similar to: Debugging Asterisk in GDB (DDD)

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 -------------- next part -------------- An HTML attachment was scrubbed...
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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040720/2975991b/attachment.htm
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 -------------- next part -------------- An HTML attachment was scrubbed... URL:
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