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История изменений [назад]

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спросил 2014-04-25 15:23:49 +0400

OrionDim Gravatar OrionDim

asterisk + sipnet + raspberry входящие звонки

Исходящие звонки работают на ура. Входящие не проходят. Дебаг показывает вот это.

<--- SIP read from UDP:212.53.40.40:5060 ---> INVITE sip:1@128.68.242.152:5060 SIP/2.0 Via: SIP/2.0/UDP 212.53.40.40:5060;branch=z9hG4bK931968-kmbdctj;cgp=etc.tario.ru;rport P-CGP-Redirector: oriondim@sipnet.ru Record-Route: <sip:212.53.40.40:5060;lr> Record-Route: <sip:192.168.40.71:5060;lr> Record-Route: <sip:rev.600405-192.168.40.71.dialog.cgatepro;lr> Max-Forwards: 10 From: <sip:+79067600406@sipnet.ru>;tag=B2C132B1-934244-1AC46CF3_kmbdctj-4D05 To: <sip:0041998912@sipnet.ru> Call-ID: IWF-0cfb771a079814f47051fc710f7bde2@trusted Contact: <sip:signode-934244-1ac46cf3_kmbdctj-4d05@212.53.40.40> CSeq: 1 INVITE Supported: 100rel,timer,replaces,histinfo,precondition Session-Expires: 7200 Min-SE: 90 User-Agent: CommuniGatePro-callLeg/6.0.10d Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,UPDATE,REFER Content-Type: application/sdp Content-Length: 460

v=0 o=CGPLeg934244 1776574855 888287428 IN IP4 212.53.40.71 s=session c=IN IP4 212.53.40.71 t=0 0 a=mediagateway:etc.tario.ru:600405:192.168.40.71 m=audio 27832 RTP/AVP 0 8 18 101 c=IN IP4 212.53.40.71 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv a=ptime:20 a=ice-pwd:52989ED534ABB4A1137D0661 a=ice-ufrag:pt0646706 a=silenceSupp:off - - - - <-------------> --- (19 headers 19 lines) --- Sending to 212.53.40.40:5060 (no NAT) Sending to 212.53.40.40:5060 (no NAT) Using INVITE request as basis request - IWF-0cfb771a079814f47051fc710f7bde2@trusted Found peer 'sipnet' for '+79067600406' from 212.53.40.40:5060 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 18 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 G729 for ID 18 Found audio description format telephone-event for ID 101 Понятно, что проблема с кодеками, но какие ставить? Напомню речь идет про raspberry Pi

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обновил 2014-04-25 17:14:31 +0400

asterisk + sipnet + raspberry входящие звонки

Исходящие звонки работают на ура. Входящие не проходят. Дебаг показывает вот это.

<--- SIP read from UDP:212.53.40.40:5060 --->
INVITE sip:1@128.68.242.152:5060 SIP/2.0
Via: SIP/2.0/UDP 212.53.40.40:5060;branch=z9hG4bK931968-kmbdctj;cgp=etc.tario.ru;rport
P-CGP-Redirector: oriondim@sipnet.ru
Record-Route: <sip:212.53.40.40:5060;lr>
Record-Route: <sip:192.168.40.71:5060;lr>
Record-Route: <sip:rev.600405-192.168.40.71.dialog.cgatepro;lr>
Max-Forwards: 10
From: <sip:+79067600406@sipnet.ru>;tag=B2C132B1-934244-1AC46CF3_kmbdctj-4D05
To: <sip:0041998912@sipnet.ru>
Call-ID: IWF-0cfb771a079814f47051fc710f7bde2@trusted
Contact: <sip:signode-934244-1ac46cf3_kmbdctj-4d05@212.53.40.40>
<sip:signode-934244-1AC46CF3_kmbdctj-4D05@212.53.40.40>
CSeq: 1 INVITE
Supported: 100rel,timer,replaces,histinfo,precondition
Session-Expires: 7200
Min-SE: 90
User-Agent: CommuniGatePro-callLeg/6.0.10d
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,UPDATE,REFER
Content-Type: application/sdp
Content-Length: 460
 
460

v=0
o=CGPLeg934244 1776574855 888287428 IN IP4 212.53.40.71
s=session
c=IN IP4 212.53.40.71
t=0 0
a=mediagateway:etc.tario.ru:600405:192.168.40.71
m=audio 27832 RTP/AVP 0 8 18 101
c=IN IP4 212.53.40.71
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
a=ptime:20
a=ice-pwd:52989ED534ABB4A1137D0661
a=ice-ufrag:pt0646706
a=silenceSupp:off - - - -
<------------->
--- (19 headers 19 lines) ---
Sending to 212.53.40.40:5060 (no NAT)
Sending to 212.53.40.40:5060 (no NAT)
Using INVITE request as basis request - IWF-0cfb771a079814f47051fc710f7bde2@trusted
Found peer 'sipnet' for '+79067600406' from 212.53.40.40:5060
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
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 G729 for ID 18
Found audio description format telephone-event for ID 101
[2014-04-25 11:58:55] NOTICE[2894][C-00000017]: chan_sip.c:10564 process_sdp: No compatible codecs, not accepting this offer!

Понятно, что проблема с кодеками, но какие ставить? Напомню речь идет про raspberry Pi

asterisk + sipnet + raspberry входящие звонки

Исходящие звонки работают на ура. Входящие не проходят. Дебаг показывает вот это.

<--- SIP read from UDP:212.53.40.40:5060 --->
INVITE sip:1@128.68.242.152:5060 sip:0041998912@128.68.242.152:5060 SIP/2.0
Via: SIP/2.0/UDP 212.53.40.40:5060;branch=z9hG4bK931968-kmbdctj;cgp=etc.tario.ru;rport
212.53.40.40:5060;branch=z9hG4bK738198-kmbdctk;cgp=etc.tario.ru;rport
P-CGP-Redirector: oriondim@sipnet.ru
Record-Route: <sip:212.53.40.40:5060;lr>
Record-Route: <sip:192.168.40.71:5060;lr>
<sip:192.168.40.72:5060;lr>
Record-Route: <sip:rev.600405-192.168.40.71.dialog.cgatepro;lr>
<sip:rev.845325-192.168.40.72.dialog.cgatepro;lr>
Max-Forwards: 10
From: <sip:+79067600406@sipnet.ru>;tag=B2C132B1-934244-1AC46CF3_kmbdctj-4D05
<sip:+79067600406@sipnet.ru>;tag=0DBC9AD4-255884-36F4E3B4_kmbdctk-4D05
To: <sip:0041998912@sipnet.ru>
Call-ID: IWF-0cfb771a079814f47051fc710f7bde2@trusted
IWF-150b445a0c09ba710c9d245b6180687@trusted
Contact: <sip:signode-934244-1AC46CF3_kmbdctj-4D05@212.53.40.40>
<sip:signode-255884-36F4E3B4_kmbdctk-4D05@212.53.40.40>
CSeq: 1 INVITE
Supported: 100rel,timer,replaces,histinfo,precondition
Session-Expires: 7200
Min-SE: 90
User-Agent: CommuniGatePro-callLeg/6.0.10d
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,UPDATE,REFER
Content-Type: application/sdp
Content-Length: 460
461

v=0
o=CGPLeg934244 1776574855 888287428 o=CGPLeg255884 2030505154 1015252578 IN IP4 212.53.40.71
212.53.40.72
s=session
c=IN IP4 212.53.40.71
212.53.40.72
t=0 0
a=mediagateway:etc.tario.ru:600405:192.168.40.71
a=mediagateway:etc.tario.ru:845325:192.168.40.72
m=audio 27832 24844 RTP/AVP 0 8 18 101
c=IN IP4 212.53.40.71
212.53.40.72
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
a=ptime:20
a=ice-pwd:52989ED534ABB4A1137D0661
a=ice-ufrag:pt0646706
a=ice-pwd:7F4DC8870527D79DC45668E8
a=ice-ufrag:pt0921457
a=silenceSupp:off - - - -
<------------->
--- (19 headers 19 lines) ---
Sending to 212.53.40.40:5060 (no NAT)
Sending to 212.53.40.40:5060 (no NAT)
Using INVITE request as basis request - IWF-0cfb771a079814f47051fc710f7bde2@trusted
IWF-150b445a0c09ba710c9d245b6180687@trusted
Found peer 'sipnet' for '+79067600406' from 212.53.40.40:5060
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
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 G729 for ID 18
Found audio description format telephone-event for ID 101
[2014-04-25 11:58:55] NOTICE[2894][C-00000017]: 13:56:15] NOTICE[2894][C-0000001e]: chan_sip.c:10564 process_sdp: No compatible codecs, not accepting this offer!

<--- Reliably Transmitting (no NAT) to 212.53.40.40:5060 --->
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP 212.53.40.40:5060;branch=z9hG4bK738198-kmbdctk;cgp=etc.tario.ru;received=212.53.40.40;rport=5060
From: <sip:+79067600406@sipnet.ru>;tag=0DBC9AD4-255884-36F4E3B4_kmbdctk-4D05
To: <sip:0041998912@sipnet.ru>;tag=as506458f3
Call-ID: IWF-150b445a0c09ba710c9d245b6180687@trusted
CSeq: 1 INVITE
Server: FPBX-2.11.0(11.8.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'IWF-150b445a0c09ba710c9d245b6180687@trusted' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:212.53.40.40:5060 --->
INVITE sip:1@128.68.242.152:5060 SIP/2.0
Via: SIP/2.0/UDP 212.53.40.40:5060;branch=z9hG4bK738200-kmbdctk;cgp=etc.tario.ru;rport
P-CGP-Redirector: oriondim@sipnet.ru
Record-Route: <sip:212.53.40.40:5060;lr>
Record-Route: <sip:192.168.40.72:5060;lr>
Record-Route: <sip:rev.845326-192.168.40.72.dialog.cgatepro;lr>
Max-Forwards: 10
From: <sip:+79067600406@sipnet.ru>;tag=0DBC9AD4-255884-36F4E3B4_kmbdctk-4D05
To: <sip:0041998912@sipnet.ru>
Call-ID: IWF-150b445a0c09ba710c9d245b6180687@trusted
Contact: <sip:signode-255884-36F4E3B4_kmbdctk-4D05@212.53.40.40>
CSeq: 1 INVITE
Supported: 100rel,timer,replaces,histinfo,precondition
Session-Expires: 7200
Min-SE: 90
User-Agent: CommuniGatePro-callLeg/6.0.10d
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,UPDATE,REFER
Content-Type: application/sdp
Content-Length: 461

v=0
o=CGPLeg255884 2030505154 1015252578 IN IP4 212.53.40.72
s=session
c=IN IP4 212.53.40.72
t=0 0
a=mediagateway:etc.tario.ru:845326:192.168.40.72
m=audio 24846 RTP/AVP 0 8 18 101
c=IN IP4 212.53.40.72
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
a=ptime:20
a=ice-pwd:517B6DA594833F62B3F2A1D9
a=ice-ufrag:pt0921459
a=silenceSupp:off - - - -
<------------->
--- (19 headers 19 lines) ---
Sending to 212.53.40.40:5060 (no NAT)
Sending to 212.53.40.40:5060 (no NAT)
Using INVITE request as basis request - IWF-150b445a0c09ba710c9d245b6180687@trusted
Found peer 'sipnet' for '+79067600406' from 212.53.40.40:5060
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
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 G729 for ID 18
Found audio description format telephone-event for ID 101
[2014-04-25 13:56:15] NOTICE[2894][C-0000001f]: chan_sip.c:10564 process_sdp: No compatible codecs, not accepting this offer!

<--- Reliably Transmitting (no NAT) to 212.53.40.40:5060 --->
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP 212.53.40.40:5060;branch=z9hG4bK738200-kmbdctk;cgp=etc.tario.ru;received=212.53.40.40;rport=5060
From: <sip:+79067600406@sipnet.ru>;tag=0DBC9AD4-255884-36F4E3B4_kmbdctk-4D05
To: <sip:0041998912@sipnet.ru>;tag=as338ed522
Call-ID: IWF-150b445a0c09ba710c9d245b6180687@trusted
CSeq: 1 INVITE
Server: FPBX-2.11.0(11.8.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'IWF-150b445a0c09ba710c9d245b6180687@trusted' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:212.53.40.40:5060 --->
ACK sip:0041998912@128.68.242.152:5060 SIP/2.0
P-CGP-Redirector: oriondim@sipnet.ru
Via: SIP/2.0/UDP 212.53.40.40:5060;branch=z9hG4bK738198-kmbdctk;cgp=etc.tario.ru;rport
Max-Forwards: 10
From: <sip:+79067600406@sipnet.ru>;tag=0DBC9AD4-255884-36F4E3B4_kmbdctk-4D05
To: <sip:0041998912@sipnet.ru>;tag=as506458f3
Call-ID: IWF-150b445a0c09ba710c9d245b6180687@trusted
CSeq: 1 ACK
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog 'IWF-150b445a0c09ba710c9d245b6180687@trusted' Method: ACK

<--- SIP read from UDP:212.53.40.40:5060 --->
ACK sip:1@128.68.242.152:5060 SIP/2.0
P-CGP-Redirector: oriondim@sipnet.ru
Via: SIP/2.0/UDP 212.53.40.40:5060;branch=z9hG4bK738200-kmbdctk;cgp=etc.tario.ru;rport
Max-Forwards: 10
From: <sip:+79067600406@sipnet.ru>;tag=0DBC9AD4-255884-36F4E3B4_kmbdctk-4D05
To: <sip:0041998912@sipnet.ru>;tag=as338ed522
Call-ID: IWF-150b445a0c09ba710c9d245b6180687@trusted
CSeq: 1 ACK
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog 'IWF-150b445a0c09ba710c9d245b6180687@trusted' Method: ACK

Понятно, что проблема с кодеками, но какие ставить? Напомню речь идет про raspberry Pi

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Проект компании "АТС Дизайн"
Asterisk® и Digium® являются зарегистрированными торговыми марками компании Digium, Inc., США.
IP АТС Asterisk распространяется под лицензией GNU GPL.