similar to: Able to tell if phone is registered?

Hi all, In sip.conf: [general] regcontext = autoreg [devabc] regexten = 543 creates "exten=> 543,1,Noop(devabc)" in context autoreg when devabc registers. But I can't use "exten=> _5XX,2,Dial(SIP/${EXTEN})" in the dialplan, because there's no device SIP/543. Now I know I can add a line like "exten=> 543,2,Dial(SIP/devabc)" for each and

Hello, This morning, I asked myself if WebRTC could be a viable alternative to softphone deployment. For me, main issue with Softphones is the amount of work needed for installation and configuration. Also, Softphones must be carefully choosen if Deskphone-like quality is expected. Now that WebRTC becomes ubiquitous, it might make sense to trade Softphone features (call history, BLF, ...) for

Hi All I'm using dovecot 2.0.16 with the pigeonhole plugin 0.25. I'm trying to use the notifiy mechanism from sieve to send notifications when a mail arrives in the mailbox. The message is checked to be a 8bit message, otherwise it is replaced by the default message "Notification of new message." How can I create a 8bit message body within the sieve script that is accepted

Hi Friends, Something I'd like to shed some light on if possible - how is it that a single ISDN conversation only uses 64K for bidirectional communication (using ulaw, correct?), but on several occasions now have seen references to ulaw voip conversations using 64K per side of the conversation, plus packet overhead (http://www.zytrax.com/tech/protocols/voip_rates.htm - seems to be down

Before I play around with this again in 1.2.1, regexten is still essentially broken, correct? The misconception seems to be that it allows you to execute a command upon registration from a SIP UA. Even the O'Reilly TFOT book erroneously states that this is what it is for. After reading the developer discussion though, it definitely seems to be broken. Is it fixed yet? Doug.

Can anyone help with the use of regexten? (* 1.4.3) I've got Asterisk creating extensions for my SIP phones using regexten but I can't seem to figure out how to make use of them once they're registered. Here's my dialplan for from-sip (the SIP's default context): asterisk*CLI> dialplan show from-sip [ Context 'from-sip' created by 'pbx_config' ]

It seems that when you use Realtime static and possibly realtime realtime for sip users, that Asterisk fails to create the regexten context for DUNDi. Someone else had the same problem back in July. Doesn't look like they ever had a resolution. <http://lists.digium.com/pipermail/asterisk-users/2006-July/160105.html>

On Wed, Sep 26, 2018 at 9:40 AM Carlos Chavez <cursor at telecomab.mx> wrote: > > On 9/26/2018 4:46 AM, Olivier wrote: > > > Hello, > > > > This morning, I asked myself if WebRTC could be a viable alternative > > to softphone deployment. > > > > For me, main issue with Softphones is the amount of work needed for > > installation and

Has anyone here ever tried using a wifi bridge to place a deskset in someplace where there was no LAN drop? If so what hardware did you use and was it succesful? Michael -- Michael Graves mgraves@pixelpower.com Sr. Product Specialist www.pixelpower.com Pixel Power Inc. mgraves@mstvp.com o713-861-4005

I am trying to construct a macro for long distance dialling. I have two internet feeds, I have all routes including Teliax on Internet A and a static route to Voxee on Internet B. I thought I could use the dialplan entry below which uses the ChanIsAvail() command to check the connection, but this returns the provider but not the username, so I don't understand how to use this for real

I've been asked for recommendations for a small call centre, an ethernet SIP deskphone with a wireless headset. Similar approach would be a mobile phone with bluetooth head set. Either I've not looked hard enough, or there isn't much on offer. Alec Davis -------------- next part -------------- An HTML attachment was scrubbed... URL:

Dear all. I have what appears to be a configuration error but I cannot for the life of me see what it is. (I am a newbie) I have searched the wikki and google etc but still none the wiser. Any help would be very gratefully received. Problem: Unable to make outgoing calls via E1 euroISDN Digium TE110p card, given congestion signal as per config, unable to open zap channel. All incoming calls work

Hi ho, is there anyone out here that is making use of the regcontext and regexten settings in sip.conf? I've tried this on two Asterisk boxes (1.2.10 and 1.2.12.1) and in both cases I don't see the Noop priority 1 being created upon SIP client registration, "show dialplan xxx" reveals no change. And yes, I have also read and checked bug 7144; if I go down that route and no

Hi my Digium FXO card isn't picking up the callerid I get from the PSTN. I have verified with a deskphone that can display the callerid that the facility works. So, it's definitely the FXO card not picking it up. As I am in Japan, I guess that NTT uses a different method to provide the callerid and so I guess that it is just a matter of configuring the FXO card so that it uses the

I have been reading a number of the past threads about G.729 licensing., about how the registration keys are linked to the network configurations, limited number of registrations etc, etc. Is there no reason why the decoding can't be done in with some Asterisk compatible hardware, so that once the adapter is bought, all licensing issues go away. In that way the owner could fiddle with the

I'm trying to setup a demo where my Asterisk box with a TDM01B (FXO) answers an incoming call and forwards that call to a SIP softphone (X-lite.) Seems all is built/installed okay: # ztcfg -vv Zaptel Configuration ====================== Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) 1 channels configured. I'm pretty new at this and the extensions.conf file is eating my

Hello List Ive been trying to use regcontext, but I cant get it to work. Ive setup my sip peers to have the regexten _[0-9]., so that I can capture all registrations in a single extension. But when they register, I can see that the dynamic extension is created, but none of the rest of the code is executed, priority 2-4. Can anyone explain how I should use the regcontext parameter, etc. am I using

Ok, I could have sworn this was fixed in Asterisk 1.2, but it seems in Asterisk 1.4.4, that doing a reload, or even an 'extensions reload' will clear any extensions that have been created by regexten. This is VERY bad! Doug. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi Anyone know how to use regcontext et regexten parameter from sip.conf and can give an example ? thx regards Harry -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090807/ef9ba45e/attachment.htm

Hello guys, Is there a decent click-to-dial CTI which works well with Asterisk? We have vanilla asterisk implementation and I have tried a few (ADA, Outcall etc) but they have poor documentation and don't work very well. We are looking for an application which can allow us to dial a number from Outlook and IE/Firefox for outbound calls and get a pop-up for inbound calls with call history