Ranganath B wrote:
> Hi,
> 
> Thanks again for your response, I guess I had a little misunderstanding 
> with working of SEMS.
> 
> Well, I had an idea to test the following things using SEMS:
> 
> 1) Voicemail.
> 2) IVR.
> 3) Conference.

> 4) Bandwidth control.
> 5) Audio path testing - Here I want to force all the RTP packets through 
> SEMS and then run some RTP test tool available free online on the same 
> machine and measure the audio quality.
try apps/examples/jukecall ( 
http://ftp.iptel.org/pub/sems/doc/current/ModuleDoc_jukecall.html ).

normally it strips 3 characters from username, so you may want to modify 
that (string callee = "sip:" + dlg.user + "@" + dlg.domain;).

then
if (method=="INVITE" && uri=~"test" && src_ip!=aa.bb.cc.dd) {
    t_relay_to_udp("aa.bb.cc.dd", "5070");
    break;
}

with aa.bb.cc.dd SEMS' IP.

what free RTP audio quality analysis tools have you found?

hth
Stefan

> 
> I guess my first 3 requirements are satisfied using SEMS :-)
> 
> I tested conference the way it was described in our earlier emails and 
> it works fine :-)
> 
> Thanks and Regards,
> Ranganath B
> 
> On Thu, Apr 17, 2008 at 1:07 AM, Stefan Sayer <[EMAIL PROTECTED] 
> <mailto:[EMAIL PROTECTED]>> wrote:
> 
>     Hello,
> 
>     Ranganath B wrote:
> 
>         Hi,
> 
>         Thank you very much for the support, I tried a lot of
>         combinations but at the end with no luck. Now I don't face the
>         problem of calls getting 500 error response :-) but still my
>         calls are peer-to-peer and not through the SEMS. When I observe
>         the INVITE message that is sent out from the SER-SEMS running
>         machine to the remote Xlite phone, I observed that the "a:
>         Connection Information:" parameter under SDP is still the
>         Caller-Xlite and not SEMS. So still my calls are not through
>         SEMS. Need help in resolving this issue.
> 
> 
>     if you want to call the other phone, then the call should go peer to
>     peer to it. why have it go through SEMS?
> 
>     if you want to call e.g. conference, or voicemail or the like,
>     modify your ser config as i wrote in the other mail.
> 
>     what are you trying to do?
> 
>     Stefan
> 
> 
>         Regards,
>         Ranganath B
> 
>     -- 
>     Stefan Sayer
>     VoIP Services
> 
>     [EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]>
>     www.iptego.com <http://www.iptego.com>
> 
>     iptego GmbH
>     Am Borsigturm 40
>     13507 Berlin
>     Germany
> 
>     Amtsgericht Charlottenburg, HRB 101010
>     Geschaeftsfuehrer: Alexander Hoffmann
> 
> 

-- 
Stefan Sayer
VoIP Services

[EMAIL PROTECTED]
www.iptego.com

iptego GmbH
Am Borsigturm 40
13507 Berlin
Germany

Amtsgericht Charlottenburg, HRB 101010
Geschaeftsfuehrer: Alexander Hoffmann
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