Re: [asterisk-users] Notifying missed calls

g: *When the called party hangs up*, continue to execute commands in the current context at the next priority On Wed, Nov 3, 2021 at 4:39 PM Luca Bertoncello wrote: > Am 03.11.2021 um 21:34 schrieb Antony Stone: > > On Wednesday 03 November 2021 at 21:29:46, Luca Bertoncello wrote: > > > >> I

Re: [asterisk-users] Community forum ?

e anything >> open source around in this field at all. >> >> > Sangoma acquired Digium. > > How this impacts Asterisk is answered by the community FAQ: > > https://wiki.asterisk.org/wiki/display/AST/Sangoma+and+ > Digium+Join+Together+FAQ > > tl;dr: it doesn't. &g

Re: [asterisk-users] Community forum ?

oon there won't be anything > open source around in this field at all. > > On Thu, 30 Aug 2018 11:14:33 -0400, > Carlos Rojas wrote: > > > > [1 ] > > [1.1 ] > > [1.2 ] > > Is the list going to be the same after sangoma take over digium? > > >

Re: [asterisk-users] Community forum ?

Is the list going to be the same after sangoma take over digium? On Thu, Aug 30, 2018 at 11:12 AM, Joshua Colp wrote: > On Thu, Aug 30, 2018, at 12:05 PM, sean darcy wrote: > > I see a lot of tag lines on posts for the Asterisk Community Forum. Is > > that forum supposed to supersede this

Re: [asterisk-users] Community forum ?

I don't think so. On Thu, Aug 30, 2018 at 11:05 AM, sean darcy wrote: > I see a lot of tag lines on posts for the Asterisk Community Forum. Is > that forum supposed to supersede this mailing list ? > > sean > > > -- > _ > --

Re: [asterisk-users] getting invites to rtp ports ??

Hi Probably somebody is trying to hack your system, you should block that ip on your firewall. Regards On Wed, Aug 29, 2018 at 9:34 AM, sean darcy wrote: > I'm getting invites to very high ports every 30 seconds from a particular > ip address: > > Retransmitting #10 (NAT) to

Re: [asterisk-users] Pass through registration / proxy

Hi You could use kamailio +asterisk On Tue, Apr 10, 2018, 9:25 PM Telium Technical Support wrote: > I need to create a SIP proxy to be placed in front of a legacy PBX. When > a phone registers with the proxy, I would like Asterisk to register with > the PBX behind it. (To

Re: [asterisk-users] Using g729 now that patents have expired

Hi You can uses: http://asterisk.hosting.lv/ On Tue, Feb 7, 2017 at 4:47 PM, Steve Edwards wrote: > Now that the g729 patents have expired, how do we use g729 in Asterisk? > > Will Digium be releasing a g729 codec for 'free' use or do we download the > 'free'

Re: [asterisk-users] sip show [general]?

Hi You can do sip show settings On Jan 11, 2017 5:32 AM, "Thufir Hawat" wrote: > I appreciate that the console lets you see the details for a peer with > "sip show peer foo". Certainly, I can look in sip.conf to see the > [general] context, but can I output those

Re: [asterisk-users] implementing call center using asterisk

Hi You can use, gnudialer, vicidial, goautodial. On Wed, Jun 22, 2016 at 12:47 PM, Goke Aruna wrote: > hello all, > I am looking for an implementation of a 10 man call center. low cost > license or GPL will be preferred. > I will be glad for your help. > Regards > > -- >

Re: [asterisk-users] SPA112 flapping

Hi It sounds like a keep alive issue On Sun, Jun 19, 2016, 4:39 PM Gergo Csibra wrote: > Friday, June 17, 2016, 11:56:34 PM, Mike wrote: > > > I've got a device that seems to become unreachable for about 2 minutes, > every > > hour. From what I can tell, it isn't due to

Re: [asterisk-users] Recommendations for free virtual server tech and Asterisk?

I have tried with xen and kvm both are working fine. On Wed, Apr 6, 2016 at 3:44 PM, Loic Chabert wrote: > Hello, > > Work well with kvm and centos 7. > Some ajustements has to be made with systemd. > > I'm using it in production since 1.5 year now, no issue to report. >

Re: [asterisk-users] PRI error "ROSE REJECT"

Hi Did you activate the pri debug on the cli asterisk? On Thu, Mar 24, 2016 at 12:59 PM, Carlos Chavez wrote: > We've been having some problems with an E1 PRI line for a few days. We > get the following errors: > > [Mar 24 10:13:39] ERROR[22009] chan_dahdi.c: PRI

Re: [asterisk-users] FAX Detection.

Hi I have used sangoma cards, but I know that openvox, is shipper than Sangoma. On Wed, Feb 24, 2016 at 1:10 PM, Aziz TestAccount wrote: > Hi All, > > I'm looking for a PSTN Card that I can use with my Asterisk Server to > achieve the following goal : > > 1. Detect FAX

Re: [asterisk-users] Looking for Asterisk Consultants & Experts

Hi I am Carlos Rojas I am asterisk dCAP, 2171 What do you need? On Wed, Sep 2, 2015 at 7:40 AM, Shahid H <shah...@gmail.com> wrote: > Hello, > > Can someone recommend me where is best place to find Asterisk > Expert/Consultant for freelance work? > > If yo

Re: [asterisk-users] Grandstream GXP2140

Hi If you are going to use only a phone, it's fine, but if you are going to install a lot of grandstream's phones, probably you network traffic is going to increase a lot. On Wed, Apr 15, 2015 at 3:12 PM, dsi...@hcmr.gr wrote: I'm working with GXP2130. About 12 phone on gigabit with PC after

Re: [asterisk-users] Gateway Eurotech

I Ricky I have worked with this gateway few years ago, it's good product, they have gateways with PRI connectors and SIP. The quality is good, and it woks good with asterisk or regular PBXs. On Thu, Mar 26, 2015 at 11:16 PM, ricky gutierrez xserverli...@gmail.com wrote: Hi, I know there are

Re: [asterisk-users] Popup URL for outgoing calls.

You can use vtiger or sugar Both are working with asterisk. On Fri, Jun 27, 2014 at 9:04 PM, Prakash N prakas...@tevatel.com wrote: What CRM your going to use? With regards N.Prakash From: Rusty Newton Sent: ‎28-‎06-‎2014 01:01 AM To: Asterisk Users Mailing List - Non-Commercial

Re: [asterisk-users] SIP Softphone

Zoiper gsm -Original Message- From: Mark Robinson vsysnetw...@gmail.com Sender: asterisk-users-bounces@lists.digium.comDate: Sun, 8 Jun 2014 17:01:54 To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List -

Re: [asterisk-users] Verbose only one context

You can do this sip set debug ip x.x.x.x On Wed, Mar 26, 2014 at 11:28 AM, Rafael dos Santos Saraiva rafaels...@gmail.com wrote: Hi It's possible in Asterisk 1.8 enable verbose only in one context or extension? thanks Att, *Rafael dos Santos Saraiva*

Re: [asterisk-users] VoiceMail Issue

Hi Could you send us the logs from the asterisk? Carlos On Sat, Mar 8, 2014 at 4:03 AM, Phil Daws ux...@splatnix.net wrote: Any ideas on why this may not be working please ? - Original Message - From: Phil Daws ux...@splatnix.net To: Asterisk Users Mailing List - Non-Commercial

Re: [asterisk-users] Integration with outlook

Hi Yes, there is, I am using http://outcall.sourceforge.net/ it's opensource. On Tue, Jan 28, 2014 at 2:13 PM, bilal ghayyad bilmar...@yahoo.com wrote: Hello; Is there a method way to be able to dial the phone number through asterisk from the outlook email contact? Regards Bilal

Re: [asterisk-users] IAX and Variables

I thunk so Let me see -Original Message- From: Mikhail Lischuk mlisc...@itx.com.ua Sender: asterisk-users-bounces@lists.digium.comDate: Tue, 08 Oct 2013 01:08:22 To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing

Re: [asterisk-users] Checking messages from outside the network

Are talking about of prepend message? Because for listening the messages, you can use VoiceMailMain Carlos Rojas On Wed, Sep 11, 2013 at 11:37 AM, jg webaccou...@jgoettgens.de wrote: Have you considered using VoiceMailMain()? jg

Re: [asterisk-users] VM notification to multiple email recipients

Hi You can do this, http://mike.eire.ca/2012/02/03/asterisk-1-8-vm-multiple-emails/ If you are using asterisk 1.8 On Wed, Sep 11, 2013 at 1:55 PM, Mike Diehl mdiehlena...@gmail.com wrote: Hi all, I've got a user who wants to receive voicemail notifications at two different email

Re: [asterisk-users] Am I being hacked?

Hi You should install something like fail2ban Regards On Sun, Aug 18, 2013 at 5:41 PM, Ira i...@extrasensory.com wrote: Hello Asterisk-users, [2013-08-18 05:56:29] NOTICE[17089][C-00a8] chan_sip.c: Failed to authenticate device 390sip:3...@xx.xx.xxx.xxx ;tag=2762c06e

Re: [asterisk-users] Freeswitch with Digium T316 timed out, T316 timed out

My friend, You are in a wrong list, this an asterisk list, you should to be in freeswitch list Kind Regards On Thu, Aug 8, 2013 at 10:39 AM, Rajat toshniwal rajat.toshni...@tekmindz.com wrote: ** Hi I am trying to deploy freeswitch with Digium TE121 card for my office setup, but it is

Re: [asterisk-users] Mysql Support int Asterik-11

Hi Asterisk 1.6 and old versions, were using asterisk-addons, since asterisk 1.8 asterisk addon, is included in the asterisk code, you must select it in menu select. Kind Regards Carlos On Wed, Jul 24, 2013 at 8:36 AM, Prashant Abhang abhang_prash...@yahoo.co.in wrote: I have done using

Re: [asterisk-users] Asterisk 1.8 Service: -r does not give CLI

Not it didn't, Did you execute asterisk -r or /usr/sbin/asterisk -r ? If not working did you execute asterisk -gc ? Kind Regards On Mon, Jul 22, 2013 at 10:41 AM, Meadows Hoa meadows_...@yahoo.com wrote: We have Asterisk1.8.11 and can not move to a newer version right

Re: [asterisk-users] Asterisk offline compiling with get_mp3_source.sh

Hi You must copy the directory mp3, to the addons directory, where you put the source asterisk code, and recompile it, again. Kind Regards On Mon, Jul 15, 2013 at 9:25 AM, leonardo collantes leonardo07...@gmail.com wrote: I need to make a Asterisk 18.0's offline compiling, SVN mp3

Re: [asterisk-users] asterisk -rx core show channels + time

Hi You can do, core show channels verbose Kind Regards On Thu, Jun 20, 2013 at 6:45 PM, Joseph syscon...@gmail.com wrote: When I type: asterisk -rx core show channels I usually get Channel Location State Application(Data) SIP/pstn--03

Re: [asterisk-users] dCAP study recommendations

Hi, If you read, O'Reilly - Asterisk - The Definitive Guide - 3rd Edition, you should be ready for take the test. Of course, you must read voip-info too. Carlos Rojas Dcap 2171 On Fri, Jun 7, 2013 at 2:20 PM, Michael Gilleran mgille...@realtyim.comwrote: Greetings. Anyone have any

Re: [asterisk-users] monitoring Asterisk 1.8

I'm using opennms and It's working fine. On Thu, May 9, 2013 at 3:23 PM, motty cruz motty.c...@gmail.com wrote: Hello, i'm looking for suggestions to monitor Asterisk Server? I installed Nagios but no success, I do prefer not to install any web server on the server running Asterisk.

Re: [asterisk-users] monitoring Asterisk 1.8

http://opennms.org/wiki/Installation:Yum On Thu, May 9, 2013 at 4:03 PM, Carlos Rojas crt.ro...@gmail.com wrote: I'm using opennms and It's working fine. On Thu, May 9, 2013 at 3:23 PM, motty cruz motty.c...@gmail.com wrote: Hello, i'm looking for suggestions to monitor Asterisk

Re: [asterisk-users] monitoring Asterisk 1.8

Asterisk, not the server Asterisk in running on. thanks, -Motty On Thu, May 9, 2013 at 1:06 PM, Carlos Rojas crt.ro...@gmail.com wrote: http://opennms.org/wiki/Installation:Yum On Thu, May 9, 2013 at 4:03 PM, Carlos Rojas crt.ro...@gmail.com wrote: I'm using opennms and It's working fine

Re: [asterisk-users] problem

Hi Are you sure that your hard drive sda, is ok? Looks like your hard drive is broken. On Wed, Feb 6, 2013 at 10:30 AM, brahim abidar abidarbah...@gmail.comwrote: Hi every body; I want to intall some softwars working with my Asterisk server and I get these erreurs : * error: cannot

Re: [asterisk-users] g723 transcoding

Hi Look at it this link http://asterisk.hosting.lv/ Kind Regards On Thu, Jan 24, 2013 at 10:34 AM, Richard Kenner ken...@gnat.com wrote: It appears that there are no transcoders from g723 to anything else in Asterisk 10.7.1. Does anybody know how to fix that? --

Re: [asterisk-users] - configure ring group

Maybe, You can do that, with queues, and ringall strategy. On Wed, Dec 5, 2012 at 4:53 PM, Leandro Dardini ldard...@gmail.com wrote: You can dial all the extensions at once, putting all them in the dial string, separated by . There is no other method. Leandro 2012/12/5 Paolo De Michele

Re: [asterisk-users] Allowing peers from specific subnet only

Hello In SIP.find you can to use Deny=0.0.0.0/0.0.0.0 Permit=192.168.1.25/255.255.255 Regards On Nov 19, 2012 7:12 AM, bilal ghayyad bilmar...@yahoo.com wrote: Hi; How I can make my configuration to allow the sip phones only from specific IP addresses range (for example from 192.168.10.1 -

Re: [asterisk-users] multitenanat third party app

Hi You will need change the names for your extensions 101-company_a 102-company_a ETC On Wed, Oct 31, 2012 at 2:23 PM, Darin Iv adari...@gmail.com wrote: Is it possible to bul multitenant system using some third party opensouce application My design is like this. Company A: Context

Re: [asterisk-users] blocking incoming call - asterisk 1.8

Hello Yes, has a berckeley database, wirh function blackllist Regards On Oct 9, 2012 12:51 AM, Joseph syscon...@gmail.com wrote: Can someone refresh my memory how blocking incoming call works based on caller ID in Asterisk 1.8? If I remember correctly in asterisk 1.4 it was possible to block

Re: [asterisk-users] Asterisk 1.8.10

Hello You should be modify the volume in the file, there are several software for that, like wavepad . Regards On Mon, Oct 1, 2012 at 2:52 PM, Danny Nicholas da...@debsinc.com wrote: AFAIK, there is still not a MOH volume control. What I did was to take my moh wav files and run them

Re: [asterisk-users] Remote SIP Extension Best Practices

Hi Ok, I think vpn is good way, but , you can use tls that uses certificates, and srtp for media encriptatio, in sip protocol. Regards On Sep 29, 2012 12:59 PM, Chris Nighswonger cnighswon...@foundations.edu wrote: On Sat, Sep 29, 2012 at 12:41 PM, Carlos Rojas crt.ro...@gmail.com wrote

Re: [asterisk-users] Hangup not detected

Hello In indications.com are the tones for several countries On Sep 18, 2012 4:34 AM, Mehdi Rahimi mrm.ci...@gmail.com wrote: Hi AJS, Thank you for your reply , I am using this in IRAN so please guide me what to do and and explain me more. Look forward to hearing from your side. Regards,

Re: [asterisk-users] Check for the voicemail

Hello Check voicemail.conf maxmsg = 100 And change it. On Tue, Aug 21, 2012 at 12:52 PM, Danilo Dionisi dionisi.dan...@gmail.com wrote: I'm sorry, I haven't been clear. I do not have to check the inbox on Asterisk, but I have to check the free space on a particular mailbox of Exchange

Re: [asterisk-users] Hosted Softswitch Integration

Hello I think you must change type = peer insecure=invite,port qualify=yes ; for monitor the ip Regards On Fri, Aug 17, 2012 at 2:11 PM, Selecstine Bucci Anukwu buchal...@gmail.com wrote: Hello Everyone, We are trying to integrate a hosted soft-switch to an Asterisks server and the error

Re: [asterisk-users] Segmenting A Configration File

Hi Have you seen thirdlane? Thirdlane has a multitenant version. Regards On Aug 11, 2012 11:11 AM, Carlos Alvarez car...@televolve.com wrote: On Sat, Aug 11, 2012 at 3:16 AM, Kannan vasdevelo...@gmail.com wrote: I am planning a multi-tenant VoIP services system with Asterisk, using

Re: [asterisk-users] Voice Mail beep / tone detection

Hello You will need to do, something like [outbound] exten = s,1,NoCDR exten = s,n,AMD exten = s,n,GotoIf($[${AMDSTATUS}=HUMAN]?humn:mach) exten = s,n(mach),WaitForSilence(2500) exten = s,n,Playback(message-when-machine) exten = s,n,Hangup exten = s,n(humn),WaitForSilence(500) exten =

Re: [asterisk-users] any working calling card solution open source

Hello a2billing works fine Regards On Mon, Jul 16, 2012 at 1:47 PM, Goke M Aruna gok...@gmail.com wrote: hi all, Can someone give me information on any open source asterisk calling card solution? I have laid my hands on astpp, astcc, asterisk-prepaid-0.3.1, agi-ccard.agi without luck. I

Re: [asterisk-users] weird dect beheaviour multiple handsets

Hello Is your server behind nat? This problems sounds me nat problems. Regards On Thu, Jul 12, 2012 at 7:53 AM, Roland o/d Akker aster...@rolandow.com wrote: I have this very specific problem with two dect sets. Problem that I have is one-way audio, in this very rare situation. I am

Re: [asterisk-users] New to Asterisk

Hello http://www.voip-info.org/wiki/view/Asterisk I prefer asterisk under linux sistem works better. Regards On Sun, Jun 17, 2012 at 12:28 PM, Jim Schultz jimschultz...@gmail.comwrote: Greetings, I am interested in learning more ablout Asterisk. Is there a recommended link for getting

Re: [asterisk-users] Line noise/hiss on Openvox A400P card on FXO

Hello Are you using a amd server? Sometimes openvox doesn't work fine with amd processor Regards On Mar 1, 2012 2:07 PM, Dave Platt dpl...@radagast.org wrote: 5. Placing ferrite cores on the phone cables. Do either of the phone lines in question have DSL on them? If so, a ferrite core

[asterisk-users] Virtual Server

Hello everybody someone in this list, has installed asterisk, in a virtual server like proxmox? I'm thinking install some asterisk servers in a machine dell xeon 64 processor, but I'm not sure, about virtual Server software. I heard, about proxmox, but I don't know if works fine. Regards

Re: [asterisk-users] asterisk problem sip

...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Carlos Rojas *Sent:* Saturday, 14 January 2012 3:37 p.m. *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] asterisk problem sip Hi everybody I have been presenting a periodic problem, do not know

[asterisk-users] asterisk problem sip

Hi everybody I have been presenting a periodic problem, do not know if anyone listed has happened something similar,I'm using the asterisk, asterisk-1.6.2.13, in different locations works well, but every so often fails, hangs on Asterisk server or simply asterisk, SIP requirements do not answer,

Re: [asterisk-users] dialplan - dial command - custom ringtone

Hello Do you use hard phone or softphone? In many ip phones you can change the ring tones or use w option in Dial command Regards On Jan 3, 2012 4:08 AM, Qqblog Qqblog qqb...@ymail.com wrote: i could add r option in dial command. this will generate a ringtone during connection. could i

Re: [asterisk-users] IAX2 woes

Hello Asterisk only says that the iax2 channel don't work maybe you look the iax.conf. you trunk. Is iax I think Regards On Dec 29, 2011 6:49 AM, --[ UxBoD ]-- ux...@splatnix.net wrote: Hello all, I attempted to make a couple of outbound calls this morning and always got the busy tone. I

Re: [asterisk-users] Monitor Command Records separate channales

Hello, Do you use monitor?, because in asterisk 1.4 to new versions, It's use mixmonitor, in asterisk 1.2 had this mistake. Regards On Wed, Dec 28, 2011 at 10:11 PM, Steve Edwards asterisk@sedwards.comwrote: Un-top-posting, snarky comments inline... On Wed, 28 Dec 2011, Faraj Khasib

Re: [asterisk-users] Interesting attack tonight fail2ban them

Hello, Do you set up, your logrotate in /etc/asterisk ? Do you test that your fail2ban work fine? Regards On Wed, Dec 28, 2011 at 11:07 PM, Michelle Dupuis mdup...@ocg.ca wrote: I happened to be in the cli tonight as some (208.122.57.58) initiated a simple attack - just trying to make long

Re: [asterisk-users] Client - registers but unreachable

Hello, Your blackberry sip client, works in your wifi network? or by blackberry internet? do you set nat=yes if your phone, register by internet? What is your sip.conf? Regards On Wed, Dec 28, 2011 at 11:16 PM, Michelle Dupuis mdup...@ocg.ca wrote: I have a softphone I'm trying on a

Re: [asterisk-users] how to stop hacking of my server

Hello I use fail2ban, and works fine, Regards On Tue, Dec 27, 2011 at 1:54 AM, virendra bhati virbh...@gmail.com wrote: Hi list someone is trying to hack my server . Is there any way by whcih I can stop hacking of my server except iptables ? I want to stop on the basis of sip.conf account

Re: [asterisk-users] GOIP GSM to SIP Gateway?

Hello It is possible but how do you have the dialplan ? In your dial plan you can do that Regards On Dec 20, 2011 2:40 PM, Matt mhop...@gmail.com wrote: Hi, Has anyone here any experiencing with linking an Asterisk PBX to a GOIP GSM to SIP Gateway? We've got inbound calls from the GSM

[asterisk-users] asterisk and heartbeat

Hello everybody I'm setting, heartbeat and asterisk, with rsync, anyone, work them fine? I've been find any information and saw heatbeat + cysnc2 and heartbeat + rdbd, any one has worked any these aplications fine? Best regards --

Re: [asterisk-users] How to monitor SIP Trunk on production server

Hello, Do you saw this solution? http://linuxnotes.us/ Regards On Sun, Dec 18, 2011 at 12:26 AM, virendra bhati virbh...@gmail.com wrote: Hi List, I have asterisk 1.6.2.20 installed at production server, I have 2 SIP voip trunk for making outgoing and DID for incoming to server. My

Re: [asterisk-users] Set Caller Number in E1 PRI ISDN Lines

Hello Did you use callerid(num) in your dial plan? On Dec 16, 2011 7:38 PM, Kaushal Shriyan kaushalshri...@gmail.com wrote: Hi, I am having an E1 PRI ISDN Lines with 30 bearer channels and 1 D Channel with hundred DIDs (Direct Inward Dialing) numbers attached to a Sangoma PRI Card on the

[asterisk-users] SLA and polycom

Hello, every body Anyone set up, the sla sharing line appearances, in asterisk, I'm setting, tha but, don't, work, I change the sla.conf, extensions.conf, and sip.cfg, but don't work fine. Any one, could setup, tha? Regards Carlos Rojas

Re: [asterisk-users] Beginner Question: Remote access

Hello, I use no-ip service, is similar than dyndns.com Best Regards asterisk-l...@puzzled.xs4all.nl wrote: On 09/07/2011 02:17 AM, A Dunor wrote: Hello list, I am a beginner at asterisk. I want to access my asterisk box from my laptop, on a different network (mobile hotspot). The asterisk

Re: [asterisk-users] how to know how many calls are on hold

Can you send the logs in cli console for help you? Regards On Tue, May 17, 2011 at 9:16 AM, virendra ban hati virbh...@gmail.comwrote: hi list, please help me how to know how many calls are on hold. -- - Thanks and regards Virendra Bhati +91-9172341457 Asterisk Engineer

Re: [asterisk-users] OUTBOUND CALLER ID

Hello Do you set your callerid in the context outgoing? [outgoing] exten = _X.,1,Set(CALLERID(num)=4663000) exten = _X.,n,Dial(.. On Mon, May 9, 2011 at 4:45 AM, mahesh katta maheshka...@flexydial.comwrote: Sir , this is not working On Mon, May 9, 2011 at 1:52 PM, A J Stiles

Re: [asterisk-users] I look ARI (Asterisk Recording Interface)

Hello, I use cri http://www.tikalnetworks.com/voip/index.php?cid=38 Best regards On Thu, Jun 24, 2010 at 3:22 AM, Mickael Monsieur mickael.monsi...@gmail.com wrote: Hello Bruce, This module is not reliable on FreePBX? You know if there is a open source web-voicemail for Asterisk? Best

Re: [asterisk-users] Individual PIN Code per Extension

Hello, I use Authenticate command in dialplan. Regards Carlos Rojas On Wed, Aug 19, 2009 at 6:33 AM, James Mutuku listmut...@gmail.com wrote: Hellos, I have astersist 1.2 working with freepbx. I want to tie pin codes to extensions(users). How do I do this? -- Best Regards, James

Re: [asterisk-users] multiple call dialing and playback an message

Hello, You need configure a queue, with agents for that. Regards. On Thu, Aug 20, 2009 at 11:22 AM, kaustuva...@bbsr.syscomes.com wrote: I have tried a lot like as exten = 123,1,Dial(SIP/114SIP/113SIP/115) and all the channels are dialing and if i answered any 3 of one, all the

Re: [asterisk-users] no ring tone

Hello, I never use externhost y use \ externip=public ip And work fine Regards On Fri, Aug 14, 2009 at 9:18 AM, Ott Rose sixfourimp...@hotmail.com wrote: how do i troubleshoot no ring tone. It was working and all i added was the lines below now it doesn't ring. Edit sip_nat.conf for

Re: [asterisk-users] no ring tone

Hello One question In sip.con or sip_additionals.conf, in freepbx, the context of your client do you put nat = yes externip = You put your public ip. Are you sure that? Regards On Fri, Aug 14, 2009 at 10:43 AM, Ott Rose sixfourimp...@hotmail.comwrote: i changed it and still didn't

[asterisk-users] Help for Alcatel asterisk

Hello everybody I have an asterisk with an integration of alcatel pbx, by sip trunk, all calls are fine, but tha calls calls that originate from a analog line, the recipient is not listening, and that if they hear the call originates, the lines are E1 in alcatel pbx. When a asteris user call to

Re: [asterisk-users] Fwd: User Authentication in sip.conf

Hello, In your sip.conf You need host=sip.xxx.com or IP don't work with dynamic Regards On Wed, Aug 12, 2009 at 8:27 AM, harry R rhm.noa...@gmail.com wrote: Dear all, I want to setup the incoming calls, that don't use authentication in sip.conf file. My configurations as

Re: [asterisk-users] Voicemail attachments not working

Hello, Your smtp server is on? Best regards Carlos Rojas On Mon, Jul 6, 2009 at 7:30 PM, Steve Anness steve.ann...@gmail.com wrote: Today I discovered that voicemail attachments are not working on our latest asterisk server (version 1.4.24.1). I have two other asterisk servers that I

Re: [asterisk-users] Trunk SIP and configuration

Carlos Rojas On Wed, Apr 1, 2009 at 10:45 AM, ludo perrot ludoper...@gmail.com wrote: hello, I am beginning to asterisk. I have a sip trunk access to operator and VPN access with operator. i booked 10 sda numbers. IP adress asterisk : 192.168.600.1 IP adress operator : 192.168.700.50 i can

Re: [asterisk-users] having problems with asterisk

Hello asterisk -vvvgc Regards On Wed, Dec 10, 2008 at 7:45 PM, Scott Berry n7...@northlc.com wrote: Hello there, I am reading Asterisk: The Future of Telephony Chapter four. I am using a Ubuntu box with Asterisk precompiled at this time so I can learn. I am finding that I am

Re: [asterisk-users] canreinvite=yes problem

Hello, canreinvite, don't work with all softphone or hardphone. Regards On Wed, Dec 3, 2008 at 12:38 PM, BERGANZ François [EMAIL PROTECTED] wrote: Someone have a solution for me ? *De :* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *De la part de* BERGANZ François *Envoyé :* mercredi

Re: [asterisk-users] ztd-ethmf

Hello, Do you download zaptel of Redfone website? Best Regards On Fri, Aug 22, 2008 at 6:28 PM, Bill Michaelson [EMAIL PROTECTED] wrote: I expected to find th module ztd-ethmf[.c...] in support of the redfone TDMoE product in my zaptel distro (I have 1.4.11). But it's not there. I am

Re: [asterisk-users] Grandstream

Hello, Do you redirected the rtp ports to your phone? usually 1 - 2 defautl rtp ports Best Regards Carlos Rojas On Thu, May 22, 2008 at 8:48 AM, Phibee Network Operation Center [EMAIL PROTECTED] wrote: I have a problem connecting a Grandstream ipphone to an asterisk

Re: [asterisk-users] Newbie: Two problems with Asterisk Config, Please Help

Hello, Do your verify, the codecs, of both clients, in your sip.conf? What codec do you use? Best Regards On Thu, Mar 20, 2008 at 12:13 AM, Pete Kay [EMAIL PROTECTED] wrote: Hi, I am sorry my questinos are too fundamental. I am new to Asterisk, and hope to catch up as fast as I can.

[asterisk-users] asterisk gateway

Hello everybody Anyone, to know a gateway that works with nextel simm cards? I'm looking for them, in internet, but I did'n look. Best regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To

Re: [asterisk-users] asterisk on Hp servers

Hello, Remember, that linux has problems with irq and pci cards of digium, do you have 3 digium card, and don't have any problems ? Best Regards On Jan 5, 2008 11:01 PM, Eric S López [EMAIL PROTECTED] wrote: Gres, Me, as an asterisk and linux newbie installed redhat 4 (without the gui)

[asterisk-users] Softswitch digim

Hello averybody, I'm looking the softswitch in digium website, anyone test the softswitch? Best Regards ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit:

Re: [asterisk-users] Copy or Make + Make Install

Hello, Only copy the configuration files, extensions.conf, sip.conf, iax.conf , Best regards On Nov 27, 2007 1:27 PM, bilal ghayyad [EMAIL PROTECTED] wrote: Hi List; If I have a running Asterisk on one machine and I need to have another Asterisk on another machine, can I copy

Re: [asterisk-users] TDM400P not answering or making calls

Heloo, I think that your error is: zaptel.conf: --- fxsks=1 loadzone= uk defaultzone = uk zapata.conf: [channels] language=en context=incoming signalling=fxs_ks busydetect=yes busycount=4 callprogress=no relaxdtmf=yes callwaiting=no

Re: [asterisk-users] Locating Asterisk documentation after installation

Hello, Do you have install doxygen? Best regards On 8/13/07, Philipp Kempgen [EMAIL PROTECTED] wrote: MOSBAH ABDELKADER wrote: After installing Asterisk, i have installed the docs by make progdocs. But i don't know where to locate this documentation. Maybe

Re: [asterisk-users] Problems building zaptel 1.4.4

Hello, In Asterisk 1.4 and zaptel 1.4, don't work make linux26, zaptel and asterisk works with kernel 26, and only work with ./configure make menuselect make make install Best Regards Carlos Rojas Lima - Peru On 7/31/07, hugolivude [EMAIL PROTECTED] wrote: Hi, I'm having trouble

Re: [asterisk-users] asterisk or asterisknow

Hello, I prefere, asterisk Best Regards On 7/31/07, Al lists [EMAIL PROTECTED] wrote: You can use both Asterisk or AsteriskNow to have meetme (conference room) On 7/30/07, fateme fatah [EMAIL PROTECTED] wrote: Hi: I want to have conference call service.You offer me use asterisk or

Re: [asterisk-users] Calling to users in other asterisk servers

? Thanks very much!!! On 7/28/07, Carlos Rojas [EMAIL PROTECTED] wrote: Hello, Do you have porf forwardin for SIP protocol in your firewall? SIP: 5060 udp rtp 1 - 2 udp (default) and IAX2 4569 udp Best Regards Carlos Rojas On 7/28/07, Ary Junior [EMAIL

Re: [asterisk-users] Calling to users in other asterisk servers

Hello, Do you have porf forwardin for SIP protocol in your firewall? SIP: 5060 udp rtp 1 - 2 udp (default) and IAX2 4569 udp Best Regards Carlos Rojas On 7/28/07, Ary Junior [EMAIL PROTECTED] wrote: Hi, Im a asterisk newbie and I've configured an asterisk server here in my

Re: [asterisk-users] Asterisk and ATA-186 question-- calling one port from the other port..

Hello, I Check this page: http://www.asterisk.net.au/general/1/ It's very interesting Best Regards Carlos Rojas On 7/18/07, Dmytro Mishchenko [EMAIL PROTECTED] wrote: Tim Reimers wrote: Hi - I need to configure Asterisk (Trixbox 2.2) and my ATA-186 with both ports. I need to be able

Re: [asterisk-users] open source screen pop software for asterisk

Hi, I work with gnudialer vicidal Best Regards On 7/14/07, Todd H [EMAIL PROTECTED] wrote: I like ADM as it has a URL popup feature (open a URL with a DID or CallerID in URL). The problem is that for each call, I tend to get 4 or 5 popups... But as the other author said, there are

Re: [asterisk-users] simple dial plan question

Hello, In your sip.conf you don't have the user for you provider: [yourprovider] username=1234 secret=sdfdsf host=sip.yourprovider.com type=peer ... In yor extensions.conf [mycontext] exten = 2000,1,Dial(SIP/2000,20) exten = 2000,103,Hangup exten = 2001,1,Dial(SIP/2001,20) exten =

Re: [asterisk-users] Connect two Asterisk boxes through IVR Menu

Hello, I take the example: exten = 300,1,Dial(IAX2/user:[EMAIL PROTECTED]/${EXTEN},30) Best Regards On 5/26/07, Alex Balashov [EMAIL PROTECTED] wrote: Matt, On Sat, 26 May 2007, Matt Darnell wrote: exten = _3xx,1,dial(IAX2/{$EXTEN}) exten = 300,1,dial(IAX2/301) You do not appear

Re: [asterisk-users] TDM400P usada?

Hey Look http://www.asterisk-es.org Best Regards On 5/5/07, Cesar Benjamin Garcia Martinez [EMAIL PROTECTED] wrote: Bueno, no esta en chile, yo no estoy en chile, pero hablo español, soy de mexico, asi que en parte tienes razón, pero tb creo que deberías haber puesto de donde eres.

Re: [asterisk-users] Re: Balancing interrupts.

Hello And lspci -vb ?? Regards On 5/4/07, Daniel Pittman [EMAIL PROTECTED] wrote: Steve Edwards [EMAIL PROTECTED] writes: I see the following on one of my new servers: -ts10::sedwards:~$ cat /proc/interrupts CPU0 CPU1 CPU2 CPU3 0:2979045

Re: [asterisk-users] Yeastar Cards

Hello I'd like to know too On 4/2/07, Gustavo Felisberto [EMAIL PROTECTED] wrote: I am in the process of buying a TDM800 card from Yeastar ( http://www.yeastar.com/products.asp?TypeName=TDM800%20PCI%20CardcTypeName=1) Any one has tested this cards? How reliable are them? I am specially

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