similar to: Asterisk 11 and pulse

Has any messed with getting Asterisk to work using the Voice Pulse Open Access plan? I currently have 2 numbers with Voice Pulse, 1 is a number that is assigned to their hardware device (Sipura SPA-2000), the other is a Open Access number that uses SIP from any device (you must authenticate with them). I want to be able to use the Open Access number on my Asterisk server here at home with no FXO

Hi, all Is there someway ability to insert custom Header to "SIP 486" message, when HANGUP application is invoked? Our use case is to set that Header, when call-limit is reached, to analyze elsewhere, but we do not want to set some custom causecode in HANGUP application because this can confuse a calling equipment.

I have been picking at Asterisk for about a week, and I think I'm close. I was hoping for a little guidance to bring this on home. I want to be able to make outgoing calls from my SJPhone clients using my VoicePulse Connect account. I have the two requisite items from Voice Pulse, but I've had no luck successfully integrating the VoicePulse settings into iax.conf. My current config:

>>I'm not sure that your question is clear. You'll probably want to be more specific. >> What is pulse? You mention "as a user", are you talking about voicepulse.com ? >> What are you trying to do with pulse? >> What problem are you running into? Sorry Rusty... I am trying to get Asterisk 11 to co-exist with a centos 7 box that has pulse audio running as

When I use pulse audio and Asterisk 11.4.0 on the Console/Dsp port I get a motorboating sound or warble - or - just not clear audio. When I switch that to ALSA direct it sounds just fine. What might be happening with pulse audio that it does not sound clear??? asound.conf below. Thanks, Jerry more /etc/asound.conf # # Place your global alsa-lib configuration here... # @hooks [ {

Hi, I am thinking of signing up with voice pulse connect to connect to my asterisk server and using it as a regular line. Is it good? Or should I go with vonage or others ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050124/16792f10/attachment.htm

Hi list! I'm trying to configure Asterisk to record incoming calls, if the called press *3. I added in features.conf: automixmon => *3 then, in my dialplan: exten => 1,n,Dial(SIP/00493511111111,20,RcxX) Well, if I **CALL** a number I'm able to record the call, but if I'll be called, and press *3 nothing happens... In the console I can't see anything, too. Could you

hello, is it possible simultaneously use chan_sip and chan_pjsip? if yes, can you recommend settings i'm thinking about - chan_sip - for sip hardphones/softphones (sip udp 5060) - chan_pjsip - for webrtc -- --------------------------------------- Marek Cervenka =======================================

Hi I'm using a macro to dial in a AEL dialplan. The problem is the macro do not set the field CDR(dst), showing only ~~s~~. I tried various configurations, but without solutions. This is the macro: macro dial-out(destno,dialstring,route_descr,interno) { __TRANSFER_CONTEXT=ipbx; if(${interno} = 1) { Set(__PICKUPMARK=${destno}); if(${ODBC_verify_user(${CALLERID(num)})} > 0) { t = tT; }

Hi, despite some searching I haven't found an answer to this question: Is there a way I can see in the log, or find out in the dialplan, what codec has been negotiated for a SIP channel? If possible, I'd like to do this in both Asterisk 11 and in an old 1.2 system. What I'm specifically trying to do is to determine historically the usage of the G.729 licences installed in a system,

Hello! I've got a little problem with Asterisk (11.14.1), the voicemessages are kinda limited to 40 seconds (average) aproximately; because when a message reach this long I got a cut in the file (*.wav) after I got this message: WARNING[15035][C-000021ef]: format_wav_gsm.c:418 wav_read: Short read (20) (Resource temporarily unavailable)! Does anyone got this problem, any idea of what is

Hi, Sorry if off topic, but is anyone here on this list using it? I am currently searching for a good router for my home network wich supports SIP. Many thanks!

We have a couple of positions open, please contact me off-list if interested. http://www.ntegratedsolutions.com/voice-engineer-dallas/ These are full time positions in Dallas, no telecommuters please. Thanks. JR -- JR Richardson Engineering for the Masses Chasing the Azeotrope

Hello, Is there a definitive guide on how SIP peers could be authenticated using LDAP in asterisk 11 and up? It seems https://wiki.asterisk.org/wiki/display/AST/LDAP+Realtime+Driver is not updated as there are mis-matched parameters in the configuration samples and ldap schema files. This is because I get: *Command: *sip show peer "1000" load *Output: **ERROR*: res_config_ldap.c:1389

On Thu, Apr 2, 2015 at 10:43 AM, Rusty Newton <rnewton at digium.com> wrote: > On Wed, Apr 1, 2015 at 9:08 AM, Trey Hilyard <kctrey at gmail.com> wrote: > >> Hello - >> >> I am trying to decide if I have stumbled across a bug in PJSIP or I am >> just missing something. My Asterisk has two interfaces, an "internal" eth0 >> and an

Hi, I am using ubuntu-12.04 and installed asterisk from repository (apt-get install asterisk). I have configured it to work with odbc, *CLI> odbc show ODBC DSN Settings ----------------- Name: asterisk DSN: asterisk-connector Last connection attempt: 1970-01-01 05:30:00 Pooled: No Connected: Yes But it still show me the following error [Jul 31 12:36:18]

In the absence of "The Definitive Guide to Asterisk Dial Plans" book, I'd like to do something possibly unique with the formatting of extensions in my dial plan, and am having trouble. We use VoicePulse connect, which gives us local DID for inbound and outbound calls (even though DTMF tones are not working in Voice Pulse Connect at the moment). To dial local numbers, you have to

hello liste i have creat i trunk sip and inboun route for inbound calls the issue whe i use the DID in inboud route i have a error No DID or CID Match. but when i leave this DID field blank i can route the call without any issue how can ido in order to use DID in route inboud "i use elastix" Executing [s at from-trunk:1] NoOp("SIP/358-106-000000c0", "No DID or CID

Hello: I am new to the list. I am trying to set up asterisk with voicepulse. I have a voicepulse username + password, and SIP DID. When I login to voicepulse, I have this under my devices tab: Devices *Login:* Sysxxxxxxx *Password:* xxxxxxxxxx *Context:* VPWS *Connects to:* gw5.voicepulse.com My question is: Do I need a 2.4.x kernel? Currently I am running Debian/stable stock 2.2.x ? Has

Hello - I am trying to decide if I have stumbled across a bug in PJSIP or I am just missing something. My Asterisk has two interfaces, an "internal" eth0 and an "external" eth1. In pjsip.conf, I define the following transports: [trusted] type=transport protocol=udp bind=10.xx.yy.zz:5060 [untrusted] type=transport protocol=udp bind=12.4.aa.bb:5060 My internal endpoints use