Hi Adrian,
I was able to successfully setup and run the Sylk-Server in a Debian machine.
Now I am trying to get connected to the server with my Openphone based endpoint.
Is there any documentation for this regard that I can refer to?
(How to get connected to the server, etc.)
Thanx,
Kapi
> Thank
Hi Leon,
I'm having this same problem, you implemented this feature?
Best regards,
--
Guilherme Bessa Rezende
Security and Telecom Developer
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I have an issue with Snom Phones. They support the SIP method "MESSAGE" and
when my SIP SIMPLE clients IM each other the Snom phone also receives the
IM. This is annoying because you will always need to press cancel on the
phone to get rid of the messages. I've posted on the Snom forum and I
I can reproduce this for every call.
Complete SIP call trace and media proxy logs are attached.
Thanks,
On Thu, May 10, 2012 at 3:41 AM, Saúl Ibarra Corretgé [via OpenSIPS
(Open SIP Server)] wrote:
> Hi,
>
> Can you paste the dispatcher and relay logs from the start of the call?
> Also, can you
Depending on your setup there are a couple of ways to do it.
See this post
http://opensips-open-sip-server.1449251.n2.nabble.com/Global-Variables-td3764430.html
I've used cache_store and cache_fetch without issue.
http://www.opensips.org/Resources/DocsCoreFcn18
On , prasad kelkar wrote:
h
hello,
I want to add one global variable.
which script contains global variable?
thank you
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I am trying to do load balancig with softphones as register.
I ad to softphones to resources table with resource field set to cc=1
(refered to current call), then I add this to opensips.cfg
if ($rU=~"^1002") {
load_balance("1","cc");
}
1002 is a registred user. then when I dialed 1002 from a regist
Hi,
On May 10, 2012, at 10:35 AM, Jacek Konieczny wrote:
> Hello,
>
> I have an opensips with mediaproxy configured as a SIP gateway between
> our PBXes and PSTN trunk providers. I would like to hide our internal
> network topology, though I would prefer not to change whole
> configuration to B2
Hello,
I have an opensips with mediaproxy configured as a SIP gateway between
our PBXes and PSTN trunk providers. I would like to hide our internal
network topology, though I would prefer not to change whole
configuration to B2B. Opensips 1.7 provided the topology_hiding()
function in the dialog m
Hi,
Can you paste the dispatcher and relay logs from the start of the call? Also,
can you reproduce this consistently? The call SIP trace would also help.
Thanks!
On May 10, 2012, at 12:50 AM, whglee wrote:
> I have a problem with media proxy not rewriting SDP in the 200 OK. I got the
> follo
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