Displaying 20 results from an estimated 2000 matches similar to: "RE: dialing out"
2004 Aug 17
0
RE: RE: dialing out
Nevermind. Figured this out. I needed the following in extensions.conf
to enable outbound dial.
exten => _9.,1,Dial(Zap/2/${EXTEN:1},70,Tt)
Thanks
-----Original Message-----
From: Info [mailto:info@psgsite.com]
Sent: Tuesday, August 17, 2004 9:27 AM
To: 'asterisk-users@lists.digium.com'
Subject: RE: dialing out
Thanks to Greg Hill for pointing me to the 'sip debug on'
2003 Nov 06
2
Dialing an outside number -- QUESTION --
Hello--
I'd like to do a little processing on external phone numbers from within
the asterisk pbx. Fairly simple stuff, but... devilishly hard to make it
work so far!
1. I'd like to dial 9 to get an outside line.
2. If the number dialed after the 9 is 754XXXX, I'd like it to go thru
unmodified. It's the only local number available here.
3. I'd like all 1 XXX XXX XXXX numbers
2003 Jun 10
1
Slow Faxing
I currently have two fax machines on my system.
Both of them seem to send and receive very slowly. My end users
are complaining; saying it was faster before we moved to * (Straight
Analog Lines)
Any help would be great.
PS: I already have the d option on the Dial line.
Both fax machines are in their own context:
[faxes]
exten => _9NXXXXXX,1,StripMSD,1
exten =>
2003 Mar 21
8
Help with linejack as a trunk?
I have a linejack and a phone jack in my asterisk server working well
between the SIP phones and the phonejack. what I cannot get to work is
the outbound linejack Phone/phone0 trunk line? how can I get a SIP or
Phone/phone1 phonejack phone to dial 9 then outside number and pickup
Phone/phone0 and dial it? right now it accepts a 95551212 but busy's on
the last digit 2. no outside dial.
2003 Jul 08
5
Using multiple iconnecthere accounts
Has anybody out there tried to use two different iconnecthere accounts
with Asterisk?
What I want to do is use a second account if the first is busy.
I have tried the following:
exten=>_91NXXNXXXXXX,1,StripMSD,1
exten=>_1NXXNXXXXXX,2,Dial,SIP/BYEXTENSION@iconnect ;iconnect is the
first account
exten=>_1NXXNXXXXXX,3,Dial,SIP/BYEXTENSION@iconnect2 ;iconnect2 is
the second account
But that
2003 Dec 20
2
BYEXTENSION and DBPut
Hey I need another pair of eyes on this!
I would like to add phones numbers to the blacklist from any handset so I
did this:
exten => _*66XXXXXXXXXX,1,StripMSD,3
exten => _XXXXXXXXXX,2,DBPut,blacklist/BYEXTENSION/1
exten => _XXXXXXXXXX,3,Hangup
However what I get in the database is:
/blacklist/BYEXTENSION : 1
And BYEXTENSION is not replaced with the actual number
2003 Jun 18
2
== Everyone is busy at this time problem
hi,
i installed asterisk and works very well, the only problem is that
when i try to call a direct number of a company that has a normal PBX
i got this error:
to 10.8.210.153:5060
== Accepting call on 'SIP/a.sampietro-f7be' (a.sampietro)
-- Executing Goto("SIP/a.sampietro-f7be", "doisdn|BYEXTENSION|1") in new stack
-- Goto (doisdn,00115601992,1)
--
2003 Apr 09
0
can't use both controllers...
hi
when two calls are active on controller 2, chan_capi won't use controller 1.
this is with AVM C2
roy
-- Executing Goto("SIP/torgeir-b476", "capiring|BYEXTENSION|1") in new
stack
-- Goto (capiring,90044875,1)
-- Executing Dial("SIP/torgeir-b476",
"CAPI/22545066:bBYEXTENSION|120|Ttr") in new stack
== data = 22545066:b90044875
==
2003 May 06
2
capi + bri ?
Hello,
I have som problems with my BRI/capi setup. I manage to call in to the system (some rows below).
----------------
-- Executing Dial("CAPI[contr1/16453]", "SIP/BYEXTENSION@janm|10") in new stack
-- Called s@janm
-- SIP/janm-63f5 is ringing
-- SIP/janm-63f5 is ringing
-- SIP/janm-63f5 is ringing
----------------
But I can't make outgoing calls from
2005 Mar 26
1
Dialout handler with/without leading 1
If this handles the case where 10 digits are required:
exten => _9NXXXXXXXXX,1,StripMSD,1
exten => _NXXXXXXXXX,2,Dial,Zap/4/BYEXTENSION
How do you create a handler which works for either this or
the case with a leading '1' plus 10 digits?
tnx
-kim
--
w8hdkim@gmail.com
2003 Sep 20
0
Asterisk with Samsung SKP 816H PBX !
Hi,
Having Asterisk-0.4.0 with 2FXO port and Samsung SKP 816H PBX in 2 offices.
I am able to make call between two offices. But the problem is that call
dosen't hangup.
Office A [Asterisk+2FXO+SamsungPBX] <------------- I A X ------------>
Office B [Asterisk+2FXO+SamsungPBX]
Configuration files are given here..............
------------------------
zapata.conf
2004 Apr 19
1
capi_request: didn't find capi device with outgoing msn =
Hi,
I can't make outgoing calls with CAPI (passive ISDN Fritz card). See
Asterisk error below.
Incoming calls and SIP to SIP calls do work. It looks like a msn
mismatch in extensions.conf
and capi.conf, but I can't find it.
Can anyone help me find the problem?
Thanks,
Rob
*CLI>
-- Executing Dial("SIP/8112-1be9", "CAPI/356666666:BYEXTENSION") in
new stack
2003 Sep 03
1
SIP to PSTN gateway
Hello all,
taking examples from various pointers, I am attempting to put together an outbound dialing example using SIP (Cisco 7960) with 2 X100P. Everything seems to be working without generating errors, but the problem is the phone hangs up (102/Bye). Any pointers/advice are much appreciated
Here is the section in extensions.conf:
extensions.conf
; From CISCO at work
;
exten =>
2003 Nov 19
1
FXO card still won't pick up...
I recently updated (fresh checkout) to the newest zaptel and Asterisk.
The one I was using before was a couple of months old.
After updating, my zap channels don't work. They won't pick up incoming
calls or dial out. When I try to dial out I get:
-- Executing Dial("SIP/3064-564c", "Zap/g1/ww954.......") in new stack
NOTICE[245776]: File app_dial.c, Line 698
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
2003 Jul 25
3
chan_capi error
hello,
sometimes my capi_channel stop works - e.g. when i try to call number
which does not exist ( typo error ) and i must restart asterisk.
following lines appears in the log files :
ERROR[393234]: File chan_capi.c, Line 1050 (capi_request): no free
channel on controller 1! will continue searching.
ERROR[393234]: File chan_capi.c, Line 1050 (capi_request): no free b channel on controller 1!
2004 Apr 05
3
ZAP channels
I have made bri-stuff.0.0.2rc19 to work (I think) but I can not achieve
any in-dialing nor I can dial out;
this is what I have from "pri intense debug span 1" command
----------
*CLI> pri intense debug span 1
Enabled EXTENSIVE debugging on span 1
-- Executing Playback("SIP/201-a862", "tt-weasels") in new stack
-- Playing 'tt-weasels' (language
2005 Jan 18
0
RE: mgcp <-> h323 problem
Hello,
I try to place a call between CISCO IP phone 7905 (h323) and an analogue
telephone on DLINK DG-104S (mgcp).
Call setup is handled ok - both way, but after connection of the call I
get both site no-audio, even codec are
Set to alow-64 both.
The call is terminated with this debug output - Didn't get a frame from
channel: MGCP/aaln/1@dlink-1 and is
Hanged up.
Jan 18 20:32:55
2003 Jul 07
1
overlap dialing on a pri span
Hi,
I am lost trying to figure out how to enable overlap dialing for calls
coming in and coing out on a pri span. DISA looked promising at first,
but does not seem to support overlap dialing. Just picking up a call by
and trying to dial out does not seem the way to do it either. I tried:
[dialincontext]
exten => 12341234,1,Goto(dialoutcontext,s,1)
[dialoutcontext]
exten => s,1,Wait,1
2005 Aug 05
0
call outside from FXS through FXO
Hi,
I am trying to make an outbound call from phone attached to FXS port.
My telephone (VoIP) line is connected to FXO port (Zap/4)
Default context for channel # 4 is 'directdial'
here is part of my extension.conf
[directdial]
ignorepat => 9
exten => 9,1,Dial,Zap/4/
exten => 9,2,Congestion
include => international
[international]
ignorepat => 9
exten =>