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 tr

Re: [asterisk-users] Community forum ?

7;t be 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&

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 mailin

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 > > > -- > _ > -- Ban

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 5.199.133.128:5273

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 tell the PBX to send

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' codec off the Internet now that

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 settings, and only those > s

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 network or server issue

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. > > Regards. > Le 6 avr. 2

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 Span: 2 ROSE REJECT: > [Mar 24

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 signal and route it to a

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 wrote: > Hello, > > Can someone recommend me where is best place to find Asterisk > Expert/Consultant for freelance work? > > If you are interested to work as a freelan

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, wrote: > I'm working with GXP2130. > About 12 phone on gigabit with PC after phone. > With

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 wrote: > Hi, I know there are people with much experi

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 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 Discussion > Subject

Re: [asterisk-users] SIP Softphone

Zoiper gsm -Original Message- From: Mark Robinson Sender: asterisk-users-bounces@lists.digium.comDate: Sun, 8 Jun 2014 17:01:54 To: Asterisk Users Mailing List - Non-Commercial Discussion Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] SIP So

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 wrote: > Any ideas on why this may not be working please ? > > - Original Message - > From: "Phil Daws" > To: "Asterisk Users Mailing List - Non-Commercial Discussion" < > asterisk-users

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 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 Sender: asterisk-users-bounces@lists.digium.comDate: Tue, 08 Oct 2013 01:08:22 To: Asterisk Users Mailing List - Non-Commercial Discussion Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [as

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 wrote: > Hi all, > > I've got a user who wants to receive voicemail notifications at two > different email addresses. I could probably

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 wrote: > Have you considered using VoiceMailMain()? >

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 wrote: > Hello Asterisk-users, > > [2013-08-18 05:56:29] NOTICE[17089][C-00a8] chan_sip.c: >Failed to authenticate device 390 >;tag=2762c06e > [2013-08-18 05:56:34] NOTICE[17089][C-00a9

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

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 wrote: > We have Asterisk1.8.11 and can not move to a newer version right now. But > when we run A

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 wrote: > I need to make a Asterisk 18.0's offline compiling, SVN mp3 support > sources downloading

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 wrote: > When I type: asterisk -rx "core show channels" > I usually get > Channel Location State Application(Data) > SIP/pstn--03 7807574622@internal: Up >

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 wrote: > Greetings. Anyone have any recommendations for

Re: [asterisk-users] monitoring Asterisk 1.8

> the server Asterisk in running on. > > thanks, > -Motty > > > On Thu, May 9, 2013 at 1:06 PM, Carlos Rojas wrote: > >> http://opennms.org/wiki/Installation:Yum >> >> >> On Thu, May 9, 2013 at 4:03 PM, Carlos Rojas wrote: >> >>> I&

Re: [asterisk-users] monitoring Asterisk 1.8

http://opennms.org/wiki/Installation:Yum On Thu, May 9, 2013 at 4:03 PM, Carlos Rojas wrote: > I'm using opennms and It's working fine. > > > > > > On Thu, May 9, 2013 at 3:23 PM, motty cruz wrote: > >> Hello, >> >> i'm looking for sugg

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 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. > > > Thanks in adv

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 wrote: > Hi every body; > > I want to intall some softwars working with my Asterisk server and I get > these erreurs : > > * > error: cannot seek `/dev/sda'.

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 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 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 > >> hi all,

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" 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 - 192.168.10.50

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 wrote: > Is it possible to bul multitenant system using some third party opensouce > application My design is like this. > > Company A: > Context Company_A > IVR Co

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" 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 caller ID > fr

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 wrote: > AFAIK, there is still not a MOH volume control. What I did was to take my > moh wav files and run them through sox like thi

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" wrote: > On Sat, Sep 29, 2012 at 12:41 PM, Carlos Rojas > wrote: > > Hello. > > &

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

Hello. Vpn is good idea, is more secure, you can use tls with srtp as well. Are you using asterisk 1.8? Right? Regard On Sep 29, 2012 9:57 AM, "Chris Nighswonger" wrote: > What are best practices for allowing connection by remote SIP > extensions over the internet? I'm thinking of putting the

Re: [asterisk-users] asterisk ip authentication

Hello Yes, there is, in sip.conf you should be using Insecure=invite,port for that Regards On Sep 25, 2012 4:06 PM, "jin jan" wrote: > Hi all, > I've tried to sen calls to asterisk from different soft switch. > I want to define ip authentication(not register) to an extension for make > call thr

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" 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, > Mehdi > > O

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 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 software. > It's possibl

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 wrote: > Hello Everyone, > > We are trying to integrate a hosted soft-switch to an Asterisks server and > the error received on the

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" wrote: > On Sat, Aug 11, 2012 at 3:16 AM, Kannan wrote: > >> I am planning a multi-tenant VoIP services system with Asterisk, using >> configuration tweaks. Having all the tenant

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) exte

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 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 guess my p

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 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 calling with a Siemens N

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 wrote: > Greetings, > > I am interested in learning more ablout Asterisk. Is there a recommended > link for "getting started". Can I set u

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" 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 (which will bl

[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 Carl

Re: [asterisk-users] Dialing multiple numbers

Hello, Maybe exten => 100,1,Dial(Local/14153456789@provider&Local/15606305670@provider) Regards On Sun, Feb 5, 2012 at 12:16 PM, Matt Hamilton wrote: > Hi, > > When dialing multiple numbers, we get connected to the first channel that > answers and others are hung up. > > exten => 100,1,Dial(S

Re: [asterisk-users] Dialing multiple numbers

Hello, Maybe exten => 100,1,Dial(Local/14153456789@provider&Local/15606305670@provider) Regards On Sun, Feb 5, 2012 at 12:16 PM, Matt Hamilton wrote: > Hi, > > When dialing multiple numbers, we get connected to the first channel that > answers and others are hung up. > > exten => 100,1,Dial(S

Re: [asterisk-users] asterisk problem sip

> > > -- > *From:* asterisk-users-boun...@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:* [ast

[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, a

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" wrote: > i could add "r" option in dial command. this will generate a ringtone > during connection. could i change this de

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 ]--" wrote: > Hello all, > > I attempted to make a couple of outbound calls this morning and always got > the busy tone. I checked the As

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 wrote: > I have a softphone I'm trying on a blackberry, that reg

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 wrote: > I happened to be in the cli tonight as some (208.122.57.58) initiated a > simple attack - just trying to make long distance call

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 wrote: > Un-top-posting, snarky comments inline... > > > On Wed, 28 Dec 2011, Faraj Khasib wrote: > > I am trying

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 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 only. bcoz I can

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" 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 network > working

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 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 question is how

[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] 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" 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 server, > I am using

[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? Regard

Re: [asterisk-users] Asterisk call transfers not working

> > *Ramiro PAZ > MASTERLINE LOGISTICS > > * > ** > On Mon, Oct 24, 2011 at 3:33 PM, Carlos Rojas wrote: > >> Hello, >> >> That sound a tones problem, what do you seting, dtmf in your sip.conf? >> >> Regards >> >> >> On Mon,

Re: [asterisk-users] Asterisk call transfers not working

Hello, That sound a tones problem, what do you seting, dtmf in your sip.conf? Regards On Mon, Oct 24, 2011 at 2:15 PM, Ramiro Paz wrote: > Hi everibody: > > Sorry, I want to relive this issue. I still have the problem, if somebody > could help me will be appreciated. Tks. > > *Ramiro PAZ > MAS

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

Hello, I use no-ip service, is similar than dyndns.com Best Regards 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 box doesn't have a stat

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 wrote: > 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 wrote: > Sir , > > this is not working > > > On Mon, May 9, 2011 at 1:52 PM, A J Stiles > wrote: > >> On

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? >

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, wrote: > I have tried a lot like as > exten => 123,1,Dial(SIP/114&SIP/113&SIP/115) > > and all the channels are dialing and if i answered any 3 of one, all the > channels except which one i a

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 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,

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 wrote: > i changed it and still didn't ring. however it did ring

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 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 proper NAT: > localne

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 wrote: > Dear all, >> I want to setup the incoming calls, that don't use authentication in >> sip.conf file. >> My configurations as follows, >> >> [

[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 a

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 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 maint

Re: [asterisk-users] Trunk SIP and configuration

Regards Carlos Rojas On Wed, Apr 1, 2009 at 10:45 AM, ludo perrot 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

Re: [asterisk-users] having problems with asterisk

Hello asterisk -vvvgc Regards On Wed, Dec 10, 2008 at 7:45 PM, Scott Berry 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 having a problem

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 :*

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 UNSU

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: htt

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 > c

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 callwaitingca

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 /usr/src/aste

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 a

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, >

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

> users in another 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 > >

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 asteri

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 w

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