similar to: why not forwarding to this number?

Hello all, HI i am very new to asterisk and my boss needs me to investigate setting up asterisk for a new client. I have downloaded and installed (make, make install and make progdocs)asterisk on my personal computer and when i try to run it (./asterisk -vvvc) i get the following output below: NOTE: i am running REDHAT 9.0 on a 796MHz cpu machine: I am excited to be able to work with asterisk

Hi: I deleted old modules in /usr/lib/asterisk/modules before make install. I built zaptel and libpri before asterisk. Modprobe zaptel and modprobe -v wctdm executed witiout complaint. Starting asterisk produced the output below with several warnings and a failure. Can someone help, please. I double-spaced the warnings in the text below. The first warning is about music on hold because it

Hi, First i would like to thank the kind people of the list who have answered my previuos mail, but i am still stuck as asterisk still crashes upon startup, i have read the install article at http://www.voip-info.org/wiki-Asterisk+Step-by-step+Installation and i have search the asterisk archives, but i still cant get asterisk to work, i have tried reinstalling asterisk but it still complains and

I have an Ubuntu system and went into Synaptic and checked asterisk for installation. Once installed, I started it with /usr/sbin/asterisk -vvvgc and got the following output with several errors and notices. Do I need to do more or are these ok? I expected to have some conf files in /etc/asterisk but there is nothing there. Thanks! Created by Mark Spencer <markster@digium.com>

Greetings, Running CVS HEAD about 3 weeks old, I have been beating my head trying to get this to work properly.. Or at least figure out what's going on. Maybe I have done things wrong... I have created a 'catch all' extension at the end of our last context where all phones & voicemail extension exist. This catch all is included in all and works quite nicely except when voicemail

This seems basic but something is missing..... I dial from my cell phone to my DID and enter the context in extensions.conf I am hoping to cascade through the plan and successfully automatically dial the 1444 number listed. But it fails. And, I dpon't know why? Should I removed the Hangup application? Syntax issue somewhere? I have a good SIP registration with the vendor, voipvoip.

How do you enable call waiting on sip phones? Ive looked and googled and can only find call waiting pstn phones butnot for sip. Is their a way of setting this up within the dailplan?

After upgrading from asterisk-10.5.0 to asterisk-11.1.2, I am getting a Segmentation fault. [root at localhost asterisk-11.1.2]# asterisk -vvvvvvc Asterisk 11.1.2, Copyright (C) 1999 - 2012 Digium, Inc. and others. Created by Mark Spencer <markster at digium.com> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components

Hi all, I have a connect between a siemens hipath & Asterisk system over PRI The connection works perfectly I can call from the Hipath to an Asterisk Extension. I want to allow the hipath extensions to dial out over a SIP trunk on asterisk but I keep getting "The number you have dialed is not in service" In this CLI I'm dialing from hipath extension 9 (prefix for sip trunk)

Has anyone had any luck with PostgreSQL and Realtime? The realtime instructions on voip-info seem pretty straight forward... just not woking for me. I've included all of my config files below, and my console output. Entire console bootup output: [root@abox asterisk]# /usr/sbin/asterisk -vvvvvvc == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing

I have the following senerio. Call file calls 1st party. When connected give called party option to connect to second party. Issue Dial to second party. Caller answers and the two are bridged together. My issue is that 4 out of 5 calls fail to bridge the audio. Am I missing something or is there some kind of bug? Here is my test dialplan ;Dialer Base Code Files. ;Variables

Here is a very very basic config. But, not working (: I simply want to dial the DID that is registered with the SIP provider. then, as you can see the call should dial the 703111 number Hints please? sip.conf ;register => 908366554:396444 at carrier.jazzey.com register => 908366554:396444 at sip.jazzey.com [jazzey] type=friend host=sip.jazzey.com username=908366554 secret=396444

Hi, Ive been struggling with asterisk for a few days now. I understand pretty much how it works and how to tie things together (SIP -> SIP internally works fine etc), but just cannot get asterisk to work with an X100P clone (its a Ambient MD3200, if that means anything to you guys). I have tried (initially) asterisk 1.07 with zaptel 1.07, and now Asterisk CVS-HEAD with zaptel cvs. Both give

why does this not work? i simply want to hear the recorded message exten => s,1,Answer() ;exten => s,n,Record(zipcodegutter1.gsm) ;zcg1 exten => s,n,Playback(zipcodegutter1) exten => s,n,Dial(SIP/c000001s/12222222259,120,A,(demo-thanks))

I want to create a script for IVR that compiles responses, aggregates them to a total number. Then, run an equation based on the result. Press 1 for X (X is a positive number 500) Press 2 for Y (Y is a positive number 200) Press 3 for Z (Z is a positive number 300) Press 20 to calculate the results = 500+200+300 =1000 then, exten => s,n,Read(NUMBER,,1000) exten => s,n,SayDigits(${NUMBER})

Folks, I've just successfully set up Asterisk (as part of the Asterisk Management Portal installation). When I say "successfully", I mean that I have gone through all the steps detailed for the installation of AMP and not hit any snags there. I can connect to my asterisk server via ssh and can also connect via Http to the portal to change settings in AMP. Now I'm trying to

Installing DevkitLite hardware (Very similar to John Lange's post on Tue Oct 08 2002) I cannot get anything to work on the phone connected to the s100u. I dont know what to do. Can someone please help me? I used the sample configuration files from digium documentaion that was supposed to be "sane" defaults for the kit. Very similar to John Lange's post on Tue Oct 08 2002 Here

After doing an update from SUSE 9.2 to 9.3 and Checking out the latest from CVS, Asterisk crashes on startup with an apparent MySQL (res_config_register) error: # asterisk -vvvgc > asterisk_startup_error1.log asterisk: symbol lookup error: /usr/lib/asterisk/modules/res_config_mysql.so: un defined symbol: ast_cust_config_register The log is shown below. I've seen the posts

Hello, Occasionally, I get the following warning in my asterisk log, pbx.c: We were unable to say the number [n], is it too large? n is two or one digit number, which doesn't look like large to me! Could anybody please tell more about this warning, like in what scenario I may have this warning. Thanks, Mohammad -------------- next part -------------- An HTML attachment was scrubbed...

Hi, How could I possibly inform incoming callers that the number they'd dialed is monitored and recorded. I wanted that when a call-in or call-out is made, a playback will be played to inform caller & callee that thier line is monitored prior to start conversation. Thanks. Angel __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best