You can set the contact manually, look up set_contact_alias() ... I'm not sure whether this would be advisable though... you need someone more kamailio-knowledgeable...<br><br>I think the problem is the linphone config, i sometimes use it and have never seen that... <br><div class="gmail_quote"><div dir="ltr">On Wed, May 17, 2017 at 8:45 PM Jean Cérien <<a href="mailto:cerien.jean@gmail.com">cerien.jean@gmail.com</a>> wrote:<br></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><br><div>Thanks for the help. </div><div>I have reverted to the default config file (<a href="https://github.com/sipwise/kamailio/blob/master/etc/kamailio.cfg" target="_blank">https://github.com/sipwise/kamailio/blob/master/etc/kamailio.cfg</a>), and trying to place a call between two ua (linphone & zoiper). I am testing totally on a LAN, clients & kamailo on the same subnet, no nat.</div><div><br></div><div>Register is fine, Invite is fine, I receive the 200OK from called, then I get the ACK from the calling, and while processing it, I get the following errors in the log: </div><div><div><font face="monospace, monospace"><br></font></div><div><font face="monospace, monospace">May 17 14:12:32 kamailio : ERROR: <core> [resolve.c:1694]: sip_hostport2su(): could not resolve hostname: "(null)"</font></div><div><font face="monospace, monospace">May 17 14:12:32 kamailio : ERROR: <core> [forward.c:495]: forward_request(): bad host name (null), dropping packet</font></div><div><font face="monospace, monospace">May 17 14:12:32 kamailio : ERROR: sl [sl_funcs.c:363]: sl_reply_error(): ERROR: sl_reply_error used: Unresolvable destination (478/SL)</font></div></div><div><br></div><div>Digging a bit more, I've noticed that the calling party, using Linphone, has the Contact field a bit weird: </div><div>Contact: <sip:user@(null)></div><div><br></div><div>Changing to another softphone that populates correctly this field works ok. Is there a way to mitigate this external issue ? The softphone works ok directly connected to asterisk for instance</div><div><br></div><div>Rgds</div></div><div dir="ltr"><div>J.</div><div><br></div></div><div class="gmail_extra"><br><div class="gmail_quote">On Tue, May 16, 2017 at 6:15 PM, David Villasmil <span dir="ltr"><<a href="mailto:david.villasmil.work@gmail.com" target="_blank">david.villasmil.work@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">There isn't an ACK received, check in kamailio side to make sure it is received. This is most probably a nat issue.<br><div class="gmail_quote"><div><div class="m_-7672058114087510294h5"><div dir="ltr">On Tue, May 16, 2017 at 11:20 PM Jean Cérien <<a href="mailto:cerien.jean@gmail.com" target="_blank">cerien.jean@gmail.com</a>> wrote:<br></div></div></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div class="m_-7672058114087510294h5"><div dir="ltr"><br><div>Hello</div><div><br></div><div>I am getting started with Kamailio (or restarted, used it briefly years ago), with the final objective to do load balancing.</div><div><br></div><div>For the time being, I am just trying to have one asterisk and one kamailio, on the same box. I have setup a box with an asterisk 11.3, and kamailio 4.4. I've taken the config file from <a href="http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb" target="_blank">http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb</a>. </div><div><br></div><div>My idea is that asterisk runs on port 5080, while kamailio on port 5060. Client interacts with Kamailio on port 5060. </div><div><br></div><div>It almost works... Registration is fine, but when I send an invite, it is properly acknowledged (by asterisk 100 trying then 200 OK) - but the OK message gets repeated multiple times and asterisk issues its infamous 'Retransmission timeout reached ...' - as if Kamailio wasnt processing it. See below ngrep traces between asterisk and kamailio</div><div><br></div><div>Any ideas where to look ?</div><div><br></div><div>Thanks</div><div>J.</div><div><br></div><div><div><font face="monospace, monospace">#</font></div><div><font face="monospace, monospace">U +18.289105 <a href="http://192.168.2.228:5060" target="_blank">192.168.2.228:5060</a> -> <a href="http://192.168.2.228:5080" target="_blank">192.168.2.228:5080</a></font></div><div><font face="monospace, monospace"> INVITE <a href="mailto:sip%3A102@192.168.2.228" target="_blank">sip:102@192.168.2.228</a> SIP/2.0..Record-Route: <sip:192.168.2.228;lr=on;ftag=1034946464>..Via: SIP/2.0/UDP 192.168.2.228;branch=z9hG4bK9e22.bff83e1026a667df63b9922</font></div><div><font face="monospace, monospace"> 4198ef593.0..Via: SIP/2.0/UDP 192.168.2.200:5085;received=192.168.2.200;rport=5085;branch=z9hG4bK1247964647..From: <<a href="mailto:sip%3A199@192.168.2.228" target="_blank">sip:199@192.168.2.228</a>>;tag=1034946464..To: <sip:102@</font></div><div><font face="monospace, monospace"> 192.168.2.228>..Call-ID: 1571382735..CSeq: 21 INVITE..Contact: <sip:iper@(null)>..Content-Type: application/sdp..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY</font></div><div><font face="monospace, monospace"> , MESSAGE, SUBSCRIBE, INFO..Max-Forwards: 69..User-Agent: Linphone/3.6.1 (eXosip2/4.1.0)..Subject: Phone call..Content-Length: 437....v=0..o=199 2799 2990 IN IP4 192.</font></div><div><font face="monospace, monospace"> 168.2.200..s=Talk..c=IN IP4 192.168.2.200..t=0 0..m=audio 7078 RTP/AVP 124 111 110 0 8 101..a=rtpmap:124 opus/48000..a=fmtp:124 useinbandfec=1; usedtx=1..a=rtpmap:111 s</font></div><div><font face="monospace, monospace"> peex/16000..a=fmtp:111 vbr=on..a=rtpmap:110 speex/8000..a=fmtp:110 vbr=on..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-11..m=video 9078 RTP/AVP 103 99..a=rtpmap:103</font></div><div><font face="monospace, monospace"> VP8/90000..a=rtpmap:99 MP4V-ES/90000..a=fmtp:99 profile-level-id=3.. </font></div><div><font face="monospace, monospace">#</font></div><div><font face="monospace, monospace">U +0.000683 <a href="http://192.168.2.228:5080" target="_blank">192.168.2.228:5080</a> -> <a href="http://192.168.2.228:5060" target="_blank">192.168.2.228:5060</a></font></div><div><font face="monospace, monospace"> OPTIONS <a href="http://sip:199@192.168.2.228:5060" target="_blank">sip:199@192.168.2.228:5060</a> SIP/2.0..Via: SIP/2.0/UDP 192.168.2.228:5080;branch=z9hG4bK7c2d71fb..Max-Forwards: 70..From: "asterisk" <<a href="http://sip:199@192.168.2.228:5080" target="_blank">sip:199@192.168.2.228:5080</a>>;</font></div><div><font face="monospace, monospace"> tag=as57c98c4b..To: <<a href="http://sip:199@192.168.2.228:5060" target="_blank">sip:199@192.168.2.228:5060</a>>..Contact: <<a href="http://sip:199@192.168.2.228:5080" target="_blank">sip:199@192.168.2.228:5080</a>>..Call-ID: 52fa034372ce18ca2b93fc1817ad38a5@192.168.2.228:5080..CSeq: 102 OPTIONS</font></div><div><font face="monospace, monospace"> ..User-Agent: Asterisk PBX 11.3.0..Date: Tue, 16 May 2017 21:08:29 GMT..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH..Supported: re</font></div><div><font face="monospace, monospace"> places, timer..Content-Length: 0.... </font></div><div><font face="monospace, monospace">#</font></div><div><font face="monospace, monospace">U +0.001035 <a href="http://192.168.2.228:5080" target="_blank">192.168.2.228:5080</a> -> <a href="http://192.168.2.228:5060" target="_blank">192.168.2.228:5060</a></font></div><div><font face="monospace, monospace"> OPTIONS <a href="http://sip:199@192.168.2.228:5060" target="_blank">sip:199@192.168.2.228:5060</a> SIP/2.0..Via: SIP/2.0/UDP 192.168.2.228:5080;branch=z9hG4bK4248158e..Max-Forwards: 70..From: "asterisk" <<a href="http://sip:199@192.168.2.228:5080" target="_blank">sip:199@192.168.2.228:5080</a>>;</font></div><div><font face="monospace, monospace"> tag=as30d9cfb4..To: <<a href="http://sip:199@192.168.2.228:5060" target="_blank">sip:199@192.168.2.228:5060</a>>..Contact: <<a href="http://sip:199@192.168.2.228:5080" target="_blank">sip:199@192.168.2.228:5080</a>>..Call-ID: 4331da391bca02965b2af65254717a18@192.168.2.228:5080..CSeq: 102 OPTIONS</font></div><div><font face="monospace, monospace"> ..User-Agent: Asterisk PBX 11.3.0..Date: Tue, 16 May 2017 21:08:29 GMT..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH..Supported: re</font></div><div><font face="monospace, monospace"> places, timer..Content-Length: 0.... </font></div><div><font face="monospace, monospace">#</font></div><div><font face="monospace, monospace">U +0.000407 <a href="http://192.168.2.228:5080" target="_blank">192.168.2.228:5080</a> -> <a href="http://192.168.2.228:5060" target="_blank">192.168.2.228:5060</a></font></div><div><font face="monospace, monospace"> SIP/2.0 100 Trying..Via: SIP/2.0/UDP 192.168.2.228;branch=z9hG4bK9e22.bff83e1026a667df63b99224198ef593.0;received=192.168.2.228..Via: SIP/2.0/UDP 192.168.2.200:5085;rec</font></div><div><font face="monospace, monospace"> eived=192.168.2.200;rport=5085;branch=z9hG4bK1247964647..Record-Route: <sip:192.168.2.228;lr=on;ftag=1034946464>..From: <<a href="mailto:sip%3A199@192.168.2.228" target="_blank">sip:199@192.168.2.228</a>>;tag=1034946464..To: <sip</font></div><div><font face="monospace, monospace"> :<a href="mailto:102@192.168.2.228" target="_blank">102@192.168.2.228</a>>..Call-ID: 1571382735..CSeq: 21 INVITE..Server: Asterisk PBX 11.3.0..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS</font></div><div><font face="monospace, monospace"> H..Supported: replaces, timer..Contact: <<a href="http://sip:102@192.168.2.228:5080" target="_blank">sip:102@192.168.2.228:5080</a>>..Content-Length: 0.... </font></div><div><font face="monospace, monospace">#</font></div><div><font face="monospace, monospace">U +0.003961 <a href="http://192.168.2.228:5080" target="_blank">192.168.2.228:5080</a> -> <a href="http://192.168.2.228:5060" target="_blank">192.168.2.228:5060</a></font></div><div><font face="monospace, monospace"> SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.2.228;branch=z9hG4bK9e22.bff83e1026a667df63b99224198ef593.0;received=192.168.2.228..Via: SIP/2.0/UDP 192.168.2.200:5085;receive</font></div><div><font face="monospace, monospace"> d=192.168.2.200;rport=5085;branch=z9hG4bK1247964647..Record-Route: <sip:192.168.2.228;lr=on;ftag=1034946464>..From: <<a href="mailto:sip%3A199@192.168.2.228" target="_blank">sip:199@192.168.2.228</a>>;tag=1034946464..To: <sip:102</font></div><div><font face="monospace, monospace"> @<a href="http://192.168.2.228" target="_blank">192.168.2.228</a>>;tag=as497f35c0..Call-ID: 1571382735..CSeq: 21 INVITE..Server: Asterisk PBX 11.3.0..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, I</font></div><div><font face="monospace, monospace"> NFO, PUBLISH..Supported: replaces, timer..Contact: <<a href="http://sip:102@192.168.2.228:5080" target="_blank">sip:102@192.168.2.228:5080</a>>..Content-Type: application/sdp..Content-Length: 312....v=0..o=root 350189084 350189084 I</font></div><div><font face="monospace, monospace"> N IP4 192.168.2.228..s=Asterisk PBX 11.3.0..c=IN IP4 192.168.2.228..t=0 0..m=audio 17148 RTP/AVP 0 8 101..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:101 telep</font></div><div><font face="monospace, monospace"> hone-event/8000..a=fmtp:101 0-16..a=silenceSupp:off - - - -..a=ptime:20..a=sendrecv..m=video 0 RTP/AVP 103 99.. </font></div><div><font face="monospace, monospace">#</font></div><div><font face="monospace, monospace">U +0.087859 <a href="http://192.168.2.228:5060" target="_blank">192.168.2.228:5060</a> -> <a href="http://192.168.2.228:5080" target="_blank">192.168.2.228:5080</a></font></div><div><font face="monospace, monospace"> SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.2.228:5080;branch=z9hG4bK7c2d71fb..From: "asterisk" <<a href="http://sip:199@192.168.2.228:5080" target="_blank">sip:199@192.168.2.228:5080</a>>;tag=as57c98c4b..To: <<a href="http://sip:199@192.168.2.228:506" target="_blank">sip:199@192.168.2.228:506</a></font></div><div><font face="monospace, monospace"> 0>;tag=524348182..Call-ID: 52fa034372ce18ca2b93fc1817ad38a5@192.168.2.228:5080..CSeq: 102 OPTIONS..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, SUBSCRIBE, NOTIFY,</font></div><div><font face="monospace, monospace"> INFO..Accept: application/sdp..User-Agent: Linphone/3.6.1 (eXosip2/4.1.0)..Content-Length: 0.... </font></div><div><font face="monospace, monospace">#</font></div><div><font face="monospace, monospace">U +0.000213 <a href="http://192.168.2.228:5060" target="_blank">192.168.2.228:5060</a> -> <a href="http://192.168.2.228:5080" target="_blank">192.168.2.228:5080</a></font></div><div><font face="monospace, monospace"> SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.2.228:5080;branch=z9hG4bK4248158e..From: "asterisk" <<a href="http://sip:199@192.168.2.228:5080" target="_blank">sip:199@192.168.2.228:5080</a>>;tag=as30d9cfb4..To: <<a href="http://sip:199@192.168.2.228:506" target="_blank">sip:199@192.168.2.228:506</a></font></div><div><font face="monospace, monospace"> 0>;tag=939659485..Call-ID: 4331da391bca02965b2af65254717a18@192.168.2.228:5080..CSeq: 102 OPTIONS..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, SUBSCRIBE, NOTIFY,</font></div><div><font face="monospace, monospace"> INFO..Accept: application/sdp..User-Agent: Linphone/3.6.1 (eXosip2/4.1.0)..Content-Length: 0.... </font></div><div><font face="monospace, monospace">#</font></div><div><font face="monospace, monospace">U +0.011138 <a href="http://192.168.2.228:5080" target="_blank">192.168.2.228:5080</a> -> <a href="http://192.168.2.228:5060" target="_blank">192.168.2.228:5060</a></font></div><div><font face="monospace, monospace"> SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.2.228;branch=z9hG4bK9e22.bff83e1026a667df63b99224198ef593.0;received=192.168.2.228..Via: SIP/2.0/UDP 192.168.2.200:5085;receive</font></div><div><font face="monospace, monospace"> d=192.168.2.200;rport=5085;branch=z9hG4bK1247964647..Record-Route: <sip:192.168.2.228;lr=on;ftag=1034946464>..From: <<a href="mailto:sip%3A199@192.168.2.228" target="_blank">sip:199@192.168.2.228</a>>;tag=1034946464..To: <sip:102</font></div><div><font face="monospace, monospace"> @<a href="http://192.168.2.228" target="_blank">192.168.2.228</a>>;tag=as497f35c0..Call-ID: 1571382735..CSeq: 21 INVITE..Server: Asterisk PBX 11.3.0..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, I</font></div><div><font face="monospace, monospace"> NFO, PUBLISH..Supported: replaces, timer..Contact: <<a href="http://sip:102@192.168.2.228:5080" target="_blank">sip:102@192.168.2.228:5080</a>>..Content-Type: application/sdp..Content-Length: 312....v=0..o=root 350189084 350189084 I</font></div><div><font face="monospace, monospace"> N IP4 192.168.2.228..s=Asterisk PBX 11.3.0..c=IN IP4 192.168.2.228..t=0 0..m=audio 17148 RTP/AVP 0 8 101..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:101 telep</font></div><div><font face="monospace, monospace"> hone-event/8000..a=fmtp:101 0-16..a=silenceSupp:off - - - -..a=ptime:20..a=sendrecv..m=video 0 RTP/AVP 103 99.. </font></div><div><font face="monospace, monospace">#</font></div><div><font face="monospace, monospace">U +0.200291 <a href="http://192.168.2.228:5080" target="_blank">192.168.2.228:5080</a> -> <a href="http://192.168.2.228:5060" target="_blank">192.168.2.228:5060</a></font></div><div><font face="monospace, monospace"> SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.2.228;branch=z9hG4bK9e22.bff83e1026a667df63b99224198ef593.0;received=192.168.2.228..Via: SIP/2.0/UDP 192.168.2.200:5085;receive</font></div><div><font face="monospace, monospace"> d=192.168.2.200;rport=5085;branch=z9hG4bK1247964647..Record-Route: <sip:192.168.2.228;lr=on;ftag=1034946464>..From: <<a href="mailto:sip%3A199@192.168.2.228" target="_blank">sip:199@192.168.2.228</a>>;tag=1034946464..To: <sip:102</font></div><div><font face="monospace, monospace"> @<a href="http://192.168.2.228" target="_blank">192.168.2.228</a>>;tag=as497f35c0..Call-ID: 1571382735..CSeq: 21 INVITE..Server: Asterisk PBX 11.3.0..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, I</font></div><div><font face="monospace, monospace"> NFO, PUBLISH..Supported: replaces, timer..Contact: <<a href="http://sip:102@192.168.2.228:5080" target="_blank">sip:102@192.168.2.228:5080</a>>..Content-Type: application/sdp..Content-Length: 312....v=0..o=root 350189084 350189084 I</font></div><div><font face="monospace, monospace"> N IP4 192.168.2.228..s=Asterisk PBX 11.3.0..c=IN IP4 192.168.2.228..t=0 0..m=audio 17148 RTP/AVP 0 8 101..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:101 telep</font></div><div><font face="monospace, monospace"> hone-event/8000..a=fmtp:101 0-16..a=silenceSupp:off - - - -..a=ptime:20..a=sendrecv..m=video 0 RTP/AVP 103 99.. </font></div><div><font face="monospace, monospace">#</font></div><div><font face="monospace, monospace">U +0.400478 <a href="http://192.168.2.228:5080" target="_blank">192.168.2.228:5080</a> -> <a href="http://192.168.2.228:5060" target="_blank">192.168.2.228:5060</a></font></div><div><font face="monospace, monospace"> SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.2.228;branch=z9hG4bK9e22.bff83e1026a667df63b99224198ef593.0;received=192.168.2.228..Via: SIP/2.0/UDP 192.168.2.200:5085;receive</font></div><div><font face="monospace, monospace"> d=192.168.2.200;rport=5085;branch=z9hG4bK1247964647..Record-Route: <sip:192.168.2.228;lr=on;ftag=1034946464>..From: <<a href="mailto:sip%3A199@192.168.2.228" target="_blank">sip:199@192.168.2.228</a>>;tag=1034946464..To: <sip:102</font></div><div><font face="monospace, monospace"> @<a href="http://192.168.2.228" target="_blank">192.168.2.228</a>>;tag=as497f35c0..Call-ID: 1571382735..CSeq: 21 INVITE..Server: Asterisk PBX 11.3.0..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, I</font></div><div><font face="monospace, monospace"> NFO, PUBLISH..Supported: replaces, timer..Contact: <<a href="http://sip:102@192.168.2.228:5080" target="_blank">sip:102@192.168.2.228:5080</a>>..Content-Type: application/sdp..Content-Length: 312....v=0..o=root 350189084 350189084 I</font></div><div><font face="monospace, monospace"> N IP4 192.168.2.228..s=Asterisk PBX 11.3.0..c=IN IP4 192.168.2.228..t=0 0..m=audio 17148 RTP/AVP 0 8 101..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:101 telep</font></div><div><font face="monospace, monospace"> hone-event/8000..a=fmtp:101 0-16..a=silenceSupp:off - - - -..a=ptime:20..a=sendrecv..m=video 0 RTP/AVP 103 99.. </font></div><div><font face="monospace, monospace">#</font></div></div><div><br></div></div></div></div>
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