So say RTPProxy can successfully do MOH, if your current VoIP infrastructure
uses Mediaproxy then you would need to set up a RTPProxy server and have the
customers that wish to have MOH use RTPProxy instead of Mediaproxy correct?
I guess the only caveat is that RTPProxy doesn't work with CDRTool
On Thursday 28 January 2010 11:22:52 Saúl Ibarra Corretgé wrote:
> Hi Mike,
>
> I received your trace, but let me answer you on the list, I won't expose
> anything:)
>
> Here is what happens:
>
> - Zoiper sends an INVITE with audio proposal.
> - Asterisk answers with 200OK and audio answer (m=audio
Hi Marie,
You are being very unclear in fact. Were have you tried to put the
parameter? When calling b2b_init_function? If so, see the documentation
about the script variables here:
http://www.opensips.org/Resources/DocsCoreVar#toc66.
Regards,
--
Anca Vamanu
www.voice-system.ro
marie.grem
Hello list,
I'm using:
Solaris 11 x86 (nv-b91)
OpenSIPS 1.6.0 with TLS
...and I see this in the log:
opensips[2717]: WARNING:core:get_send_socket: protocol/port mismatch
some hundreds of times per day (about once every 20 minutes
per registered UAC.)
I have this in the route script:
Hi Saúl
Thanks, too tell you the truth I was seeing this in the asterisk logs
but was not sure what a correct trace should look like.
I'll look into the correcting some other way.
Thanks
On 28/01/10 9:52 PM, Saúl Ibarra Corretgé wrote:
> Hi Mike,
>
> I received your trace, but let me answer
Hi Mike,
I received your trace, but let me answer you on the list, I won't expose
anything:)
Here is what happens:
- Zoiper sends an INVITE with audio proposal.
- Asterisk answers with 200OK and audio answer (m=audio on the SDP)
- Zoiper sends a re-INVITE to change to T.38 with a proposal (m=im
Hi,
On 28/1/10 10:29 AM, Mike O'Connor wrote:
> Hi All
>
> I've been trying to get T38 faxing going and for testing I've been using
> the t38 client in zoiper and asterisk 1.6.
>
> The process is that the call goes to opensips which has a usr_pref which
> indicates this is a fax number, the call i
Hi All
I've been trying to get T38 faxing going and for testing I've been using
the t38 client in zoiper and asterisk 1.6.
The process is that the call goes to opensips which has a usr_pref which
indicates this is a fax number, the call is then forwarded to Asterisk.
Asterisk answers the call af
Hi Bodgan,
I forgot to mention that files are not storing by callee or caller number.
Moreover it is taking its own unique caller id. How to over come this in
order to modify the recording file name as callee<->caller and time stamp
format.
--
View this message in context:
http://n2.nabble.com/
Hi Bodgan,
Yes. These files are raw rtp files. When ever call is ended rtp proxy
storing the 2 raw rtp files in to specified destination folder. One for
callee and other for caller. The problem i faced is i have to merge these 2
raw rtp files of each call and convert into wav file to hear the
con
Hello,
An jeu., janv 21, 2010, Alex Balashov schrieb:
>On 01/21/2010 12:00 PM, Mehdi Bouchefra wrote:
>> I'd like to implement the function Music on hold on Opensips.
>>
>OpenSIPS is a SIP proxy, not a media endpoint. So, it doesn't do
>that.
>
If while OpenSIPS routes a call a person presses 'h
To simplify the call flow, I omitted the 100 Trying sent upstream by
OpenSIPS for both INVITE (1) and (2).
You will find below the complete call-flow
Only one INVITE will be accepted, but I could not guess which one.
The upstream proxy provides the REGISTRAR function, thus it forks initial
To simplify the call flow, I omitted the 100 Trying sent upstream by
OpenSIPS for both INVITE (1) and (2).
You will find below the complete call-flow
Only one INVITE will be accepted, but I could not guess which one.
The upstream proxy provides the REGISTRAR function, thus it forks initial
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