[SR-Users] Problem with ACK
Stoyan Mihaylov
stoyan.v.mihaylov at gmail.com
Thu Jun 6 17:06:33 CEST 2013
I dont know what caused problem. I just found working solution. I used
fireshark to get messages, and I saw that some ACK and BYE messages
"reenter" kamailio and keep growing - as I remember (not sure). As I
remember (not sure) - VIA started to grow for next messages. For me it
looked like message for some reason reenter kamailio, adding new VIA record.
My whole solution is:
If I receive ACK or BYE message, I process them next way:
route[ACKBYE] {
#!ifdef WITH_MYFORWARD
if(($sht(forw=>$ft))=~$td){
$du=$sht(forw=>$ft);
}else if((($td=="sip.OurCompany.com
")||($td=="xxx.xxx.xxx.xxx"))&&($si=="xxx.xxx.xxx.xxx")){
$du=$sht(forw=>$ft);
return;
}
#!endif
return;
}
Of course: sip.OurCompany.com=xxx.xxx.xxx.xxx
Here I initialize $sht(forw=>$ft)
route[PSTNINVITE] {
#!ifdef WITH_MYFORWARD
if(is_method("INVITE")){
ds_select_dst("1","4");
$sht(forw=>$ft)=$du;
sl_send_reply("100","Trying");
route(RELAY);
exit();
}
#!endif
return;
}
On Thu, Jun 6, 2013 at 5:51 PM, Daniel-Constantin Mierla
<miconda at gmail.com>wrote:
>
> On 6/6/13 4:34 PM, Stoyan Mihaylov wrote:
>
> We had some similar problems.
>
>
> But what was the actual problem? At least in the two ACKs provided below,
> loose routing handling with looks correct.
>
> Is something that Asterisk doesn't like?
>
> Cheers,
> Daniel
>
> Our configuration is:
> SIP client 1 <-> Kamailio <-> Asterisk <->Kamailio<->SIP client 2
> My solution was to check $td and $si and if they are same as Kamailio, to
> forward call to Asterisk.
> Because I planed to use more then 1 Asterisk, I keep in variable which one
> to use.
>
>
>
> On Thu, Jun 6, 2013 at 5:28 PM, Daniel-Constantin Mierla <
> miconda at gmail.com> wrote:
>
>> Hello,
>>
>> the incoming ACK has the top Route with lr parameter, meaning is loose
>> routing. By that, the proxy removes the top route header, preserves the
>> R-URI and sends to the URI in the next Route header.
>>
>> From what I can see in the Route stack, it seems a spiral back to the
>> proxy because ip 81.21.38.34 is two times there.
>>
>> If you can't sort it out, send the full SIP trace taken on the proxy from
>> the initial INVITE to the ACK. Then we can see how Record-Route headers are
>> set and the signaling flow.
>>
>> Cheers,
>> Daniel
>>
>> On 6/6/13 3:30 PM, phillman25 wrote:
>>
>> Dear list further to the above problem i observed the following:
>>
>> ACK message coming from PABX1:
>>
>> U +0.001877 192.168.10.189:5060 -> 81.21.38.34:5060
>> ACK sip:94294294 at 81.21.38.55 SIP/2.0*
>> Via: SIP/2.0/UDP 192.168.10.189:5060;branch=z9hG4bK76f8f103;rport*
>> Route: <sip:81.21.38.34;lr=on;ftag=as181922af;did=c83.f641>,<
>> sip:94294294 at 81.21.38.5;pgw-call=call-2aa6>,
>> <sip:81.21.38.34;lr=on;ftag=as181922af>*
>> Max-Forwards: 70*
>> From: "22498045" <sip:22498045 at 192.168.10.189>;tag=as181922af*
>> To: <sip:94294294 at 81.21.38.34
>> >;tag=3d95248-37261551-13c4-50022-1c3096-5cc2673d-1c3096*
>> Contact: <sip:22498045 at 192.168.10.189:5060>*
>> Call-ID: 696cfa577f42395767bc812e7e8a38a5 at 192.168.10.189:5060*
>> CSeq: 102 ACK*
>> User-Agent: FPBX-2.8.1(1.8.21.0)*
>> Content-Length: 0*
>>
>>
>>
>> ACK message sent to PGW from Kamailio1
>>
>> U +0.001254 81.21.38.34:5060 -> 81.21.38.5:5060
>> ACK sip:94294294 at 81.21.38.55 SIP/2.0*
>> Via: SIP/2.0/UDP
>> 81.21.38.34;branch=z9hG4bKc526.9402c2edbc3ef96d9e405408364506a9.0*
>> Via: SIP/2.0/UDP 192.168.10.189:5060;branch=z9hG4bK76f8f103;rport=5060*
>> Route: <sip:94294294 at 81.21.38.5;pgw-call=call-2aa6>,
>> <sip:81.21.38.34;lr=on;ftag=as181922af>*
>> Max-Forwards: 16*
>> From: "22498045" <sip:22498045 at 192.168.10.189>;tag=as181922af*
>> To: <sip:94294294 at 81.21.38.34
>> >;tag=3d95248-37261551-13c4-50022-1c3096-5cc2673d-1c3096*
>> Contact: <sip:22498045 at 192.168.10.189:5060>*
>> Call-ID: 696cfa577f42395767bc812e7e8a38a5 at 192.168.10.189:5060*
>> CSeq: 102 ACK*
>> User-Agent: FPBX-2.8.1(1.8.21.0)*
>> Content-Length: 0*
>>
>>
>>
>>
>> Shouldn't the ACK message to the PGW have the header ACK
>> sip:94294294 at 81.21.38.5;pgw-call=call-2aa6 and the Route:
>> <sip:81.21.38.34;lr=on;ftag=as181922af>* ???
>>
>>
>>
>>
>> Your help is much appreciated!!
>>
>> Phillip
>>
>>
>>
>> On Thu, Jun 6, 2013 at 12:26 PM, phillman25 <phillman25 at gmail.com> wrote:
>>
>>> Dear List
>>>
>>> I upgraded from Kamailio v 3.3 to 4.0.1 and am now facing an issue for
>>> the below scenario:
>>>
>>> PABX1 ==> Kamailio1 ==> Cisco PGW ==> Kamailio1 ==> PABX2
>>>
>>>
>>> I understand that this is a hairpin scenario but was working normally
>>> on v 3.3.
>>>
>>> Checking in the syslog i see:
>>> ERROR: <core> [receive.c:230]: ERROR: receive_msg: no via found in reply
>>>
>>> Checking the sip trace i see that when calling from PABX1 to PABX2.
>>> After PABX2 answers and the the 200 OK is eventually sent to PABX1 ,
>>> PABX1 answers with ACK but seems like its not sent back to PABX2 as a
>>> result PABX resends a 200 OK and the cycle continues until PABX2 sends a
>>> BYE message. Please see below the ACK received from PABX1:
>>>
>>> ACK sip:94294294 at 81.21.38.55 SIP/2.0
>>> Via: SIP/2.0/UDP 192.168.10.189:5060;branch=z9hG4bK6bffe37c;rport
>>> Route: <sip:81.21.38.34;lr=on;ftag=as1cd4f8f1;did=e36.c471>,<
>>> sip:94294294 at 81.21.38.5;pgw-call=call-26eb>,
>>> <sip:81.21.38.34;lr=on;ftag=as1cd4f8f1>
>>> Max-Forwards: 70
>>> From: "22498045" <sip:22498045 at 192.168.10.189>;tag=as1cd4f8f1
>>> To: <sip:94294294 at 81.21.38.34
>>> >;tag=3d94f08-37261551-13c4-50022-1c1e67-87fe958-1c1e67
>>> Contact: <sip:22498045 at 192.168.10.189:5060>
>>> Call-ID: 03042a717e27a87e759f7f4879e70377 at 192.168.10.189:5060
>>> CSeq: 102 ACK
>>> User-Agent: FPBX-2.8.1(1.8.21.0)
>>> Content-Length: 0
>>>
>>>
>>> Is there an issue with the above ACK message? Is there any way to
>>> solve this issue quickly perhaps by disabling loose route?
>>> I have observed that this issue occurs only when hairpinned.
>>>
>>>
>>> Thanking you in advance!
>>>
>>> Phillip
>>>
>>>
>>
>>
>> _______________________________________________
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users at lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
>> --
>> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
>> Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013
>> * http://asipto.com/u/katu *
>>
>>
>> _______________________________________________
>> 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
>>
>>
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users at lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
> --
> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
> Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013
> * http://asipto.com/u/katu *
>
>
> _______________________________________________
> 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
>
>
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