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Pieter Claassen

2006-May-22 04:04 UTC

[Asterisk-Users] Recommended SIP phones?

I am dying here with linphone (not sure if it is crap software or just me 
being an idiot) but out of the box debian installations of two linphones fail 
with a "Got SIP response 415 "Unsupported Media Type" back from
192.168.1.3"

Can anybody recommend a particular SIP soft phone that broadly satisfies the 
following criteria?
1. Run on linux.
2. Simple to use and setup.
3. Is preferably packaged in APT.
4. OSS compliant license.

Any recommendations (or let me know if I am doing something wrong with 
linphone) are welcome.

Thanks
Pieter

M.Hockings

2006-May-22 08:58 UTC

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[Asterisk-Users] Re: Recommended SIP phones?

Pieter Claassen wrote:> I am dying here with linphone (not sure if it is crap software or just me 
> being an idiot) but out of the box debian installations of two linphones
fail
> with a "Got SIP response 415 "Unsupported Media Type" back
from 192.168.1.3"
> 
> Can anybody recommend a particular SIP soft phone that broadly satisfies
the
> following criteria?
> 1. Run on linux.
> 2. Simple to use and setup.
> 3. Is preferably packaged in APT.
> 4. OSS compliant license.
> 
> Any recommendations (or let me know if I am doing something wrong with 
> linphone) are welcome.
> 
> Thanks
> Pieter
I've been using both the windows and linux versions of xten lite and 
they work fine when configured correctly...

Mike

Pieter Claassen

2006-May-22 14:25 UTC

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[Asterisk-Users] Recommended SIP phones?

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@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@192.168.1.31
> Contacting  sip:rachel@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@192.168.1.31 SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.3:5060;rport;branch=z9hG4bK825087290
> From: <sip:pieter@192.168.1.3>;tag=981113905
> To: <sip:rachel@192.168.1.31>
> Call-ID: 1727161342@192.168.1.3
> CSeq: 20 INVITE
> Contact: <sip:pieter@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@192.168.1.3>;tag=981113905
> To: <sip:rachel@192.168.1.31>
> Call-ID: 1727161342@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@192.168.1.3>;tag=981113905
> To: <sip:rachel@192.168.1.31>;tag=1828902728
> Call-ID: 1727161342@192.168.1.3
> CSeq: 20 INVITE
> Contact: <sip:rachel@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@192.168.1.3>;tag=981113905
> To: <sip:rachel@192.168.1.31>;tag=1828902728
> Call-ID: 1727161342@192.168.1.3
> CSeq: 20 INVITE
> Contact: <sip:rachel@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@192.168.1.3>;tag=981113905
> To: <sip:rachel@192.168.1.31>;tag=1828902728
> Call-ID: 1727161342@192.168.1.3
> CSeq: 20 INVITE
> Contact: <sip:rachel@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@192.168.1.31:5060 SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.3:5060;rport;branch=z9hG4bK2000556490
> From: <sip:pieter@192.168.1.3>;tag=981113905
> To: <sip:rachel@192.168.1.31>;tag=1828902728
> Call-ID: 1727161342@192.168.1.3
> CSeq: 20 ACK
> Contact: <sip:pieter@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
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