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Dear, I have asterisk 1.6.2.13 in a small hardware PBX (512 GB RAM and Celeron), and last days when I call from one extension to another of the same PBX after I dial the number the rings sound after 20 seconds. In the CLI log, when I debug the AGI, I see always goes good until dialparties.agi, and after that there are 20 seconds without any log, and so the ring sound. I've read

Hello: We have a TDM2433E Digium Card (12 FXS, 12 FXO) and Asterisk 1.8.7.0 running. Everything seems to be ok but call transfers. This is the issue: *A, B, C and D are in FXS ports*. 1) A calls B. B anwers. 2) B tries to transfer the call to C dialing *2 (code for attended transfer). 3) A hears MOH. B dials number C. 4) Asterisk says the dialed number is incorrect or non existing. We tried

...to display "Front Desk" <1600> on internal calls, and the DID associated with extension 1600 on external calls. This also means I don't need any extra AGI's or db lookups, etc. If you have any questions, please feel free to ask. -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com <http://www.selbytech.com> --00163649982d63313b04a55fe58d Content-Type: text/html; charset=ISO-8859-1 Content-Transfer-Encoding: quoted-printable <div class=3D"gmail_quote">On Fri, Jun 10, 2011 at 12:52 PM, Warren Selby <= span dir=3D"ltr">&lt;<a h...

Hello, For those customers with only analog lines, who ask for fax2email and email2fax, whats the most reliable solution available and tested with Asterisk? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120529/3e28b56e/attachment.htm>

Hi All; If I set a context other than the default context, then I do not see a generation for a configuration in the extensions_additional.conf for this context, but always the generation for the configuration is for the default context (from-internal). Normally, I have to put some Phones in a context and another Phones in a context, and give each context a privilages, but if I do this, then I

Hey Guys! I am getting this wired error when i am calling IAX trunk. Everything works! but i want to get rid on these RED WARNING messages.. what is wrong here ? I have func_aes.so module loaded. also i remove and test but still same error. -Satish == Using SIP RTP CoS mark 5 -- Executing [7623 at from-sip:1] Macro("SIP/7527-0000000d", "orasebcamdial,7623") in

...ng the way the process starts up (commit r376428), but I'm not enough of a coder to understand if those would cause what I'm seeing. Is anyone else seeing this issue? Should I open an issue on the tracker? Anyone see something obvious I missed? -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com <http://www.selbytech.com> -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130116/33e98470/attachment.htm>

be able to setup a SIP trunk. I've been able to successfully integrate a Cisco CallManager 7.x system with Asterisk using SIP trunking, so I imagine you should be able to do the same here. -- Thanks, --Warren Selby, dCAP http://www.selbytech.com --0016e651f0a6bbe47b04a303939e Content-Type: text/html; charset=ISO-8859-1 Content-Transfer-Encoding: quoted-printable <div class=3D"gmail_quote">On Wed, May 11, 2011 at 12:30 PM, Darrin Henshaw= <span dir=3D"ltr">&lt;<a href=3D"mailto:darrin.aster...

When I receive a call, I want to automatically park it from the dialplan so that I can retrieve it later. However, I don't want callers to be aware that they are being parked, so I want to play a ringing tone to the caller. Park() is supposed to be able to do this: Park([timeout][,return_context[,return_exten[,return_priority[,options[,parking_lot_name]]]]]) options r: Send ringing

Hi, I've been meaning to fix my non-working paging feature here for a while, and I've just spent the last 5 hours looking at many, many web pages that all say the same thing. I am using Asterisk 1.6.2.18 and Polycom phones, both older (501 with "latest" legacy 3.1.7 firmware) and newer (335 and 650 with latest 3.3.1f). I have changed the correct values in sip.cfg like

...me time reading > and playing. Getting these phones working is not rocket > science and there are similarities with how to do firmware / > config pushes. > > Not to sound mean but RTFM > > Sent from my iPhone > > On Jun 21, 2011, at 7:45 PM, Warren Selby <wcselby at selbytech.com> > wrote: > > > On Tue, Jun 21, 2011 at 5:35 PM, bilal ghayyad <bilmar_gh at yahoo.com> > wrote: > > Dear Warren; > > > > Please, keep all discussions to the list.? > There's no need to email me personally about this. > > > > <...

Hi All; When the call coming via the E1 dahdi and I handle the call (as first step) by exten => 5631040,1,Goto(OrangeCMG,s,1) it is not working and the call will be disconnected instead of queued. But, when I handle the call (as first step) by playing any sound file and then send for the queue, then it is working fine, WHY? exten => 5631040,1,Playback(WelcomeMessage) exten =>

someone have been installed Asterisk (Trixbox) on VirtualBox which is installed on a Linux host (Ubuntu server 10.04 specifically). I want to know if it is convenient or not, and the reaseons if i should on shouldn't do it. Thanks in advance.! -- Esteban L. Cacavelos de Amoriza Cel: 0981 220 429 -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I'm getting one way audio when calling over the SIP trunk i.e. end device B (remote end of SIP trunk) can hear device A (softphone registered with Asterisk) but device A can't hear device B. Even though I configured same NAT configurations on other servers and they are working good. The NAT configuration is listed below; localnet=130.0.0.0/130.0.0.0 externhost=12.131.12.13

NEWBIE alert: i'm a linux person, not an asterisk person so i'm certainly capable of handling any linux-flavoured solution you can suggest. here's a note i got from a local company i know (some proper names removed): ===== start ===== Now and again our ISP goes down and when it does give us a hicup, the Asterisk system shuts down (not very forgiving). When it shuts down our phone

...be able to do that during a failure on their side. During a recent outage (I think they had some major issues at one of their switches), they were not able to send the calls to our box which was online. > >Thanks, >Matt > >Date: Tue, 30 Apr 2013 20:38:19 -0500 >From: wcselby at selbytech.com >To: asterisk-users at lists.digium.com >Subject: Re: [asterisk-users] multiple provider for incomingOn Tue, Apr 30, 2013 at 7:50 PM, David Wessell <david at ringfree.biz> wrote:Hi Matt, You can't have multiple providers for inbound traffic. You can have multiple providers for o...

takes precedence over a queue's defined moh class. --=20 Thanks, --Warren Selby, dCAP http://www.selbytech.com --000e0ce0494051d402049b4247c1 Content-Type: text/html; charset=windows-1252 Content-Transfer-Encoding: quoted-printable <div class=3D"gmail_quote">On Tue, Feb 1, 2011 at 10:20 AM, Danny Nicholas = <span dir=3D"ltr">&lt;<a href=3D"mailto:danny at de...

Hello, I have an internal extension, e.g. 1005 which is being called from an external/public number like 123456789. Now when it comes to the spoken voicemail information it says something like "number 1000 not available", however it should say "number 123456789 not available". How can I configure this? I already googled and I guess this is really easy, but I just couldn't

Hi All; Using the FreePBX, after I added the extension from the GUI, I discover that it is automatically added in the extensions_additional.conf in the context [ext-local] and [from-did-direct-ivr] How I can change these context name? I need to determine this. How? Regards Bilal

Hello everyone, How can I let Digium Wildcard TDM800P work successfully with DAHDI? Because the Centos recognizes the card but I can't get the analog card working with DAHDI. Thanks in advance, Gilberto -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130309/d82ddd47/attachment.htm>