Update:
No luck with versions 1.6 and 1.8.7 I had to revert back to 1.4 which
worked with no problem.
Probably if I have some time, I will do more testing with version 1.8.7 to
see what the difference is and what changes need to be made for this kind
of setup to work in 1.8.7
Joseph
On Mon,
Thanks.
exten = s,n,Set(foo=${CHANNEL(peerip)}) ; Doesn't return anything
exten = s,n,Set(foo=${CHANNEL(recvip)}) ; Doesn't return anything
exten = s,n,Set(foo=${SIPCHANINFO(peerip)}) ; Returns public IP when
calling from a SIP device
Strange that CHANNEL doesn't return anything.
--
Hi everyone,
I'm currently trying to play a little with WebRTC using sipml5 client and
Asterisk trunk (370821)
It seems the the WebRTC implementation for Asterisk 11 is already available
in the trunk? Am I right?
http://lists.digium.com/pipermail/asterisk-dev/2012-July/055940.html
I'm having
mailsvb wrote:
Hi everyone,
Hola!
I'm currently trying to play a little with WebRTC using sipml5 client
and Asterisk trunk (370821)
It seems the the WebRTC implementation for Asterisk 11 is already
available in the trunk? Am I right?
On 12-08-07 03:31 AM, ml asterisk wrote:
Hi,
I used to install asterisk on debian squeeze with digium repository.
The last build of asterisk available is 1.8.11.1.
Is this repository discontinued ?
Since leaving Digium they have become unmaintained. If you are
interested in helping out, you
I apologize as I am quite new to working with SNMP. I have compiled the SNMP
module for Asterisk, configured snmpd, and successfully added the nodes to
OpenNMS for a couple of my systems running on Ubuntu Server 10.04 LTS. I am
able to see the site
information and the resource graphs such as
On 8/7/2012 7:27 AM, Paul Belanger wrote:
On 12-08-07 03:31 AM, ml asterisk wrote:
Hi,
I used to install asterisk on debian squeeze with digium repository.
The last build of asterisk available is 1.8.11.1.
Is this repository discontinued ?
Since leaving Digium they have become unmaintained.
On 2/08/2012, at 2:27 PM, Arstan Jusupov arst...@gmail.com wrote:
Dear list,
I am looking for an open source SIP client(or any SDK) that can work on a
browser. It may be based html5, javascript, flash, adobe air. I have done
some research myself and I would like to ask the community if they
Correct me if I'm wrong but phono works with voxeo tropo.
Sent from my iPhone
On Aug 8, 2012, at 7:24 AM, Matt Riddell li...@venturevoip.com wrote:
On 2/08/2012, at 2:27 PM, Arstan Jusupov arst...@gmail.com wrote:
Dear list,
I am looking for an open source SIP client(or any SDK) that can
Hi,
How many Public IPs connect to you ? If they are less than 15 or 10 , I
suggest you make sip.conf peers for them with host=Publicip and then decide
if you want that to be blocked or rerouted to some other direction !
If that isn't doable then try extracting/parsing some IP using the
10 matches
Mail list logo
