This idea is quite standard in SBCs, typically called codec profiles, where
you permit only certain codecs to be passed through the SBC in an INVITE and
all the rest are stripped out.
We use it to get around interop issues with certain codecs. E.g. we have
some end devices/customers that have
Hi,
I want to disable CDRTool's quota functionality completely including SELECTs
with quota column from subscriber table, UPDATEs during cdrs normalization
etc.
How can I do this?
--
Best regards,
Ikar
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Hello,
I am hoping someone can point me in the right direction. I have
configured my OpenSIPs server to load balance 10+ asterisk servers
using the dispatcher module. To date I have not been able to
implement the probe functionality because the OPTIONS and INFO
methods both cause asterisk to
Hi Ikar,
If your data source does not have UserQuotaClass defined, the scripts
that check the quota are not performing any actions. This exclude the
SELECT in the opensips.subscriber table so you do not need to add a
quota column either.
Internally the cdrtool.quota_usage MySQL table is
The correct way is this...
UAC Proxy UAS
|INVITE | |
|---| |
|100 Trying | |
|---| INVITE |
|
Hi Geoff,
It's very strange that Asterisk answers OPTIONS pings with a 4xx error,
because OPTIONS is the method Asterisk uses to do its own availability
pings -- that's what the qualify= setting for peers in sip.conf enables.
What exactly is the 4xx error? Is it 403 Forbidden? Might it have
Iñaki Baz Castillo wrote:
El Domingo, 1 de Febrero de 2009, Alex Balashov escribió:
It's very strange that Asterisk answers OPTIONS pings with a 4xx error,
because OPTIONS is the method Asterisk uses to do its own availability
pings -- that's what the qualify= setting for peers in sip.conf
El Lunes, 2 de Febrero de 2009, Geoffrey Mina escribió:
Thanks for pointing me in the right direction. I didn't have an s
extension defined in my default context, so asterisk was returning a
404 error because OpenSIPs doesn't specify an extension in the OPTIONS
packet. The s is apparently
How would I configure the ruri in opensips to provide an extension
similar to sip:p...@asterisk.mydomain.com?
I couldn't get anything other than sip:asterisk.mydomain.com
Thanks.
Geoff
On 2/1/09, Iñaki Baz Castillo i...@aliax.net wrote:
El Lunes, 2 de Febrero de 2009, Geoffrey Mina escribió:
Hi everyone,
I have installed OpenSIPS 1.4.4 recently on a ubuntu 8.04, I have 3 UAC
configured. Two of them are SJphone which are within the same network. One
of them (Xlite) is outside the network and behind NAT. I am facing two
problems.
1. The SJphone within network works fine but user hear
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