[OpenSIPS-Users] B2BUA Ripping/Truncating Callid

Ovidiu Sas osas at voipembedded.com
Thu Dec 15 21:24:04 CET 2011 

It was not fixed.  It is the same bug.

Regards,
Ovidiu Sas

On Thu, Dec 15, 2011 at 3:20 PM, Logan <voipmaster at me.com> wrote:
> 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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-- 
VoIP Embedded, Inc.
http://www.voipembedded.com

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