search for: exension

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 =>

...01. If both extensions are in use, I want that 3rd call to be queued. I don't think that the config below will direct calls to extension 1001 because the second line states that any incomming calls should be routed to extension 1000. How do I change this so that calls are directed to?all of my?exensions? extensions.conf [from-myprovider] exten => *DID number*,1,Answer exten => *DID number*,2,Dial(SIP/1000) exten => *DID number*,3,Queue(support) ;not sure if this line belongs here exten => *DID number*,4,Hangup ? queues.conf ? [general] [support] musicclass=default strategy=rrmemory...

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

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 Redirect while a playback, the Redirect fails disconnecting the call: http://docs.google.com/Doc?id=ahfnfrcrh3rr_31ghh84bkd Regards -------------- next part ---...

...ground() 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 in: exten => *,VoicemailMain() -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20230617/8ea509f2/attachment.html>

hi everyone, I'm trying to get vlans tagged in libvirt as my switch's end (yes traffic will be leaving the host and into network switches) allows only tagged vlans. But with network as such: ...   <portgroup name='vlan-55'>     <vlan trunk='yes'>       <tag id='55'/>     </vlan>   </portgroup> </network> and guest as:    

...a fax machine the fax machine just says 'no carrier'. Looking at the hylafax logs I see t38modem answer but then get 'no carrier'. If I make t38modem dial an extension all I hear is a beeep.......beeep......beeep of a fax machine. Also when I try to use fax tone detection in my exensions.conf it doesnt work - the call is always routed to the normal sip device and not the t38modem (oh323) device. I have followed the examples on the wiki. Am I doing something fundamentally wrong here? Do I also need to load faxcapi? I have not yet tried using the sipaura 3000 to pick up the in...

...low 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 in: > I'd have to assume yes.  I don't use WaitExten() and I set autofallthrough=no in the /etc/asterisk.conf, since that is the way I've always expected Asterisk to work; my dialplan examples are based on that. The below example shows a call coming into a DID, playing backgr...

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 th...

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. Sample: exten => s,300,MYSQL(Connect connid ${HOST} ${USER} ${PASSWORD} ${DATABSE}) exten => s,301,MYSQL(Execute resultid...

...rking 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 workaround? Thanks, Scott BTW: I am aware of the presence bug with the IP Phones, and this is while the phones are actually monitoring.

...thods into a lib file (taggable_extensions.rb) which mirrors the acts_as_x plugin setup, and so in my taggable classes i now say acts_as_taggable include TaggableExtensions This is fine for those classes, although it would be nice if there was some way i could get aatos to automatically use these exensions without going in and hacking the plugin. However, i''m not sure what to do with my Tag and Tagging class methods. Should i put those in the same module, in taggable_extensions? If so then how do i get them to be loaded? I guess these two questions are the same, ie "Can i push some...

...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 a series of DTMF codes that OldPhoneSystem understands. That bit works with no problems. Once it's connected, the context I've placed the call into looks like this: exten => s,1,Answer exten => s,2,Monitor(wav,test...

...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 in: >> > > I'd have to assume yes.  I don't use WaitExten() and I set > autofallthrough=no in the /etc/asterisk.conf, since that is the way > I've always expected Asterisk to work; my dialplan examples are based > on that. > > The below example shows a...

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

...; > > the hylafax logs I see t38modem answer but then get 'no carrier'. If I > > > make t38modem dial an extension all I hear is a > > > beeep.......beeep......beeep of a fax machine. > > > > > > Also when I try to use fax tone detection in my exensions.conf it > > > doesnt work - the call is always routed to the normal sip device and > > > not the t38modem (oh323) device. I have followed the examples on the > > > wiki. > > > > > > Am I doing something fundamentally wrong here? Do I also need t...