[SR-Users] Repeat of packets

David Villasmil david.villasmil.work at gmail.com
Wed May 17 21:23:55 CEST 2017 

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...

I think the problem is the linphone config, i sometimes use it and have
never seen that...
On Wed, May 17, 2017 at 8:45 PM Jean Cérien <cerien.jean at gmail.com> wrote:

>
> Thanks for the help.
> I have reverted to the default config file (
> https://github.com/sipwise/kamailio/blob/master/etc/kamailio.cfg), 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.
>
> 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:
>
> May 17 14:12:32 kamailio : ERROR: <core> [resolve.c:1694]:
> sip_hostport2su(): could not resolve hostname: "(null)"
> May 17 14:12:32 kamailio : ERROR: <core> [forward.c:495]:
> forward_request(): bad host name (null), dropping packet
> May 17 14:12:32 kamailio : ERROR: sl [sl_funcs.c:363]: sl_reply_error():
> ERROR: sl_reply_error used: Unresolvable destination (478/SL)
>
> Digging a bit more, I've noticed that the calling party, using Linphone,
> has the Contact field a bit weird:
> Contact: <sip:user@(null)>
>
> 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
>
> Rgds
> J.
>
>
> On Tue, May 16, 2017 at 6:15 PM, David Villasmil <
> david.villasmil.work at gmail.com> wrote:
>
>> There isn't an ACK received, check in kamailio side to make sure it is
>> received. This is most probably a nat issue.
>> On Tue, May 16, 2017 at 11:20 PM Jean Cérien <cerien.jean at gmail.com>
>> wrote:
>>
>>>
>>> Hello
>>>
>>> I am getting started with Kamailio (or restarted, used it briefly years
>>> ago), with the final objective to do load balancing.
>>>
>>> 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
>>> http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb
>>> .
>>>
>>> My idea is that asterisk runs on port 5080, while kamailio on port 5060.
>>> Client interacts with Kamailio on port 5060.
>>>
>>> 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
>>>
>>> Any ideas where to look ?
>>>
>>> Thanks
>>> J.
>>>
>>> #
>>> U +18.289105 192.168.2.228:5060 -> 192.168.2.228:5080
>>>   INVITE sip:102 at 192.168.2.228 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
>>>   4198ef593.0..Via: SIP/2.0/UDP 192.168.2.200:5085;received=192.168.2.200;rport=5085;branch=z9hG4bK1247964647..From:
>>> <sip:199 at 192.168.2.228>;tag=1034946464..To: <sip:102@
>>>   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
>>>   , 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.
>>>   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
>>>   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
>>>    VP8/90000..a=rtpmap:99 MP4V-ES/90000..a=fmtp:99 profile-level-id=3..
>>>
>>>
>>> #
>>> U +0.000683 192.168.2.228:5080 -> 192.168.2.228:5060
>>>   OPTIONS sip:199 at 192.168.2.228:5060 SIP/2.0..Via: SIP/2.0/UDP
>>> 192.168.2.228:5080;branch=z9hG4bK7c2d71fb..Max-Forwards: 70..From:
>>> "asterisk" <sip:199 at 192.168.2.228:5080>;
>>>   tag=as57c98c4b..To: <sip:199 at 192.168.2.228:5060>..Contact: <
>>> sip:199 at 192.168.2.228:5080>..Call-ID:
>>> 52fa034372ce18ca2b93fc1817ad38a5 at 192.168.2.228:5080..CSeq: 102 OPTIONS
>>>   ..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
>>>   places, timer..Content-Length: 0....
>>>
>>>
>>> #
>>> U +0.001035 192.168.2.228:5080 -> 192.168.2.228:5060
>>>   OPTIONS sip:199 at 192.168.2.228:5060 SIP/2.0..Via: SIP/2.0/UDP
>>> 192.168.2.228:5080;branch=z9hG4bK4248158e..Max-Forwards: 70..From:
>>> "asterisk" <sip:199 at 192.168.2.228:5080>;
>>>   tag=as30d9cfb4..To: <sip:199 at 192.168.2.228:5060>..Contact: <
>>> sip:199 at 192.168.2.228:5080>..Call-ID:
>>> 4331da391bca02965b2af65254717a18 at 192.168.2.228:5080..CSeq: 102 OPTIONS
>>>   ..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
>>>   places, timer..Content-Length: 0....
>>>
>>>
>>> #
>>> U +0.000407 192.168.2.228:5080 -> 192.168.2.228:5060
>>>   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
>>>   eived=192.168.2.200;rport=5085;branch=z9hG4bK1247964647..Record-Route:
>>> <sip:192.168.2.228;lr=on;ftag=1034946464>..From: <sip:199 at 192.168.2.228>;tag=1034946464..To:
>>> <sip
>>>   :102 at 192.168.2.228>..Call-ID: 1571382735..CSeq: 21 INVITE..Server:
>>> Asterisk PBX 11.3.0..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
>>> SUBSCRIBE, NOTIFY, INFO, PUBLIS
>>>   H..Supported: replaces, timer..Contact: <sip:102 at 192.168.2.228:5080>..Content-Length:
>>> 0....
>>>
>>> #
>>> U +0.003961 192.168.2.228:5080 -> 192.168.2.228:5060
>>>   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
>>>   d=192.168.2.200;rport=5085;branch=z9hG4bK1247964647..Record-Route:
>>> <sip:192.168.2.228;lr=on;ftag=1034946464>..From: <sip:199 at 192.168.2.228>;tag=1034946464..To:
>>> <sip:102
>>>   @192.168.2.228>;tag=as497f35c0..Call-ID: 1571382735..CSeq: 21
>>> INVITE..Server: Asterisk PBX 11.3.0..Allow: INVITE, ACK, CANCEL, OPTIONS,
>>> BYE, REFER, SUBSCRIBE, NOTIFY, I
>>>   NFO, PUBLISH..Supported: replaces, timer..Contact: <
>>> sip:102 at 192.168.2.228:5080>..Content-Type:
>>> application/sdp..Content-Length: 312....v=0..o=root 350189084 350189084 I
>>>   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
>>>   hone-event/8000..a=fmtp:101 0-16..a=silenceSupp:off - - -
>>> -..a=ptime:20..a=sendrecv..m=video 0 RTP/AVP 103 99..
>>>
>>> #
>>> U +0.087859 192.168.2.228:5060 -> 192.168.2.228:5080
>>>   SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.2.228:5080;branch=z9hG4bK7c2d71fb..From:
>>> "asterisk" <sip:199 at 192.168.2.228:5080>;tag=as57c98c4b..To: <
>>> sip:199 at 192.168.2.228:506
>>>   0>;tag=524348182..Call-ID:
>>> 52fa034372ce18ca2b93fc1817ad38a5 at 192.168.2.228:5080..CSeq: 102
>>> OPTIONS..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, SUBSCRIBE,
>>> NOTIFY,
>>>    INFO..Accept: application/sdp..User-Agent: Linphone/3.6.1
>>> (eXosip2/4.1.0)..Content-Length: 0....
>>>
>>> #
>>> U +0.000213 192.168.2.228:5060 -> 192.168.2.228:5080
>>>   SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.2.228:5080;branch=z9hG4bK4248158e..From:
>>> "asterisk" <sip:199 at 192.168.2.228:5080>;tag=as30d9cfb4..To: <
>>> sip:199 at 192.168.2.228:506
>>>   0>;tag=939659485..Call-ID:
>>> 4331da391bca02965b2af65254717a18 at 192.168.2.228:5080..CSeq: 102
>>> OPTIONS..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, SUBSCRIBE,
>>> NOTIFY,
>>>    INFO..Accept: application/sdp..User-Agent: Linphone/3.6.1
>>> (eXosip2/4.1.0)..Content-Length: 0....
>>>
>>> #
>>> U +0.011138 192.168.2.228:5080 -> 192.168.2.228:5060
>>>   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
>>>   d=192.168.2.200;rport=5085;branch=z9hG4bK1247964647..Record-Route:
>>> <sip:192.168.2.228;lr=on;ftag=1034946464>..From: <sip:199 at 192.168.2.228>;tag=1034946464..To:
>>> <sip:102
>>>   @192.168.2.228>;tag=as497f35c0..Call-ID: 1571382735..CSeq: 21
>>> INVITE..Server: Asterisk PBX 11.3.0..Allow: INVITE, ACK, CANCEL, OPTIONS,
>>> BYE, REFER, SUBSCRIBE, NOTIFY, I
>>>   NFO, PUBLISH..Supported: replaces, timer..Contact: <
>>> sip:102 at 192.168.2.228:5080>..Content-Type:
>>> application/sdp..Content-Length: 312....v=0..o=root 350189084 350189084 I
>>>   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
>>>   hone-event/8000..a=fmtp:101 0-16..a=silenceSupp:off - - -
>>> -..a=ptime:20..a=sendrecv..m=video 0 RTP/AVP 103 99..
>>>
>>> #
>>> U +0.200291 192.168.2.228:5080 -> 192.168.2.228:5060
>>>   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
>>>   d=192.168.2.200;rport=5085;branch=z9hG4bK1247964647..Record-Route:
>>> <sip:192.168.2.228;lr=on;ftag=1034946464>..From: <sip:199 at 192.168.2.228>;tag=1034946464..To:
>>> <sip:102
>>>   @192.168.2.228>;tag=as497f35c0..Call-ID: 1571382735..CSeq: 21
>>> INVITE..Server: Asterisk PBX 11.3.0..Allow: INVITE, ACK, CANCEL, OPTIONS,
>>> BYE, REFER, SUBSCRIBE, NOTIFY, I
>>>   NFO, PUBLISH..Supported: replaces, timer..Contact: <
>>> sip:102 at 192.168.2.228:5080>..Content-Type:
>>> application/sdp..Content-Length: 312....v=0..o=root 350189084 350189084 I
>>>   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
>>>   hone-event/8000..a=fmtp:101 0-16..a=silenceSupp:off - - -
>>> -..a=ptime:20..a=sendrecv..m=video 0 RTP/AVP 103 99..
>>>
>>> #
>>> U +0.400478 192.168.2.228:5080 -> 192.168.2.228:5060
>>>   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
>>>   d=192.168.2.200;rport=5085;branch=z9hG4bK1247964647..Record-Route:
>>> <sip:192.168.2.228;lr=on;ftag=1034946464>..From: <sip:199 at 192.168.2.228>;tag=1034946464..To:
>>> <sip:102
>>>   @192.168.2.228>;tag=as497f35c0..Call-ID: 1571382735..CSeq: 21
>>> INVITE..Server: Asterisk PBX 11.3.0..Allow: INVITE, ACK, CANCEL, OPTIONS,
>>> BYE, REFER, SUBSCRIBE, NOTIFY, I
>>>   NFO, PUBLISH..Supported: replaces, timer..Contact: <
>>> sip:102 at 192.168.2.228:5080>..Content-Type:
>>> application/sdp..Content-Length: 312....v=0..o=root 350189084 350189084 I
>>>   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
>>>   hone-event/8000..a=fmtp:101 0-16..a=silenceSupp:off - - -
>>> -..a=ptime:20..a=sendrecv..m=video 0 RTP/AVP 103 99..
>>>
>>> #
>>>
>>> _______________________________________________
>>> Kamailio (SER) - Users Mailing List
>>> sr-users at lists.kamailio.org
>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>>
>>
>> _______________________________________________
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>> sr-users at lists.kamailio.org
>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
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