Displaying 20 results from an estimated 1000 matches similar to: "Debugging Asterisk in GDB (DDD)"
2004 Jul 17
1
Using a group variable for a groupofextension to dial
Actually doing both sounds good to me. Can you explain further about
ringing them all at once?
Here is how I tried to make mine work and failed...
{global}
PHONES0=SIP/2000
PHONES1=SIP/2001
[local]
exten => 6001,1,Dial(${PHONES0&PHONES1),20,trf)
When I dial 6001 I see my debugger tell me that I am using the wrong
syntax.
Do you know the correct syntax for ringing them all at once?
I
2004 Jul 20
10
Installing X100P
I attempted to install an X100P card but it was not correctly recognized
by my Redhat 9 install. I had a test install running without any cards
which was working great minus the outward dialing since no cards
existed. Now that I have a card, I want to add it to the system. Do I
have to scratch the whole current install in order to get the X100P
running on my system or is there a way to get it
2004 Nov 23
4
Spandsp and Asterisk
Does anyone have an update patch file to get Spandsp installed?
I'm running asterisk CVS-HEAD-11/19/04-21:53:37 on redhat 9.0
I installed spandsp-0.0.2
when runnig the patch I get
patching file Makefile
Hunk #1 FAILED at 41.
Hunk #2 FAILED at 69.
2 out of 2 hunks FAILED -- saving rejects to file Makefile.rej
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2004 Jul 17
1
Using a group variable for a group ofextension to dial
That could be it. What I want to do is set a group of callers and have
the event cause the phone to ring them in order. I will tie it to my
IVR portion and thus I can make sure peole in sales get calls based on
our hierarchy in the office. So if I am reading your example right the
syntax is....
Exten => 501,1,Dial(SIP/PHONE1&SIP/PHONE2&SIP/PHONE3), rtf)
Is that a valid way to cause
2004 Aug 18
27
SpanDSP
Anyone knows where can I find spandsp? Official site seems permanently
down...
TIA,
Simone.
2004 Nov 22
8
Patching asterisk for spandsp
When I try to patch the Makefile for asterisk with the
Apps_makefile.patch from Spandsp I get the following error.
patching file Makefile
Hunk #1 FAILED at 47.
Hunk #2 FAILED at 76.
2 out of 2 hunks FAILED
Has anybody seen this.
2004 Jul 04
2
music on hold question with asterisk
hello I'm trying to figure out if anyone's accomplished putting someone on
hold with a hardphone that doesn't have a hold button or multiple lines. I'm
thinking transferring the caller to a specific extension or something...is
this possible? Has it been done?
thanks
hank
2004 Jul 10
2
Looking for a patch that was post May 1 2004
Hello group
I'm working on getting festival installed and working on my FC1. I ran
into a problem and after searching Google I found this message talking
about a patch for Speech Tools and Festival
http://lists.digium.com/pipermail/asterisk-users/2004-May/045134.html
The above site does not have the files.
Does anyone in the group have this patch?
Marc Sutter & Reed Wade do you still
2004 Jul 13
1
Asterisk don't listen to my phones
Hello,
First, sorry for my english. I'm a french student.
I have a problem with asterisk.
I use Budgetone SIP phones.
When I dial 555 (VoicemailMain), I hear "You have 5 new messages,
1- Read your messages, 2- , etc ... )
But when I dial 1 or 2 or everything else, nothing happen.
Are they some lines wich do that asterisk listen my phones ?
Thanks for your help,
have a nice day
Thomas
2004 Jul 17
1
voicemail broadcast feature
Using CVS from 7/12/04 and trying to get the voicemail broadcast feature
to work.
Voicemail.conf has
[mycontext]
3722 => 1234,BroadCast Test,,,cc=*@mycontext
.
then many other voicemail boxes.
-----
whenever I leave voicemail at box 3722, only box 3722 gets the
voicemail. It is not expanding it to other voicemail boxes in the
[mycontext] context.
Even if I replace the cc= line with
2004 Jul 17
1
Using a group variable for a group of extension to dial
I ahve been searching to no avail for a referenc eon how to setup a part
of my dial plan that will ring certain groups of number based upon the
context. Essentually, I want to be able to designate 3 people as sales
and have my IVR handoff and ring their extensions in order. Then maybe
I will ahve a couple of people I group together and have them ring if
someone selects 2 on the IVR for tech
2004 Jul 19
2
callparking vs calltransfer
HI ALL;
Anybody can explain the difference between "call parking " vs "call transfer"
Regards
mohammad
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2004 Jul 19
2
codec translate
HI ALL;
Is astersik enable to translate between different codecs.
I have couple of SIP-UA , one with (a-law) and the other with (g729), registered with my astersik box.Can astersik translate between alaw-g729 and vice varsa.
Regards
mohammad
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2004 Jul 20
3
New CVS version
I yesterday brought up to date the version of * the CVS and now I have a
problem.
I cannot effect the RELOAD that * it breaks.
Somebody can help or say as to load new users without stopping * ?
Thank?s
Excuse my English
Joao Carlos Moura
2004 Jul 20
1
Up to date?
Hi,
before you start throwing stones to me let me tell you that I am a bit new
to Linux. I downloaded Asterisk from the cvs server on Wednesday 15 July
2004, as described in Andy Powell's "Getting Started with Asterisk"
(http://www.automated.it/guidetoasterisk.htm). Thanks Andy! I read about the
Asterisk 1.0 RC1, and I would like to download it and install it.
Could someone tell
2004 Jul 23
1
addmailbox
Hi,
I am a new user to both Linux and Asterisk and would be grateful
for any help and advice anyone has to offer. I have installed Linux and
asterisk as per Andy Powell's excellent getting started guide. The problem I
have is that the addmailbox utility does not work and I cannot find the file
anywhere on the machine. I downloaded the files via CVS so assume I have the
current
2004 Aug 02
3
App.c
Can someone tell me where I can get just app.c from. Mine somehow got
corrupted, and no updates or anything else will fix it. I just need the one
file from the latest cvs. 8-1-04. Please help
2004 Aug 03
1
Emailing phone messages?
Where do you set the outgoing mail server for use with asterisks mail
system? I have entered the info in the voicemail.conf file correctly,
but I am still unable to get the voicemail messages via email. I ran a
tcpdump on the system while calling in and leaving a voicemail and I
don't even see the system try and contact a mail server.
HELP!!! Thank you all in advance.
Sean Garland
2004 Aug 09
2
ChangeMonitor syntax
I'm trying to use the ChangeMonitor command on the asterisk manager API, but I
can't find the syntax anywhere. Asterisk only tells me:
Action: ChangeMonitor
But I don't know the parameters. Can anybody help me?
2004 Aug 09
1
called and callers buttons on bt100
is there something that needs to be set up to make the 'called' and
'callers' buttons work on this phone?
all i get is the backlight to switch on and off.
Jason Kawakami