similar to: Delay when ringing internal extensions on incoming zap call

Ok here is one for you. I know we all do the this for 911: exten => _911,1,Dial(Zap/1/911) exten => _9911,1,Dial(Zap/1/911) And this probably is more then acceptable for most of us. However I have a system setup that uses SIP for most calls and 1 POTS line. We use a "least cost" routing that uses the POTS line for local calls AND SIP when appropiate. What I want to do is

Well I dont want it to operate as a group. I understand that it is *8 on most PBX systems but I would like it to work as dialing *8+extension-thats-ringing to have it go over to my extension. I am having a lot of trouble finding more information from the wiki and google. Thanks! -James -----Original Message----- From: asterisk-users-bounces@lists.digium.com

Hi, I have an asterisk installation that has been happily working in production for some time (E100P and UK BT ISDN30). Recently I upgraded to HEAD-07/29/04. Now, incoming callers don't hear ringing while calling in. As far as I can tell, my config files haven't changed from what was working before. Can anyone please help before my boss shoots me? JC zaptel.conf

Hi. I have a huge list called twitter: > dim(twitter) NULL > str(twitter) List of 1 $ :Classes 'PlainTextDocument', 'TextDocument', 'character' atomic [1:35575] 11999;10:47:14;20;10;2009;ObamaLouverture;Trails Mixed Lessons For Governance From Campaigner-in-chief: President obama jumps campaign 09 tuesday..

I've setup SPA-3000 and when the calls come through my phone is rining almost instantly but the [demo] doesn't answer till after about 13 seconds. So I have about 13 seconds delay and I don't know what setting is causing it; here is a part of my settings from extension.conf. [from_pstn] exten => 1000,1,Goto(demo,s,1) [demo] exten => s,1,Answer ; Answer the

http://bugs.freedesktop.org/show_bug.cgi?id=15350 Summary: randr12: VGA-0 always reported as "disconnected" Product: xorg Version: git Platform: Other OS/Version: All Status: NEW Severity: normal Priority: medium Component: Driver/nouveau AssignedTo: nouveau at lists.freedesktop.org

Dear all I have TDM808B 8 port FXO it is configure perfectly but i got some problem of incomming phone Hangup and callerid display problem i am going to explain you the issue i have install asterisk 1.4 and i have 100 of SIP phone now everything is fine but problem is when i incoming call on FXO and dial sip extention SIP phone is rining but when i disconnect my incoming

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Hello, I am working on a CentOS Linux 2.6.32-71.el6.x86_64 system. I noticed that the system came with httpd-2.2.15-6 installed. After I run 'yum update' I get httpd-2.2.15-9. I did some research on the Internet, but can't find the following information: are most of the security fixes that appear in httpd-2.2.21 applied to the update on httpd-2.2.15-9? There was a security fix just

I am running * with a Junghanns quadbri that should allow us to integrate our ISDN house telephone system with VOIP. Preferably I would like to run a setup, so that our internal ISDN phones on an S bus are not aware that * is sitting in between. With the configuration below I run into the following problems: 1. On outbound calls, I get the normal rining call progress tone althought the

Using this link http://www.oldskoolphreak.com/tfiles/voip/installing_app_cepstral.txt This is a Dell PowerEdge 1950 running Whitbox 4 and Asterisk 1.4.0-beta3 I get the following errors on make install Any help would be GREAT! Thanks [CC] app_cepstral.c -> app_cepstral.o In file included from /usr/src/asterisk/include/asterisk/linkedlists.h:23, from

Hey everyone, I was frustrated with the existing app_cepstral/app_swift TTS modules I've found on the net, so I hacked up my own. It's been working really well for me so I thought I'd share. In developing this, I wanted to avoid: * the startup delay incurred writing TTS output to a temp file then playing it back * gluing cepstral/swift to the asterisk module in a way that fubard

Dear all I am going to setup Asterisk Call center solution and i have setup my queue and agent i have 2 SNOM ip phone but when i call to queue my agent phone is rining without musicnhold or when both phone is busy then i call to queue its directy hangup without musicnhole means my call not goes in to queue what is the problem my queue.conf [root at pbx asterisk]# cat

what's the easiest way of using cepstral voices with asterisk ? On their website, in the ssml page (http://www.cepstral.com/cgi-bin/support?page=ssml), they say "Asterisk PBX SSML can be used with Cepstral voices in Asterisk by simply embedding the markup into the input text." what input text ? To what application ? Thanks ! Julian

I just got cepstal working fine in the dial plan using code like: exten => 511,5,AGI(cepstral.pl|Welcome to my house finder. At the beep enter your zip code.) The php script it calls is based on the nerdvittles weather one so it calls a webpage which prints to the screen, the nerdvittles code uses system to generate the .wav file then has the dial plan call it via: //php script $retcode2 =

I try to use a SIP trunk from a VOIP provider to make land to mobile calls. If I do these from a ZAP channel, using an analogue phone, after few seconds of silence (I don't like to generate fake [r]inging) I ear the ringing tone from the mobile operator along with any message the mobile operator decide to say me. If I try to use a SIP phone (or a IAX phone) attached to my asterisk box, I

Hi all, I'm researching IP phones for a new office setup. We will need 30 phones. I have read the wiki and the polycom site for the phones, but I still had some questions about real world experience with these phones. -According to the documentation, the 500 series ( and 600, according to a polycom rep ) have built in hubs. Has anybody noticed performance issues in this setup, when

Hi all, can somebody explain me why i can't hear a ringing tone (alerting) if i'am going to connect to my destination end point? Is it basically so that i have to configure like: exten => xxx,1,Dial,ChanTec/number|timout|r Is it really nessesary to use the "r" option everytime if i want to indicate a ringing tone? This suggest a wrong call flow for the user ... Thanks for

Dear All, All these days I was ahpppily using Asterisk with TDM11B, but from today all of a sudden asterisk has started acting strange. The telephone device connected to channel 1 rings continously, following info is displayed on console -- Starting simple switch on 'Zap/4-1' Jan 28 15:38:58 ERROR[77024176]: callerid.c:261 callerid_feed: fsk_serie made mylen < 0 (-8) Jan 28

As soon I do a reload I see contant ringing like this on the CLI: -- Zap/14-1 is ringing -- Zap/23-1 is ringing -- Zap/22-1 is ringing -- Zap/20-1 is ringing -- Zap/19-1 is ringing -- Zap/14-1 is ringing -- Zap/23-1 is ringing -- Zap/22-1 is ringing -- Zap/20-1 is ringing -- Zap/19-1 is ringing This goes on continuously and no phones are ringing. I am using a digium T1 card and ADIT 600. Does