Displaying 20 results from an estimated 5000 matches similar to: "Call Waiting broken on ZAP"
2007 Dec 22
0
Dead Incoming call - Sangoma A200
Hello List,
I am having a strange issue with a trixbox system we installed for a client, and I would appreciate any help on this one. The issue is that occasionally when they go to answer an inbound call from the Sangoma A200 - there is no one there, and they are presented with dial tone. The calling party is hung up.
A bit of background:
The client actually has two systems install (one at
2006 Apr 07
1
Inbound PRI calls drop after 5 seconds using Sangoma A101
Hi Folks,
I'm have Asterisk version 1.2.1 with a A101 PRI card. I'm working with the
CLEC to bring up the PRI and inbound calls are hanging up at his end after
a few seconds. I ran PRI debug but it only gives me minimal insight.
" Ext: 1 Cause: Unknown (16), class = Normal Event (1)"
He ran a trace and the only difference he is seeing is a
"ISDN interface explicitly
2008 Apr 04
0
Problem about calling from atrixbox to pbx extension
I have a trixbox 2.2 and Nortel santral that are speak each other. I use
digium TDM100M 2 fxs-2fxo. After I made yum update I had met with some
problems when I want to make any call from extension of trixbox to
extension of nortel. When I attend to log (/var/log/messages) I meet
with these messages as you see below.
When I try to make any call from trixbox extension the call seems
established but
2005 Sep 01
1
dialparties.agi is returning no extensions to dial
Hi,
I set up a ring group. I would like for people who select a certain voice
menu option to ring a list of extensions (I have just one extension in there at
the moment) and if it doesn't answer to go to an extension's voice mail. I am
using a version of asterisk from CVS, last updated a couple of weeks ago.
This line in extensions_addtional.conf sends the call to ringgroup 3 if
2009 May 16
1
Queue Load, Asterisk Disconnected
I have Asterisk 1.2.29, Zaptel 1.2.24 and Freepbx Setup for
a queue up to 15 agents through a PRI line, it was working fine for more than 1
year, suddenly, when there is a load on the queue, the asterisk service
disconnects and the calls are dropped. And the service starts again after few
seconds, and so on.
I am not using fax.
I checked PRI by zttool and there are no alarms.
The cdr logs
2010 Jun 21
1
How to tell if a dropped call is my fault
I just had a user report that they called out to someone on a cell phone this morning, and after a short conversation, the call was dropped/lost. The person on the cell phone says that this is very rare, and would not suspect the dropped/lost call to be on their side. I have looked at the asterisk/full log as thoroughly as I can, and have pasted the lines which seem relevant to that call below.
2006 Feb 10
0
Yuck! Asterisk Crash...
Hi,
I'm currently running CVS-HEAD 2005-09-03
I do plan to upgrade to the newest version, but need to do some
testing with it first. In the mean time... does anyone know what
these messages below are about? I've never seen it before, but when
it happened it locked Asterisk up pretty good.
Feb 10 10:16:51 DEBUG[14917] chan_zap.c: Echo cancellation already on
Feb 10 10:16:57
2006 Feb 28
1
FW: Re: Delay on Phone ringing
Skipped content of type multipart/alternative-------------- next part --------------
asterisk1*CLI> soft hangup Zap/1-1
Requested Hangup on channel 'Zap/1-1'
== Spawn extension (macro-exten-vm, s-BUSY, 2) exited non-zero on 'Zap/1-1' in macro 'exten-vm'
== Spawn extension (ext-local, 220, 1) exited non-zero on 'Zap/1-1'
-- Hungup 'Zap/1-1'
--
2009 Mar 17
0
Weird issue with outbound calls and MOH
Hi,
We have a PRI Trunk (physical E1) and we are getting
some rather weird and very isolocated problems. On outbound calls to
specific numbers, it would seem to me that DTMF from the remote side is
affecting the local asterisk system. Basically what happens:
- We make a OUTBOUND call via the PSTN (PRI Trunk) to a remote System
- Remote Answers, and converse
- Remote sends DTMF on their site to
2006 Feb 02
0
Agents, queues and zombies
Hi all,
Have been experimenting with agents and queues instead of placing calls
direct to a user's phone extension, but I've run into problems with calls to
both the agent and the extension which creates a zombie and double records
calls abandoned etc. We're using a unique queue for each agent (only a
handful of users) to try and get some agent/queue information to see what
the
2005 Jul 14
0
Zap channel billing on busy tone!
Here is a log from a recent call made out on a ZAP channel from a SIP phone
inside my network.
For some reason, CDR is billing time even though the "busy tone" was
detected.
It's also logging the call as ANSWERED.
Is this normal behavior? Seems a little odd to me.
I have this as the first 3 lines of my zapata.conf
[channels]
busydetect=1
busycount=3
CVS HEAD updated late
2009 Sep 18
0
Blind Transfer Won't Hangup
I'm using FreePBX 2.5.2.2 with Asterisk 1.6.1.4. If I make a call and
then decide to blind transfer them using ## my side of the call is not
hung up. Instead it sends me to voicemail. If somebody calls me and
then I blind transfer them with ## I am hung up on as expected.
I called from 8678 to 28688. I then transferred the call to 8532.
Asterisk acts like it wants to hang up, but then
2009 Oct 08
4
Dialplan problem
Hi people,
I have the following dialplan, but it doesn't have the behavior that I think it should have.
[default]
exten => 2001,1,Answer
exten => 2001,n,Dial(local/3005)
exten => 2001,n,Hangup
exten => 3005,1,Set(__RINGTIMER=10)
exten => 3005,n,Macro(exten-vm,novm,3005)
exten => 3005,n,Hangup
When I execute the Originate (AMI) with the argument Channel=local/2001, It rings
2005 Sep 27
1
Extensions go straight to voicemail
Hello,
I have setup a test server with asterisk/AMP and have several 7960's
connected to it. The asterisk server has a public ip and all the
7960's are behind nat'd routers. When I try to call from extension
to extension I get directed straight to voicemail. I do not have any
cards installed and instead direct everything to an Ondo server. I
have been told it's not an AMP
2006 Jun 09
2
No CID on ZAP
I am using asterisk version 1.2.6 with Zaptel version 1.2.5. I have a POTs
line coming into a Digium TDM01B. It appears to not be getting CID at all.
If I hook up a POTS phone to the line CID comes through fine. Inbound and
outbound calls work fine but there is just no CID on inbound for this
channel.The incoming route for the channel is Zaptel Channel 0. No DID or
CID settings applied. My IP
2005 Aug 02
0
Hang up as soon as other party picks up call
Hello,
I have an Asterisk box with a TE410P connected to a PRI line and agents with
X-Lite installed on the same LAN as the Asterisk server. Sometimes, when I
make outbound calls it hangs up as soon as other party tries to picks up the
call. Does someone ever experienced this situation? On X-Lite, only
G711-ulaw is enabled and here is what i put in sip.conf:
[4001]
type=friend
username=4001
2007 Jun 08
1
call problem...
Hi, i got Ubuntu 6.06 installed and theres a problem with asterisk.
I've sucessfully installed it with the command:
#apt-get install asterisk
Then after installing FreePBX i get this error when restarting asterisk:
root@hernandezz-laptop:/home/hernandezz# asterisk -rvvvvvvvvvv
Unable to connect to remote asterisk (does
/var/run/asterisk/asterisk.ctl exist?)
After looking at the logs i
2006 Mar 26
0
hang up when pickup analog phone
Hello,
I have a system with two cards: a HFC-PCI ISDN and a TDM21B (2 FXO and 1
FXS), running Asterisk 1.2.4-BRIstuffed-0.3.0-PRE-1l with freePBX beta5
dialplan.
I have connected an analog phone to TDM FXS port, but when I pickup the
phone to make a call, Asterisk "hangs up" the call. Let me explain:
In another system, when I pickup the phone, Asterisk give me tone to dial:
>---
2007 Oct 31
0
Problem with flash hook
Hi,
I facing a problem with flash hook. When ever I do a flash hook to place an
extsing call on hold, the call gets disconnected. The debugs on Asterisk
shows that 'on hook event detected' when I press the flash button on the
phone. The setup is like this
Asterisk box with T1 cards and FXS cards. The T1 card is connected to an IAD
and configured for ISDN PRI lines. Analog phones come
2006 Jan 08
1
spandsp, rxfax, TDM400/zaptel, missed frames, any help?
Sorry in advance if this is a FAQ...
I've got a working Asterisk setup based on A@H 2.2. I have a TDM400
card with 2 FXS and 2 FXO ports; PSTN connections come in via the TDM
card.
I haven't been able to get inbound fax with spandsp and rxfax to work.
Occasionally an all-text fax will come in, though it's usually badly
corrupted, but in most cases, it would appear that the call is