[asterisk-users] gigasets A580IP Recall Button

Hi I am trying to get the recall button working for the gigasets What settings do i need to set in the advance settings?   Zakir-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us

[asterisk-users] Any way to stop Playtones(dial) when the user presses a key, emulating a CO's behavior?

The subject says it all. I'm betting there's a way to do it, but so far I haven't found the dialplan runestone via web searching. Thanks. b. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to As

Re: [asterisk-users] Soundpoint IP 430 -- discontinued.

On Nov 5, 2010, at 11:13 PM, Michael Graves wrote: > On Fri, 5 Nov 2010 23:09:19 -0400, Fred Posner wrote: > >> Curious Michael... Why won't you subject people to the 335's? I love these >> phones for a call center deployment. The are a fantastic agent phone... let >> alone a great phone for ki

Re: [asterisk-users] Soundpoint IP 430 -- discontinued.

On Fri, 5 Nov 2010 23:09:19 -0400, Fred Posner wrote: >Curious Michael... Why won't you subject people to the 335's? I love these >phones for a call center deployment. The are a fantastic agent phone... let >alone a great phone for kitchens, break rooms, lobby, etc. But I love them as >an agent

Re: [asterisk-users] Soundpoint IP 430 -- discontinued.

On Nov 5, 2010, at 10:45 PM, Michael Graves wrote: > On Fri, 5 Nov 2010 19:02:43 -0400, Mike wrote: >> >> On Nov 5, 2010, at 5:35 PM, Ken D'Ambrosio wrote: >> >>> Hey, all. I'm in the middle of a rollout, and just learned that the >>> SoundPoint IP 430 -- my favorite mid-range phone -- has bee

Re: [asterisk-users] Soundpoint IP 430 -- discontinued.

On Fri, 5 Nov 2010 19:02:43 -0400, Mike wrote: > >On Nov 5, 2010, at 5:35 PM, Ken D'Ambrosio wrote: > >> Hey, all. I'm in the middle of a rollout, and just learned that the >> SoundPoint IP 430 -- my favorite mid-range phone -- has been discontinued. >> The heir apparent is the SoundPoint IP 450

Re: [asterisk-users] Short rings for extensions when part of the Queue

On Fri, 5 Nov 2010 01:18:30 -0400 Bruce B wrote: > > What I noticed is that when someone misses a call on the queue, it > > switches over to the next person, but almost immediately (after a > > short ring) it breaks off and cancels that call because of a > > timeout. I think it's a matter of tim

Re: [asterisk-users] Call using password

On Fri, 5 Nov 2010 22:05:28 -0200 Flavio Miranda wrote: > What is the easier way to make call using a password? I have > A2billing but its authentication is too big, I would like four > digits long. Something like that: In any extensons, the user dial the > password and make call. Thanks in ad

[asterisk-users] Call using password

Hi, What is the easier way to make call using a password? I have A2billing but its authentication is too big, I would like four digits long. Something like that: In any extensons, the user dial the password and make call. Thanks in advanced! Att, Flavio Roberto Miranda MSN:flaviormira

Re: [asterisk-users] GROUP_COUNT not counting correctly

On Fri, 05 Nov 2010 15:27:11 +0100 Jonas Kellens wrote: > this is a test to add a channel to multiple GROUPs. It can't be done. $ sudo asterisk -rx 'core show function GROUP' -= Info about function 'GROUP' =- [Syntax] GROUP([category]) [Synopsis] Gets or sets the channel group. [Descrip

Re: [asterisk-users] Soundpoint IP 430 -- discontinued.

-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Fred Posner Sent: Friday, November 05, 2010 5:38 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Soundpoint IP 430 --

Re: [asterisk-users] res_ais Error

Hi, after some test the system don't crash but no members: CLI> ais show clm members = === Cluster Members = = === === -

Re: [asterisk-users] Soundpoint IP 430 -- discontinued.

On Nov 5, 2010, at 5:35 PM, Ken D'Ambrosio wrote: > Hey, all. I'm in the middle of a rollout, and just learned that the > SoundPoint IP 430 -- my favorite mid-range phone -- has been discontinued. > The heir apparent is the SoundPoint IP 450 -- for a low, low, low $130 > more/handset. AND it doe

[asterisk-users] Soundpoint IP 430 -- discontinued.

Hey, all. I'm in the middle of a rollout, and just learned that the SoundPoint IP 430 -- my favorite mid-range phone -- has been discontinued. The heir apparent is the SoundPoint IP 450 -- for a low, low, low $130 more/handset. AND it doesn't look as nice. Ouch. Does anyone have any recommenda

Re: [asterisk-users] Funky IAX behavior between 1.4 and 1.8

_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren Selby Sent: Friday, November 05, 2010 4:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Funky IAX behavior between 1.4 and 1.

Re: [asterisk-users] Funky IAX behavior between 1.4 and 1.8

On Fri, Nov 5, 2010 at 10:34 AM, Mark Deneen wrote: > On Fri, Nov 5, 2010 at 11:04 AM, Danny Nicholas wrote: > > Hi Gang, > > > > My production box with my DAHDI cards is a 1.4.26 build. I > > have 3 test machines that I do IAX communication with. > > > > Machine 1 is a real Dell P

Re: [asterisk-users] res_ais Error

On Fri, Nov 5, 2010 at 4:44 PM, bakko wrote: > When I load res_ais.so module, the pbx crash (boths) > Generate a backtrace[1] and upload to this thread. [1] https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybe

[asterisk-users] res_ais Error

Hi, I'm trying distributed events with Openais but don't work. I made the test with two asterisk box in the same LAN box A: 192.168.142.246 asterisk 1.6.2.13 BoxB: 192.168.142.248 asterisk 1.8.0 openais.conf: # Please read the openais.conf.5 manual page totem { version: 2 secauth: off threads

Re: [asterisk-users] Alternative to Proxmox

- "Tim Nelson" wrote: > > Hi Everyone, >Is there other comparable products to Proxmox to be used for Asterisk >instances? Ease of use, web interface, and Asterisk/CentOS support would be >ideal. > > There is OpenNode: > http://opennode.activesys.org/ > I've heard good things thus far

Re: [asterisk-users] Alternative to Proxmox

- "Bruce B" wrote: > Hi Everyone, > Is there other comparable products to Proxmox to be used for Asterisk > instances? Ease of use, web interface, and Asterisk/CentOS support would be > ideal. There is OpenNode: http://opennode.activesys.org/ I've heard good things thus far but ha

[asterisk-users] Alternative to Proxmox

Hi Everyone, Is there other comparable products to Proxmox to be used for Asterisk instances? Ease of use, web interface, and Asterisk/CentOS support would be ideal. Thanks -- _ -- Bandwidth and Colocation Provided by http://www

Re: [asterisk-users] Unable to place 2 or more calls to a DID

_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Frager Sent: Friday, November 05, 2010 2:33 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Unable to place 2 or more calls to a DID Hello, I'm having a

[asterisk-users] Unable to place 2 or more calls to a DID

Hello, I'm having a problem trying to do this, and it used to work with Asterisk 1.4. Since Asterisk 1.6 series I have not been able to place more than one call to a DID. I get this message: Skipping dialing interface 'SIP/16034817...@flowroute' again since it has already been dialed I'm

[asterisk-users] Polycom WEB UI configuration - What needs to be put in for basic SIP registration?

Hi Everyone, Configuring a Polycom conference bridge IP 5000 to connect to Asterisk. For some reason I don't see any SIP packets coming in to Asterisk at all. I don't want to use XML or ftp etc for now and just use the Web Interface to get it going with basic features. But the Web UI is a bit conf

[asterisk-users] Unable to place 2 or more calls to a DID

Hello, I'm having a problem trying to do this, and it used to work with Asterisk 1.4. Since Asterisk 1.6 series I have not been able to place more than one call to a DID. I get this message: Skipping dialing interface 'SIP/16034817...@flowroute' again since it has already been dialed I'm

Re: [asterisk-users] Elementary question - accessing feature codes from cell phone

We use DISA (http://www.voip-info.org/wiki/view/Asterisk+cmd+DISA) to access our entire [features] context from our cell phones. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Regal Sent: Friday, November 05, 2010 11:11 AM To: ast

Re: [asterisk-users] Short rings for extensions when part of the Queue

Sorry, I am not following. If an extension rings for 15 or 16 seconds and then waits for 2 or three seconds what difference does the being divisible make? Is there something internal to Asterisk that makes the Retry time dependent on Time Out (also known as Ring Time)? P.S. I think the 15 seconds

Re: [asterisk-users] Asterisk 1.8 Installation Problem

I`ve disabled chan_ooh323 and res_adsi and it worked . Bogdan -Original Message- From: Bogdan Sarandan Sent: Friday, November 05, 2010 12:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 1.8 Installation Problem Thanks for the answ

Re: [asterisk-users] Asterisk 1.8 Installation Problem

Thanks for the answer. All of those libraries are already installed and it's still not working. Package libstdc++-devel-4.1.2-48.el5.i386 already installed and latest version Package matching libxml2-devel-2.6.26-2.1.2.8.i386 already installed. Checking for update. Package openssl-devel-0.9.8e-

Re: [asterisk-users] Elementary question - accessing feature codes fromcell phone

On Fri, Nov 05, 2010 at 12:49:45PM -0400, John Regal wrote: >Anyway, I think the idea of replicating the function into an extension will >work. Any pointers on the best way to accomplish this? I created a new >extension but am unsure what to do next. I thought about the FollowMe >feature but I wou

[asterisk-users] No audio with gtalk client behind http proxy

Hi all, I'm trying to establish jingle call in this network scenario: Asterisk -> NAT -> Internet -> HTTP_PROXY -> GTalk client The call is received and answered in gtalk but there is no audio in the call. I suppose it could be related to the support for relay candidates in asterisk jingle

Re: [asterisk-users] Elementary question - accessing feature codes fromcell phone

Thanks for the quick response! I have had a lot of issues in the past with DTMF. Anyway, I think the idea of replicating the function into an extension will work. Any pointers on the best way to accomplish this? I created a new extension but am unsure what to do next. I thought about the FollowMe

Re: [asterisk-users] Phones slow to ring

It worked! I['ll have to figure out how to add the dial string to the phone. Thanks a bunch for your help On Thu, Nov 4, 2010 at 9:04 PM, Mark Phillips wrote: > I would second that. > > If you don't set a dial string in your handset then it waits for N > seconds before submitting the call. Pre

Re: [asterisk-users] Elementary question - accessing feature codes fromcell phone

DTMF sent from cell phones are usually not well recognized at the asterisk end. The main reason for this is that cell phones transmit out-of-band DTMF, which by the time reaches an asterisk server traveling through cell towers, their equipment, various VoIP carriers etc. is usually drifted away fro

Re: [asterisk-users] Funky IAX behavior between 1.4 and 1.8

On Fri, Nov 5, 2010 at 11:04 AM, Danny Nicholas wrote: > Hi Gang, > > My production box with my DAHDI cards is a 1.4.26 build.  I > have 3 test machines that I do IAX communication with. > > Machine 1 is a real Dell POWEREDGE 1500 running CENTOS running 1.4.30. > Machine 2 is a SUSE 1

Re: [asterisk-users] Short rings for extensions when part of the Queue

On Fri, Nov 5, 2010 at 10:38 AM, Bruce B wrote: > Yeah, I think I had it set to 2 seconds and that creates that short ring on > another extension. > Thanks, The point was that 14 and 16 are divisible by 2 (evenly) while 15 is not. -- _

Re: [asterisk-users] Asterisk default sound files

_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Erol Demir Sent: Friday, November 05, 2010 10:22 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk default sound files Hi, I installed BigBlueButton

[asterisk-users] Asterisk 1.8 Installation Problem

Hi, We want to upgrade both our servers to asterisk 1.8, the one from Romania and the one from Chicago, but for the moment I`m trying to install Asterisk 1.8 on a test machine running CentOS 5.5 with the kernel: Linux asterisk3 2.6.18-194.17.4.el5PAE #1 SMP Mon Oct 25 16:35:27 EDT 2010 i686 i68

[asterisk-users] Asterisk default sound files

Hi, I installed BigBlueButton and I want to change default conference playbacks. Is it possible ? if Yes, how :) Thank you. Bu elektronik postada bulunan tum fikir ve gorusler ve ekindeki dosyalar sadece adres sahip/sahiplerine ait olup, Yasar Toplulugu Sirketleri bu mesajin icerigi ile il

Re: [asterisk-users] Elementary question - accessing feature codes fromcell phone

_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Regal Sent: Friday, November 05, 2010 10:11 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Elementary question - accessing feature codes fromcell phone H

[asterisk-users] Elementary question - accessing feature codes from cell phone

Hi, please forgive me for this (hopefully) simple question. I cannot seem to find an answer or solution while searching around. I want to be able to call in to my server using my cell phone and be able to set call forwarding for my extension and enter a phone number and also be able to call in to

[asterisk-users] Funky IAX behavior between 1.4 and 1.8

Hi Gang, My production box with my DAHDI cards is a 1.4.26 build. I have 3 test machines that I do IAX communication with. Machine 1 is a real Dell POWEREDGE 1500 running CENTOS running 1.4.30. Machine 2 is a SUSE 11.1 VM running 1.4.30. Machine 3 is another SUSE 11.1 VM running 1.

[asterisk-users] Using Dial() but no CDR is generated for this outcall

Hi, as far as I know my problem is not a bug but wanted behaviour. Let's assume the following dialplan: exten => 123,1,Answer exten => 123,n,Dial(DAHDI/g0/00492112233,20,g) [...] exten => 123,n,Hangup I do an dial-in with my SIP-Client (or phone). The Dial-Application starts the outdial and I g

[asterisk-users] How to append custom option to Contact: header on outgoing SIP INVITE msgs?

Hi list, My need is to append a site specific parameter to the Contact: header on all INVITEs exiting * via a SIP trunk. I'd like it to look something like this: Contact: where SITE-ID=us.here is set in a config file that * parses on startup. Or in a Dial() command option? Or I don't care ex

[asterisk-users] Asterisk in the third world - Astricon 2010 keynote follow-up

Friends, After listening to Mark Summer's keynote at Astricon (hopefully soon on the Astricon web site) I think we should come back to the discussion he started. Mark talked about using Open Source in general and Asterisk in particular in third world projects as well as in emergencies in other c

[asterisk-users] CDMA Media gateway EVRC codec

Hi, I want to use asterisk as a media gateway for a CDMA application. I need support for EVRC codec. Anyone know which cards support EVRC? Thanks, Thanks, Dave -- _ -- Bandwidth and Colocation Provided by ht

Re: [asterisk-users] Short rings for extensions when part of the Queue

Yeah, I think I had it set to 2 seconds and that creates that short ring on another extension. Thanks, On Fri, Nov 5, 2010 at 9:47 AM, Mark Deneen wrote: > On Fri, Nov 5, 2010 at 1:18 AM, Bruce B wrote: > > Chad, > > You are absolutely right on this one. I had setup the Queue time out for > >

[asterisk-users] GROUP_COUNT not counting correctly

Hello, this is a test to add a channel to multiple GROUPs. this is my dialplan : exten => s,n,NoOp(groepcount = ${GROUP_COUNT(40)}) exten => s,n,Set(GROUP(40)=40) exten => s,n,NoOp(This channel is member of : ${GROUP_LIST()}) exten => s,n,NoOp(groepcount = ${GROUP_COUNT(40)}) exten => s,n,NoOp

[asterisk-users] VoIP Uses Conference: Friday November 5th: Cloud Computing

This is just a brief reminder that today's VUC call will be about cloud computing with some emphasis on voice applciations: We have assembled a small panel of experienced people to discuss the mattering, including: Eric Chamberlain, Founder of RF.com, Presenter to Astricon 2009 on running Asteri

Re: [asterisk-users] MixMonitor

Hi, marked -> noticed. I do not know where it comes from, my CPU goes from 2% to 60-70% at a command Dial (sip) + MixMonitor. I have an Intel (R) Core (TM) 2 Duo CPU e4...@2.40ghz 2010/11/5 Norbert Zawodsky > Am 05.11.2010 10:16, schrieb Mickael MONSIEUR: > > none ? > > > > > > 2010/11/5 Micka

Re: [asterisk-users] Short rings for extensions when part of the Queue

On Fri, Nov 5, 2010 at 1:18 AM, Bruce B wrote: > Chad, > You are absolutely right on this one. I had setup the Queue time out for > agent set to 15 seconds and retry to 2 seconds. So, I think during those two > seconds Asterisk for some crazy reason hits another extension and then comes > back to

[asterisk-users] MoH streamers for asterisk

El 04/11/10 17:14, Tzafrir Cohen escribió: > In the 'files' mode Asterisk plays the music separately for each > channel. If you use mpg123 or any other streamer, there is a single > stream per class. > A single stream per class sounds like good efficiency. Could you please tell me what streamers c

Re: [asterisk-users] upgrade 1.6 -> 1.8: wrong password!

Dear Paul, I submitted the issue to the tracker. ID 0018263 Thanks pepesz On Thu, Nov 4, 2010 at 8:46 PM, Paul Belanger wrote: > On Thu, Nov 4, 2010 at 3:24 PM, pepesz wrote: > > > WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", > > nonce="5fcd5fa1 " > > > > I'm surprised to

Re: [asterisk-users] Mobile Phones and Asterisk

Hi, one way to solve the problem with Mailbox or that Message that get's played when busy/not available (same happens with Orange in Austria and other providers) you can implement something similar to what Elastix/FreePBX has. "Confirm call" - this will let the caller think it's still ringing wh

Re: [asterisk-users] MixMonitor

On 5 Nov 2010, at 01:22, Mickael MONSIEUR wrote: > Have you noticed a marked increase in CPU load when using MixMonitor? Since when? 1.6.2.9-1? 1.6.2.8? 1.0? S -- _ -- Bandwidth and Colocation Provided by http://www.api-digita

Re: [asterisk-users] MixMonitor

Am 05.11.2010 10:16, schrieb Mickael MONSIEUR: > none ? > > > 2010/11/5 Mickael MONSIEUR > > > Hi, > Have you noticed a marked increase in CPU load when using MixMonitor? > > I use PHPAgi and Asterisk 1.6.2.9-2. > > Mickael. > > Obviously, if th

Re: [asterisk-users] MixMonitor

none ? 2010/11/5 Mickael MONSIEUR > Hi, > Have you noticed a marked increase in CPU load when using MixMonitor? > > I use PHPAgi and Asterisk 1.6.2.9-2. > > Mickael. > -- _ -- Bandwidth and Colocation Provided by http://www.ap