[OpenSIPS-Users] B2BUA Ripping/Truncating Callid

Ovidiu Sas osas at voipembedded.com
Wed Dec 7 23:18:30 CET 2011


Add a log and print out what are you adding before adding it and you
will see if it's good or not.

On Wed, Dec 7, 2011 at 5:13 PM, Logan <voipmaster at me.com> wrote:
> This is the extent of my local route. If the $var is not present, I do not
> add it. Do you see any issue with what I'm doing here?
>
>
> local_route {
>         #xlog("L_INFO","***** IN LOCAL ROUTE ********\n");
>
>         if (is_method("INVITE")) {
>                 if($var(pai_userpart)) {
>                         append_hf("P-Asserted-Identity:
> \"$var(pai_display)\" <sip:$var(pai_userpart)@$Ri>\r\n");
>                 }else{
>                         xlog("L_INFO","PAI is not present, not adding\n");
>                 }
>         }
>
>
> }
>
> On Dec 07, 2011, at 04:57 PM, Ovidiu Sas <osas at voipembedded.com> wrote:
>
> You need to be careful when you alter requests in B2B mode (the
> received INVITE and the sent INVITE belong to different transactions).
> Make sure that you have something valid in those vars before applying
> any changes to the outgoing message.
>
> Regards,
> Ovidiu Sas
>
> On Wed, Dec 7, 2011 at 4:49 PM, Logan <voipmaster at me.com> wrote:
>> I'm storing some $vars in route[0] prior to calling b2b_init_request("top
>> hiding");
>>
>> Then in my local route Im appending a P-Asserted-Identity header.
>>
>> I can't use the custom_headers modparam because it's going to preserve the
>> PAI as it comes in. Most of the time it's not present, or is in the wrong
>> format so I'm adding it in local route.
>>
>>
>> On Dec 07, 2011, at 04:31 PM, Ovidiu Sas <osas at voipembedded.com> wrote:
>>
>> Are you trying to perform any msg manipulations during b2b scenarios?
>> Also, keep in mind that the b2b server functionality must be kept
>> isolated from the proxy server functionality (proxy mode is not
>> compatible with b2b mode).
>>
>> Regards,
>> Ovidiu Sas
>>
>> -- VoIP Embedded, Inc.http://www.voipembedded.com
>> On Wed, Dec 7, 2011 at 3:41 PM, Logan <voipmaster at me.com> wrote:
>>> Hello list this is the second odd thing I've seen with b2bua in opensips
>>> 1.7.1 It looks like the b2bua module is mangling the cancel message and
>>> is
>>> ripping out the callid when sending upstream:
>>>
>>>
>>> U 2011/12/07 20:15:05.895915 192.168.1.143:5060 -> 192.168.1.145:5090
>>>
>>> CANCEL sip:9993518045551212 at 192.168.1.145:5090 SIP/2.0.
>>>
>>> Via: SIP/2.0/UDP 192.168.1.143;branch=z9hG4bKac0e.5a3d2bf1.0.
>>>
>>> From: "8669800222" <sip:8669800222 at 192.168.1.1>;tag=3532277698-944952.
>>>
>>> Call-ID: 494823-3532277698-944947 at 192.168.1.1.
>>>
>>> To: "18045551212" <sip:18045551212 at 192.168.1.143>.
>>>
>>> CSeq: 1 CANCEL.
>>>
>>> Max-Forwards: 70.
>>>
>>> User-Agent: Opensips.
>>>
>>> Content-Length: 0.
>>>
>>> .
>>>
>>>
>>>
>>> U 2011/12/07 20:15:05.896027 192.168.1.145:5090 -> 192.168.1.143:5060
>>>
>>> SIP/2.0 200 canceling.
>>>
>>> Via: SIP/2.0/UDP 192.168.1.143;branch=z9hG4bKac0e.5a3d2bf1.0.
>>>
>>> From: "8669800222" <sip:8669800222 at 192.168.1.1>;tag=3532277698-944952.
>>>
>>> Call-ID: 494823-3532277698-944947 at 192.168.1.1.
>>>
>>> To: "18045551212"
>>>
>>> <sip:18045551212 at 192.168.1.143>;tag=3330ae74b9cf9aed85afbc9203dd6238-e6b7.
>>>
>>> CSeq: 1 CANCEL.
>>>
>>> Server: Opensips.
>>>
>>> Content-Length: 0.
>>>
>>> .
>>>
>>>
>>>
>>> U 2011/12/07 20:15:05.896097 192.168.1.145:5090 -> 10.2.3.210:5060
>>>
>>> CANCEL sip:+18045551212 at 65.211.120.23 SIP/2.0.
>>>
>>> Via: SIP/2.0/UDP 192.168.1.145:5090;branch=z9hG4bK0299.252f8e61.0.
>>>
>>> .
>>>
>>> From: <sip:7324812444 at 66.29.74.37>;tag=418802140f6308e008db76a1e1de765b.
>>>
>>> CSeq: 2 INVITE54.7172739.
>>>
>>> Content-Lengt
>>>
>>> To: sip:+18045551212 at 65.211.120.237.
>>>
>>> Call- CANCEL.
>>>
>>> User-Agent: OpenSIPS (1.7.1-notls (x86_64/linux)).
>>>
>>> Max-Forwards: 70.
>>>
>>> Init-CallID: 494823-3532277698-944947 at 192.168.1.1.
>>>
>>> Contact: <sip:192.168.1.145:5090>.
>>>
>>> .
>>>
>>>
>>>
>>> U 2011/12/07 20:15:05.910842 10.2.3.210:5060 -> 192.168.1.145:5090
>>>
>>> SIP/2.0 400 Missing Mandatory Header Call-Id.
>>>
>>> v: SIP/2.0/UDP
>>> 192.168.1.145:5090;branch=z9hG4bK0299.252f8e61.0;received=192.168.1.145.
>>>
>>> l: 0.
>
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-- 
VoIP Embedded, Inc.
http://www.voipembedded.com



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