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@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@sip.vendor.es
<mailto:sip%3A50026896@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@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@192.168.0.1:5060
<http://4bf8effb45b0ae8e049366297924cbba@192.168.0.1:5060>
From: <sip:28999999999@192.168.0.2
<mailto:sip%3A28999999999@192.168.0.2>;tag=k0eci3x3-CC-30
To: 34888888888 <sip:50026896@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@no{soy,ns#005#007] old=[sip:34888888888@8.8.8.8
<mailto:sip%3A34888888888@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@192.168.0.1
<mailto:sip%3A28999999999@192.168.0.1>]
old=[sip:999999999@sip.vendor.es <mailto:sip%3A999999999@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