[SR-Users] (no subject)

[Daniel-Constantin Mierla]

Hello,

On 26/06/15 13:05, Nelson Migliaro wrote:
> Hello everybody,
>
> My SIP vendor request me to replace FROM before sending the traffic.
> In order to achieve this I use uac_replace_from.
>
> UAC module is setup in restore_mode = auto.
>
> In my insfrastructure I have an Asterisk and then a Kamailio that
> connects to vendor via internet.
>
> Softphone -> Asterisk -> Kamailio -> Internet -> SIP vendor
>
> If caller ID is setup in Asterisk using CALLERID(num)=34888888888 and
> then INVITE is forwarded to Kamailio, the call is established and
> finished correctly but the URI in TO field in BYE request from
> Kamailio to Asterisk contains garbage. In the scenario the callee
> hangs up the call.
>
> Example of TO Field with garbage: 34888888888 <sip:50026896 at no{soy,ns^^>
>
> What I do see is that the number "50026896" that is part of the URI is
> the same I use in:
> uac_replace_from("50026896", "sip:50026896 at sip.vendor.es
> <mailto:sip%3A50026896 at sip.vendor.es>");
>
> Something else that I have found is that vsf field is the same in the
> INVITE and in the BYE.
>
> ----------------------------------------------------------------------------------------------------------------
> 2015/06/23 17:48:38.552442 192.168.0.2:5060 <http://192.168.0.2:5060>
> -> 192.168.0.1:5060 <http://192.168.0.1:5060>
> BYE sip:34888888888 at 192.168.0.1:5060
> <http://sip:34888888888@192.168.0.1:5060> SIP/2.0
> Via: SIP/2.0/UDP
> 192.168.0.2;branch=z9hG4bKfb49.73c6517609cdb6f7ec00b1f40a05dbe9.0
> Via: SIP/2.0/UDP
> 8.8.8.8.8;rport=5060;branch=z9hG4bKfcdf.3767c59c2d0e3d8ab695669845ce4cea.0
> branch=z9hG4bK04boo6104o5hcso0c2a1sd0000g00.1
> Call-ID: 4bf8effb45b0ae8e049366297924cbba at 192.168.0.1:5060
> <http://4bf8effb45b0ae8e049366297924cbba@192.168.0.1:5060>
> From: <sip:28999999999 at 192.168.0.2
> <mailto:sip%3A28999999999 at 192.168.0.2>;tag=k0eci3x3-CC-30
> To: 34888888888 <sip:50026896 at no{soy,ns^^>;tag=as041b7d84
> CSeq: 1 BYE
> Reason: Q.850;cause=16;text="normal call clearing"
> Max-Forwards: 67
> Content-Length: 0
>
> ------------------------------------------------------------------------------------------------------------------
> DEBUG: uac [replace.c:525]: restore_uri(): getting 'vsf' Route param
> DEBUG: uac [replace.c:533]: restore_uri(): route param is
> 'AAAAAAEECQkCAgsNAXBeL0NGQUsfVl02Ni44Mw--' (len=40)
> DEBUG: uac [replace.c:607]: restore_uri(): decoded uris are:
> new=[sip:50026896 at no{soy,ns#005#007] old=[sip:34888888888 at 8.8.8.8
> <mailto:sip%3A34888888888 at 8.8.8.8>]
> DEBUG: uac [replace.c:525]: restore_uri(): getting 'vst' Route param
> DEBUG: uac [replace.c:533]: restore_uri(): route param is
> 'AAAAAAQPAw8MDgsAAHZBKRVdAhoVHQ4XH1BdYWJhbnRlLmVz' (len=48)
> DEBUG: uac [replace.c:607]: restore_uri(): decoded uris are:
> new=[sip:28999999999 at 192.168.0.1
> <mailto:sip%3A28999999999 at 192.168.0.1>]
> old=[sip:999999999 at sip.vendor.es <mailto:sip%3A999999999 at sip.vendor.es>]
> -----------------------------------------------------------------------------------------------------------------------------------
>
>

Can you check if the From/To display name and URI are changed by end
devices comparing with what Kamailio is sending? If yes, then you should
use uac module with the option of storing the original URIs via dialog
module.

Cheers,
Daniel

-- 
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Book: SIP Routing With Kamailio - http://www.asipto.com

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