search for: exensively

I have a phone (a panasonic globalrange phone) which always sends a fully qualified phone number. That is, for a local Canadian number, even if I key in 6135551212 it actually sends to asterisk 01116135551212. This means of course, along with "normal" phones I end up having twice as many extensions for outdialed numbers. Is there any way I could canonicalize this down to the more

Calling from PSTN let extension 601 ring twice, hang up and starts over again to ring twice, ... If I pickup I do not hear on extension 601, and on the PSTN it is still signaling to ring. Can anybody enlighten me, please? extension.conf [incoming_88097074] exten => s,1,Wait(1) ;wait to get caller ID in. exten => s,2,Dial(SIP/102,20) exten => s,3,Voicemail(u102) exten =>

? Hi, I am working on a small inbound call center solution that uses an ACD system. I might add an IVR system later on. I only have 2 extensions set up (extensions 1000 and 1001), I want the system to put new calls in a queue if both extensions are busy. I am currently subscribed with a SIP trunk provider and can successfully recieve calls. I want?to design a system where customers?can call my

I'm about to present a report (for internal use of governmental agency). I used extensively R , contibuted packages, as well as communications on the R-list As well as citing R, I would like to know how to cite the contributed packages (it is not so easy, as some have been used exensively, other marginally, some are called from another package and some were not used as softwares but gave me insight of how to proceed), as well as thank the persons who generously responded to my demands on the list... Is there an established way of doing that? Thanks I take the occasion of thanking...

Hi, I'm bringing this discussion here from http://www.moythreads.com/wordpress/2007/12/24/asterisk-asynchronous-agi/ about how to manage stopping a playback on a extension previously launched with AsyncAGI and redirecting the call to another exension. If I make the Redirect without a playback, the Redirect works: http://docs.google.com/Doc?id=ahfnfrcrh3rr_30f7fzq4hd But if I make the

Doug, This is where the weeds start growing. On 6/17/2023 4:55 AM, Doug Lytle wrote: > > For both capabilities, you can use Background() instead of Playback() > for audio prompts.  Background() allows for interrupting the prompts > and continue on with your dialplan. > > Understood. From the book: The most common use of the Background() application is to create basic

...           : "ovsbr0" other_config        : {} qos                 : [] statistics          : {} status              : {} tag                 : [] trunks              : [] vlan_mode           : [] No tags, no trunks, no vlan mode??? Is there something I missed (in docs though I sroogled exensively) ? I also tried to add mode='trunk' into <tag id='55'/> and virsh does not complains but next time I edit the guest the mode bit is gone. My vSwitch's bridge has only one phys iface (into the net switch) and I tried setting that iface with tag/no tag, with vlan_mode/no v...

Hi All, I know that this may not strictly be an asterisk issue, but if anyone has any ideas on this prob it would be appreciated. I have a working asterisk setup using Suse 9.1, a BT ISDN BRI card (Fritz?), sipura 3000 and a few cisco 7940's. I wanted to do add faxing into the equation, so I thought I would try and setup the following: PSTN->asterisk->hylafax->pc At the moment

On 6/17/23 08:47, Steve Matzura wrote: > > Both Background() and WaitExten()  allow the caller to enter DTMF > digits. Asterisk then attempts to find an extension in the current > context that matches the digits that the caller entered. If Asterisk > finds a match, it will send the call to that extension. > > > My question then is, is "*" a valid exension, as

I want to include a new context in my exensions.conf I have read this page http://www.voip-info.org/wiki-Asterisk+howto+dial+plan and I can sort of follow it?! I have a context [local] that I know zapata.conf points to, I have edited extensions.conf and put in my phone, sip and iax extensions. I want to add an sms context. I understand that all calls go through my [local] context and I have

We're getting our feet more and more wet with VOIP at work. We want to experiment with a good wireless (as in WiFi) phone. What would be a good phone to impress my boss with? I'm personally drooling over the UTStarcom F3000, but compatibility and shipping ETA info is a bit sketchy. Phil

On 6/16/23 20:29, Steve Matzura wrote: > As always, thanks in advance for a kick in the right direction. For both capabilities, you can use Background() instead of Playback() for audio prompts.  Background() allows for interrupting the prompts and continue on with your dialplan. Doug -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, For those interested in using MySQL directly from extensions.conf, there's already a source file floating around for using a MYSQL application to do SELECT queries. We're using the MYSQL app a lot in our exensions.conf, but we missed support for queries that don't return a result like UPDATE or INSERT. Here's an updated app_mysql.c which introduces the Execute command.

I am working with some Polycom IP600's and for outgoing calls from the phones, the presence features work fine. I am utilizing the 100,hint,SIP/100 for these calls. The problem that I am seeing is that any inbound calls that originate from a IVR in which the extension is dialed, does not show the exension busy on the other phone. Has anyone seen this before? Does anyone know of a

hey folks I''ve installed the plugin acts_as_taggable_on_steroids (henceforth aatos), and i already have some tagging functionality set up, using the same approach (ie a Tag and a Tagging class with associated tables) as aatos. So, i want my existing Tag and Taggings methods, plus some other methods i have in a module which i include with the taggable classes (ie the ones that would call

Hi all, We're in a transition between OldPhoneSystem and Asterisk. One of the things that's needed to be done right now with OldPhoneSystem is the ability to record calls. I thought "Asterisk can record calls", so I set about to make it happen. And it does, sort of. I made a .call file that rings the exension that I want to have recorded, and barges into the conversation, using

OK, this is how I thought it's supposed to work. It just confounded me why the book would say the Playback() and Background() syntax were the same, then in the very next paragraph give an example that belied that claim. On 6/17/2023 1:46 PM, Doug Lytle wrote: > On 6/17/23 08:47, Steve Matzura wrote: >> >> Both Background()  and WaitExten()  allow the caller to enter DTMF

Hello! I am using Asterisk as transparent voice recorder for calls (isdn <-> asterisk <-> pbx). Voice recording (therefore voice forwarding) is working great but seems that Asterisk does not route/bridge/forward D-Channel messages which means PBX cannot get time synchronization answer from provider and tarification impulse too. With direct connection PBX works great and use both

Dear R-helpers, apologies if this is somewhere in a manual, I have not been able to find anything relevant. I run Windows Vista. I have some Fortran code in a subroutine, and have no problem calling this from R with .Fortran, compiling the code either with 'R CMD SHLIB' or independently with gfortran. But is it possible to have more than one subroutine in my source file, one depending

Hi, I think this is one area that needs to be developed. I am curently implementing a system for my home so cannot really justify the cost of financially supporting the development of this when all I really need to do is buy a telephone extension lead for my existing fax modem!!! I am more than willing to devote some testing/documentation time (I am not really a programmer) if that helps.