <sip:123456@172.24.26.40:5060>
Call-ID: 5f129c263cb7a99719b046bb5549e391@172.24.26.40:5060
CSeq: 102 ACK
User-Agent: FPBX-2.8.1(1.8.11.0)
Content-Length: 0
<--- SIP read from UDP:172.24.26.40:5060 --->
BYE sip:79046528473@172.24.26.48:5060 SIP/2.0
Via: SIP/2.0/UDP 172.24.26.40:5060;branch=z9hG4bK3f226292;rport
Max-Forwards: 70
From: "123456" <sip:123456@172.24.26.40>;tag=as1fb4a99f
To: <sip:79046528473@172.24.26.48:5060>;tag=as4fe5084a
Call-ID: 5f129c263cb7a99719b046bb5549e391@172.24.26.40:5060
CSeq: 103 BYE
User-Agent: FPBX-2.8.1(1.8.11.0)
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
Sending to 172.24.26.40:5060 (NAT)
Scheduling destruction of SIP dialog '5f129c263cb7a99719b046bb5549e391@172.24.26 .40:5060' in 32000 ms (Method: BYE)
<--- Transmitting (NAT) to 172.24.26.40:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.24.26.40:5060;branch=z9hG4bK3f226292;received=172.24.26.40; rport=5060
From: "123456" <sip:123456@172.24.26.40>;tag=as1fb4a99f
To: <sip:79046528473@172.24.26.48:5060>;tag=as4fe5084a
Call-ID: 5f129c263cb7a99719b046bb5549e391@172.24.26.40:5060
CSeq: 103 BYE
Server: FPBX-2.8.1(1.8.11.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H
Supported: replaces, timer
Content-Length: 0
\\\\\\\\\\дамп с астериска, который с екстеншенами(астер1)\\\\\\\\\\\\\\\\\\\\\\
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Audio is at 17318
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 172.24.26.48:5060:
INVITE sip:79046528473@172.24.26.48:5060 SIP/2.0
Via: SIP/2.0/UDP 172.24.26.40:5060;branch=z9hG4bK5bd2d834;rport
Max-Forwards: 70
From: "123456" <sip:123456@172.24.26.40>;tag=as1fb4a99f
To: <sip:79046528473@172.24.26.48:5060>
Contact: <sip:123456@172.24.26.40:5060>
Call-ID: 5f129c263cb7a99719b046bb5549e391@172.24.26.40:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.8.1(1.8.11.0)
Date: Fri, 28 Sep 2012 08:09:56 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 282
v=0
o=root 829674207 829674207 IN IP4 172.24.26.40
s=Asterisk PBX 1.8.11.0
c=IN IP4 172.24.26.40
t=0 0
m=audio 17318 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
---
-- Called SIP/out_test2/79046528473
<--- SIP read from UDP:172.24.26.48:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.24.26.40:5060;branch=z9hG4bK5bd2d834;received=172.24.26.40;rport=5060
From: "123456" <sip:123456@172.24.26.40>;tag=as1fb4a99f
To: <sip:79046528473@172.24.26.48:5060>
Call-ID: 5f129c263cb7a99719b046bb5549e391@172.24.26.40:5060
CSeq: 102 INVITE
Server: FPBX-2.8.1(1.8.11.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
ontact: <sip:79046528473@172.24.26.48:5060>
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
<--- SIP read from UDP:172.24.26.48:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.24.26.40:5060;branch=z9hG4bK5bd2d834;received=172.24.26.40;rport=5060
From: "123456" <sip:123456@172.24.26.40>;tag=as1fb4a99f
To: <sip:79046528473@172.24.26.48:5060>;tag=as4fe5084a
Call-ID: 5f129c263cb7a99719b046bb5549e391@172.24.26.40:5060
CSeq: 102 INVITE
Server: FPBX-2.8.1(1.8.11.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:79046528473@172.24.26.48:5060>
Content-Type: application/sdp
Content-Length: 282
v=0
o=root 284048546 284048546 IN IP4 172.24.26.48
s=Asterisk PBX 1.8.11.0
c=IN IP4 172.24.26.48
t=0 0
m=audio 15688 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (12 headers 13 lines) ---
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.24.26.48:15688
list_route: hop: <sip:79046528473@172.24.26.48:5060>
set_destination: Parsing <sip:79046528473@172.24.26.48:5060> for address/port to send to
set_destination: set destination to 172.24.26.48:5060
Transmitting (NAT) to 172.24.26.48:5060:
ACK sip:79046528473@172.24.26.48:5060 SIP/2.0
Via: SIP/2.0/UDP 172.24.26.40:5060;branch=z9hG4bK1c2f8dfd;rport
Max-Forwards: 70
From: "123456" <sip:123456@172.24.26.40>;tag=as1fb4a99f
To: <sip:79046528473@172.24.26.48:5060>;tag=as4fe5084a
Contact: <sip:123456@172.24.26.40:5060>
Call-ID: 5f129c263cb7a99719b046bb5549e391@172.24.26.40:5060
CSeq: 102 ACK
User-Agent: FPBX-2.8.1(1.8.11.0)
Content-Length: 0
---
-- SIP/out_test2-000000d3 answered SIP/3333-000000d2
-- Locally bridging SIP/3333-000000d2 and SIP/out_test2-000000d3
-- Executing [h@macro-dialout-trunk:1] Macro("SIP/3333-000000d2", "hangupcall,") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("SIP/3333-000000d2", "1?endmixmoncheck") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] NoOp("SIP/3333-000000d2", "End of MIXMON check") in new stack
-- Executing [s@macro-hangupcall:10] GotoIf("SIP/3333-000000d2", "1?nomeetmemon") in new stack
-- Goto (macro-hangupcall,s,15)
-- Executing [s@macro-hangupcall:15] NoOp("SIP/3333-000000d2", "MEETME_RECORDINGFILE=") in new stack
-- Executing [s@macro-hangupcall:16] GotoIf("SIP/3333-000000d2", "1?noautomon") in new stack
-- Goto (macro-hangupcall,s,18)
-- Executing [s@macro-hangupcall:18] NoOp("SIP/3333-000000d2", "TOUCH_MONITOR_OUTPUT=") in new stack
-- Executing [s@macro-hangupcall:19] GotoIf("SIP/3333-000000d2", "1?noautomon2") in new stack
-- Goto (macro-hangupcall,s,25)
-- Executing [s@macro-hangupcall:25] NoOp("SIP/3333-000000d2", "MONITOR_FILENAME=") in new stack
-- Executing [s@macro-hangupcall:26] GotoIf("SIP/3333-000000d2", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,29)
-- Executing [s@macro-hangupcall:29] GotoIf("SIP/3333-000000d2", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,32)
-- Executing [s@macro-hangupcall:32] GotoIf("SIP/3333-000000d2", "1?theend") in new stack
-- Goto (macro-hangupcall,s,34)
-- Executing [s@macro-hangupcall:34] Hangup("SIP/3333-000000d2", "") in new stack
== Spawn extension (macro-hangupcall, s, 34) exited non-zero on 'SIP/3333-000000d2' in macro 'hangupcall'
== Spawn extension (macro-dialout-trunk, h, 1) exited non-zero on 'SIP/3333-000000d2'
Scheduling destruction of SIP dialog '5f129c263cb7a99719b046bb5549e391@172.24.26.40:5060' in 32000 ms (Method: INVITE)
set_destination: Parsing <sip:79046528473@172.24.26.48:5060> for address/port to send to
set_destination: set destination to 172.24.26.48:5060
Reliably Transmitting (NAT) to 172.24.26.48:5060:
BYE sip:79046528473@172.24.26.48:5060 SIP/2.0
Via: SIP/2.0/UDP 172.24.26.40:5060;branch=z9hG4bK3f226292;rport
Max-Forwards: 70
From: "123456" <sip:123456@172.24.26.40>;tag=as1fb4a99f
To: <sip:79046528473@172.24.26.48:5060>;tag=as4fe5084a
Call-ID: 5f129c263cb7a99719b046bb5549e391@172.24.26.40:5060
CSeq: 103 BYE
User-Agent: FPBX-2.8.1(1.8.11.0)
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
---
== Spawn extension (macro-dialout-trunk, s, 19) exited non-zero on 'SIP/3333-000000d2' in macro 'dialout-trunk'
== Spawn extension (from-internal, 79046528473, 7) exited non-zero on 'SIP/3333-000000d2'
<--- SIP read from UDP:172.24.26.48:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.24.26.40:5060;branch=z9hG4bK3f226292;received=172.24.26.40;rport=5060
From: "123456" <sip:123456@172.24.26.40>;tag=as1fb4a99f
To: <sip:79046528473@172.24.26.48:5060>;tag=as4fe5084a
Call-ID: 5f129c263cb7a99719b046bb5549e391@172.24.26.40:5060
CSeq: 103 BYE
Server: FPBX-2.8.1(1.8.11.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '5f129c263cb7a99719b046bb5549e391@172.24.26.40:5060' Method: INVITE
-- Remote UNIX connection
-- Remote UNIX connection disconnected
\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\
внешне ситуаия выглядит так: звонок уходит с первого астера, приходит во второй, и уже второй его принимает, но не может как я понял номер передать, пришедший с первого. 2012-09-28 10:12:31 123456 s SIP/out_test-000032c3 ANSWERED 4s пустой канал приемник.
Re: как использовать провайдера телефонии с другого астериск
Этот принцип начинаеш понимать из классических систем где на Е1 есть тайм слоты со своими номерами . и если правильно знать их - то можно творить такие чудеса маршрутизации в ручную 
но все таки их в своих системах надо знать наизусть .( ну или по крайней мере знать где поссмотреть ) - все знать и помнить не возможно .
но все таки их в своих системах надо знать наизусть .( ну или по крайней мере знать где поссмотреть ) - все знать и помнить не возможно .
Своими вопросами , вы загоняете меня в ГУГЛЬ.
