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...ms talking about changing 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>

...This setup allows me 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"l...

...that allows me to see every IP address of every sip registration attempt, along with details about the sip reg attempt (I.e user name tried, success or failure, user agent, etc). I haven't found a way to do this yet, I'm hoping I've just missed something simple? Thanks, Warren Selby

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

..."20100713-110853 - 2010 - 07 - 13") in new stack 20100713-110853 - 2010 - 07 - 13 Is what I'm trying to do possible? It seems like it's at least recognizing that I'm trying to grab a date, but it's not taking the date format parameters that I want. -- Thanks, --Warren Selby http://www.selbytech.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100713/68f58207/attachment.htm

I am using Elastix. Asterisk is used for PBX application in Elastix. I want to make customize MOH. But not able to use MOH. I make simple extension in asterisk conf file but no success :( Below are the details of configuration files. Even default MOH is also not working.... *Asterisk Version 1.6.2.17.2 * *1) Extension.conf* [incoming] exten => 6000,1,Answer exten =>

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

Can you say a Virtualized Asterisk with a PRI card! http://www.phoronix.com/scan.php?page=news_item&px=OTY0OQ Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety."

...eeds to just spend some 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 =>

I am running Asterisk 1.8.10.1. I am trying to set up some routing in my dial plans and having some issues (the issue being that I don't quite understand all of the syntax and patterns that can be used: Examples: DID1 = 6140000000 DID2 = 6140000001 CNAME1 = 6149999999 CNAME2 = 6149999998 CNAME3 = 6149999997 context1 context2 context3 I have looked at several examples (patterns) and I

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>

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

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

The Asterisk Development Team has announced the release of Asterisk 1.8.2.3. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.8.2.3 resolves the following issue: * Reimplemented fax session reservation to reverse the ABI breakage introduced in r297486. (Reported by Jeremy Kister on the asterisk-users

The Asterisk Development Team has announced the release of Asterisk 1.8.2.3. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.8.2.3 resolves the following issue: * Reimplemented fax session reservation to reverse the ABI breakage introduced in r297486. (Reported by Jeremy Kister on the asterisk-users

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

Hi List, I have another issue on allowing outgoing calls to PSTN on Asterisk via Avaya Phones, I hope that anyone could help me fix this issue: *When I dial through Avaya phone i just here a "good bye message" reply from asterisk server. And here is the log:* == Starting OOH323/(null)-b7db8aa0 at internal,s,1 failed so falling back to exten 's' == Starting

Thank you. That's a nice looking radio app. On that link below, I only see links to download pre-compiled APK files (apps). Are there links to the actual source code available? Date: Fri, 20 Jun 2014 10:36:58 +0200 From: sardylan at gmail.com To: icecast-dev at xiph.org Subject: Re: [Icecast-dev] Android Java source for icecast streaming Hi jSelbie, try to have a look to