Displaying 20 results from an estimated 4000 matches similar to: "Macro: s-NOANSWER, _s-."
2007 Aug 27
4
Prepaid Billing: A2Billing, AstBill, ASTCC
Hi List;
I need to use an prepaid billing system with Asterisk,
and I do not know which one is more stable and
integrated with Asterisk?
A2Billing or AstBill or ASTCC?
Also, from where I can download it and ready about its
configuration?
Regards
ITS
IP Telephony and Contact Center Engineer
Eng. Bilal Ghayad
Mobile: 009659849460
2007 Sep 09
3
canreinvite
Hi List;
If I need traffic to be directly between the
endpoints, then I have to set the canreinvite = yes?
If I did not configure the canrenvite at all, then by
default it will pass the traffic via Asterisk and not
directly between the endpoints?
What if one endpoint was SIP and configured with
canreinvite=yes while other endpoint was IAX2 and
configured with canreinvite=yes, then they can send
2009 Jul 17
2
How do I create an IVR/Dial Group that works properly?
Hi all,
I am trying to understand how I can get a simple IVR scenario to work
properly (having already removed most of my hair...).
The basic requirement is as follows:
* Caller arrives at our main number
* Caller is greeted and then told they can enter an extension number, if
known, or wait and their call will be connected to an available rep.
* The IVR then dials a group of extensions (if
2007 Jul 23
2
Upgrade and keep the configuration
Hi List;
How to upgrade the Asterisk, Zaptel and LibPri and
keep the configuration the same? I do not need to
remove current asterisk, zaptel and libpri and
download new one and write new configuration.
Regards,
--------------
ITS
IP Telephony and Contact Center Engineer
Eng. Bilal Ghayad
Mobile: 00965 9849460
2007 Nov 29
2
Using existing extensions.conf macros, and co-habitation
I'm trying to set up my extensions.conf file using some of the existing
macros like stdexten, etc. while at the same time having two logically
separate virtual PBX's (with no "default" context) and two trunks coming
into separate contexts, i.e. one for residence and one for my at-home
business.
I noticed, however, that macro-stdexten depends on the "default" context:
2007 Aug 02
1
H.323
Hi List;
Did any one tried the H.323 module? How much it is
stable and work fine?
Regards,
------------
ITS
IP Telephony and Contact Center Engineer
Eng. Bilal Ghayad
Mobile: 00965 9849460
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2007 Jan 19
1
Red: Sip Phone CID
Here is what I have in my extensions.conf file now. Trustrcid and
sendrcid are turned to "yes" in the conf file.
I'm not fully sure how the SIPCalledRPID works though. The example I
found seems to try and provide the stuff automatically (id and name),
but does the SIPPEER stuff even exist?
I think this is probably the right track though. Any insight would be
much appreciated.
2004 Dec 20
1
Example config for SPA-1001
Hi,
Has anyone managed to create a setup with a Sipura SPA-1001 as a client?
Right now I can connect to the device by dialing the extension number
but when I try to connect from the phone handset to make an outbound
call it gives an unavailable tone.
I'm using Line 2 on the SPA-1001 to connect to the local asterisk
server, line 1 is used to connect to my VOIP provider until I can get
the
2007 Aug 02
4
Receiving SIP calls without registeration and dynamic IP address
Hi List;
How can I configure asterisk to receive a call from
SIP end point without being registered at asterisk and
its IP address is dynamic, and authentication to be
based on the username and password or any other
string?
I know that if I place the host with static IP then no
need to register, but what if the voip gateway was
having dynamic IP and I do not need to register on
asterisk, but I
2006 Jan 06
1
Problem with integrating ISDN PBX using NT mode
Hi,
I'm just in the process of replacing a crappy Siemens PBX with a new and
shiny Asterisk system. To connect Legacy equipment I hooked up a small
ISDN PBX (DeTeWe OpenCom 36) to one port on a Junghanns.net quadBRI
card. That port is configured for NT Point to Multipoint
(Mehrgeraeteanschluss) mode. Now I can place calls from the ISDN PBX to
the other Asterisk extensions but the other way
2005 Jan 18
1
Asterisk and IAX softphone (firefly) problem/question
Quick question from a newbie,
I have asterisk configured to dial IAX extensions (which works). When
dialing from one IAX extension (using Firefly) to another IAX extension
(also using Firefly), the Firefly client rings on the receiving end and
gives the option of accepting or denying the call. However, when I dial in
to Asterix using a VoicePulse number and dial the same extension Firefly
2008 Mar 13
5
Newbie One-touch Recording: Does not work
I thought it was quite easy to implement but I cannot get one-touch
recording to work. Here are the changes what I did:
I restarted Asterisk after the change (because reload does not work for
changes in features.conf).
I press *1 on the Polycom IP600 phone to record a conversation but no
new wav file appear in /var/spool/asterisk/monitor or elsewhere.
Any suggestions?
Here is the console log:
2005 Mar 03
3
Why ${EXTEN} variable changes after Goto ?
Hi,
I'm trying to implement dynamic routing of incoming calls to local extension
if previous outgoing call was unanswered.
But after I do Goto to s-NOANSWER, variable ${EXTEN} changes to
's-NOANSWER'. I guess this is normal, but I don't understand why ? How to
workaround on this one ?
Thanks in advance,
regards,
Rob.
[outbound-capi-ISDN]
exten => _0.,1,NoOp(Calling ISDN
2009 Feb 12
4
Asterisk Queue and URL Calling
Dear All
I want to integrate sugarcrm and asterisk , so when customer call the call
center the agent or extension which answers the call , before pickup the
phone and talk to customer , view his/her information if it is available.
I do this as described below :
1-Setup login username for sugarcrm for each extension
2-Extension Users will login to the sugarcrm.
3-Develop php script to handle new
2007 Aug 23
3
Asterisk Prompt
Hi List;
I read the following sentence:
"The CLI prompt is set with the ASTERISK_PROMPT UNIX
environment variable"
In the following link:
http://www.voip-info.org/wiki/index.php
page=Asterisk+CLI+prompt
The question is: what is the ASTERISK_PROMPT UNIX
environment variable and where I can access it to
change it? Also where I can find information about it?
Regards
Bilal Ghayad
2006 Jun 19
2
Asterisk voicemail problem with isdn avm fritz!card
Hello everyone,
I have Asterisk SVN-trunk-r7498 installed on a server (celeron 2.4 Ghz,
256MB) with a TDM40b a TDM04b and an avm fritz!card pci.
I experience a problem with voicemail: my messages are good unless the
incoming call comes from isdn, which means via the avm fritz!card. In
this case (and in this case only) the message is disjointed and I can
hear at most 1 second out of a 1 minute
2007 May 01
10
Digital Phones
Hi List;
Asterisk does not have any kind of cards that can work
with it to be used with Digital Phones (digital phones
differ than analoge phone and differ than IP Phones).
Anyone can advise about this as I did not find this on
Diguim
Regards
Bilal Ghayad
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2007 Jul 01
4
Not able to find the file zaptel.conf after compiling asterisk and zaptel
Hi List;
I compiled Zaptel 1.4 and Asterisk 1.4 after
downloading them using svn, but when I checked the
file zaptel.conf under etc/asterisk, I did not find
this file. Any help?
By the way: How can I know the asterisk and zaptel
version extactly that I compiled them? In other words,
asterisk 1.4.... and zaptel 1.4.... ?
Regards
-------------
ITS
IP Telephony and Contact Center Engineer
Eng.
2005 May 16
1
Dial plan - does not stop after first match
My dial plan seems to work great - in that when I call extensions 1234
it connects to 1234. Strangely, after the call terminates (the other
side hangs up first), Asterisk continues in the same context and then
matches to extensions _. which causes an invalid extension error!
Why does asterisk not leave the context (called internalmenu) after the
remote hangup? Instead, it continues to the
2004 Feb 27
6
Video Conference
Is Asterisk capable of handling video conference? I am wondering if there is anybody in the list who tried it with NetMeeting(s). If it is possible, is the * required to register in the GK for this purpose? or making it as h323gw only is enough.
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