Hi Guys
Has anyone got this working on Centos 6?
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Hi All,
OS : Cent OS 5 64Bit
Asterisk : 1.8.0-rc2
AMR Source Link : http://sourceforge.net/projects/aterisk-amr/files/
When I tried to call or start asterisk, I found Segmentation Fault. Below
I paste same for AMR
Loaded symbols for /usr/lib/asterisk/modules/app_db.so
Core was generated by
Hi David,
The old app_fax code, which allowed spandsp to be used with Asterisk
before Digium introduced the new modules supported the features you
want. Maybe someone can go through that code and port the feature into
the current res-fax code.
Steve
On 07/03/2012 09:57 AM, David Cunningham
Dear All,
I have two server 'A' and 'B' . In Server 'A', five different ivr (Sevices)
is playing and call is *forwarding *into Server 'B'. Server 'B' basically
use for agent login(Extension).
I want to play different hold music(Server 'B') bases on the corresponding
services which is running into
Hi,
if possible for you put some header in SIP which mentions the music on hold
flag on Server-A. The Dial the call to Server-B. On Server-B extract the
value of that header and change the music on hold class based on the value.
Regards,
Sammy
On Tue, Jul 3, 2012 at 5:30 PM, akhilesh chand
I just tried it on asterisk 1.8.13 with agi set debug on. The last log
line reveals it - streamfile return the endpos.
[2012-07-03 15:16:39] VERBOSE[7046] res_agi.c:
SIP/tgoellner-0002AGI Rx STREAM FILE /audio1/dtmf_detector/2.0
1234567890*#
[2012-07-03 15:16:39] VERBOSE[7046]
That is all. Thank you.
FAQ.
welcome to read the
believe me, you are
If you do not
to *bottom*.
we read from *top*
In this mailing list,
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Since you're using IAX2 to contact Server B, you can use channel variables
to control the moh class. There was a good thread in June on this. An
easier way however would be to have each service dial a different IAX
number, then have each IAX number on server B select it's MOH Class.
Server A
Sorry, but I am using a self developed PHP-Library where I parse STDIN
myself. So I have no problem on this side. You are using a Perl-API?
There should be a method available for getting the AGI-Result-String?! I
never used Perl myself ...
Am 03.07.2012 16:13, schrieb CDR:
Yes, ai saw that
I administer a group of Asterisk servers running a mix of 10.3, 10.4, and
1.8.8.1 (mostly 10.4). One of those servers is a call concentrator/relay for
E911 service. All of the other servers make an IAX connection to the relay
server, which then hands off to a SIP trunk to my E911 provider.
- Original Message -
From: Noah Engelberth n...@directlinkcomputers.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, July 3, 2012 10:56:10 AM
Subject: [asterisk-users] IAX trunking stopped working
I administer a group
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Matthew Jordan
Sent: Tuesday, July 03, 2012 2:27 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] IAX trunking
Dear;
What is the setting to be done on freepbx to let the voicemail go for hangup
after while (or after leaving the message) even if the caller did not dial #.
It is very important for me to be sure of the hangup status.
Regards
Bilal
--
I am using call files to make calls to a remote machine but can't seem to
quite understand the directmedia options that are set by default in
Asterisk.Is there any way i can specify the directmedia options using call
files?
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On Tue, 2012-07-03 at 17:13 +0530, Chandrakant Solanki wrote:
Hi All,
OS : Cent OS 5 64Bit
Asterisk : 1.8.0-rc2
AMR Source Link : http://sourceforge.net/projects/aterisk-amr/files/
When I tried to call or start asterisk, I found Segmentation Fault.
Without trying to be pedantic, but
I have been doing some testing with queues. I have been experiencing some
strange behavior and I wanted to see if anyone else sees this. I am using
1.8.11-cert2.
It is my understanding that I cannot directly tell from the dial plan that a
member is dynamic or static unless I check RQMSTATUS
Hi,
The method Danny suggested is simple except I guess he swapped the priority
and exten field.
The idea is to dial a different extension on B server if you need to use
some other MOH class. If you don't want to change the dialled extension you
can always add a single digit prefix in Server-A
So, is http://sourceforge.net/projects/aterisk-amr/files/ same patch also
works in 1.8.13.0??
On Wed, Jul 4, 2012 at 3:18 AM, Hans Witvliet aster...@a-domani.nl wrote:
On Tue, 2012-07-03 at 17:13 +0530, Chandrakant Solanki wrote:
Hi All,
OS : Cent OS 5 64Bit
Asterisk : 1.8.0-rc2
AMR
I don't think you can set SIP properties in some variables anywhere in
asterisk dialplan or call file. What you can do is change the directmedia
options of the SIP or any other channel you're using. i.e if your call file
has
CHANNEL=SIP/12345@latestgateway
Then change the properties of the
hi,
Server A
extentsion.conf
exten = N,n,Set(Service_name=Test)
exten = N,n,Dial(IAX2/
server2:server2@192.168.14.112/${result},${Service_name})
but Server B doesn't identify service_name.
Server B
iax.conf
[general]
register = server1:server1@192.168.14.110
[server2]
type=friend
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