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

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

From:	Magnus Sandin
Subject:	Re: [Linphone-users] Problem with incoming calls when busy...
Date:	Wed, 05 Oct 2005 22:52:09 +0200
User-agent:	Mozilla Thunderbird 1.0.6 (X11/20050912)
Hello,

It now works as expected.

Good work! ;-)
I also see that a separate sound device can be configured for the
ringing, which is great.

Best Regards
// Magnus Sandin

*Magnus Sandin
*Cache miss - please take better aim next time




Simon Morlat wrote:

>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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Re: [Linphone-users] Problem with incoming calls when busy..., Magnus Sandin, 2005/10/02
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