Displaying 20 results from an estimated 10000 matches similar to: "dialparties.agi is returning no extensions to dial"
2006 Feb 28
1
FW: Re: Delay on Phone ringing
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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'
--
2006 Mar 30
3
asterisk doesn't wait for whole extension
Hi,
maybe a dumb question, but it seems that some calls are directed to our
central dial in number despite the extensions the callers say they dialled.
E.g. they dial 1234-567, asterisk recognizes 12345, it says this is an unknown
extension, where it is right, and redirects the call to the central dial in
extension 1234-0. This only seems to happen when the numbers are dialled
manually. When
2006 Jan 31
5
Queue() with timeout=0
Hello,
i've recently switched over from 1.0.9 to 1.2.3.
I've experienced some (to me) weird behaviour.
This is the config for an example queue.conf:
[654]
wrapuptime=30
timeout=20
strategy=ringall
retry=5
queue-youarenext=queue-youarenext
queue-thereare=queue-thereare
queue-thankyou=queue-thankyou
queue-callswaiting=queue-callswaiting
music=default
monitor-join=yes
monitor-format=
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
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
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
2006 Feb 27
3
Asterisk with HT 488 FXO
Hi, i have a HT 488 and I want using this like an FXO for Asterisk.
I have find some configuration in the list archive & google but my HT
with these config not work.
my sip.conf
[HT-488]
username=400
type=peer
secret=wowowow
qualify=yes
port=5062
nat=no
host=192.168.1.157
fromuser=400
disallow=all
context=from-pstn
allow=g711u
allow=ulaw
allow=alaw
my sip debug:
2004 Feb 17
5
chan_capi problem
Hi to all
I've mada up my mind and i tried to change from i4l to chan_capi, following some councelling from the gurus.
I compiled it up, and when i try to load it in modules.conf, i get that wonderful message and Asterisk does not start:
[chan_capi.so]Feb 17 09:21:40 WARNING[16384]: loader.c:239 ast_load_resource: /usr/lib/asterisk/modules/chan_capi.so: undefined symbol: ast_get_group
Feb
2006 Jan 27
5
External IAX2 phone defined as internal behaving as from PSTN
Have asterisk@home 1.2.1
The server is on an internal network eg 10.10.10.10
It is NAT'd 1:1 via Checkpoint firewall to external public IP eg
50.50.50.50
The remote IAX2 phone (ATCOM320) is configured to call 50.50.50.50 on
extension 1055.
Outbound calls to 1055 work perfectly.
Inbound calls from 1055 get picked up as if it were an external call
(see below) and goes straight to the ring
2006 Mar 31
4
cannot set outgoing cid
Hi,
sorry for the long debug output below. I configured Asterisk with AMP to send
the whole number including the extensions of the callers to the called party.
Whatever I configure in AMP it looks like it is used, In my eyes it is ok, but
doesn't seem to work.
033811234451 is the call id i configured, and it seems to use them, but the
caller will only see a 0338189040 instead of my
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
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
2007 May 15
1
Asterisk is not showing the correct Incomming CallerID
Hi Everyone,
I have an asterisk box in my office. It does not display the correct Incomming Caller id.
For incomming we are using ISDN Bri line which is terminated in a Digium 4 port bri card (B410P).
Like if a number say 02 12345678 calls to our line asterisk displays it 12 12345678.
Similarlay if a mobile number say 0416 123456 dials us , asterisk displays 1416 123456.
I am not sure where the
2007 Feb 02
0
Call Waiting broken on ZAP
Problem: *Call* *waiting* comes in, I press flash to answer it, and the
first caller gets disconnected after 3 seconds. This is all ZAP - no VOIP.
System:
Analog stations and trunks running on a pair of TDM400's. It does NOT have *
call* *waiting* from the phone company, and I have enabled it in all my conf
files. The trunks are set to FXSKS and the stations are FXOKS. I am not
using *call*
2006 Feb 08
1
incoming call release after 1 ring
Hello,
Can somebody please assist me with my problem.
Currently I am using a Asterisk@HOme version 2.4 with
a TE406P digium card. One the E1 is connected to a
telco switch via an ISDN. May issue is that may
incoming calls in the zap channels gets disconnected
or release after 1 ring. I really dont know what
setting should I change to increase the timeout of the
ring. I have even tried upgrading
2005 Feb 11
2
transferring a IAX call into a conference
When I make a call out on the Faktortel number I am then able to
transfer to call to my asterisk meetme room of 801 by hitting 'transfer'
then '801' then 'send' on my grandstream phone.
This connects my faktortel trunk (and who ever is on the other end) to
my conference room I can then make another call using my local pstn
service and set up a 3 way (or whatever number
2005 Jul 11
1
ASterisk@home + Broadvoice = Almost working installation...
Hello Guys,
I'm somewhat of a newbie and am desperately seeking for some help...
I've managed to get asterisk up and running on my server, and signed up with a
broadvoice account...
I'm having no problem dialing and communicating between extensions, but whenever
anyone tries to call my broadvoice account, they are greeted by no ring or
anything, but rather simply a direct to
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.
2005 Jan 17
1
Attempting native bridge
ERROR CONDITION
---------------
-- Executing Dial("SIP/2001-f6c4", "SIP/2000|20") in new stack
-- Called 2000
-- SIP/2000-0ead is ringing
-- SIP/2000-0ead answered SIP/2001-f6c4
-- Attempting native bridge of SIP/2001-f6c4 and SIP/2000-0ead
Have searched web and archive w/o good results.
Thks in advance for any help,
Dave
sip.conf
--------
[general]
port =
2007 Mar 26
1
Asterisk incoming caller id problem
Hi, guys,
For my server, if i use my handphone to call in the PSTN line by TDM400p
card, the server could not receive the caller id correctly. anyone knows the
problem? I am currently using asterisk 1.2.14 with freepbx 2.2.1. The CLI is
as below, "Caller ID name is 'zap1' number is '4521'" , this 4521 is one of
my FXS zap extension created.
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