[Asterisk-Users] Recommended SIP phones?
Pieter Claassen
pieter at claassen.co.uk
Mon May 22 14:25:44 MST 2006
Hi Steve,
Included is the trace as per your request.
This is what appears on the asterisk CLI:
-- Executing Dial("SIP/pieter-3079", "SIP/rachel||r") in new stack
-- Called rachel
-- SIP/rachel-ede0 is making progress passing it to SIP/pieter-3079
-- SIP/rachel-ede0 is ringing
-- SIP/rachel-ede0 answered SIP/pieter-3079
-- Attempting native bridge of SIP/pieter-3079 and SIP/rachel-ede0
-- Got SIP response 415 "Unsupported Media Type" back from 192.168.1.31
== Spawn extension (internal, rachel, 1) exited non-zero
on 'SIP/pieter-3079'
Regards,
Pieter
On Monday 22 May 2006 14:19, you wrote:
> An Ethereal trace would be useful.. I don't see any "415 Unsupported Media
> Type" messages in the text you sent.
>
> -----Original Message-----
> From: Pieter Claassen [mailto:pieter at claassen.co.uk]
> Sent: 22 May 2006 13:13
> To: Steve Langstaff
> Subject: Re: [Asterisk-Users] Recommended SIP phones?
>
> On Monday 22 May 2006 13:20, Steve Langstaff wrote:
> > Can you get a trace of the SIP traffic between the phone(s) and the
> > server, using something like Ethereal? - It might be possible to simply
> > solve your "Unsupported Media Type" problem.
>
> Included is the whole conversation (as per linphonec -d 5). THis call was
> done from linphonec to linphone and not via Asterisk.
>
> Any ideas? If required, I can also get an ethereal trace.
>
> THanks,
> Pieter
>
>
> linphonec> call sip:rachel at 192.168.1.31
> Contacting sip:rachel at 192.168.1.31
>
> | INFO1 | <eXutils.c: 416> Outgoing interface to reach 15.128.128.93 is
>
> 192.168.1.3.
>
> | INFO1 | <eXutils.c: 416> Outgoing interface to reach 192.168.1.31 is
>
> 192.168.1.3.
>
> | INFO1 | <eXutils.c: 416> Outgoing interface to reach 192.168.1.31 is
>
> 192.168.1.3.
>
> | INFO2 | <osip_transaction.c: 131> allocating transaction ressource 1
>
> 1727161342
>
> | INFO2 | <ict.c: 34> allocating ICT context
>
> linphonec> | INFO2 | <eXutils.c: 492> IPv4 address detected: 192.168.1.31
>
> | INFO2 | <eXutils.c: 541> DNS resolution with 192.168.1.31:5060
> | INFO1 | <jcallback.c: 148> Message sent:
>
> INVITE sip:rachel at 192.168.1.31 SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.3:5060;rport;branch=z9hG4bK825087290
> From: <sip:pieter at 192.168.1.3>;tag=981113905
> To: <sip:rachel at 192.168.1.31>
> Call-ID: 1727161342 at 192.168.1.3
> CSeq: 20 INVITE
> Contact: <sip:pieter at 192.168.1.3:5060>
> Max-Forwards: 5
> User-Agent: Linphone-1.2.0/eXosip
> Subject: Phone call
> Expires: 120
> Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE
> Content-Type: application/sdp
> Content-Length: 352
>
> v=0
> o=pieter 123456 654321 IN IP4 192.168.1.3
> s=A conversation
> c=IN IP4 192.168.1.3
> t=0 0
> m=audio 7078 RTP/AVP 0 3 8 110 111 115 101
> b=AS:20
> a=rtpmap:0 PCMU/8000/1
> a=rtpmap:3 GSM/8000/1
> a=rtpmap:8 PCMA/8000/1
> a=rtpmap:110 speex/8000/1
> a=rtpmap:111 speex/16000/1
> a=rtpmap:115 1015/8000/1
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-11
> (len=16 sizeof(addr)=128 28)
>
> | INFO1 | <jcallback.c: 515> cb_sndinvite (id=1)
> | INFO1 | <eXosip.c: 340> eXosip: timer sec:0 usec:496643!
> | INFO1 | <udp.c: 2193> Received message:
>
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP 192.168.1.3:5060;rport=5060;branch=z9hG4bK825087290
> From: <sip:pieter at 192.168.1.3>;tag=981113905
> To: <sip:rachel at 192.168.1.31>
> Call-ID: 1727161342 at 192.168.1.3
> CSeq: 20 INVITE
> Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, MESSAGE, INFO
> Content-Length: 0
>
> | INFO3 | <osip_event.c: 89> MESSAGE REC. CALLID:1727161342
> | INFO1 | <jcallback.c: 601> cb_rcv1xx (id=1)
> | INFO1 | <eXosip.c: 333> eXosip: Reseting timer to 15s before waking up!
> | INFO1 | <udp.c: 2193> Received message:
>
> SIP/2.0 101 Dialog Establishement
> Via: SIP/2.0/UDP 192.168.1.3:5060;rport=5060;branch=z9hG4bK825087290
> From: <sip:pieter at 192.168.1.3>;tag=981113905
> To: <sip:rachel at 192.168.1.31>;tag=1828902728
> Call-ID: 1727161342 at 192.168.1.3
> CSeq: 20 INVITE
> Contact: <sip:rachel at 192.168.1.31:5060>
> Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, MESSAGE, INFO
> Content-Length: 0
>
> | INFO3 | <osip_event.c: 89> MESSAGE REC. CALLID:1727161342
> | INFO1 | <jcallback.c: 601> cb_rcv1xx (id=1)
> | INFO1 | <eXosip.c: 333> eXosip: Reseting timer to 15s before waking up!
>
> LinphoneCore-Message: CALL_PROCEEDING
>
> | INFO1 | <udp.c: 2193> Received message:
>
> SIP/2.0 180 Ringing
> Via: SIP/2.0/UDP 192.168.1.3:5060;rport=5060;branch=z9hG4bK825087290
> From: <sip:pieter at 192.168.1.3>;tag=981113905
> To: <sip:rachel at 192.168.1.31>;tag=1828902728
> Call-ID: 1727161342 at 192.168.1.3
> CSeq: 20 INVITE
> Contact: <sip:rachel at 192.168.1.31:5060>
> Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, MESSAGE, INFO
> Content-Length: 0
>
> | INFO3 | <osip_event.c: 89> MESSAGE REC. CALLID:1727161342
> | INFO1 | <jcallback.c: 601> cb_rcv1xx (id=1)
> | INFO1 | <eXosip.c: 333> eXosip: Reseting timer to 15s before waking up!
>
> LinphoneCore-Message: CALL_RINGING
>
> LinphoneCore-Message: Remote ringing...
> MediaStreamer-Message: alsa_set_params: blocksize=512.
> MediaStreamer-Message: ms_filter_add_link: ringplay,0 -> OssWrite,0
> MediaStreamer-Message: Opening sound card [SiS SI7012 (Advanced Linux Sound
> Architecture)] in playback mode with stereo=0,rate=8000,bits=16
> MediaStreamer-Message: alsa_set_params: blocksize=512.
>
> | INFO1 | <udp.c: 2193> Received message:
>
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 192.168.1.3:5060;rport=5060;branch=z9hG4bK825087290
> From: <sip:pieter at 192.168.1.3>;tag=981113905
> To: <sip:rachel at 192.168.1.31>;tag=1828902728
> Call-ID: 1727161342 at 192.168.1.3
> CSeq: 20 INVITE
> Contact: <sip:rachel at 192.168.1.31:5060>
> Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, MESSAGE, INFO
> Content-Type: application/sdp
> Content-Length: 302
>
> v=0
> o=rachel 123456 654321 IN IP4 192.168.1.31
> s=A conversation
> c=IN IP4 192.168.1.31
> t=0 0
> m=audio 7078 RTP/AVP 3 110 111 115 101
> b=AS:20
> a=rtpmap:3 GSM/8000/1
> a=rtpmap:110 speex/8000/1
> a=rtpmap:111 speex/16000/1
> a=rtpmap:115 1015/8000/1
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-11
>
> | INFO3 | <osip_event.c: 89> MESSAGE REC. CALLID:1727161342
> | INFO1 | <jcallback.c: 1213> cb_rcv2xx (id=1)
> | INFO1 | <eXutils.c: 416> Outgoing interface to reach 192.168.1.31 is
>
> 192.168.1.3.
>
> | INFO2 | <eXutils.c: 492> IPv4 address detected: 192.168.1.31
> | INFO2 | <eXutils.c: 541> DNS resolution with 192.168.1.31:5060
> | INFO1 | <jcallback.c: 148> Message sent:
>
> ACK sip:rachel at 192.168.1.31:5060 SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.3:5060;rport;branch=z9hG4bK2000556490
> From: <sip:pieter at 192.168.1.3>;tag=981113905
> To: <sip:rachel at 192.168.1.31>;tag=1828902728
> Call-ID: 1727161342 at 192.168.1.3
> CSeq: 20 ACK
> Contact: <sip:pieter at 192.168.1.3:5060>
> Max-Forwards: 5
> User-Agent: Linphone-1.2.0/eXosip
> Content-Length: 0
>
> (len=16 sizeof(addr)=128 28)
>
> | INFO1 | <jcallback.c: 189> cb_ict_kill_transaction (id=1)
> | INFO1 | <eXosip.c: 333> eXosip: Reseting timer to 15s before waking up!
>
> LinphoneCore-Message: CALL_ANSWERED
>
> Connected.
> MediaStreamer-Message: Mediastreamer processing thread is exiting.
> MediaStreamer-Message: Closing writing channel of soundcard.
> MediaStreamer-Message: ms_filter_add_link: OssRead,0 -> GSMEncoder,0
> MediaStreamer-Message: ms_filter_add_link: GSMEncoder,0 -> RTPSend,0
> MediaStreamer-Message: ms_filter_add_link: RTPRecv,0 -> GSMDecoder,0
> MediaStreamer-Message: ms_filter_add_link: GSMDecoder,0 -> OssWrite,0
> MediaStreamer-Message: Opening sound card [SiS SI7012 (Advanced Linux Sound
> Architecture)] in capture mode with stereo=0,rate=8000,bits=16
> MediaStreamer-Message: alsa_set_params: blocksize=512.
> MediaStreamer-Message: Opening sound card [SiS SI7012 (Advanced Linux Sound
> Architecture)] in playback mode with stereo=0,rate=8000,bits=16
> MediaStreamer-Message: alsa_set_params: blocksize=512.
> LinphoneCore-Message: CALL_STARTAUDIO
>
> oRTP-Message: payload type changed to 3(GSM) !
>
> | INFO1 | <eXosip.c: 333> eXosip: Reseting timer to 15s before waking up!
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