similar to: Toll Call Voicemail Ring Timeout (new module????)

[answeringsvc] exten => 0,1,Wait,1 exten => 0,2,Dial(ZAP/g1/7135551212,${RING_TIMEOUT},r) exten => 0,3,Dial(ZAP/g1/7135551212,${RING_TIMEOUT},mr) exten => 0,103,Goto(0,3) exten => 0,104,Goto(0,3) This should call 713-555-1212. If there are no ZAP lines available it should kick back around and play music on hold until a zap line is available, correct? I'd like the

Always Learning wrote: > >> Used a VCR or Cassette Player lately? > > My VCR broke. Replaced it with a DVD/HDD & USB3 unit. Replaced cassette > player and tape recorders with broadcast quality handheld recorder > DR-100mk3 and an amazingly good Sony PX440. But how do you play all your old VCR tapes? As I said, I want to burn them to disk, but I still have a working VCR.

Dear The Expert, I am very new with this, I have installed AsteriskNow, X-Lite as my SoftPhone, I am using SPA-3102. I have 3 extensions, me at 250, 998 is my Linksys SPA-3102 and 999 for PSTN Line (see below) My problem is, I am unable to call 998, I thought this is registration problem, (because the Linksys screen info said Registration Failed) Could any body please help? Many thanks in

Hi, I have these two records which are saved in two different tables. The problem is unless the first record (Cr) is saved the second (Crmapping) doesn''t get created because of referencing. It''s my assumption that create() does not need explicit save() method. It instantiates the model obj and saves it into db. But then any ideas on why the Cr.create is not creating a record? if

Is there a way to override sip peers defined in users.conf with respect to their context and hints? Every extension I have defined in users.conf always gets an @default for the hint priority. Below are asterisk outputs and users.conf entries. In peer 1203 I've set a subscribecontext, which is completely ignored. Thanks for any help. nurscarepbx*CLI> core show version Asterisk 1.4.22

Hi, installing Office 2007 was a breeze, I then added the override for riched20.dll. Word works fine AFAICS except for font anti-aliasing, but that's not crucial for now. Powerpoint can import .wav files but not .mp3 files. More crucially, I'm unable to import .mpg or any other video files. http://ubuntuforums.org/archive/index.php/t-470842.html has maybe a similar problem (year-old

Johnny Hughes wrote: <snip> > I get it .. but no one needed a hand held cell phone before 1973 and no > one needed a smart phone before 2007. Now, almost everyone has a smart > cell and land lines are dying. Technology moves forward. People want > integrated cloud, container, SDN technology, etc. Used a VCR or > Cassette Player lately? I have no intention of *ever*

On 02/15/2017 07:34 AM, Leroy Tennison wrote: > Too much temptation to resist, I don't know which one of us is older but I have a feeling it's a "horse race". Like you, I still have a land line, WiFi is too slow and "WiFi security" seems to be an oxymoronic phrase. Why people text (or IM for that matter) anything other than a one-liner is beyond me. > > Now

> Used a VCR or Cassette Player lately? My VCR broke. Replaced it with a DVD/HDD & USB3 unit. Replaced cassette player and tape recorders with broadcast quality handheld recorder DR-100mk3 and an amazingly good Sony PX440. Still retain the original functionality. C7 doesn't retain all the original functionality :-) -- Regards, Paul. England, EU. England's place is in the

On Wed, February 15, 2017 7:34 am, Leroy Tennison wrote: > Too much temptation to resist, I don't know which one of us is older but I > have a feeling it's a "horse race". Like you, I still have a land line, > WiFi is too slow and "WiFi security" seems to be an oxymoronic phrase. > Why people text (or IM for that matter) anything other than a one-liner is

Too much temptation to resist, I don't know which one of us is older but I have a feeling it's a "horse race". Like you, I still have a land line, WiFi is too slow and "WiFi security" seems to be an oxymoronic phrase. Why people text (or IM for that matter) anything other than a one-liner is beyond me. Now for the real issue, what happens when Network Manager

Hello, everyone. Today I've installed an asterisk svn trunk (r61667). The problem I'm having is no matter what context I set in the config file for that peer, "default" is always being used. The output of "sip show peers" shows the context correctly, but when I try to make a call, using that peer, I can only dial the numbers set in the "default" context.

Hi, when I'm trying to call the following extension exten => 6002,1,Verbose(1|Extension 6002) exten => 6002,n,Dial(Agent/6002) exten => 6002,n,Hangup() the call is terminated and I get the following warning from asterisk: app_dial.c:1106 dial_exec_full: Unable to create channel of type 'Agent' (cause 17 - User busy) When calling the agent with Dial(SIP/6002) no problem

Hi all, I have a demo machine I'm running up on Lenny - it has the packaged Asterisk version installed (1.4.21.2+stuff). I'm trying to add an extension to leave a voicemail message, just with Voicemail(1234), which I've done before (on 1.2 at least), but it's saying "no application 'Voicemail' ". "module show like voi" shows

Hello, I'm trying to get calls working between websocket clients and sip clients. For clients I have sip.js based clients on chrome, Zoipers and a Grandstream phone. Challenge here is I'd like to have Kamailio and rtpengine to handle the bridging between different rtp profiles but Asterisk changes them in the sdp bodies along the way. I'm using Asterisk 11.11.0. Is there a way to

Hello, I noticed a strange thing while testing my Asterisk-Kamailio Realtime setup. In an INVITE the From and To headers contain the same number when calling through a Realtime integration setup. This happens when the INVITE leaves Asterisk. Can you guys tell me what might be causing this? I have 660 at testers.com as a websocket client and 700 at testers.com (caller) using a Zoiper client (db

On Sun, 1 Oct 2017 20:02:08 -0400 Mark LaPierre wrote: > What repo did you find ffmpeg-libs in? Version : 2.6.8 Release : 3.el7.nux Architecture: x86_64 Install Date: Wed 27 Apr 2016 06:23:09 PM CST Group : Unspecified Size : 13562904 License : GPLv2+ Signature : RSA/SHA1, Wed 27 Apr 2016 06:35:00 AM CST, Key ID e98bfbe785c6cd8a Source RPM :

I recently upgraded from asterisk 1.2.13 to 1.4.2 and am looking at using the users.conf file to setup my users, before i was using real time SIP which worked fine. However when i create a user in users.conf i am unable to register the user form a softphone, however that same softphone can still register a different the users i currently have setup form the sip.conf from real time. i've

I've downloaded R-2.6.0 I want to install it on Solaris, so, I run the configure command, it to be fine, but once I run make it give me the following error: *ld: fatal : fichier Rmain.o : type de machine ELF erronV : EM_386 ld: fatal : Erreurs dans le traitement des fichiers. Aucun rVsultat n'a VtV Vcr it dans R.bin collect2: ld returned 1 exit status *** Error code 1 make: Fatal error:

Hello, I have a problem with a call between 2 webrtc clients. Asterisk removes the ice-related lines from the sdp when it sends the INVITE out, and the called webrtc client rejects the INVITE due to the missing ice lines. Both webrtc clients are defined exactly the same way, same values in all fields except the number of the peer. There's probably something I've changed that causes this