[SR-Users] Problem with Kamailio not routing ACK to a 200 OK
Daniel-Constantin Mierla
miconda at gmail.com
Thu Nov 11 20:20:10 CET 2010
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
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