similar to: Multiple telephone participants

Hello Group, Current Setup: - Eicon Quad BRI ISDN Card. - 4 ISDN BRI (Telco: Telstra) Onramp2 services. - Mode: Point2Point. - 100 Indial Number ranges. Full National Number (9 digit format): BAAAAAAXX where: B (Area code): 2/3/7/8 A (Normal Numbers) X (99 Indial extensions) eg: BAAAAAA00 BAAAAAA20 etc Requirement: - To be able send Indial numbers as Caller ID when dialing out. Configration:

We have a 100 number indial range and every so often get fax calls on our voice numbers (our fax number isn't in the 100 number range). If you just hang up the sending fax will often try a few times before finally giving up. Our outgoing fax is connected to the PBX (not asterisk), and we can do a blind transfer to that which will print it out, but right now the fax is printing a misdialled

Hello, I am using chan_capi and would like * to behave differently depending on the MSN the caller dials. Is there a way to route incoming ISDN calls to different contexts based on the MSN dailled? I have tried something like msn=1234 incomingmsn=1234 context=msn1 msn=4567 incomingmsn=4567 context=msn2 in capi.conf but with no results. Thanks for any hints. -Walter -- Walter Doerr

Is there a way of temporarily suspending participants in a conference? Say I have 5 users, A,B,C,D,E and I wish A, B and C to have a discussion in the confbridge session that D and E can't hear, is there a way to suspend D and E for a while (whilst they are played music or whatever) and later join them back in? Failing that, I was considering kicking them and using an AGI script to rejoin

Hi Asterisk Users, we are using Asterisk 1.4.18.1 on debian 4.0 etch, pwlib 1.10 and openh323 1.18. We are using MeetMe for conference calls and with two participants there is no echo problems, but with more than two participants there is a lot of echo that sometimes disappear for a short time and all function well. Someone have some suggestions?? Do you ever used app_conference

The meetmeadmin() dialplan function lets you specify a user to mute, un-mute or kick. But how do you get a list of users in your dialplan? When a user joins a conference, the user number assigned is "the last user number +1." If you have a long running conference with callers joining and leaving all the time, this can grow to be a large number. I want to be able to

Hey Guys I have an indial range of 61211118[01234]X being trunked sip to xxx.yyy.189.65 Now I want to break this down into 612111180x going to xxx.yyy.188.145 and 612111184x going to xxx.yyy.189.199 reminder being used for fax->email etc etc etc I have created the outbound routes and sip trunks I can see that all the sip trunks are up I can see the outbound routes are there and

Hi, folks. I have a problem using Asterisk 1.2. I create conferences using app_meetme and Zap channels, and for every participant I run the script defined by AGI_BACKGROUND_SCRIPT to be able to listen and react to DTMF tones. As the docs tell me, when using the AGI background script one loses the ability to control the meetme conference via the command line so for muting conference participants I

This is an email to all New York based Asterisk users. For some time it's been bugging me that we don't have a local contact point/user community. If you are involved in Asterisk and in NY/NJ shoot me an email, I'm going to try and revitalize either meetup.com or some other shared environment for Asterisk users in NY. Shoot me an email and once I get an idea of how many

Hello all, I have the following working (somewhat) setup: TELCO | | E1 (30 Chan -- TE210 SPAN 2) | | Asterisk box 1.6 with DAHDI drivers loaded Digium TE210p | | E1 (30 Chan -- TE210 SPAN 1) | | NEC PBX

Has anybody had a problem registering their IAXtel account? I just signed up for an account and followed the documentation on iaxtel.org and my registration is always rejected. When I type "iax show registry", I get the following output: Host Username Perceived Refresh State 12.37.165.130:5036 xxxxxxxx <Unregistered> 60 Rejected

Grandstream owners, I just noticed that there is a new firmware release, for those that are interested: http://www.grandstream.com/BETATEST/ Doug

Hi All, The issue appearing at the random for confbridge module i.e. in some cases if a participant joins the confbridge, he/she unable to hear others which make him/her to hangup the call and redial the bridge again. By joining the bridge second time, participant able to hear the other participants. Any ideas which may causing this issue? As Asterisk version I'm using is 11.2.1. Is it a

Hi, Looking at https://wiki.asterisk.org/wiki/display/AST/ConfBridge+Configuration there is an option for admin_toggle_mute_participants however the non admin users can still toggle toggle_mute. Is there any option for the admin to disallow non admins from using toggle_mute to unmute themselves? If there isn't such an option on there any devs here that can ping me off line what it would

On 4/26/20 10:48 AM, Dovid Bender wrote: > Hi, > > Looking at > https://wiki.asterisk.org/wiki/display/AST/ConfBridge+Configuration there > is an option for admin_toggle_mute_participants however the non admin > users can still toggle toggle_mute. Is there any option for the admin > to disallow non admins from using toggle_mute to unmute themselves? If > there

http://bugs.digium.com/view.php?id=4760 If you use the L() option on dial and say the latest CVS-HEAD in the past month you're potentially getting screwed out of a lot of money. We originally wrote the L() option for dial and it worked great till someone came along and hijacked the timer for something else thus causing the L option to fail/reset the timer to zero thus causing it to

Software zur Erkennung von "Spam" auf dem Rechner priamus.teamware-gmbh.de hat die eingegangene E-mail als m?gliche "Spam"-Nachricht identifiziert. Die urspr?ngliche Nachricht wurde an diesen Bericht angeh?ngt, so dass Sie sie anschauen k?nnen (falls es doch eine legitime E-Mail ist) oder ?hnliche unerw?nschte Nachrichten in Zukunft markieren k?nnen. Bei Fragen zu diesem

I am having problems getting incoming caller id to work on a Telstra Onramp 10. I have changed "/DEFAULT_CIDRINGS 2"/ Is there something i'm missing ? My Cisco 7960 just shows "asterisk" Thanks, Nathan [zapata.conf] context=incoming usecallingpres=yes relaxdtmf=no rxgain=0.0 txgain=0.0 busydetect=no pridialplan=local usecallerid=yes callerid=asreceived

Hello, We have a quadBRI in NT mode with bri_cpe_ptmp signalling and when connect a ISDN telephone to this nothings happen. What can I do? My config files are this: Zaptel.conf: loadzone=es defaultzone=es # qozap span definitions # most of the values should be bogus because we are not really zaptel span=1,1,3,ccs,ami span=2,1,3,ccs,ami span=3,1,3,ccs,ami span=4,1,3,ccs,ami bchan=1,2 dchan=3

I connect Asterisk to my analog PBX using X100P. In my analog PBX, I need to dial 0 (zero) to pick up the line. How can I use Dial command to dial (0 + telephone number) directly? I used exten => 10,1,Answer() exten => 10,2,Dial(Zap/1/0) exten => 10,3,Hangup It works, but I need to dial 10 and after the ring tone, the telephone number How can I do?