Displaying 20 results from an estimated 1000 matches similar to: "FW: Re: Delay on Phone ringing"
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 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
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
2005 Aug 04
1
no ring to callers?
OK, i've got asterisk @ home 1.3 up and running with Broadvoice.
BUT I have one nagging problem to sort out. When you call my BV # the
calling party gets no ring indication, just silence until either I
answer the phone, or the call bounces over to voicemail. below is the
console output when a call is recieved. what am i missing here?
thanks
Bernie
-- Executing
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=
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
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:
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
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
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
2006 Jan 27
0
Digium Wildcard TDM400P call pickup timing
I have an analogue trunk to an AT&T Definity.
It has a DISA context defined.
From a Definity handset call the analogue port extension 1008 and wait
for dial tone from asterisk. It takes between 3&4 rings.
Likewise from Asterisk SIP handset <PBX Access No><PBX Extn> takes
nearly 10 secs to ring.
Is this configurable?
Ian Cowley
-----Original Message-----
From:
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 Aug 02
12
WHat does it take
How many times do you ask for help here before getting a respone? Every
single thing I do No matter what I get busy extensions. I am willing to pay
someone to help here. Anybody got a clue?
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050802/d0d1326c/attachment.htm
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 Feb 06
12
Cisco 2620 as PRI gateway
I just inherited a Cisco 2621 with a VWIC-1MFT-T1 card in it. Can I make
this thing into MGCP gateway or even a SIP gateway for asterisk? Seems like
it should bee useful for something!
I'm perfectly happy to do my homework, but also don't feel thee need to
reinvent the wheel! So, links with relevant info would be appreciated. If
there is a config for a 2621 being used as a gateway
2006 Mar 10
0
Voice Mail woe
Hi
i have installed AAH 2.6 and configured some extensions
the calls are working fine. but if i dont answer a call then
it says " the person at extension " and hangs up .
it doesnt spell out the extesion number nor it goes to voice mail box.
*************************** Asterisk CLI log ****************************
dialparties.agi: Extension 200 is available...skipping checks
--
2005 Feb 28
2
Fax Failing
Hello All,
I am trying to set up faxing using Asterisk@home 0.6. I have followed
the instructions to the best of my knowledge. When a fax comes in, the
system seems to detect OK but does ot manage to make the fax to pdf to
email leap. Here is what I saw in the CLI when I tested. Any help
would be appreciated.
Thanks!
Wiley
-- Starting simple switch on 'Zap/2-1'
-- Executing