[OpenSIPS-Users] B2BUA Ripping/Truncating Callid

Logan voipmaster at me.com
Thu Dec 15 21:20:38 CET 2011 

Out of curiosity, based on the feedback in this bug; is this something that's being fixed? I notice this bug was for 1.6.4, but my experience is in 1.7.1. so I want to make sure if this was fixed, I report a new bug for 1.7.1


Hi Bogdan,

This bug fix requires further work in tm module, in local_route processing,
so as to update the shortcuts in tm when lumps are applied for headers
also. The fix that was committed last week solved this problem only when
body lumps were applied. 
Unfortunately, I don't have time to work on this, so I have removed the
assignation to me for this bug report.

Regards,
Anca

On Dec 12, 2011, at 09:05 AM, Ovidiu Sas <osas at voipembedded.com> wrote:

Yes, indeed. Thanks for pointing out.

Regards,
Ovidiu Sas

On Mon, Dec 12, 2011 at 1:26 AM, Ryan Bullock <rrb3942 at gmail.com> wrote:
> I think this is related to a bug that is already open:
> http://sourceforge.net/tracker/?func=detail&aid=3316230&group_id=232389&atid=1086410
>
>
> On Fri, Dec 9, 2011 at 5:46 PM, Ovidiu Sas <osas at voipembedded.com> wrote:
>> Yeah, it's the first request after the modified INVITE that is
>> malformed (I reproduced this running a snapshot from trunk).  Please
>> open a bug report.
>>
>> Regards,
>> Ovidiu Sas
>>
>> --
>> VoIP Embedded, Inc.
>> http://www.voipembedded.com
>>
>>
>> On Fri, Dec 9, 2011 at 1:52 PM, Logan <voipmaster at me.com> wrote:
>>> I added the log and everything looks fine. It's only adding the PAI to the
>>> initial invite which is what I want. The odd thing is there are no issues
>>> with the invites, it just looks like the Cancel messages that are being
>>> mangled. I posted a separate issue to the list prior to this report but no
>>> one responded, I'm not sure it went through correctly but resulting cancel
>>> coming out of the B2BUA looked like this:
>>>
>>>  Reference:
>>>
>>> 192.168.1.146 = Opensips Proxy
>>> 192.168.1.145 = Opensips B2BUA
>>> 10.2.3.245 = Carrier
>>>
>>>
>>>
>>> U 2011/12/01 22:51:11.558887 192.168.1.146:5060 -> 192.168.1.145:5090
>>> CANCEL sip:9993512125551212 at 192.168.1.145:5090 SIP/2.0.
>>> Via: SIP/2.0/UDP 192.168.1.146;branch=z9hG4bK2df7.78db1d81.0.
>>> From: "James Logan" <sip:8884442222 at 192.168.1.137>;tag=as06eabdcd.
>>> Call-ID: 40c30c6459b3eaa4683991082381cadb at 192.168.1.137.
>>> To: "12125551212" <sip:12125551212 at 192.168.1.146>.
>>> CSeq: 102 CANCEL.
>>> Max-Forwards: 70.
>>> User-Agent: Opensips.
>>> Content-Length: 0.
>>> .
>>>
>>>
>>> U 2011/12/01 22:51:11.559378 192.168.1.145:5090 -> 192.168.1.146:5060
>>> SIP/2.0 200 canceling.
>>> Via: SIP/2.0/UDP 192.168.1.146;branch=z9hG4bK2df7.78db1d81.0.
>>> From: "James Logan" <sip:8884442222 at 192.168.1.137>;tag=as06eabdcd.
>>> Call-ID: 40c30c6459b3eaa4683991082381cadb at 192.168.1.137.
>>> To: "12125551212"
>>> <sip:12125551212 at 192.168.1.146>;tag=3330ae74b9cf9aed85afbc9203dd6238-715f
>>> CSeq: 102 CANCEL.
>>> Server: B2BUA.
>>> Content-Length: 0.
>>> .
>>>
>>>
>>> U 2011/12/01 22:51:11.559527 192.168.1.145:5090 -> 10.2.3.245:5060
>>> CANCEL ............i...............i.. SIP/2.0.
>>> Via: SIP/2.0/UDP 192.168.1.145:5090;branch=z9hG4bK5421.22999dd2.0.
>>> ........B2B.256.3572553sip:+12125551212 at 10.2.3.245sip:8884442222 at 192.168.1.1379120d3`.....p..i...........................................q.i............
>>> ........ CANCEL.
>>> User-Agent: OpenSIPS (1.7.1-notls (x86_64/linux)).
>>> Max-Forwards: 70.
>>> User-Agent: Opensips.
>>> Init-CallID: 40c30c6459b3eaa4683991082381cadb at 192.168.1.137.
>>> Contact: <sip:192.168.1.145:5090>.
>>> .
>>>
>>> On Dec 07, 2011, at 05:18 PM, Ovidiu Sas <osas at voipembedded.com> wrote:
>>>
>>> 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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