Yep, looks like Google changed something. Try this:
https://issues.asterisk.org/jira/browse/ASTERISK-18301
Fixed it for me.
On Fri, Aug 19, 2011 at 11:09 PM, Jim Boykin wrote:
> We have been using gtalk channel from a long time now. It was working
> fine so far but from yesterday we are having
We have been using gtalk channel from a long time now. It was working
fine so far but from yesterday we are having problem. When gtalk
destination is dialed and even answered, channel remains in Ringing
state. Is there anything changed on google side?
Jim
--
__
I am not sure you even read my mail, no music on hold option - it
should work dynamically with any file.
On Fri, Aug 19, 2011 at 6:18 PM, bakko wrote:
> Hi,
>
> you can configure a new music on hold, example:
>
> nano /etc/asterisk/musiconhold.conf
>
> [default1]
> mode=files
> directory=moh1
>
>
This is a simple call file task. First of all, I would convert the MP3 file
to wav format. Then just use this call file and you're good to go
(1 file for each PRI line)
Line 1
extension: 170
Channel: DAHDI/R1/#
app: Playback
Application: Playback
Data: /var/lib/asterisk/sounds/en/tt-monkeys
Line
Hi,
I have 8 E1 PRI Lines and i have 200 phone numbers and 200 channels
(25 channels per PRI). is there a utility available in Asterisk to
dial out 200 numbers and run a campaign for 200 numbers concurrently
and play a mp3 file ?
Please suggest/guide
Regards
Kaushal
--
Danny,
Now that I notice, MaxRetries has a default of 0 anyway, so not setting
it should have not retried at all anyway. Although I did set it to 0
manually and still get the double calls.
Brandon
On 08/19/2011 09:31 AM, Brandon Phelps wrote:
We are running version 1.8.5.0. I'll try the Ma
Danny,
Now that I notice, MaxRetries has a default of 0 anyway, so not setting
it should have not retried at all anyway. Although I did set it to 0
manually and still get the double calls.
Brandon
On 08/19/2011 09:31 AM, Brandon Phelps wrote:
We are running version 1.8.5.0. I'll try the Ma
We are running version 1.8.5.0. I'll try the Maxtries and see what happens.
Brandon
On 08/19/2011 09:19 AM, Danny Nicholas wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Brandon Phelps
Sent: Friday,
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Brandon Phelps
Sent: Friday, August 19, 2011 8:15 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Possible Bug? .call files executing multiple time
Hello all,
We are setting up an auto-dialer to call customers based on the opening
of tickets in our internal ticketing system. Everything is going fine
so far except for one snag:
To test the system we are implementing I am manually moving .call files
into the /var/spool/asterisk/outgoing
any answer on below..
On 08/18/2011 03:50 PM, Nikhil wrote:
Hi
Using AMI how can I get the presence feature.Below are the requirement.
--> List of all users in the PBX including analog and SIP
including registration status.
--> Status(BUSY or available ) of all users both analog a
Hi,
you can configure a new music on hold, example:
nano /etc/asterisk/musiconhold.conf
[default1]
mode=files
directory=moh1
and put the audio file in this directory; then change your dialplan like:
exten => 500,1,NoOp
exten => 500,2,Dial(SIP/14085551234@myprovider,m(default1))
exten => 503,3
I was using Asterisk with SmartVoip and JustVoip for a long time running
on CentOS 5. Recently I upgraded to CentOS 6 with a fresh build of
Asterisk 1.8.5.0 Using the same firewall and SIP configuration as before
now I can't dial out:
retrans_pkt: Retransmission timeout reached on transmission
A(x) does not accomplish this. It completes the playback and then
dials. What I would like is that dialing should start in parallel and
playback should stop as soon as early media or ringing starts.
Similarly, music-on-hold is not an option, it's too hard coded, I like
to be able to change playbac
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