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:[EMAIL PROTECTED]:5060 SIP/2.0 >>Via: SIP/2.0/UDP aa.bb.cc.dd:5060;branch=z9hG4bK4285f30d >>From: "Unknown" <sip:[EMAIL PROTECTED]>;tag=as266b87dc >>To: <sip:[EMAIL PROTECTED]:5060> >>Contact: <sip:[EMAIL PROTECTED]> >>Call-ID: [EMAIL PROTECTED] >>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:[EMAIL PROTECTED]>;tag=as266b87dc >>To: <sip:[EMAIL PROTECTED]:5060> >>Call-ID: [EMAIL PROTECTED] >>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:[EMAIL PROTECTED]>;tag=as266b87dc >>To: <sip:[EMAIL PROTECTED]:5060>;tag=1745034882 >>Call-ID: [EMAIL PROTECTED] >>CSeq: 102 INVITE >>Contact: <sip:[EMAIL PROTECTED]: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:[EMAIL PROTECTED]>;tag=as266b87dc >>To: <sip:[EMAIL PROTECTED]:5060>;tag=1745034882 >>Call-ID: [EMAIL PROTECTED] >>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:[EMAIL PROTECTED]:5060 SIP/2.0 >>Via: SIP/2.0/UDP aa.bb.cc.dd:5060;branch=z9hG4bK4285f30d >>From: "Unknown" <sip:[EMAIL PROTECTED]>;tag=as266b87dc >>To: <sip:[EMAIL PROTECTED]:5060>;tag=1745034882 >>Contact: <sip:[EMAIL PROTECTED]> >>Call-ID: [EMAIL PROTECTED] >>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:[EMAIL PROTECTED]:5060 SIP/2.0..Via: SIP/2.0/UDP >>>>aa.bb.cc.dd:5060;branch=z9hG4bK26faa6bc..From: "031123456" >>>> <sip:[EMAIL PROTECTED]>;tag=as69bd634d..To: >>>><sip:[EMAIL PROTECTED]:5060>..Contact: <sip:[EMAIL PROTECTED]>. >>>> .Call-ID: [EMAIL PROTECTED]: 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:[EMAIL PROTECTED] >>>> cc.dd>;tag=as69bd634d..To: <sip:[EMAIL PROTECTED]:5060>..Call-ID: >>>>[EMAIL PROTECTED]: >>>> 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 >>>> [EMAIL PROTECTED]>;tag=as69bd634d..To: >>>><sip:[EMAIL PROTECTED]:5060>;tag=1089067986..Call-ID: >>>>0250c4a667e0657f51d43da >>>> [EMAIL PROTECTED]: 102 INVITE..Contact: >>>><sip:[EMAIL PROTECTED]: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:[EMAIL PROTECTED] >>>> bb.cc.dd>;tag=as69bd634d..To: >>>><sip:[EMAIL PROTECTED]:5060>;tag=1089067986..Call-ID: >>>>[EMAIL PROTECTED] >>>> .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:[EMAIL PROTECTED]:5060 SIP/2.0..Via: SIP/2.0/UDP >>>>aa.bb.cc.dd:5060;branch=z9hG4bK26faa6bc..From: "031123456" <s >>>> ip:[EMAIL PROTECTED]>;tag=as69bd634d..To: >>>><sip:[EMAIL PROTECTED]:5060>;tag=1089067986..Contact: <sip:[EMAIL PROTECTED] >>>> bb.cc.dd>..Call-ID: >>>>[EMAIL PROTECTED]: 102 ACK..User-Agent: >>>>Asterisk PBX..Content-L >>>> ength: 0.... >>>> >>>> >>>>Any ideas why this happens? >>>> >>>>Regards >>>>// Magnus Sandin >>>> >>>> _______________________________________________ Linphone-users mailing list [email protected] http://lists.nongnu.org/mailman/listinfo/linphone-users
