[SR-Users] Problem with Kamailio not routing ACK to a 200 OK

Morten Isaksen misak at misak.dk
Mon Nov 15 21:28:36 CET 2010


Hi Daniel,

Thank you very much for the help.

I will report the bug to Aastra.

/Morten

On Mon, Nov 15, 2010 at 4:22 PM, Daniel-Constantin Mierla
<miconda at gmail.com> wrote:
> Hello,
>
> I got the pcap file and had the time too check it. There seems to be an
> extra LF at the end of ACK:
>
> 0230  73 65 72 2d 41 67 65 6e  74 3a 20 41 61 73 74 72   ser-Agen t: Aastr
> 0240  61 20 49 6e 74 65 6c 6c  69 67 61 74 65 0d 0a 43   a Intell igate..C
> 0250  6f 6e 74 65 6e 74 2d 4c  65 6e 67 74 68 3a 20 30   ontent-L ength: 0
> 0260  0d 0a 0d 0a 0a                                     .....
>
> Once the last header is finished and ended with CRLF, there must be another
> CRLF and that's it if content length is 0.
>
> According to wireshark and the capture you sent, there is an extra 0x0a
> (LF), so instead of ending in CRLFCRLF, the ACK ends in CRLFCRLFLF
>
> You can remove the content-length check in sanity function, but I recomend
> you report to vendor to get the issue fixed there.
>
> Cheers,
> Daniel
>
>
> On 11/11/10 11:28 PM, Daniel-Constantin Mierla wrote:
>>
>> Hello,
>>
>> On 11/11/10 11:02 PM, Morten Isaksen wrote:
>>>
>>> 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.
>>
>> it happens sometime that some module parameters control the behavior of
>> exported functions and it is not mentioned in description. This one was
>> discovered pretty recently and the description of sanity_check() don't refer
>> to autodrop parameter. I will try to update asap.
>>
>>> I have solved the problem by removing the sanity_check.
>>>
>>> I am just a bit curious why it failed.
>>
>> That should be found to see where is the failure. As a second guess based
>> on checks, it may be that the ACK has some whitespace in the body. Do you
>> have pcap version of this ACK trace?
>>
>> Daniel
>>
>>> 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
>>>>
>>>
>>>
>>
>
> --
> Daniel-Constantin Mierla
> http://www.asipto.com
>
>



-- 
Morten Isaksen



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