[SR-Users] Problem with Kamailio not routing ACK to a 200 OK
Daniel-Constantin Mierla
miconda at gmail.com
Mon Nov 15 16:22:00 CET 2010
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
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