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


From: Simon Morlat
Subject: Re: [Linphone-users] Problem with incoming calls when busy...
Date: Wed, 5 Oct 2005 18:09:01 +0200
User-agent: KMail/1.8.2

Hello,

Thanks to this call log I could identify the bug and fix it.
Please try linphone-1.2.0pre5 from 
http://simon.morlat.free.fr/download/unstable/source
And tell me if it also works for you.
Thanks a lot.

Simon

Le Dimanche 2 Octobre 2005 14:25, Magnus Sandin a écrit :
> Hello Simon.
>
> This is the scenario:
>
> 1. I start linphone
> 2. I'm placing a call from linphone
> 3. I call linphone from another phone
>
> I have included the logs from step 2 above:
> | INFO1 | <eXosip.c: 339> eXosip: timer sec:0 usec:149!
> | INFO1 | <jcallback.c: 321> cb_nist_kill_transaction (id=7)
> | INFO1 | <eXosip.c: 339> eXosip: timer sec:0 usec:10354!
> | INFO1 | <jcallback.c: 321> cb_nist_kill_transaction (id=8)
> | INFO1 | <eXosip.c: 339> eXosip: timer sec:0 usec:13840!
> | INFO1 | <eXosip.c: 339> eXosip: timer sec:0 usec:422!
> | INFO1 | <jcallback.c: 321> cb_nist_kill_transaction (id=9)
> | INFO1 | <eXosip.c: 339> eXosip: timer sec:0 usec:21008!
> | INFO1 | <jcallback.c: 321> cb_nist_kill_transaction (id=10)
> | INFO1 | <eXosip.c: 339> eXosip: timer sec:0 usec:10275!
> | INFO1 | <jcallback.c: 321> cb_nist_kill_transaction (id=11)
> | INFO1 | <eXosip.c: 332> eXosip: Reseting timer to 15s before waking up!
> | INFO1 | <eXosip.c: 339> eXosip: timer sec:4 usec:2520!
> | INFO1 | <eXosip.c: 339> eXosip: timer sec:0 usec:126!
> | INFO1 | <jcallback.c: 189> cb_ict_kill_transaction (id=13)
> | INFO1 | <eXosip.c: 332> eXosip: Reseting timer to 15s before waking up!
> | INFO1 | <udp.c: 2193> Received message:
>
> INVITE sip:address@hidden:5060 SIP/2.0
> Via: SIP/2.0/UDP aa.bb.cc.dd:5060;branch=z9hG4bK4285f30d
> From: "Unknown" <sip:address@hidden>;tag=as266b87dc
> To: <sip:address@hidden:5060>
> Contact: <sip:address@hidden>
> Call-ID: address@hidden
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX
> Date: Sun, 02 Oct 2005 12:21:04 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Content-Type: application/sdp
> Content-Length: 215
>
> v=0
> o=root 2289 2289 IN IP4 aa.bb.cc.dd
> s=session
> c=IN IP4 aa.bb.cc.dd
> t=0 0
> m=audio 14034 RTP/AVP 3 101
> a=rtpmap:3 GSM/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
>
> | INFO3 | <osip_event.c: 84> MESSAGE REC.
>
> CALLID:3bf5cecb35000d767d0e53fd6e9f0456
>
> | INFO1 | <udp.c: 2228> This is a request
> | INFO2 | <osip_transaction.c: 129> allocating transaction ressource 15
>
> 3bf5cecb35000d767d0e53fd6e9f0456
>
> | INFO2 | <ist.c: 32> allocating IST context
> | INFO1 | <eXutils.c: 416> Outgoing interface to reach aa.bb.cc.dd is
>
> 10.104.2.17.
>
> | INFO1 | <jcallback.c: 332> cb_rcvinvite (id=15)
> | INFO2 | <eXutils.c: 492> IPv4 address detected: aa.bb.cc.dd
> | INFO2 | <eXutils.c: 541> DNS resolution with aa.bb.cc.dd:5060
> | INFO1 | <jcallback.c: 148> Message sent:
>
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP aa.bb.cc.dd:5060;branch=z9hG4bK4285f30d
> From: "Unknown" <sip:address@hidden>;tag=as266b87dc
> To: <sip:address@hidden:5060>
> Call-ID: address@hidden
> CSeq: 102 INVITE
> Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, MESSAGE, INFO
> Content-Length: 0
>
>  (len=16 sizeof(addr)=128 28)
>
> | INFO1 | <jcallback.c: 1669> cb_snd1xx (id=15)
> | INFO2 | <eXutils.c: 492> IPv4 address detected: aa.bb.cc.dd
> | INFO2 | <eXutils.c: 541> DNS resolution with aa.bb.cc.dd:5060
> | INFO1 | <jcallback.c: 148> Message sent:
>
> SIP/2.0 101 Dialog Establishement
> Via: SIP/2.0/UDP aa.bb.cc.dd:5060;branch=z9hG4bK4285f30d
> From: "Unknown" <sip:address@hidden>;tag=as266b87dc
> To: <sip:address@hidden:5060>;tag=1745034882
> Call-ID: address@hidden
> CSeq: 102 INVITE
> Contact: <sip:address@hidden:5060>
> Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, MESSAGE, INFO
> Content-Length: 0
>
>  (len=16 sizeof(addr)=128 28)
>
> | INFO1 | <jcallback.c: 1669> cb_snd1xx (id=15)
> | INFO1 | <eXosip.c: 332> eXosip: Reseting timer to 15s before waking up!
> | INFO1 | <eXosip.c: 332> eXosip: Reseting timer to 15s before waking up!
>
> LinphoneCore-Message: CALL_NEW
>
> | INFO2 | <eXutils.c: 492> IPv4 address detected: aa.bb.cc.dd
> | INFO2 | <eXutils.c: 541> DNS resolution with aa.bb.cc.dd:5060
> | INFO1 | <jcallback.c: 148> Message sent:
>
> SIP/2.0 486 Busy Here
> Via: SIP/2.0/UDP aa.bb.cc.dd:5060;branch=z9hG4bK4285f30d
> From: "Unknown" <sip:address@hidden>;tag=as266b87dc
> To: <sip:address@hidden:5060>;tag=1745034882
> Call-ID: address@hidden
> CSeq: 102 INVITE
> Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, MESSAGE, INFO
> Content-Length: 0
>
>  (len=16 sizeof(addr)=128 28)
>
> | INFO1 | <jcallback.c: 1728> cb_snd4xx (id=15)
> | INFO1 | <eXosip.c: 339> eXosip: timer sec:0 usec:499962!
> | INFO1 | <udp.c: 2193> Received message:
>
> ACK sip:address@hidden:5060 SIP/2.0
> Via: SIP/2.0/UDP aa.bb.cc.dd:5060;branch=z9hG4bK4285f30d
> From: "Unknown" <sip:address@hidden>;tag=as266b87dc
> To: <sip:address@hidden:5060>;tag=1745034882
> Contact: <sip:address@hidden>
> Call-ID: address@hidden
> CSeq: 102 ACK
> User-Agent: Asterisk PBX
> Content-Length: 0
>
> | INFO3 | <osip_event.c: 84> MESSAGE REC.
>
> CALLID:3bf5cecb35000d767d0e53fd6e9f0456
>
> | INFO1 | <jcallback.c: 337> cb_rcvack (id=15)
> | INFO1 | <eXosip.c: 339> eXosip: timer sec:4 usec:999968!
>
> LinphoneCore-Message: CALL_CLOSED or CANCELLED
>
> LinphoneCore-Message: Call terminated...
> MediaStreamer-Message: Mediastreamer processing thread is exiting.
> MediaStreamer-Message: Closing reading channel of soundcard.
> MediaStreamer-Message: Closing writing channel of soundcard.
> oRTP-stats-Message:
>    Global statistics :
>  number of rtp packet sent=1700
>  number of rtp bytes sent=56100 bytes
>  number of rtp packet received=1697
>  number of rtp bytes received=76365 bytes
>  number of incoming rtp bytes successfully delivered to the
> application=76005
>  number of times the application queried a packet that didn't exist=1702
>  number of rtp packets received too late=0
>  number of rtp packets skipped=1
>  number of bad formatted rtp packets=0
>  number of packet discarded because of queue overflow=0
>
> | INFO1 | <eXosip.c: 339> eXosip: timer sec:0 usec:686!
> | INFO2 | <udp.c: 2452> eXosip: eXosip_release_aborted_calls answered
>
> with a 4xx
>
> | INFO1 | <udp.c: 2620> Release a terminated transaction
> | INFO2 | <osip_transaction.c: 286> free transaction ressource 15
>
> 3bf5cecb35000d767d0e53fd6e9f0456
>
> | INFO2 | <ist.c: 84> free ist ressource
> | INFO1 | <eXosip.c: 332> eXosip: Reseting timer to 15s before waking up!
> | INFO1 | <eXosip.c: 332> eXosip: Reseting timer to 15s before waking up!
>
> LinphoneCore-Message: CALL_RELEASED
>
>
> Regards
> // Magnus Sandin
>
>
> *Magnus Sandin
> *Cache miss - please take better aim next time
>
> Simon Morlat wrote:
> >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




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