[asterisk-users] Asterisk with PJSIP
Marie Fischer
marie at vtl.ee
Tue Sep 23 11:31:45 CDT 2014
Hi,
yes, the 9002 Linphone answers "400 Bad Request" to Asterisk's INVITE.
Does it work in the other direction (9002 calling 9001)?
Have you checked your codecs (Linphone is offering PCMA, PCMU and GSM, Asterisk just PCMU)?
Apparently, for debug logging Linphone, you should
• open a windows shell prompt
• go to c:\Program Files\Linphone
• start Linphone like this: bin/linphone --logfile "c:\Temp\logs.txt"
So maybe this way you can see some more information.
--
marie
On 10.09.2014, at 13:00, エムディーシー太郎 <mdc.taro at gmail.com> wrote:
> Thank you for your reply.
>
> After setting "pjsip set logger on",
> the following message is displayed.
>
> It seems that the 9002(SIP client) refuse INVITE message.
> Are SIP methods too many?
>
> Thanks,
> MMEEGGAA
>
> --------------------
> <--- Transmitting SIP request (449 bytes) to UDP:192.168.177.180:16060 --->
> OPTIONS sip:9001 at 192.168.177.180:16060 SIP/2.0
> Via: SIP/2.0/UDP 192.168.177.190:5060;rport;branch=z9hG4bKPj3c090235-9385-4cea-9c56-83f2755606d1
> From: <sip:1e00dac6-cbe1-4bfa-96f4-b24085b848df at 192.168.177.190>;tag=ef5b7ffd-6d54-4c46-8100-635d862f699b
> To: <sip:9001 at 192.168.177.180>
> Contact: <sip:1e00dac6-cbe1-4bfa-96f4-b24085b848df at 192.168.177.190:5060>
> Call-ID: 6294a204-c645-41a8-ac27-e5aaa5f6c08c
> CSeq: 44261 OPTIONS
> Content-Length: 0
>
>
> <--- Received SIP response (333 bytes) from UDP:192.168.177.180:16060 --->
> SIP/2.0 200 Ok
> Via: SIP/2.0/UDP 192.168.177.190:5060;rport;branch=z9hG4bKPj3c090235-9385-4cea-9c56-83f2755606d1
> From: <sip:1e00dac6-cbe1-4bfa-96f4-b24085b848df at 192.168.177.190>;tag=ef5b7ffd-6d54-4c46-8100-635d862f699b
> To: <sip:9001 at 192.168.177.180>;tag=EF1my
> Call-ID: 6294a204-c645-41a8-ac27-e5aaa5f6c08c
> CSeq: 44261 OPTIONS
>
>
> <--- Transmitting SIP request (443 bytes) to UDP:192.168.177.191:5060 --->
> OPTIONS sip:9002 at 192.168.177.191 SIP/2.0
> Via: SIP/2.0/UDP 192.168.177.190:5060;rport;branch=z9hG4bKPj98982748-5a3d-4a4c-9d90-ad4988e8d933
> From: <sip:b4bf1474-a5c6-475b-985b-5ff48c550a7f at 192.168.177.190>;tag=73ae5dfe-d5b7-4c9b-9529-5198492f5c49
> To: <sip:9002 at 192.168.177.191>
> Contact: <sip:b4bf1474-a5c6-475b-985b-5ff48c550a7f at 192.168.177.190:5060>
> Call-ID: eb7a8bee-b06f-4a30-a5d9-eb437e104f76
> CSeq: 16803 OPTIONS
> Content-Length: 0
>
>
> <--- Received SIP response (333 bytes) from UDP:192.168.177.191:5060 --->
> SIP/2.0 200 Ok
> Via: SIP/2.0/UDP 192.168.177.190:5060;rport;branch=z9hG4bKPj98982748-5a3d-4a4c-9d90-ad4988e8d933
> From: <sip:b4bf1474-a5c6-475b-985b-5ff48c550a7f at 192.168.177.190>;tag=73ae5dfe-d5b7-4c9b-9529-5198492f5c49
> To: <sip:9002 at 192.168.177.191>;tag=hSl7b
> Call-ID: eb7a8bee-b06f-4a30-a5d9-eb437e104f76
> CSeq: 16803 OPTIONS
>
>
> <--- Received SIP request (1170 bytes) from UDP:192.168.177.180:16060 --->
> INVITE sip:9002 at 192.168.177.190 SIP/2.0
> Via: SIP/2.0/UDP 192.168.177.180:16060;branch=z9hG4bK.G4FYrVaiY;rport
> From: <sip:9001 at 192.168.177.190>;tag=~yIpJRFo9
> To: sip:9002 at 192.168.177.190
> CSeq: 20 INVITE
> Call-ID: 2c1KLd1INo
> Max-Forwards: 70
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
> Content-Type: application/sdp
> Content-Length: 633
> User-Agent: Linphone/3.6.99 (belle-sip/1.2.4)
> Contact: <sip:9001 at 192.168.177.180:16060>;+sip.instance="<urn:uuid:aa35e441-baa3-4e07-af2f-cd1ff2dc99a2>"
>
> v=0
> o=9001 2189 3894 IN IP4 192.168.177.180
> s=Talk
> c=IN IP4 192.168.177.180
> t=0 0
> a=ice-pwd:000030810000373f00004003
> a=ice-ufrag:00004c02
> m=audio 17590 RTP/AVP 0 110 3 8 101
> c=IN IP4 61.117.138.218
> a=rtpmap:110 speex/8000
> a=fmtp:110 vbr=on
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-11
> a=rtcp:21548
> a=candidate:1 1 UDP 2130706431 192.168.177.180 7078 typ host
> a=candidate:1 2 UDP 2130706430 192.168.177.180 7079 typ host
> a=candidate:2 1 UDP 1694498815 XXX.XXX.XXX.XXX 17590 typ srflx raddr 192.168.177.180 rport 7078
> a=candidate:2 2 UDP 1694498814 XXX.XXX.XXX.XXX 21548 typ srflx raddr 192.168.177.180 rport 7079
>
> <--- Transmitting SIP response (431 bytes) to UDP:192.168.177.180:16060 --->
> SIP/2.0 401 Unauthorized
> Via: SIP/2.0/UDP 192.168.177.180:16060;rport;received=192.168.177.180;branch=z9hG4bK.G4FYrVaiY
> Call-ID: 2c1KLd1INo
> From: <sip:9001 at 192.168.177.190>;tag=~yIpJRFo9
> To: <sip:9002 at 192.168.177.190>;tag=z9hG4bK.G4FYrVaiY
> CSeq: 20 INVITE
> WWW-Authenticate: Digest realm="asterisk",nonce="1410336707/cd97e01134333d7d5769e49872f750a4",opaque="58e109d10f49a371",algorithm=md5,qop="auth"
> Content-Length: 0
>
>
> <--- Received SIP request (373 bytes) from UDP:192.168.177.180:16060 --->
> ACK sip:9002 at 192.168.177.190 SIP/2.0
> Via: SIP/2.0/UDP 192.168.177.180:16060;branch=z9hG4bK.G4FYrVaiY;rport
> Call-ID: 2c1KLd1INo
> From: <sip:9001 at 192.168.177.190>;tag=~yIpJRFo9
> To: <sip:9002 at 192.168.177.190>;tag=z9hG4bK.G4FYrVaiY
> Contact: <sip:9001 at 192.168.177.180:16060>;+sip.instance="<urn:uuid:aa35e441-baa3-4e07-af2f-cd1ff2dc99a2>"
> Max-Forwards: 70
> CSeq: 20 ACK
>
>
> <--- Received SIP request (1428 bytes) from UDP:192.168.177.180:16060 --->
> INVITE sip:9002 at 192.168.177.190 SIP/2.0
> Via: SIP/2.0/UDP 192.168.177.180:16060;branch=z9hG4bK.00879CXMH;rport
> From: <sip:9001 at 192.168.177.190>;tag=~yIpJRFo9
> To: sip:9002 at 192.168.177.190
> CSeq: 21 INVITE
> Call-ID: 2c1KLd1INo
> Max-Forwards: 70
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
> Content-Type: application/sdp
> Content-Length: 633
> User-Agent: Linphone/3.6.99 (belle-sip/1.2.4)
> Contact: <sip:9001 at 192.168.177.180:16060>;+sip.instance="<urn:uuid:aa35e441-baa3-4e07-af2f-cd1ff2dc99a2>"
> Authorization: Digest realm="asterisk", nonce="1410336707/cd97e01134333d7d5769e49872f750a4", opaque="58e109d10f49a371", username="9001", uri="sip:9002 at 192.168.177.190", response="a822c66fe1c1d30492beeb08e6daaae5", cnonce="faaa92d5", nc=00000001, qop=auth
>
> v=0
> o=9001 2189 3894 IN IP4 192.168.177.180
> s=Talk
> c=IN IP4 192.168.177.180
> t=0 0
> a=ice-pwd:000030810000373f00004003
> a=ice-ufrag:00004c02
> m=audio 17590 RTP/AVP 0 110 3 8 101
> c=IN IP4 61.117.138.218
> a=rtpmap:110 speex/8000
> a=fmtp:110 vbr=on
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-11
> a=rtcp:21548
> a=candidate:1 1 UDP 2130706431 192.168.177.180 7078 typ host
> a=candidate:1 2 UDP 2130706430 192.168.177.180 7079 typ host
> a=candidate:2 1 UDP 1694498815 XXX.XXX.XXX.XXX 17590 typ srflx raddr 192.168.177.180 rport 7078
> a=candidate:2 2 UDP 1694498814 XXX.XXX.XXX.XXX 21548 typ srflx raddr 192.168.177.180 rport 7079
>
> <--- Transmitting SIP response (256 bytes) to UDP:192.168.177.180:16060 --->
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP 192.168.177.180:16060;rport;received=192.168.177.180;branch=z9hG4bK.00879CXMH
> Call-ID: 2c1KLd1INo
> From: <sip:9001 at 192.168.177.190>;tag=~yIpJRFo9
> To: <sip:9002 at 192.168.177.190>
> CSeq: 21 INVITE
> Content-Length: 0
>
>
> -- Executing [9002 at internal:1] Dial("PJSIP/9001-00000006", "PJSIP/9002,20") in new stack
> -- Called PJSIP/9002
> debug
> == debug1 (0|0:0/0/0)
> == debug2 (2|1:0/0/0)
> <--- Transmitting SIP request (910 bytes) to UDP:192.168.177.191:5060 --->
> INVITE sip:9002 at 192.168.177.191 SIP/2.0
> Via: SIP/2.0/UDP 192.168.177.190:5060;rport;branch=z9hG4bKPjf90aa758-0538-4145-b2c9-7d945da6f124
> From: <sip:9001 at 192.168.177.190>;tag=215003c4-08ad-472f-b366-bfaa8bfe303b
> To: <sip:9002 at 192.168.177.191>
> Contact: <sip:b9b2034d-e72e-4a18-bcd5-4e84d967afbc at 192.168.177.190:5060>
> Call-ID: 8a59fbf2-9cae-48b7-88de-1a2fed4ec7c9
> CSeq: 18942 INVITE
> Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
> Supported: 100rel, timer, replaces, norefersub
> Session-Expires: 1800
> Min-SE: 90
> Content-Type: application/sdp
> Content-Length: 273
>
> v=0
> o=- 278980317 278980317 IN IP4 localhost.localdomain
> s=Asterisk
> c=IN IP4 192.168.177.190
> t=0 0
> m=audio 10338 RTP/AVP 0 101
> c=IN IP4 192.168.177.190
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
> a=maxptime:150
> a=sendrecv
>
> <--- Received SIP response (309 bytes) from UDP:192.168.177.191:5060 --->
> SIP/2.0 400 Bad request
> Via: SIP/2.0/UDP 192.168.177.190:5060;rport;branch=z9hG4bKPjf90aa758-0538-4145-b2c9-7d945da6f124
> From: <sip:9001 at 192.168.177.190>;tag=215003c4-08ad-472f-b366-bfaa8bfe303b
> To: <sip:9002 at 192.168.177.191>;tag=bajXh
> Call-ID: 8a59fbf2-9cae-48b7-88de-1a2fed4ec7c9
> CSeq: 18942 INVITE
>
>
> <--- Transmitting SIP request (339 bytes) to UDP:192.168.177.191:5060 --->
> ACK sip:9002 at 192.168.177.191 SIP/2.0
> Via: SIP/2.0/UDP 192.168.177.190:5060;rport;branch=z9hG4bKPjf90aa758-0538-4145-b2c9-7d945da6f124
> From: <sip:9001 at 192.168.177.190>;tag=215003c4-08ad-472f-b366-bfaa8bfe303b
> To: <sip:9002 at 192.168.177.191>;tag=bajXh
> Call-ID: 8a59fbf2-9cae-48b7-88de-1a2fed4ec7c9
> CSeq: 18942 ACK
> Content-Length: 0
>
> -- PJSIP/9002-00000007 answered PJSIP/9001-00000006
> -- PJSIP/9002-00000007 answered PJSIP/9001-00000006
>
> == Everyone is busy/congested at this time (1:0/0/1)
> -- Auto fallthrough, channel 'PJSIP/9001-00000006' status is 'CHANUNAVAIL'
> <--- Transmitting SIP response (334 bytes) to UDP:192.168.177.180:16060 --->
> SIP/2.0 503 Service Unavailable
> Via: SIP/2.0/UDP 192.168.177.180:16060;rport;received=192.168.177.180;branch=z9hG4bK.00879CXMH
> Call-ID: 2c1KLd1INo
> From: <sip:9001 at 192.168.177.190>;tag=~yIpJRFo9
> To: <sip:9002 at 192.168.177.190>;tag=aead10f9-2194-48dd-bf38-0cc78bff561f
> CSeq: 21 INVITE
> Reason: Q.850;cause=34
> Content-Length: 0
>
>
> <--- Received SIP response (306 bytes) from UDP:192.168.177.191:5060 --->
> SIP/2.0 400 Bad request
> Via: SIP/2.0/UDP 192.168.177.190:5060;rport;branch=z9hG4bKPjf90aa758-0538-4145-b2c9-7d945da6f124
> From: <sip:9001 at 192.168.177.190>;tag=215003c4-08ad-472f-b366-bfaa8bfe303b
> To: <sip:9002 at 192.168.177.191>;tag=bajXh
> Call-ID: 8a59fbf2-9cae-48b7-88de-1a2fed4ec7c9
> CSeq: 18942 ACK
> --------------------
>
> 2014-09-05 19:24 GMT+09:00 Joshua Colp <jcolp at digium.com>:
> エムディーシー太郎 wrote:
> Hi All,
>
> I installed Astreisk 13beta with pjproject 2.3(2.2.1) from source code
> on CentOS7.
> --https://wiki.asterisk.org/wiki/display/AST/Building+and+Installing+pjproject
>
> <snip>
>
>
> ----------
> 2. dial from 9001 to 9002
>
> *CLI> -- Executing [9002 at internal:1] Dial("PJSIP/9001-00000000",
> "PJSIP/9002,20") in new stack
> -- Called PJSIP/9002
> == Everyone is busy/congested at this time (1:0/0/1)
> -- Auto fallthrough, channel 'PJSIP/9001-00000000' status is
> 'CHANUNAVAIL'
>
> What is shown if you do "pjsip set logger on" and then try to place the call?
>
> --
> Joshua Colp
> Digium, Inc. | Senior Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
> Check us out at: www.digium.com & www.asterisk.org
>
> --
> _____________________________________________________________________
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>
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