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


From: Clement Chen
Subject: Re: [Linphone-users] Problem with incoming calls when busy...
Date: Mon, 26 Sep 2005 18:21:17 -0800

>From previous discussions, Linphone is not well supported for reinvite messages.

On 9/26/05, Magnus Sandin <address@hidden> wrote:
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

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

Get Firefox... <http://www.getfirefox.com>


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