to take effect. When I reboot the server, the trunk is gone!
WTF???
--
Gary Baribault
Courriel: g...@baribault.net
GPG Key: 0xFA812835
GPG Fingerprint: 8597 4D3D 3C3D 4247 077C 9FF9 E412 CAC4 FA81 2835
___
-- Bandwidth and Colocation Provided by http
Hi All,
I'm new to Asterisk, but am a relatively accomplished Linux guy
(RH5.1) .. I'm using the Asterisk-GUI to try and create a menu for
incoming calls on an Analog Trunk. I have recorded some .WAV files for
the menu, but when I try to upload the files, I get an AG101 message. So
I
Hi All,
I'm new to Asterisk, but am a relatively accomplished Linux guy
(RH5.1) .. I'm using the Asterisk-GUI to try and create a menu for
incoming calls on an Analog Trunk. I have recorded some .WAV files for
the menu, but when I try to upload the files, I get an AG101 message. So
I
for open source, the server is not
commercial, and I have very little budget.
Thanks
Gary B
--
Gary Baribault
Courriel: g...@baribault.net
GPG Key: 0xFA812835
GPG Fingerprint: 8597 4D3D 3C3D 4247 077C 9FF9 E412 CAC4 FA81 2835
___
-- Bandwidth
Hi Folks, sorry for the delay ... I found that the documentation was
rather iffy .. I finally found the defines.php in the lib subdirectory
and figured out how to give the MySQL port with the host and it all
works fine now.
Gary Baribault
Courriel: g...@baribault.net
GPG Key: 0xFA812835
GPG
work fine, sound is correct.
Voice mail works fine as well, the IVR works great.
Any ideas?
Gary Baribault
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live
Incomming calls are on TDM lines connected to the Digium card. Calls
between extentions are on the LAN for SIP registered users/ip phones.
Gary Baribault
On 06/01/2010 03:32 PM, Zeeshan Zakaria wrote:
Incoming and outgoing calls are on SIP or on ZAP?
Zeeshan A Zakaria
--
Sent from my
This is done while the calls are active? I just issued the command and
got nothing, but there where no active calls.
Gary Baribault
On 06/01/2010 03:45 PM, Danny Nicholas wrote:
My assumption is that inbound/outbound calls are DAHDI and that internal
calls are SIP. Can OP post core show
As I stated, the incoming calls are on TDM DS0s connected to the Digium
card, and the extensions are on the same local network as the Asterisk
server. There is currently no NAT anywhere.
Gary Baribault
On 06/01/2010 05:22 PM, Zeeshan Zakaria wrote:
Output of 'iptables -L -n' would also
I have remote access to the server so I checked the canreinvite .. they
are all set to no. I can't try the call from here, I will get back to you.
Gary Baribault
On 06/01/2010 07:24 PM, Zeeshan Zakaria wrote:
Do you agree something is blocking the audio in one direction? Can you
do a 'rtp
I have checked, the users have ulaw, then alaw, the phones are set to
711u then 711a which is the same thing (I think).
Gary Baribault
On 06/02/2010 08:32 AM, taimur hasan wrote:
Also check the codecs as if you are using g729 or g723, there is a
chance that they are not available in codecs
I don't know if this makes any difference, I created a lot of this
configuration with the Asterisk-GUI (SVN-branch-2.0-r4980) and when I
edit the users.conf file, there are two entries 'type = peer' for each
extension and they are highlighted in red!
Gary Baribault
On 06/02/2010 08:32 AM, taimur
Hi All,
I have a server running Fedora 14, kernel 2.6.31.14, Asterisk
1.6.2.17.2 and Dahdi 2.4.1.
I have the wctdm24xxp+ loaded with a Wildcard TDM800P with 8 FXO ports
When a call is placed extension to extension, there is no
problem... When an extension is used to dial out, the
13 matches
Mail list logo
