Displaying 20 results from an estimated 2000 matches similar to: "Zap channel configuration problem"
2005 Jun 30
3
AMP - recording call
Hi,
I'm using the new AMP which provides a call recording. The options of
recording call Always and Never are well working.
But how to use the On-Demand option ? Should I press a pad ? Is this
configured in the featuremap of features.conf ? Why my modifications in
that features.conf have no effects ?
Please advice me.
Alexis.
2004 Feb 03
2
Dialling Hook Flash on Zaptel
Hi,
I'm trying to get my X100P to Dial the following sequence to gain access to
speed dial numbers on my Norstar PBX that the X100 is connected to...
[FLASH] [*] [0] [22] (where 22 is the speed dial number)
But so far I've had no luck, with the following extension:-
exten => 922,1,Flash(${DIALOUTANALOG})
exten => 922,2,Dial(${DIALOUTANALOG}/*022)
exten => 922,3,Congestion
2005 Jun 14
2
Features.conf for secretary function
Hi,
I am trying to use the attended transfer. So I put this in my feature.conf:
[general]
[featuremap]
atxfer => *0
blindxfer => #0
I completly restart asterik, and not just make a RELOAD. But during a
call, when I press # it runs a blind transfer and if I press * I am
disconnected.
I am using the CVS version of * get as explain here
2003 May 23
12
Unable to create channel of type 'Zap'
I've just installed an X100P, built the kernel module, and tried to use it
to make an outgoing call (via a phone connected to an ATA-186). However, I
just get a reorder tone and see this on the console:
-- Executing Dial("SIP/ata1-4409", "Zap/1/5551212") in new stack
NOTICE[1200825920]: File app_dial.c, Line 481 (dial_exec): Unable to
create channel of type
2004 Dec 22
2
Can't Receive/Send Calls
Hi,
I can't receive/send calls with Asterisk. Could someone please give me a
few pointers on my configuration?
Regards,
Norman Zhang
; sip.conf
[general]
disallow=all
allow=ulaw
port=5060
bindaddr=0.0.0.0
externip=x.x.x.x
localnet=192.168.22.0
mask=255.255.255.0
context=inbound-sip
maxexpirey=180
defaultexpirey=160
tos=reliability
srvlookup=yes
register =>
2004 Sep 14
1
Requested device 'ttyI1' does not exist
Hello List!
I finally got asterisk with capi working, and its already answering my
call as well! :)
Now i would like to call a number from my shoft phone (kphone).
This is my extentions.conf:
---
[general]
static=yes
writeprotect=no
[globals]
CONSOLE=Console/dsp ; Console interface for demo
IAXINFO=guest ; IAXtel username/password
2004 Aug 20
3
Strange problem with Dial
I'm trying to add an emergency dial to my context. However, when I try to
dial it, I get caught in an endless loop.
For debugging, I have pared out nearly all the control flow and just have
ChanIsAvail() and Dial() called. Using two different extensions to call teh
same number, I get two different actions by *.
Here is the vvverbose output:
-- Starting simple switch on
2003 Apr 19
0
Unexpected behavior of X100P and * in no-dialtone situations
I have some strange behavior happening with call flow when analog
line errors are encountered. This may be due to the way that the
X100P detects "busy" signals, or it may be something in the software.
Could someone with more in-depth knowledge make a comment on the
items below?
My dialing logic says "dial local area code numbers out of the analog
line, and if the analog line
2004 Jan 29
4
dialing wrong numbers
hi,
I am new to * and setting up a test system.
here my setup :
- debian (from knoppix 3.3)
- Asterisk 0.7.1 (from the debian package)
- AVM Fritz card used with i4l
- softphone I use for testing SJphone on windows
- I can make great softphone - softphone calls
- I can call from an outside line * and get connected to a softphone
here my problem:
I can not make outbound calls. I place a call
2004 Aug 10
11
CAPI call transfer
Hi,
I am having trouble configuring CAPI so that call transfers work.
I make a SIP call to asterisk which goes out on ISDn via CAPI. Then
I
try to do a transfer from the SIP phone which doesn't work and
results
in the call being disconnected.
The error message given by asterisk is that it chan_capi can't find
an
entry for the outgoing msn for the transfer however the outgoing msn
is the
2004 Aug 06
3
E1 monochannel :-(
Hola!
I'm using asterisk as H.323 -> PRI gateway. First call goes
thru ok, second concurrent call fails with:
Aug 6 11:52:30 DEBUG[737292]: chan_h323.c:1038 setup_incoming_call: Sending to context [ip2pri]
-- Executing Dial("H323/ip$192.168.32.25:60271/984", "Zap/1/9541163107100") in new stack
Aug 6 11:52:30 NOTICE[753677]: app_dial.c:554 dial_exec: Unable to
2004 Jul 18
18
Polycom IP 500 Voicemail
Hello All,
I have some Polycom IP 500 phones that I would like to have configured
for direct dialing to our voice mail system. So far I have been unable
to get the hard button labeled Voice Mail to connect to Asterisk without
first passing through the message center prompts. I have followed all
the Admin Guide instructions regarding the phones .cfg files and using
2004 Mar 31
5
3-4 port FXO card recommendations
*This message was transferred with a trial version of CommuniGate(tm) Pro*
In setting up Asterisk, I'm looking to dump my current phone system (Nortel
Venture). I presently have three POTS lines.
I would use a VOIP provider, but now are presently available in the Toronto, ON,
CANADA area that support user owned hardware/software. I need a 416/647 area
code number.
In looking at FXO cards
2007 Jun 04
2
FX Dialing Odd
Here's a possible bug, or more likely, I'm just missing something.
We have a pots card in one of our asterisk boxes. Its a simple asterisk
setup with one FXO/FXS card and basic static extensions file, etc. When
we dial out over the pots line, 4 out of 5 times, it will work. However,
every 4 or 5 times, we get an error back from the provider that says
"The number you have dialed.....
2004 Apr 20
3
Pattern matching rules for least cost routing
I've got two patterns I want to match on making an outgoing call...
(one day - to do Least Cost Routing for Cell/Mobile calls)
Firstly - I prefer '0' rather than '9' to get an outside line...
Either its a call to a mobile No... (072 -or- 082 -or- 083 -or- 084)
or its just another number to dial...
I added the following... the playback just advises me which 'route' is
2004 Feb 03
2
Pictures of new multiport FXO/FXS from digum
Does the voicetronix card work with Asterisk?
-----Original Message-----
From: asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of
woody+asterisk@solutionsfirst.com.au
Sent: Monday, February 02, 2004 11:06 PM
To: asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users] Pictures of new multiport FXO/FXS from
digum
> -----Original Message-----
2008 Apr 05
2
IAX IP Phone
Hi All;
Till now I am not able to find a good IAX IP Phone or
Gateway that can be used with good quality.
Anyone can advise for good one?
Regards
Bilal
____________________________________________________________________________________
You rock. That's why Blockbuster's offering you one month of Blockbuster Total Access, No Cost.
2004 Apr 09
3
Ignorepat with capi
Hi to all,
I'm trying to make outside call in this way :
ignorepat => 0
exten => _0.,1,Dial(CAPI/xxxxxxx:b${exten})
But the first number 0 is not ignored.
I'm doing something wrong ?
Bye
2004 May 24
2
Newbie extensions.conf I need to include [SMS] context.
I want to include a new context in my exensions.conf
I have read this page http://www.voip-info.org/wiki-Asterisk+howto+dial+plan
and I can sort of follow it?!
I have a context [local] that I know zapata.conf points to, I have edited
extensions.conf and put in my phone, sip and iax extensions. I want to add
an sms context.
I understand that all calls go through my [local] context and I have
2005 Jan 08
3
ASTCC questions
Hello.
I have set up ASTCC properly, calling it like this:
DeadAGI(${ACCOUNTCODE},${EXTEN})
It seems to be working correctly, but I have two questions:
- Although the cards' credit seems to be maintained correctly, I cannot
see the call details in astcc-admin. When I try to view information on
the card, it's just blank. Any ideas?
- When does the 2nd, 3rd and 4th trunk get used? I have