Re: [Linphone-users] Problem with incoming calls when busy...

[Simon Morlat]

Hello,

I've reviewed the code and I don't see how possible linphone can hangup after 
replying a 486 Busy.
It would help me if you could send me a complete log: use linphone --verbose.
Thanks

Simon

Le Lundi 26 Septembre 2005 21:58, Magnus Sandin a écrit :
> Hello!
>
> I use Linphone version 1.1.0 on Ubuntu 5.04 and I think it works really
> good, but there seems to be one big problem.
>
> I have the Linphone registered to our Aterisk PBX and I can call in and
> out, it just works great. However if I call anyone and during that call
> anyone else calls my extension (which Linphone is registered to)
> Linphone hangs up (Communication ended) but Asterisk is never told about
> it!?
>
> The strange part is that I did a sniff on port 5060 and discovered that
> Linphone actually tells Asterisk that the line is busy. This is also
> indicated because the calling party is transferred to the voicemail,
> which is the correct behaviour by Asterisk if an extension is busy.
>
> This is a log from SIP port 5060 when the second call comes in:
>
>
> 192.168.31.4 is my Linphone
> aa.bb.cc.dd is the Asterisk PBX
>
> #
> U aa.bb.cc.dd:5060 -> 192.168.31.4:5060
>   INVITE sip:address@hidden:5060 SIP/2.0..Via: SIP/2.0/UDP
> aa.bb.cc.dd:5060;branch=z9hG4bK26faa6bc..From: "031123456"
>    <sip:address@hidden>;tag=as69bd634d..To:
> <sip:address@hidden:5060>..Contact: <sip:address@hidden>.
>   .Call-ID: address@hidden: 102
> INVITE..User-Agent: Asterisk PBX..Date: Mon, 26 Sep
>   2005 19:47:09 GMT..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE,
> REFER..Content-Type: application/sdp..Content-Length: 215..
>   ..v=0..o=root 3232 3232 IN IP4 aa.bb.cc.dd..s=session..c=IN IP4
> aa.bb.cc.dd..t=0 0..m=audio 17836 RTP/AVP 3 101..a=r
>   tpmap:3 GSM/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101
> 0-16..a=silenceSupp:off - - - -..
> #
> U 192.168.31.4:5060 -> aa.bb.cc.dd:5060
>   SIP/2.0 100 Trying..Via: SIP/2.0/UDP
> aa.bb.cc.dd:5060;branch=z9hG4bK26faa6bc..From: "031123456"
> <sip:address@hidden
>   cc.dd>;tag=as69bd634d..To: <sip:address@hidden:5060>..Call-ID:
> address@hidden:
>    102 INVITE..Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE,
> NOTIFY, MESSAGE, INFO..Content-Length: 0....
> #
> U 192.168.31.4:5060 -> aa.bb.cc.dd:5060
>   SIP/2.0 101 Dialog Establishement..Via: SIP/2.0/UDP
> aa.bb.cc.dd:5060;branch=z9hG4bK26faa6bc..From: "031123456" <sip:03
>   address@hidden>;tag=as69bd634d..To:
> <sip:address@hidden:5060>;tag=1089067986..Call-ID:
> 0250c4a667e0657f51d43da
>   address@hidden: 102 INVITE..Contact:
> <sip:address@hidden:5060>..Allow: INVITE, ACK, OPTIONS, CANCEL, B
>   YE, SUBSCRIBE, NOTIFY, MESSAGE, INFO..Content-Length: 0....
> #
> U 192.168.31.4:5060 -> aa.bb.cc.dd:5060
>   SIP/2.0 486 Busy Here..Via: SIP/2.0/UDP
> aa.bb.cc.dd:5060;branch=z9hG4bK26faa6bc..From: "031123456"
> <sip:address@hidden
>   bb.cc.dd>;tag=as69bd634d..To:
> <sip:address@hidden:5060>;tag=1089067986..Call-ID:
> address@hidden
>   .bb.cc.dd..CSeq: 102 INVITE..Allow: INVITE, ACK, OPTIONS, CANCEL, BYE,
> SUBSCRIBE, NOTIFY, MESSAGE, INFO..Content-Lengt
>   h: 0....
> #
> U aa.bb.cc.dd:5060 -> 192.168.31.4:5060
>   ACK sip:address@hidden:5060 SIP/2.0..Via: SIP/2.0/UDP
> aa.bb.cc.dd:5060;branch=z9hG4bK26faa6bc..From: "031123456" <s
>   ip:address@hidden>;tag=as69bd634d..To:
> <sip:address@hidden:5060>;tag=1089067986..Contact: <sip:address@hidden
>   bb.cc.dd>..Call-ID:
> address@hidden: 102 ACK..User-Agent:
> Asterisk PBX..Content-L
>   ength: 0....
>
>
> Any ideas why this happens?
>
> Regards
> // Magnus Sandin