Has anyone any idea ?
Thanks in advance,
BR,
Mehdi BOUDOU
2013/8/9 Mehdi BOUDOU
> Hello,
>
>
> I'am already testing the perl module and opensips 1.9 in proxy mode.
> I use it from failure route and the perl script retrieve the initial
> INVITE.
> What is the bes
Hello,
I'am already testing the perl module and opensips 1.9 in proxy mode.
I use it from failure route and the perl script retrieve the initial INVITE.
What is the best way to also retrieve the Failure response from perl
script? avp value ?
Thanks,
Mehdi B
ooks like the 487 is lost - the terminating party is
> sending it, but the originating does not get it.
>
> Do you have opensips logs in debug 4 ?
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
>
> On 07/
Hello,
Has anyone any idea on what to do ?
Thanks in advance
2013/7/12 Mehdi BOUDOU
> Hello,
>
> I'm testing the B2BModule with this sipp Scenario (A simple CANCEL Call).
>
> This is the results of the execution script throught Opensips with a proxy
> configuration fi
Hello,
I'm testing the B2BModule with this sipp Scenario (A simple CANCEL Call).
This is the results of the execution script throught Opensips with a proxy
configuration file
Sipp Originating
Messages Retrans Timeout
Unexpected-Msg
INVITE -->
her side with Custom header modparam)
Thanks
2013/7/8 Mehdi BOUDOU
> Hello,
>
>
> I'm already testing the b2bua module with the simple topology Hiding test,
> with a sipp scenario of a simple call.
> The Call is OK but I set the modparam("b2b_entities", "repl
Hello,
I'm already testing the b2bua module with the simple topology Hiding test,
with a sipp scenario of a simple call.
The Call is OK but I set the modparam("b2b_entities", "replication_mode",
0) and the call id of new leg is the same for the new INVITE ?
Thanks in advance
Here my config fil
Hello,
I'm trying to do callfow with sipp ,Opensips Proxy and another Application
Server.
the callflow consist in
--if there is no cpl file forwarding the call in stateless Mode to
the other Application Server,
--else executed the cpl file in statefull mode.
I detect if there is no
the modules
> after update - compile ? install ? restart ?
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
>
> On 02/06/2013 01:31 PM, Mehdi BOUDOU wrote:
>
> Hello Bogdan,
>
> Thanks for the patch :)
&g
hello,
I'm actually using opensip 1.5 with cpl module.
I would like to redirect the SIP packets if there is no cpl script to
execute for the client in database.
Can i use a special return of cpl_run_script ? or is there another way to
do that ?
is the send function ( send(destination [, headers
In english please :-)
Le 4 juil. 2011 15:00, a écrit :
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> or, via email, send a message with subject or body '
call of the third
destination?
The Scenario was sequantial ringing with 3 destinations during 5 sec.
Regards,
Mehdi BOUDOU
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trigg B2BUA behaviour to modify headers?
Is it the great interpretation of spiral with b2bua and call back to proxy ?
Thanks,
Regards
Mehdi BOUDOU
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Hello,
to be more accurate with my older post: is it possible to combine B2BUA
logic with Cpl module ?
Thanks,
Regards.
Mehdi BOUDOU
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AS and
thanks to the tag modify the Call-sequence ?
Or just do +1 to Cseq in Cpl-module when generated the different Invite?
Thanking you in anticipation.
Regards.
Mehdi BOUDOU
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ps[8169]:
DBG:core:parse_via_param: found param type 232, =
; state=6
Jul 19 15:54:28 intbims2 /usr/local/sbin/opensips[8169]:
DBG:core:parse_via_param: found param type 236, = <1>; state=16
Must i set the T1 timer to another value to stop the overflow timer? How can
i do to permit the call wi
Hello Bogdan,
I don't formulated well my question, how can i interact with the Opensips
server to not send the 408 Timeout because it 'is an overflow of one timer?
I will send you a callflow and the log to understand better the situation.
Thanks
Regards,
Me
o the IMS
core which Broke the Call.
How can i do to permit this type of scenario and block the sent of the 408
Timeout packet?
Thanks,
Regards,
Mehdi BOUDOU
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Hi Bogdan,
Thanks for your answer.
It works well :).
Regards,
Mehdi BOUDOU
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changed the value of the proxy timeout it changes
nothing ???
Is it the mode that i have executed the script("incoming,is_stateful mode")
an other timer in opensips that i have to set ??
How can i do to force opensips to take the value of proxy timeout in the cpl
file?
Thanks,
Regards,
Me
Hello,
I found the function record_route_preset("FQDN:port") which is ok for my
request.
Regards,
Mehdi BOUDOU
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Hello,
When the Opensips server answer to the S-CSCF in the record route, he answer
with IP adresse and not with his FQDN even i defined "listen=FQDN"
How can I do to force Opensips Server to use FQDN in his answer and not his
IP Adresse ?
Thanks for suggestions.
Regards,
Me
Hello Bogdan,
In C things are not easy :),but i am willing to take the risk of C-Coding
just for fun and it will be on an integration server no-risk for the
service.
Thanks for hints for how to do ...
Regards,
Mehdi BOUDOU
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Hello Jeff,
Thanks, i will try it! I will send you the answer to know if it works.
Regards,
Mehdi BOUDOU
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Hello Bogdan,
I will try to do it in the cpl-module, my methodology will be to find where
the different tag are and call the function
add_diversion(reason) at this time, I must include all libraries of
diversion header.
Is it the good methodology?
Regards,
Mehdi BOUDOU
Hello,
I would like to delete Supported: 100rel from every INVITE header sent by
the server.
How can i do?
Thanks,
Mehdi BOUDOU
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found one message with the great header and one without
Regards,
Mehdi BOUDOU
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gards
Mehdi BOUDOU
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Hello Bogdan,
Thanks I will see the example.
Regards,
Mehdi BOUDOU
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Hello Bogdan,
Thanks, I will search also in this way.
Regards,
Mehdi BOUDOU
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Ok thanks i will try with sequantial and multiple forking have you some
simple examples with sequantial and multi ringing operation in the
configuration file?
i read the fonction serialize_branches(clear) but it's not easy to use in my
opinion
Thanks,
Mehdi B
Hello,
Is it possible to do multi and sequential ringing just with the
configuration file without any other module ?
Thanks,
Mehdi BOUDOU
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Hello Bogdan,
i make this treatment if it is a tel uri and there is no loose route i set
$du=scscf-ourdomaine.net
if there is a loose route $du will be set by the top route of the
looseroute. it seem's to work
I think that i can't do more to root tel uri ?
Thanks for help,
Me
Hello Bogdan,
it is the great solution!!!, i set $du="scscf-ourdomaine.net:5090" if it is
a tel uri and it seem's to work :)
Can i keep this adresse from the top record route to be more dynamic?
Thanks for your help!!!!
Mehdi BOUDOU
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Hello Bogdan,
Have you any track or solutions for this error ?
Thanks,
Mehdi BOUDOU
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Hello Bogdan,
Are informations (invite and request ) ok for you or need you more
informations?
Thanks,
Mehdi BOUDOU
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p:+33351431...@ims.mncourmnc.mccourmcc.3gppnetwork.org
;user=phone>;tag=1450008788-1274374035967-
CSeq: 458116352 INVITE
Server: OpenSIPS (1.5.0-notls (i386/linux))
Content-Length: 0
*
*Thanks,
Mehdi BOUDOU
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xx /usr/local/sbin/opensips[5214]:
ERROR:tm:t_forward_nonack: failure to add branches
Tomorrow i will post the INVITE request and reply
Regards,
Mehdi BOUDOU
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Hello,
Has anyone any idea or track for my problem (error code 476)?
Thanks,
Mehdi BOUDOU
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You have to pay attention to your dtd path, you may change
http://www.w3.org/2001/XMLSchema-instance";
<http://www.w3.org/2001/XMLSchema-instance%22>
xsi:schemaLocation="urn:ietf:params:xml:ns:cpl cpl.xsd" > by
Regards
Mehdi BOUDOU
_
solution.
Is possible to forward tel_uri client to scscf which contact AS (default
route for tel_uri to scscf for example) ? Is dns_enum resolution
obligatory?
Now I have error 476 Unresolved destination when a tel_uri contact AS.
Regards
Mehdi BOUDOU
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