:44 -0500
Subject: Re: [asterisk-users] Calling a demo menu after voicemail
authintication
From:
Asmaa Ahmed asabatg...@hotmail.com
To:
asterisk-users@lists.digium.com
asterisk-users@lists.digium.com,
Date:
10/09/2013 10:36 AM
Subject:
[asterisk-users
Hello,
I wonder if it is configurable possible to add a new menu demo to run within
voicemail context dialogue!
I want to run am interactive menu before or within the normal voicemail
dialogue to run a script based on the subscriber selection.
My point is to get use from the authenticated
Hello,
While connecting to my voicemail, I noticed that Asterisk may perform some
tasks isn't included in the options levels that currently played!
For ex: I am listening to the main menu which asking me to press 1 for new
messages, 2 change folders, 0 mailbox options which is OK for these
Subject: Re: [asterisk-users] Invalid options
Hi Ahmed,
Seems to be that you have a problem recognizing DTMF digits.
Do you have RFC2833 as DTMF protocol in both Asterisk and soft/hard-phone ?
Rgds,
On Wed, 2013-10-02 at 11:24 +0200, Asmaa Ahmed wrote:
Hello
Thanks a lot the logs were handy. I thought just activating it through the CLI
would be enough!
The script was executed correctly, but couldn't see the results because it was
running in a different shell.
Once I fixed it, I started to see some output!
--
: don't copy the file but MOVE the file. if you copy the file asterisk may
execute partial file.
system (/bin/mv $filename /var/spool/asterisk/outgoing);
you can use any scripting language.
Hope this will help you
On Sun, Sep 29, 2013 at 2:01 AM, Asmaa Ahmed asabatg...@hotmail.com wrote
Hello,
I am trying to get MWI working after integrating Asterisk with CCM.I have
followed the instructions in
http://www.voip-info.org/wiki/view/Asterisk+Cisco+CallManager+Voicemail+IntegrationMy
problem is that I don't see externnotify's script being called at all in the
logs, and not sure if
Hello,
Now externnotify is called after adding mwi_from=asteriskBut now I am having
these error in the logs[Sep 28 19:58:44] WARNING[21156]: pbx_spool.c:278
safe_append: Unable to set utime on
/var/spool/asterisk/outgoing/201309NaVI5844-: Operation not permitted--
Attempting call on
Hello,
It looks that I got this in the logs while running the scripts manually by
mistake, so back to the starting point I can't figure why externnotify doesn't
run? My target is to have MWI (Message waiting indicator) running.Also still
can see the debug logs in CLI/asterisk logs even with
Hello,For the time being I am using the following line to play the original
saved message by Asteriskexten = 7001,n,Playback(vm-nobodyavail)Now I am
trying to use the other features for Asterisk's voicemail. I have recorded a
message, and I can see it saved on the system, but still Asterisk
On Thu, Sep 19, 2013 at 12:54 PM, Asmaa Ahmed asabatg...@hotmail.com wrote:
Hello,
I am trying to make my first call on Asterisk to succeed. I have Asterisk
1.8.10.1 running on Ubuntu machine.The configuration is quite simple just for
my first test, Trying to have a call between two X-lite
is established but without exchanged
voice packets
Hello,If Asterisk version is 1.6 use nat=force_rport,comedia
On Fri, Sep 20, 2013 at 4:01 PM, Asmaa Ahmed asabatg...@hotmail.com wrote:
Hello,
I have set the direct media to be off, but still doesn't work. I am not sure
about NAT
extension context.
On Fri, Sep 20, 2013 at 4:21 PM, Asmaa Ahmed asabatg...@hotmail.com wrote:
Hello,
I have Asterisk 1.8.10.1Moving to nat=force_rport,comedia hasn't solved the
problem. Still having the same error!
I am not sure if this is related to the problem here, but I was trying to test
Hi Matthew,
Indeed I missed your previous message!After changing the externip, it worked
successfully... The sip session is established with the complete three-way
handshake, and the voice packet is exchanged with no problem!
Many thanks.
Date: Fri, 20 Sep 2013 10:01:52 -0500
From:
Hello,
No, another installation haven't solved the problem!
It looks more like something related to the configuration in setting the
running environment!
Thanks. --
_
-- Bandwidth and
Hello,
I am trying to make my first call on Asterisk to succeed. I have Asterisk
1.8.10.1 running on Ubuntu machine.The configuration is quite simple just for
my first test, Trying to have a call between two X-lite sipphone. The
subscribers succeeded to register and the call is established, but
Hello,
I have started using Asterisk recently on my Ubuntu server. I installed it
first using apt-get and it worked fine sort of, but still couldn't hear voice
during the call!
I read that this problem solved by reinstalling it, so I decided to reinstall
the latest version from the source as
It looks this is because Asterisk isn't startedwhen tried to start it, I got a
core dump!$ /etc/init.d/asterisk start * Starting Asterisk PBX: asteriskIllegal
instruction (core dumped)
From: asabatg...@hotmail.com
To: asterisk-users@lists.digium.com
Subject: Can't connect to Asterisk cli
Date:
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