[SR-Users] Problem with Kamailio not routing ACK to a 200 OK
Morten Isaksen
misak at misak.dk
Thu Nov 11 23:02:46 CET 2010
Hi Daniel,
The Via line is OK, it was the email formating.
I am using Kamailio 3.0.3 and the sanity docs says:
This function makes a row of sanity checks on the given request. The
function returns false (-1) if one of the checks failed. If one of the
checks fails the module sends a precise error reply via sl_send_reply.
Thus there is no need to reply with a generic error message.
I have solved the problem by removing the sanity_check.
I am just a bit curious why it failed.
But thank you very much for your help.
/Morten
On Thu, Nov 11, 2010 at 8:20 PM, Daniel-Constantin Mierla
<miconda at gmail.com> wrote:
> Hello,
>
> looking now again at the trace you sent first time, the ACK is:
>
> U 2010/10/28 10:51:13.267863 178.21.248.20:5060 -> 178.21.248.7:5060
> ACKsip:1105 at 178.21.248.56:5060 SIP/2.0.
> Record-Route:<sip:178.21.248.20;lr;ftag=AI8DA85D59B9B6634F>.
> Via: SIP/2.0/UDP 178.21.248.20;branch=z9hG4bK690c.97354e4.2.
> Via: SIP/2.0/UDP
> 87.104.233.108:5060;rport=5060;branch=z9hG4bK07103fe69c8af048b9b8216eb2f7233f.
> To:<sip:86987106 at sip.uni-tel.dk>;tag=1c2073920452.
> From:<sip:87776688 at sip.uni-tel.dk>;tag=AI8DA85D59B9B6634F.
> Call-ID:AI231CA9BD0A4A1C53 at 10.0.0.150.
> CSeq: 2 ACK.
> Max-Forwards: 69.
> Route:<sip:178.21.248.7;lr=on;ftag=AI8DA85D59B9B6634F;did=f48.5fd1db02>.
> User-Agent: Aastra Intelligate.
> Content-Length: 0.
>
> I thought that it may be the email body formatting so that the second Via
> header body gets on next line after SIP/2.0/UDP. Can you check your trace,
> is it on next line (i.e., there is a new line)? If the Via is on two lines
> like it is presented, then it is invalid. A header body can continue on a
> new line, but it as to start with whitespace.
>
> Regarding sanity, the module drops silently broken messages if you don't set
> autodrop to 0:
> http://kamailio.org/docs/modules/stable/modules/sanity.html#autodrop
>
> Note that you need latest version of branch 3.1/master for it.
>
> Cheers,
> Daniel
>
> On 11/11/10 1:50 PM, Morten Isaksen wrote:
>>
>> Hi,
>>
>> I narrowed it down to the sanity_check.
>>
>> if(!sanity_check("1511", "7"))
>> {
>> xlog("L_WARN", "sanity check - M=$rm RURI=$ru F=$fu
>> T=$tu IP=$si ID=$ci\n");
>> exit;
>> }
>>
>> The sanity_check fails but does not send a reply back or log the above
>> line. I have commented it out and now the ACK is forwarded.
>>
>> /Morten
>>
>> On Mon, Nov 8, 2010 at 3:00 PM, Morten Isaksen<misak at misak.dk> wrote:
>>>
>>> Hi,
>>>
>>> On Fri, Oct 29, 2010 at 10:59 AM, Daniel-Constantin Mierla
>>> <miconda at gmail.com> wrote:
>>>>
>>>> Hello,
>>>>
>>>> On 10/28/10 1:37 PM, Morten Isaksen wrote:
>>>>>
>>>>> Hi,
>>>>>
>>>>> I have a strange problem with Kamailio 3.0.2. When one of our end
>>>>> users makes a call Kamailio does not route the ACK (in response to the
>>>>> 200 OK). For all other end users it works fine.
>>>>>
>>>>> For me it looks the the has_totag() checks for some reason fails and
>>>>> then t_check_trans() thinks it is a ACK to a local transactions and
>>>>> then terminates the script. Otherwise there should be more lines in
>>>>> the log file.
>>>>
>>>> if you add an xlog() after the if with has_totag(), do you get the
>>>> message
>>>> in the logs?
>>>
>>> Sorry for the delay, but a had to wait for the customer to make a test
>>> call.
>>>
>>> I placed a xlog("L_WARN", "has_totag after - M=$rm RURI=$ru F=$fu
>>> T=$tu IP=$si ID=$ci\n"); just after the if (has_totag()) { .. } and it
>>> does not show in the log.
>>>
>>> It looks very strange to me. Do you have any ideas what is wrong.
>>>
>>> /Morten
>>>
>>>> Cheers,
>>>> Daniel
>>>>
>>>>> The conf is pretty standard.
>>>>>
>>>>> route{
>>>>>
>>>>> xlog("L_WARN", "New request - M=$rm RURI=$ru F=$fu T=$tu
>>>>> IP=$si ID=$ci\n");
>>>>> xlog("L_WARN", "ua=$ua");
>>>>> if (!mf_process_maxfwd_header("10")) {
>>>>> sl_send_reply("483","Too Many Hops");
>>>>> exit;
>>>>> }
>>>>>
>>>>> if(!sanity_check("1511", "7"))
>>>>> {
>>>>> xlog("Malformed SIP message from $si:$sp\n");
>>>>> exit;
>>>>> }
>>>>>
>>>>>
>>>>> if (has_totag()) {
>>>>> xlog("L_WARN", "has_totag start - M=$rm RURI=$ru F=$fu
>>>>> T=$tu IP=$si ID=$ci\n");
>>>>> # sequential request withing a dialog should
>>>>> # take the path determined by record-routing
>>>>> if (loose_route()) {
>>>>> xlog("L_WARN", "loose_route - M=$rm RURI=$ru
>>>>> F=$fu T=$tu IP=$si ID=$ci\n");
>>>>> route(RELAY);
>>>>> } else {
>>>>> if (is_method("SUBSCRIBE")&& uri == myself)
>>>>> {
>>>>> # in-dialog subscribe requests
>>>>> #route(PRESENCE);
>>>>> exit;
>>>>> }
>>>>> if ( is_method("ACK") ) {
>>>>> if ( t_check_trans() ) {
>>>>> xlog("L_WARN", "ACK
>>>>> t_check_trans - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");
>>>>> # non loose-route, but
>>>>> stateful ACK; must be an ACK after a 487 or e.g. 404 from upstream
>>>>> server
>>>>> t_relay();
>>>>> exit;
>>>>> } else {
>>>>> xlog("Ignoring ACK - M=$rm
>>>>> RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");
>>>>> # ACK without matching
>>>>> transaction ... ignore and discard.\n");
>>>>> exit;
>>>>> }
>>>>> }
>>>>> sl_send_reply("404","Not here");
>>>>> }
>>>>> xlog("L_WARN", "has_totag end - M=$rm RURI=$ru F=$fu
>>>>> T=$tu IP=$si ID=$ci\n");
>>>>> exit;
>>>>> }
>>>>>
>>>>> #initial requests
>>>>>
>>>>> # CANCEL processing
>>>>> if (is_method("CANCEL"))
>>>>> {
>>>>> if (t_check_trans())
>>>>> t_relay();
>>>>> exit;
>>>>> }
>>>>>
>>>>> setflag(4);
>>>>> t_check_trans();
>>>>>
>>>>> ...
>>>>>
>>>>> The log files show:
>>>>>
>>>>> Oct 28 10:51:13 sip-core-1 /usr/local/sbin/kamailio[10503]: WARNING:
>>>>> <script>: New request - M=ACK RURI=sip:1105 at 178.21.248.56:5060
>>>>> F=sip:87776688 at sip.uni-tel.dk T=sip:869
>>>>> 87106 at sip.uni-tel.dk IP=178.21.248.20 ID=AI231CA9BD0A4A1C53 at 10.0.0.150
>>>>> Oct 28 10:51:13 sip-core-1 /usr/local/sbin/kamailio[10503]: WARNING:
>>>>> <script>: ua=Aastra Intelligate
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> The trace
>>>>>
>>>>> U 2010/10/28 10:51:02.616337 178.21.248.7:5060 -> 178.21.248.56:5060
>>>>> INVITE sip:86987106 at 178.21.248.56 SIP/2.0.
>>>>>
>>>>>
>>>>> Record-Route:<sip:178.21.248.7;lr;ftag=AI8DA85D59B9B6634F;did=f48.5fd1db02>.
>>>>> Record-Route:<sip:178.21.248.20;lr;ftag=AI8DA85D59B9B6634F>.
>>>>> Via: SIP/2.0/UDP 178.21.248.7;branch=z9hG4bK690c.9cfea506.0.
>>>>> Via: SIP/2.0/UDP 178.21.248.20;branch=z9hG4bK690c.97354e4.0.
>>>>> Via: SIP/2.0/UDP
>>>>>
>>>>>
>>>>> 87.104.233.108:5060;rport=5060;branch=z9hG4bK3c70fae55c8b362ff34e6c782cc21592.
>>>>> To:<sip:86987106 at sip.uni-tel.dk>.
>>>>> From:<sip:87776688 at sip.uni-tel.dk>;tag=AI8DA85D59B9B6634F.
>>>>> Call-ID: AI231CA9BD0A4A1C53 at 10.0.0.150.
>>>>> CSeq: 2 INVITE.
>>>>> Max-Forwards: 68.
>>>>> Contact:<sip:87776688 at 87.104.233.108:5060;line=AI7EFC34995E724DD7>.
>>>>> Accept: application/sdp.
>>>>> Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER.
>>>>> P-Preferred-Identity:<sip:87776688 at sip.uni-tel.dk>.
>>>>> Privacy: none.
>>>>> User-Agent: Aastra Intelligate.
>>>>> Content-Type: application/sdp.
>>>>> Content-Length: 280.
>>>>> X-trunktype: IC.
>>>>> .
>>>>> v=0.
>>>>> o=intelligate 1194032777 1194032777 IN IP4 87.104.233.106.
>>>>> s=call.
>>>>> c=IN IP4 178.21.248.22.
>>>>> t=0 0.
>>>>> m=audio 60984 RTP/AVP 8 0 18 101.
>>>>> a=rtpmap:8 PCMA/8000.
>>>>> a=rtpmap:0 PCMU/8000.
>>>>> a=rtpmap:18 G729/8000.
>>>>> a=rtpmap:101 telephone-event/8000.
>>>>> a=fmtp:101 0-15.
>>>>> a=sendrecv.
>>>>> a=ptime:20.
>>>>>
>>>>> #
>>>>> U 2010/10/28 10:51:02.636854 178.21.248.56:5060 -> 178.21.248.7:5060
>>>>> SIP/2.0 100 Trying.
>>>>> Via: SIP/2.0/UDP 178.21.248.7;branch=z9hG4bK690c.9cfea506.0.
>>>>> Via: SIP/2.0/UDP 178.21.248.20;branch=z9hG4bK690c.97354e4.0.
>>>>> Via: SIP/2.0/UDP
>>>>>
>>>>>
>>>>> 87.104.233.108:5060;rport=5060;branch=z9hG4bK3c70fae55c8b362ff34e6c782cc21592.
>>>>> From:<sip:87776688 at sip.uni-tel.dk>;tag=AI8DA85D59B9B6634F.
>>>>> To:<sip:86987106 at sip.uni-tel.dk>;tag=1c2073920452.
>>>>> Call-ID: AI231CA9BD0A4A1C53 at 10.0.0.150.
>>>>> CSeq: 2 INVITE.
>>>>> Supported: em,timer,replaces,path,early-session,resource-priority.
>>>>> Allow:
>>>>>
>>>>> REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE.
>>>>> Server: Audiocodes-Sip-Gateway-Mediant 2000/v.5.60A.035.002.
>>>>> Content-Length: 0.
>>>>> .
>>>>>
>>>>> #####
>>>>> U 2010/10/28 10:51:04.134445 178.21.248.56:5060 -> 178.21.248.7:5060
>>>>> SIP/2.0 183 Session Progress.
>>>>> Via: SIP/2.0/UDP 178.21.248.7;branch=z9hG4bK690c.9cfea506.0.
>>>>> Via: SIP/2.0/UDP 178.21.248.20;branch=z9hG4bK690c.97354e4.0.
>>>>> Via: SIP/2.0/UDP
>>>>>
>>>>>
>>>>> 87.104.233.108:5060;rport=5060;branch=z9hG4bK3c70fae55c8b362ff34e6c782cc21592.
>>>>> From:<sip:87776688 at sip.uni-tel.dk>;tag=AI8DA85D59B9B6634F.
>>>>> To:<sip:86987106 at sip.uni-tel.dk>;tag=1c2073920452.
>>>>> Call-ID: AI231CA9BD0A4A1C53 at 10.0.0.150.
>>>>> CSeq: 2 INVITE.
>>>>> Contact:<sip:1105 at 178.21.248.56:5060>.
>>>>>
>>>>>
>>>>> Record-Route:<sip:178.21.248.7;lr;ftag=AI8DA85D59B9B6634F;did=f48.5fd1db02>,<sip:178.21.248.20;lr;ftag=AI8DA85D59B9B6634F>.
>>>>> Supported: em,timer,replaces,path,early-session,resource-priority.
>>>>> Allow:
>>>>>
>>>>> REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE.
>>>>> Server: Audiocodes-Sip-Gateway-Mediant 2000/v.5.60A.035.002.
>>>>> Content-Type: application/sdp.
>>>>> Content-Length: 259.
>>>>> .
>>>>> v=0.
>>>>> o=AudiocodesGW 2073965634 2073965290 IN IP4 178.21.248.56.
>>>>> s=Phone-Call.
>>>>> c=IN IP4 178.21.248.56.
>>>>> t=0 0.
>>>>> m=audio 6050 RTP/AVP 8 101.
>>>>> c=IN IP4 178.21.248.56.
>>>>> a=rtpmap:8 PCMA/8000.
>>>>> a=rtpmap:101 telephone-event/8000.
>>>>> a=fmtp:101 0-15.
>>>>> a=ptime:20.
>>>>> a=sendrecv.
>>>>>
>>>>> #
>>>>> U 2010/10/28 10:51:04.134748 178.21.248.7:5060 -> 178.21.248.20:5060
>>>>> SIP/2.0 183 Session Progress.
>>>>> Via: SIP/2.0/UDP 178.21.248.20;branch=z9hG4bK690c.97354e4.0.
>>>>> Via: SIP/2.0/UDP
>>>>>
>>>>>
>>>>> 87.104.233.108:5060;rport=5060;branch=z9hG4bK3c70fae55c8b362ff34e6c782cc21592.
>>>>> From:<sip:87776688 at sip.uni-tel.dk>;tag=AI8DA85D59B9B6634F.
>>>>> To:<sip:86987106 at sip.uni-tel.dk>;tag=1c2073920452.
>>>>> Call-ID: AI231CA9BD0A4A1C53 at 10.0.0.150.
>>>>> CSeq: 2 INVITE.
>>>>> Contact:<sip:1105 at 178.21.248.56:5060>.
>>>>>
>>>>>
>>>>> Record-Route:<sip:178.21.248.7;lr;ftag=AI8DA85D59B9B6634F;did=f48.5fd1db02>,<sip:178.21.248.20;lr;ftag=AI8DA85D59B9B6634F>.
>>>>> Supported: em,timer,replaces,path,early-session,resource-priority.
>>>>> Allow:
>>>>>
>>>>> REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE.
>>>>> Server: Audiocodes-Sip-Gateway-Mediant 2000/v.5.60A.035.002.
>>>>> Content-Type: application/sdp.
>>>>> Content-Length: 259.
>>>>> .
>>>>> v=0.
>>>>> o=AudiocodesGW 2073965634 2073965290 IN IP4 178.21.248.56.
>>>>> s=Phone-Call.
>>>>> c=IN IP4 178.21.248.56.
>>>>> t=0 0.
>>>>> m=audio 6050 RTP/AVP 8 101.
>>>>> c=IN IP4 178.21.248.56.
>>>>> a=rtpmap:8 PCMA/8000.
>>>>> a=rtpmap:101 telephone-event/8000.
>>>>> a=fmtp:101 0-15.
>>>>> a=ptime:20.
>>>>> a=sendrecv.
>>>>>
>>>>> #
>>>>> U 2010/10/28 10:51:04.136586 178.21.248.56:5060 -> 178.21.248.7:5060
>>>>> SIP/2.0 180 Ringing.
>>>>> Via: SIP/2.0/UDP 178.21.248.7;branch=z9hG4bK690c.9cfea506.0.
>>>>> Via: SIP/2.0/UDP 178.21.248.20;branch=z9hG4bK690c.97354e4.0.
>>>>> Via: SIP/2.0/UDP
>>>>>
>>>>>
>>>>> 87.104.233.108:5060;rport=5060;branch=z9hG4bK3c70fae55c8b362ff34e6c782cc21592.
>>>>> From:<sip:87776688 at sip.uni-tel.dk>;tag=AI8DA85D59B9B6634F.
>>>>> To:<sip:86987106 at sip.uni-tel.dk>;tag=1c2073920452.
>>>>> Call-ID: AI231CA9BD0A4A1C53 at 10.0.0.150.
>>>>> CSeq: 2 INVITE.
>>>>> Contact:<sip:1105 at 178.21.248.56:5060>.
>>>>>
>>>>>
>>>>> Record-Route:<sip:178.21.248.7;lr;ftag=AI8DA85D59B9B6634F;did=f48.5fd1db02>,<sip:178.21.248.20;lr;ftag=AI8DA85D59B9B6634F>.
>>>>> Supported: em,timer,replaces,path,early-session,resource-priority.
>>>>> Allow:
>>>>>
>>>>> REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE.
>>>>> Server: Audiocodes-Sip-Gateway-Mediant 2000/v.5.60A.035.002.
>>>>> Content-Type: application/sdp.
>>>>> Content-Length: 259.
>>>>> .
>>>>> v=0.
>>>>> o=AudiocodesGW 2073965634 2073965290 IN IP4 178.21.248.56.
>>>>> s=Phone-Call.
>>>>> c=IN IP4 178.21.248.56.
>>>>> t=0 0.
>>>>> m=audio 6050 RTP/AVP 8 101.
>>>>> c=IN IP4 178.21.248.56.
>>>>> a=rtpmap:8 PCMA/8000.
>>>>> a=rtpmap:101 telephone-event/8000.
>>>>> a=fmtp:101 0-15.
>>>>> a=ptime:20.
>>>>> a=sendrecv.
>>>>>
>>>>> #
>>>>> U 2010/10/28 10:51:04.136837 178.21.248.7:5060 -> 178.21.248.20:5060
>>>>> SIP/2.0 180 Ringing.
>>>>> Via: SIP/2.0/UDP 178.21.248.20;branch=z9hG4bK690c.97354e4.0.
>>>>> Via: SIP/2.0/UDP
>>>>>
>>>>>
>>>>> 87.104.233.108:5060;rport=5060;branch=z9hG4bK3c70fae55c8b362ff34e6c782cc21592.
>>>>> From:<sip:87776688 at sip.uni-tel.dk>;tag=AI8DA85D59B9B6634F.
>>>>> To:<sip:86987106 at sip.uni-tel.dk>;tag=1c2073920452.
>>>>> Call-ID: AI231CA9BD0A4A1C53 at 10.0.0.150.
>>>>> CSeq: 2 INVITE.
>>>>> Contact:<sip:1105 at 178.21.248.56:5060>.
>>>>>
>>>>>
>>>>> Record-Route:<sip:178.21.248.7;lr;ftag=AI8DA85D59B9B6634F;did=f48.5fd1db02>,<sip:178.21.248.20;lr;ftag=AI8DA85D59B9B6634F>.
>>>>> Supported: em,timer,replaces,path,early-session,resource-priority.
>>>>> Allow:
>>>>>
>>>>> REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE.
>>>>> Server: Audiocodes-Sip-Gateway-Mediant 2000/v.5.60A.035.002.
>>>>> Content-Type: application/sdp.
>>>>> Content-Length: 259.
>>>>> .
>>>>> v=0.
>>>>> o=AudiocodesGW 2073965634 2073965290 IN IP4 178.21.248.56.
>>>>> s=Phone-Call.
>>>>> c=IN IP4 178.21.248.56.
>>>>> t=0 0.
>>>>> m=audio 6050 RTP/AVP 8 101.
>>>>> c=IN IP4 178.21.248.56.
>>>>> a=rtpmap:8 PCMA/8000.
>>>>> a=rtpmap:101 telephone-event/8000.
>>>>> a=fmtp:101 0-15.
>>>>> a=ptime:20.
>>>>> a=sendrecv.
>>>>>
>>>>> ##############################
>>>>> U 2010/10/28 10:51:12.881179 178.21.248.56:5060 -> 178.21.248.7:5060
>>>>> SIP/2.0 200 OK.
>>>>> Via: SIP/2.0/UDP 178.21.248.7;branch=z9hG4bK690c.9cfea506.0.
>>>>> Via: SIP/2.0/UDP 178.21.248.20;branch=z9hG4bK690c.97354e4.0.
>>>>> Via: SIP/2.0/UDP
>>>>>
>>>>>
>>>>> 87.104.233.108:5060;rport=5060;branch=z9hG4bK3c70fae55c8b362ff34e6c782cc21592.
>>>>> From:<sip:87776688 at sip.uni-tel.dk>;tag=AI8DA85D59B9B6634F.
>>>>> To:<sip:86987106 at sip.uni-tel.dk>;tag=1c2073920452.
>>>>> Call-ID: AI231CA9BD0A4A1C53 at 10.0.0.150.
>>>>> CSeq: 2 INVITE.
>>>>> Contact:<sip:1105 at 178.21.248.56:5060>.
>>>>>
>>>>>
>>>>> Record-Route:<sip:178.21.248.7;lr;ftag=AI8DA85D59B9B6634F;did=f48.5fd1db02>,<sip:178.21.248.20;lr;ftag=AI8DA85D59B9B6634F>.
>>>>> Supported: em,timer,replaces,path,early-session,resource-priority.
>>>>> Allow:
>>>>>
>>>>> REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE.
>>>>> Server: Audiocodes-Sip-Gateway-Mediant 2000/v.5.60A.035.002.
>>>>> Content-Type: application/sdp.
>>>>> Content-Length: 259.
>>>>> .
>>>>> v=0.
>>>>> o=AudiocodesGW 2073965634 2073965290 IN IP4 178.21.248.56.
>>>>> s=Phone-Call.
>>>>> c=IN IP4 178.21.248.56.
>>>>> t=0 0.
>>>>> m=audio 6050 RTP/AVP 8 101.
>>>>> c=IN IP4 178.21.248.56.
>>>>> a=rtpmap:8 PCMA/8000.
>>>>> a=rtpmap:101 telephone-event/8000.
>>>>> a=fmtp:101 0-15.
>>>>> a=ptime:20.
>>>>> a=sendrecv.
>>>>>
>>>>> #
>>>>> U 2010/10/28 10:51:12.882388 178.21.248.7:5060 -> 178.21.248.20:5060
>>>>> SIP/2.0 200 OK.
>>>>> Via: SIP/2.0/UDP 178.21.248.20;branch=z9hG4bK690c.97354e4.0.
>>>>> Via: SIP/2.0/UDP
>>>>>
>>>>>
>>>>> 87.104.233.108:5060;rport=5060;branch=z9hG4bK3c70fae55c8b362ff34e6c782cc21592.
>>>>> From:<sip:87776688 at sip.uni-tel.dk>;tag=AI8DA85D59B9B6634F.
>>>>> To:<sip:86987106 at sip.uni-tel.dk>;tag=1c2073920452.
>>>>> Call-ID: AI231CA9BD0A4A1C53 at 10.0.0.150.
>>>>> CSeq: 2 INVITE.
>>>>> Contact:<sip:1105 at 178.21.248.56:5060>.
>>>>>
>>>>>
>>>>> Record-Route:<sip:178.21.248.7;lr;ftag=AI8DA85D59B9B6634F;did=f48.5fd1db02>,<sip:178.21.248.20;lr;ftag=AI8DA85D59B9B6634F>.
>>>>> Supported: em,timer,replaces,path,early-session,resource-priority.
>>>>> Allow:
>>>>>
>>>>> REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE.
>>>>> Server: Audiocodes-Sip-Gateway-Mediant 2000/v.5.60A.035.002.
>>>>> Content-Type: application/sdp.
>>>>> Content-Length: 259.
>>>>> .
>>>>> v=0.
>>>>> o=AudiocodesGW 2073965634 2073965290 IN IP4 178.21.248.56.
>>>>> s=Phone-Call.
>>>>> c=IN IP4 178.21.248.56.
>>>>> t=0 0.
>>>>> m=audio 6050 RTP/AVP 8 101.
>>>>> c=IN IP4 178.21.248.56.
>>>>> a=rtpmap:8 PCMA/8000.
>>>>> a=rtpmap:101 telephone-event/8000.
>>>>> a=fmtp:101 0-15.
>>>>> a=ptime:20.
>>>>> a=sendrecv.
>>>>>
>>>>> #####<<<<<<<<<<<<<<<<<<<<<< This is the problem
>>>>> packet>>>>>>>>>>>>>>>>>>>>>
>>>>> U 2010/10/28 10:51:13.267863 178.21.248.20:5060 -> 178.21.248.7:5060
>>>>> ACK sip:1105 at 178.21.248.56:5060 SIP/2.0.
>>>>> Record-Route:<sip:178.21.248.20;lr;ftag=AI8DA85D59B9B6634F>.
>>>>> Via: SIP/2.0/UDP 178.21.248.20;branch=z9hG4bK690c.97354e4.2.
>>>>> Via: SIP/2.0/UDP
>>>>>
>>>>>
>>>>> 87.104.233.108:5060;rport=5060;branch=z9hG4bK07103fe69c8af048b9b8216eb2f7233f.
>>>>> To:<sip:86987106 at sip.uni-tel.dk>;tag=1c2073920452.
>>>>> From:<sip:87776688 at sip.uni-tel.dk>;tag=AI8DA85D59B9B6634F.
>>>>> Call-ID: AI231CA9BD0A4A1C53 at 10.0.0.150.
>>>>> CSeq: 2 ACK.
>>>>> Max-Forwards: 69.
>>>>>
>>>>> Route:<sip:178.21.248.7;lr=on;ftag=AI8DA85D59B9B6634F;did=f48.5fd1db02>.
>>>>> User-Agent: Aastra Intelligate.
>>>>> Content-Length: 0.
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>> --
>>>> Daniel-Constantin Mierla
>>>> http://www.asipto.com
>>>>
>>>>
>>>> _______________________________________________
>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>>>> sr-users at lists.sip-router.org
>>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>>
>>>
>>>
>>> --
>>> Morten Isaksen
>>>
>>
>>
>
> --
> Daniel-Constantin Mierla
> http://www.asipto.com
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
--
Morten Isaksen
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