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...g that happens is "why is there a delay before the call connects?" and the answer is "you have to hit dial or wait 10 seconds". -Jonathan -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Tom Rymes Sent: Thursday, August 11, 2005 3:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Newbie Question: Building an Asterisk systemto replace an old PBX but using existing phone On Aug 11, 2005, at 11:49 AM, Sean Rima wrote: > Tom Rymes wrote: > &g...

Does anyone on the list have a script for notifying pagers that they would be willing to share? I have found a reference in the archive to such a script, but previous attempts to find the author of that posting have failed. Anyhow, I am looking to set up a system whereby a message is sent to a pager when a voicemail is left in a specified mailbox. (This is easy, it's built-in to

Hi list, I just installed Asterisk 1.4.38 (on an updated Centos 5.5 machine) and am getting this error : WARNING[6472]: res_musiconhold.c:856 moh_scan_files: getcwd() failed: No such file or directory with the default musiconhold.conf file. When I change musiconhold.conf to this: [default] mode=mp3 directory=/var/lib/asterisk/mohmp3 (and have converted all the wav files to mp3 and put them

...with a custom install, you should not use it in a production environment. - Mark Brooker says that both approaches are to be mastered. Anyway, you should be able to install from source to use Asterisk in a production environment. However, FreePBX is a great tool and should be used too. - Tom Rymes says that troubleshooting a GUI is much easier and Trixbox has no more problems with hardware than a custom install has. For example, on Sangoma site, there's a link to a customized Trixbox version targeted to Sangoma cards. Using Yum you can download new drivers an eventually install them...

I have a setup of asterisk 1.6 in one box and asteirsk 1.4 in another. I can send recieve faxes from both boxes fine to and from pstn. But the faxing between 1.6 and 1.4 extensions does fail. Any ideas please ? -- Thank You Amit Nepal

Folks, First off, this is messy, and I hope someone will be kind enough to help me clean this up (the part added to extensions_additional.conf). You've been warned! For those of your using AMP or A@H, there has been a lot of talk about how to route incoming calls to different places based on which trunk is ringing. The standard answer is that you can only do this by using DIDs,

hello i am using asterisk-1.0.9. i have a NAT problem. without NAT registration is ok. and if user is bhind NAT it is registring on asterisk. but SJPhone is showing "not registered". i think asterisk is properly sending request to UA. any comments............this sip.conf setting was working previously -- Registered SIP '5000' at 0.0.0.0 port 5060 expires 120 -- Saved

I have a brief from a local hotel to build a PBX using Asterisk but they want to use their exisiting telephones and wiring from an old PBX that no longer works. Basically, I can build the system but an looking for a card that will allow for upto 20 extensions to be wired into the back of the PC. Doeas anyone know of a solution to this Sean-- ICQ: 679813 FidoNet: 2:263/950 Jabber:

You are right. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Tarpo, Louie Sent: Thursday, August 11, 2005 5:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Newbie Question: Building anAsterisk systemtoreplace an old PBX but using existing phone You write out a

Hi folks, I was hoping that someone might be able to help clarify some confusion I have on DAHDI Fax detection after spending some time searching. My understanding is this: 1.) Echo cancellation is automatically disabled upon recognition of a CNG tone, regardless of the faxdetect setting. This can only be disabled at compile time. 2.) faxdetect=incoming will, upon detection of a CNG tone,

...ed > pbx*CLI> > I did the same on my friends machine and it show the > same thing. > Why is the refresh period so large and what can I do > to shorten it? > I've ruled out any ISP issues. I can receive calls > on my other VoIP > services just fine. > Mark > Tom Rymes wrote: > > Have you restarted Asterisk to see if that helps? > > > > What does 'sip show registry' show? > > > > Tom > > > > On Aug 19, 2005, at 10:42 AM, Mark Phillips wrote: > > > >> So it was all working well and then suddenly...

...from CVS and it seemed to fix the stability issue we were having. If you are using CVS (but not the latest one) you may want to try upgrading. I wouldn't worry about that message, though. However, I would also be interested in knowing what it means/what causes it. :) Jeremy Tom Rymes wrote: >Has anyone figured out what this message means: > >Don't know what to do if second ROSE component is of type 0x6 > >We are running a PRI through a Sangoma card that is handling the >D-channel natively at this point, but we go the error when zaptel was >handling the...

OK, after my last message about fax detection, I feel a bit better informed and able to press forward. I started looking into this because I was getting lots of false positive fax detection errors in the logs with faxdetect=both set in chan_dahdi.conf. Anyhow, I do not currently use fax detection, and we have a dedicated Fax DID on our PRI, so setting faxdetect=no works fine. Having said that, I

Hello all, we're using asterisk 1.2.12.1 in an Inbound callcenter using the queue application. If there are many calls in the queue, it sometimes takes up to 30 Seconds before a call is distributed to an agent. For example there are 10 callers in the queue, an Agent is finishing a call and it takes up to 30 seconds before his phone rings again. We're already set the

Egad, not again with Broadvoice! Anyhow, I recently installed AAH and configured my TDM11B and got that and some SIP phones working. I still have some issues to work out, etc, but my current problem is Broadvoice. I have checked out all of the online resources, including the recent list exchange about the recent changes made by Broadvoice. However, the one thing I have found to be consitent in

Ok, I am going for A@H with the CentOS iso disk. Installed and just checking a few things out. My other question is this, which I forgot to ask before. We have no Broadband here and more than likely will never have, so I am just looking at building Asterisk to handle inbound and outbound calls, at home via a ISDN card and for my hotel job via a PSTN line setup. Is this very complicated to setup

...ork without it, there may be a reason internally you can't see why it is required. (Ex. Putting extra stress on a component causing it to fail in 6 months) -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Tom Rymes Sent: Wednesday, August 10, 2005 3:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Is it mandatory to give power supply toTDM400Pcard Is there any reason why anyone wouldn't just plug it in anyway? Why mess around trying to figure out when it doe...

Dear all I use R for Windows 2.3.1 on a fully updated Windows XP Home SP2 machine and I have two related regular expression problems. platform i386-pc-mingw32 arch i386 os mingw32 system i386, mingw32 status major 2 minor

Can an asterisk box equipped with a Digium T1 card handle Integrated T1 circuits? I have a T1 with 768k data and the remaining channels voice, can the asterisk box do the Data routing + Voice processing? It's only going to support 4-5 users(the voice channels won't all be active obviously). ________________________________ This e-mail, facsimile, or letter and any files or attachments

Hi there, Does anyone know how to setup an overflow queue? When a call rings on the queue A, if all agents were busy, the call goes to the queue B. If all agents in queue B were busy, then the call stays on both queues until somebody answers it. I think this is a basic ACD feature available on most PBX that support ACD functionality. Does anybody knows how to do it with asterisk??