Displaying 20 results from an estimated 10000 matches similar to: "Getting Asterisk to automatically dialout"
2006 Feb 01
3
Dumb Dialout Question
I'm still trying to learn some parts of Asterisk, so sorry in advance for the dumb question!
How do I set up an extension to dial out to the PSTN through my ZAP interfaces? I want the ability to have a ring group that will ring all of the phones in an office and then ring cell phones if nobody answers. I'm sure this is simple to do but I'm at a loss.
I have tried the following
2008 Feb 25
4
TDM400P dialout problem
Using asterisk 1.4.18 and zaptel 1.4.9 on x86_64, I am having trouble dialing
out to the pstn. The call is initiated at Zap/1-1 and should exit via Zap/3.
I get the following:
-- Starting simple switch on 'Zap/1-1'
-- Executing [2111 at internal:1] Dial("Zap/1-1", "Zap/3/8801234") in new stack
[Feb 25 02:36:59] DEBUG[7194]: chan_zap.c:1954 zt_call: Dialing
2006 Oct 25
2
Choice of soundfile format
Hello
What soundfile format, is the one that uses least transcoding during playback?
As I can see, I can choose wav or gsm. What sucks least cpu power, during playback to example a Zap channel? I would guess wav, but is this correct?
Kind Regards
Jon Leren Sch?pzinsky
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2003 Jul 23
1
newbie - simple dialout server
Hello,
I am new to Asterisk, so RTFM answers welcome too (just include the FM's
link :).
I'd like to build a simple dialout server based on Asterisk.
I installed 0.4.0 from package (a Debian SID machine, "server").
The client is gnophone (a Debian SID machine too, "client").
My modem is a GVC 56k voice modem connected to the server's serial port.
I modified
2007 Feb 08
2
Asterisk outbound calling does not wait for answer before playback
Hello Asteriskers, :-)
We're trying to set up an outbound notification calling for system
alerts with Asterisk 1.4.0. We generate a call file in
/var/spool/asterisk/outgoing and the outbound call is originated through
Zap/1 (Sangoma A200D to a Canadian POTS line). The problem is that
Asterisk does not wait for the other side to answer before it starts
playing the message. So the
2007 Jun 22
1
Ring/Off-hook in strange state 6
HI I have two servers both of which get this message on one of the lines.
Ring/Off-hook in strange state 6. The one server seems to be ok with it, but
the other one when an extension picks up there is no one there and the
incoming call keeps ringing. I tried to adjust the levels in wcfxo.c like
someone had suggested, but it didn't do anything. I also upgraded zaptel to
the latest. 1.2.18 and
2005 Mar 22
1
Call file misbehaviour
Greetings *`s,
I am manually creating call files and dropping them into
/var/spool/asterisk/outgoing to be picked up by *.
Presently, when I use local/internal parameters using SIP it works..ie I
make an internal call from device to device.
However, when I try dial an outside number which I have set up in a
custom conf file, it bombs out with the following message :
2007 Nov 05
1
PRI dialout problem with some numbers...
I have an Asterisk server (1.4.13) using PRI in Monterrey, Mexico.
This is really the first server I have used with PRI in Mexico as we
normally use MFC/R2. Everything seems to be working except that some
numbers always seem to be busy when you dial them. All these numbers
belong to different phone companies. I know that with R2 this problem
is present if you have a "#define
2005 Jun 19
2
outgoing call routing
I have a Asterisk @home ver 1.0 running with a TDMB11 card. Several sip
extensions and a regular phone connected to the box. All routing works fine
from the regular phone connected to the box, whether its going to FWD,
broadvoice or the PSTN. The problem I am experiencing comes from making
calls from the sip phones. They get routed correctly to the sip and iax
trunks but when making calls
2005 Jun 15
4
Dial more then 9 digits
Could you kick me, I can't dial more then 9 digits. Is anyone some
default length of extensions or dialed number.
Thanks,
Bob.
2008 Jan 16
4
Unable to open master device '/dev/zap/ctl'
Hi,
I'm using zaptel-1.2.22.1 with asterisk-1.2.10 and following steps to
make zaptel working...
OS is gentoo linux 2006.1
Hardware:
---------
0000:05:01.0 Communication controller: Tiger Jet Network Inc. Tiger3XX
Modem/ISDN interface
Subsystem: Unknown device 8085:0003
Flags: bus master, medium devsel, latency 32, IRQ 22
I/O ports at b400
Memory at ff900000
2008 Oct 06
1
Dial out DAHDI Channel?
I'm attempting to convert from ZAP to DAHDI with 1.6.0.
I was using 1.6.0-beta9.
I followed the directions I could find.
I moved /etc/zapata to /etc/dahdi/system.conf
I moved /etc/asterisk/zapata.conf to /etc/asterisk/chan_dahdi.conf
I don't undestand how to deal with extensions.conf?
I replaced Dial (ZAP/ ...) with Dial (DAHDI/ ... )
All my inbound calls from DAHDI work the same as
2005 Sep 14
2
PRI to PRI passthrough with DID intact
I currently have: Telco-PRI ---- Panasonic DBS576 PBX ---- E&M wink
T1 ---- Asterisk.
I have configured the Panasonic to forward my Asterisk DIDs to the Asterisk
extensions over the T1.
I do not get DID nor CID on the Asterisk, so I want to use PRI between the
PBXs.
I do not want to pay for another PRI card for the Panasonic. (T1 and PRI are
different cards)
I see this as my least
2005 Sep 06
1
/dev/zap* is not showing up (gentoo, portage, asterisk 1.0.8)
hi there
I'm trying to get asterisk going on gentoo 2005.1
I'm just getting my feet wet so I thought I would just stick with the
stable portage packages. Right now that's asterisk 1.0.8
I emerge asterisk with the following make.conf file:
CFLAGS="-O2 -mcpu=i686"
CHOST="i386-pc-linux-gnu"
CXXFLAGS="${CFLAGS}"
USE="-gtk -gnome -qt -kde -dvd alsa
2011 Sep 28
2
PSTN connectivity
Hi All,
I am trying to connect my asterisk box with freepbx to PSTN. I
have purchased x100p FXO card and installed in my asterisk server. My
freepbx detected the x100p FXO card and i can see the card specific details
in command line. I have configured the following things.
1. OUTBOUND caller id and Dialing rules in Freepbx.
2. INBOUND route
When i call to the PSTN number before
2007 Apr 24
3
auto dial out multiple destinations
Hi,
I am searching for the most effective solution for the
following scenario:
Our users can call into our IVR menu and dial a
specific extension and immediately hang up. This event
should simply trigger Asterisk to make multiple
simultaneous calls through a group of zap channels
(5-10 calls). When the called parties answer, Asterisk
should simply play a message and hangup.
So I was thinking
2003 Dec 30
2
playback in [macro-stdexten] problem
I added the playback line to my [macro-stdexten] context but when I dail
an extension I don't get the "please hold while I try that extension"
message. It just dials the extexsion. Do I have a syntax problem
somewhere ?
exten => 8005,1,Macro(stdexten,8005,Zap/2)
exten => 8006,1,Macro(stdexten,8006,Sip/8006)
[macro-stdexten]
;
; Standard extension macro:
; ${ARG1} -
2004 May 25
1
dialout=fromvm
If you set "dialout=fromvm" in your voicemail.conf, how do you then go about being able to dial back out?
Is there a service feature code?
2005 May 12
1
chan_capi and chan_misdn
Could someone please comment on the current state of chan_capi,
chan_misdn and chan_modem channel drivers in terms of functionality and
stability. Specifically, which channel driver would be best for a
passive PCI HFC or W6692 ISDN card. The chan_misdn wiki claims that
chan_capi distinguishes between junghanns and non-junghans cards, and
that chan_misdn is better suited for general misdn
2004 Aug 27
2
Zap & ANSWER the Call
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Hi,
I'm using a TDM400 with one FXS and one FXO module (developer kit) and
I've been testing termination from SIP phones to PSTN and it works fine, but
asterisk accounting is doing something strange (for me).
Scenario:
1 - extension 1009 (SIP phone - BT101)
2 - Zap/4-1 (TDM400 FXO module)
extensions.conf:
[dialout]
exten =>