Displaying 20 results from an estimated 7000 matches similar to: "PPP dialout via * server"
2005 Feb 27
1
dialout with PPP on ISDN to an ISP
Hello my name is Ilija Poznic and I have a problem.
My configuration is
1. Digium TDM4000P with one FXS.
2. AVM Fritz ISDN adapter (configured with capi).
When I connect to my ISP and then start *. Asterisks is registering me to SIP
provider iconnect. After that I can call international call trough VoIP.
My problem is that I want to dialout to ISP only when I have a international
call.
2009 Jun 06
1
Teliax: where's the space in CALLERID(num) from?
I'm having trouble setting callerid with teliax. I use a simple dial-out
subroutine to set the callerid depending on the calling extension, and
then dial out. Teliax is saying they're not seeing any callerid info.
[DialOut] ; subroutine for dialing out.
exten => s,1,NoOp(Context: DialOut called with outgoing number ${ARG1} )
exten => s,n,NoOp(${CALLERID(num)}XXXX)
exten =>
2004 Nov 24
4
zap fxo hangs after upgrade to stable v1-0
so i have been running v1-0 on all of my test boxes for about a month now
testing iax/sip/res_xxx. I decided to put it into production so I updated a
box that was running 0.9.? that had been working perfectly for months and
low and behold the inbound line from telco now intermittantly doesn't clear
and none of the other channels can dial out on that line. I have tested the
line in this
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
2009 Aug 08
1
bad(?) TDM400 card?
I just had to move all my disk drives and PCI cards, including the
Digium TDM400P, into a new box when the motherboard died. Now that the
new box is up and running, my FXS ports no longer work although the FXO
port does. I am hoping someone can help me diagnose and debug. I am
running Fedora 8, asterisk asterisk-1.4.21, and zaptel-1.4.9 . Yes, I
realize that this is old software and that zaptel
2009 Mar 17
0
Weird issue with outbound calls and MOH
Hi,
We have a PRI Trunk (physical E1) and we are getting
some rather weird and very isolocated problems. On outbound calls to
specific numbers, it would seem to me that DTMF from the remote side is
affecting the local asterisk system. Basically what happens:
- We make a OUTBOUND call via the PSTN (PRI Trunk) to a remote System
- Remote Answers, and converse
- Remote sends DTMF on their site to
2007 Mar 21
5
automated dialout detect forward
Hi!
I have an automated dialout via a call file to a mobile.
Can I detect when the call is not answered but forwarded to the mobile
operator voicebox?
I would like to stop the dialout if this is the case.
TIA,
Mike
2005 Sep 21
1
Does Asterisk know if the trunks are busy?
I am planning on dabbling with some VOIP providers.
I was thinking of Teliax first.
My thinking is that the first LD call would go to teliax and the second
(etc.) calls would go out to the PSTN.
I could then verify bandwidth and quality to decide when to add more trunks
and to Internet connections.
I have been doing some concept testing with FWD for toll free calls, but I
am using 393 as a
2003 Jul 02
1
Dialout Lines ???
I've been reading the Linejack strikes again messages, and have another Newbie question
is it possible to use a Voip Product as a Dialout line for * ?
I have a Vegastream 100 Voip to PRI. box. With * can I use that as a Dialout / dialin box?
The Vega100 does either sip or h.323.
Thanks.
Bradley Greep
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?
2003 Oct 06
2
ISDN Dialout
Hi,
I am having some trouble with ISDN Dialout. Using a Netjet-s PCI Card.
When in Minicom, the only way I can dialout is if i issue ATS18=1 First.
Otherwise I get a BUSY message. So thats fine.
But when I dialout from asterisk, I get an immediate hangup, so my guess is
that asterisk is not issuing ATS18=1 to the ttyI device.
Here are my configs, any input would be greatly appriciated.
2007 Jan 29
0
Dropped call issue with IAX Trunking
Trixbox 2.2 Beta with freePBX 2.2.0rc1
I have a setup that looks something like this in ASCII art:
Teliax IAX Trunk ------+
|
V
Embarq PRI ----> Tandem switch ----> Ottawa Office Server------+
+--------------> Lima Office Server -----+|
2004 Apr 01
1
dialout with chan_capi
Hi,
When I try to dialout over chan_capi everything works fine
when I settle for
msn=* in my capi.conf and use the primary msn of my ISDN-line.
But trying to configure a different MSN the chan_capi doesn't dial
and comes with:
No one is available to answer at this time
What can be the prob?
--
Thanks,
Marc aka IzNoGood
2004 May 18
1
Linejack dialout
Dear all
I read on the list back in 2003 that * does not support IXJ LineJACK
dialout yet
is this still the case?
Thanks
2004 Jun 29
4
Getting Asterisk to automatically dialout
Hi,
I'm trying to get asterisk to auto-dail out. I created a *.call file
with the the top of it being "Channel: Zap/1/2609944", which should have
connected to Zap channel 1 and dial out to 2609944, but It did not do
so, asterisk would say a call was completed to Zap/1/2609944 but I never
heard that phone ring. So I tried just putting "Channel: Zap/1" at the
top of
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
2004 Nov 26
0
TDM22B - how to setup the extensions ??
I got this nice TDM22B with two green modules left and two modules right
/var/log/messages shows:
Nov 26 23:47:48 dns kernel: Zapata Telephony Interface Registered on major 196
Nov 27 00:37:53 dns kernel: Freshmaker version: 71
Nov 27 00:37:56 dns kernel: Freshmaker passed register test
Nov 27 00:37:56 dns kernel: Module 0: Installed -- AUTO FXS/DPO
Nov 27 00:37:56 dns kernel: Module 1: Installed
2004 Nov 27
0
Zapata: No such device or address
I got this nice TDM22B with two green modules left and two modules right
/var/log/messages shows:
Nov 26 23:47:48 dns kernel: Zapata Telephony Interface Registered on
major 196
Nov 27 00:37:53 dns kernel: Freshmaker version: 71
Nov 27 00:37:56 dns kernel: Freshmaker passed register test
Nov 27 00:37:56 dns kernel: Module 0: Installed -- AUTO FXS/DPO
Nov 27 00:37:56 dns kernel: Module 1:
2003 Sep 11
3
PROBLEM RECIVING CALLS AT FXO
Hi...
I have the next problem.. I have a FXO card with i can make calls but i cant
recive calls.
At the consol, i get the next error:
-- Zap/2-1 is ringing
-- Zap/2-1 is ringing
-- Zap/2-1 answered Zap/1-1
-- Attempting native bridge of Zap/1-1 and Zap/2-1
WARNING[262160]: File chan_zap.c, Line 2857 (zt_handle_event): Ring/Off-hook
in strange state 6 on channel
2008 Oct 20
1
Zaptel FXO offhook when connected to PSTN
I installed Trixbox and a TDM400P with 2 FXO and 2 FXS ports and am having
an annoying issue with the FXO ports. As soon as I plug either one into the
phone line it's as though the line is disconnected i.e. get disconnected
tone when trying to dial out, line is busy when dialling in.
The CLI shows the following:
trixbox1*CLI> zap show channel 4
Channel: 4
File Descriptor: 18
Span: 11*