I am experimenting with Asterisk having downloaded and installed the
FreePBX i386 CentOS-6.3 based distro and updated it. The current
package level on this system is:
asterisk11-11.3.0-49_centos6
freepbx-2.11.0beta2-112
I am using twinkle-1.4.2-7.el6 as a softphone testing tool.
There is no
31.03.2013 23:15, Barry Flanagan ?:
On 31 March 2013 18:11, Dmitriy Serov serov@gmail.com
mailto:serov@gmail.com wrote:
Hi, asterisk admin and users.
I need to SIP INVITE uri with domain via peer. And uri domain
differ then peer domain.
dialplan:
exten =
On 13-04-01 03:16 PM, Dmitriy Serov wrote:
31.03.2013 23:15, Barry Flanagan ?:
On 31 March 2013 18:11, Dmitriy Serov serov@gmail.com
mailto:serov@gmail.com wrote:
Hi, asterisk admin and users.
I need to SIP INVITE uri with domain via peer. And uri domain
differ then
On 31/03/13 23:43, Joshua Colp wrote:
Daniel Pocock wrote:
I'm trying to call from DruCall to Asterisk and I get this error:
WARNING[11021]: chan_sip.c:8687 process_sdp: Error in codec string 'F
103 104 111 0 8 107 106 105 13 126'
== Problem setting up ssl connection:
Daniel Pocock wrote:
Thanks for the fast reply. I agree backporting full support for AVPF
would not be justified for many use cases (including my own). What I
was more curious about is whether the F can be tolerated (in other
words, ignored or silently removed), as described here:
From a
Hi all,
Hopefully, I just need a second set of eyes on this one, but I just can't
figure out what I'm doing wrong. I'm using an agi script to dial a number,
check the dial result, and act accordingly.
The problem is that I'm not getting anything back from DIALSTATUS, or
HANGUPCAUSE.
Here is
On 01/04/13 22:06, Joshua Colp wrote:
Daniel Pocock wrote:
Thanks for the fast reply. I agree backporting full support for AVPF
would not be justified for many use cases (including my own). What I
was more curious about is whether the F can be tolerated (in other
words, ignored or
