Displaying 20 results from an estimated 1000 matches similar to: "ZOMBIE on att transfer"
2006 Apr 06
0
Open channels
First, I'm not sure is this Asterisk or ooh323 channel problem.
It seams that I have solved (I do hope so!) deadlock problem with ooh323 (thanks to Sean and his patch). Now I have another one. It seams that some channels stay open even they should not. This is what I see from CLI:
pbx*CLI> show channels
Channel Location State Application(Data)
SIP/302-924a
2006 Mar 02
3
Native music on hold - Error
I have tried to use native music on hold. In dir /var/lib/asterisk/moh-native/ I have some wav files (with 755 permission). In musiconhold.conf I have
[native]
mode=files
directory=/var/lib/asterisk/moh-native
And in sip.conf I have
musicclass=native
When I put call on hold this is what I get at CLI.
-- Executing Dial("SIP/341-5931", "SIP/344|20|wWtT") in new stack
2006 Mar 06
0
Set(LANGUAGE()=language) - for queue
Hi group!
How to set language for queue?
I have several queue's. In every queue, agents speaks different language. I need to announce queue-youarenext and similar on different languages.
This is what I have in my extensions.conf and it does set language, but when calls enters queue, it doesn't use that language.
exten => 313,1,Answer
exten => 313,n,Set(LANGUAGE()=de)
exten =>
2006 Mar 26
0
Asterisk add-ons upgrade
I have running Asterisk 1.2.5 with addons 1.2.1. on Fedora Core 4. I have installed ooh323 from 1.2.1 addons.
How to upgrade addons to 1.2.2 version and install new ooh323 driver? Do I need to install addons 1.2.2 if I only need new ooh323 driver?
Can I just untar addons, and run "make clean; make; make install" and then execute following
cd asterisk-ooh323c
./configure
make
make
2006 Mar 01
0
ooh323 codec's - alaw
Does ooh323 from asterisk-addons 1.2.1 support alaw codec?
This is what is written in h323.conf.sample that can be found in asterisk-addons dir.
The codecs to be used for all clients.Only ulaw and gsm supported as of now.
Default - ulaw
ONLY ulaw, gsm, g729 and g7231 supported as of now
disallow=all
allow=gsm
allow=ulaw
So, it shouldn't support alaw, but I manage to establish calls with
2006 Feb 28
0
Asterisk hangs up - h323
This is third time today that my Asterisk hangs up. It seams that I have problems with h323. I'm using ooh323 from Asterisk add-ons. I have the following configuration
Asterisk 1.2.1
Asterisk-addons 1.2.1
Fedora Core 4
I'm using SIP phones and
h323 trunk to my VoIP provider
Like I said this is third time today that he hang's up. First time, I came at work and Asterisk was down.
2006 Feb 28
0
My or provider error?
Situation. I call out from SIP phone over h323 trunk and called person decides not to pick up (on mobile phone they press red button - NO - hang-up). Until the called person press the NO button, I can hear ringing. When called person press the button, I don't hear anything. Asterisk waits until timeout and than ends the call.
How can I get busy or some other appropriate signal on SIP phone
2006 Mar 28
0
Addons 1.2.1 upgrade to 1.2.2
How should I upgrade addons form ver. 1.2.1. to 1.2.2.?
I'm particularly interested how to upgrade ooh323 channel driver.
--
Tomislav Parcina
tparcina#lama.hr
2006 Mar 28
0
h323 channel driver for production
Hi group!
I'm having problems with ooh323 (ver 0.3?!? - the one that comes with asterisk addons 1.2.1) and I need to know what h323 channel driver you use in production?
Have a nice day!
--
Tomislav Parcina
tparcina#lama.hr
2006 Apr 09
0
(no subject)
In article <1251.165.146.69.140.1144596935.squirrel@www.ecntelecoms.com>, yusuf@ecntelecoms.com says...
> Hi,
>
> I have had the exact same problem last week. I have not yet solved it.
> So instead I am using ooh323, but would prefer to use oh323. Can anyone
> help?
I'm glad that I'm not the only one :))
Hopefully we'll find solution to this problem.
--
2006 Nov 30
0
Distinctive ring
Hi list!
I need help with distinctive ring on Cisco 7940 phone. I'm using Asterisk 1.2.5 (I know, I should upgrade) and in dial plan I have:
exten => _64X,n,Set(_ALERT_INFO=Chirp2)
exten => _64X,n,Dial(SIP/${EXTEN},30,wWtT)
On Cisco in Settings => Ring type I have "Chirp1" and "Chirp2". By default phone is ringing sound "Chirp1". For internal calls
2009 Jul 14
0
ooh323 doesn't know what to do when bridging calls
Dears;
I am having same problem, that when I place a call from the H323 end point (even if it is not added in the ooh323.conf), then asterisk handle the call and play the wave file in the default context. Also I added endpoint to the ooh323.conf and same thing, it keep goes for default context whatever the context placed.
My Asterisk vesion is 1.4.25
My Asterisk add-on version is: 1.4.8
What I
2006 Apr 04
1
asterisk-ooh323, asterisk 1.2.6 and netmeeting
has anyone managed to get these three beasties to work together ? we're
using ooh323 from asterisk-addons-1.2.2, asterisk 1.2.6 and microsoft
netmeeting default from windows xp.
the symptoms are that calls from a SIP client to NetMeeting rings on
NetMeeting, but upon answering the call in NetMeeting, no audio is passed
between the two. eventually, the call times out and hangs up.
on a
2007 Feb 28
0
Using ooh323 with Gatekeeper controlled dialling
All,
I've fixed my problem getting Asterisk ooh323 channel to stay registered with my Cisco IOPS gatekeeper, now I need to get dialling working.
I have the following:
[Asterisk with ooh323] ----h323---- [Cisco IOS GK] ----h323---- [Radio system OpenH323]
192.168.1.5 192.168.1.6 192.168.1.7
the Asterisk box has numbers
2010 Jan 04
0
H323 Disconnects after 15+ minutes
I have posted my problem on the link below, but didn't get any answer. I am hoping someone here can help me with this issue. Here's my problem:
I am using H323 to talk between Asterisk and Avaya IP Office 500. For
some strange reason, when we are talking on a VoIP call, we get
disconnected after 10+ minutes. We have two other Elastix box, but none
of them are getting disconnected. From
2011 Jul 13
1
Connect Avaya to Asterisk PBX
Hi List,
I have another issue on allowing outgoing calls to PSTN on Asterisk via
Avaya Phones, I hope that anyone could help me fix this issue:
*When I dial through Avaya phone i just here a "good bye message" reply
from asterisk server. And here is the log:*
== Starting OOH323/(null)-b7db8aa0 at internal,s,1 failed so falling
back to exten 's'
== Starting
2007 Feb 23
1
ooh323 hang up after the call is answered
Hi,
I'm trying to make ooh323 works with one asterisk box running 1.2.15
version.
I can ring from a h.323 to SIP and SIP to H.323, but when the call is
finished when the phone is answered.
This is the log when I call from the H.323 device to a SIP device:
Feb 23 10:57:32 VERBOSE[6096] logger.c: -- Executing
Dial("OOH323/Telconet Mantaer-c5f8", "SIP/666|30|TtrwWC")
2006 Dec 07
1
-- Called 12127773456@OOH323 Segmentation fault (core dumped)
OOH323 Debugging Enabled
-- Executing Answer("SIP/3513-090f7d40", "") in new stack
-- Executing Wait("SIP/3513-090f7d40", "1") in new stack
-- Executing DeadAGI("SIP/3513-090f7d40", "a2billing.php|1") in new
stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/a2billing.php
a2billing.php|1: line:58 - IDCONFIG : 1
2006 Jun 20
0
ooh323 issues
Hi all.
Trying to setup H.323 via Asterisk between a PLANET H.323 box and
my SIP phones.
When calling from the SIP phones, it connects but quickly
disconnects citing the following error message:
****
--- build_peer
+++ build_peer
+++ reload_config
+++ ooh323_do_reload
-- Executing Dial("SIP/yyy-2965", "OOH323/203@xxx") in new
stack
--- ooh323_request - data
2013 Jun 08
0
H.323 Trunk between Asterisk 11 and Avaya
Hello,
I'm trying to create a H.323 trunk between Asterisk 11 and Avaya. I have
done this before between Asterisk 1.6 and Avaya but had some issues placing
external calls from the Asterisk to the Public network which is connected
to Avaya. I'm trying to create that trunk on Asterisk 11 because the 1.6 is
outdated and has no support.
On the Asterisk side I have Aastra 6731i SIP phones