[asterisk-users] Setting PJSIP header from AMI

); $action->setVariable('__SIPADDHEADER51',"Identity: $identity"); // $identity contains generated by 3rd party header Is there anything similar for $action = new OriginateAction("PJSIP/"); ??? that would work for PJSIP? Any suggestions a

Re: [asterisk-users] [External] 180 Ringing missing

rly media and ringing simultaneous. Especially automata calling will want to know the difference. Of course, a generated ringback tone by the caller should be stopped when media is received. Asterisk should indicate to the caller the same state it received from the callee. -- Ale

Re: [asterisk-users] Redis in place of astdb

I’ll second that - for CDR you want the fastest sequential writing with possible batching of CDR records Best regards, Alex www.thirdlane.com > On Jul 9, 2020, at 1:37 AM, Antony Stone > wrote: > > On Thursday 09 July 2020 at 00:50:28, Jon Bonilla (Manwe) wrote: > >

Re: [asterisk-users] WebRTC as Softphone substitute ?

payload for media paths that require that, which TURN server is not able to do. Best regards, Alex Alex Epshteyn a...@thirdlane.com +1 (415) 261 6601 www.thirdlane.com > On Oct 2, 2018, at 7:39 PM, David P wrote: > > Thanks for sharing this, Alex. It sounds like TURN, as a media

Re: [asterisk-users] WebRTC as Softphone substitute ?

/blog/nat-stun-turn-and-ice Best regards, Alex Alex Epshteyn a...@thirdlane.com +1 (415) 261 6601 www.thirdlane.com > On Oct 2, 2018, at 6:08 PM, Nasir Iqbal wrote: > > @Olivior > I agree that seting up WebRTC is hard, however when done it is smooth to use. > For replication you

[asterisk-users] Asterisk kafka connector feedback

l time. Please have a look and let me know if you have any feedback or use cases. Cheers, Alex -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at:

Re: [asterisk-users] Dial to FastAGI application appears as 1-second CDR - how do I fix?

El 29/05/18 a las 05:24, Tony Mountifield escribió: In article <3a005ff6-19a4-215b-4751-bee616ec7...@palosanto.com>, Alex Villací­s Lasso wrote: In my application, I am using AMI to run an Originate command between a channel and a dialplan application (NOT a context). In m

[asterisk-users] Dial to FastAGI application appears as 1-second CDR - how do I fix?

In my application, I am using AMI to run an Originate command between a channel and a dialplan application (NOT a context). In my case, the application I want to invoke is FastAGI. The Originate AMI command works correctly, but Asterisk generates a very short (0-1s) duration for the CDR that

Re: [asterisk-users] softphone instead of desktop phones

Thomas was asking how to save money and I was just offering an option. I am sorry if my post was inappropriate. That said, Thirdlane Connect itself is free, and we do offer a free version for companies with up to 10 users. -- Alex Epshteyn email: a...@thirdlane.com web: www.thirdlane.com

Re: [asterisk-users] softphone instead of desktop phones

, and screen sharing. It integrates with a variety of applications and CRMs such as Salesforce, Zoho, Zendesk, Redmine, etc. Try it out! -- Alex Epshteyn web: www.thirdlane.com - Original Message - > From: "Amit Patkar" <a...@avhan.com> > To: "Asterisk Users Ma

Re: [asterisk-users] PBX selection

ing Asterisk based PBX in 2002 and Multi Tenant PBX in 2005 - we do this as our core business and are still finding areas for improvement :). As your experience with VoIP is minimal I would side with your CTO - you should find a solution high enough in the stack to avoid the complexity of bui

[asterisk-users] Mail rejected from secur...@asterisk.org

: host asterisk.org.inbound10.mxlogic.net[208.65.145.3] said: 451 Could not load DRD for domain (asterisk.org) rcpt (secur...@asterisk.org) (in reply to RCPT TO command) I want to inform about a potential issue but my mail to secur...@asterisk.org is bouncing

[asterisk-users] AMI ManagerAction UpdateConfig on file with #include may corrupt the file

I have seen the following scenario that may lead to a corrupted and possibly invalid configuration file after using UpdateConfig through AMI, at least with Asterisk 11.25: There is a configuration file a.conf that contains several sections and also contains a #include "b.conf", which defines a

[asterisk-users] Representation of category templates and template inheritance in ARA Static Realtime

https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+ManagerAction_UpdateConfig According to the link above, the NewCat action allows for options that specify whether the newly created category is a template, and whether it should inherit from an existing set of templates.

Re: [asterisk-users] Asterisk dahai install centos 7

El 10/12/16 a las 10:15, christopher kamutumwa escribió: Package kernel-devel-3.10.0-327.36.3.el7.x86_64 already installed and latest version but i still receive the same error [root@localhost dahdi-linux-complete-2.11.1+2.11.1]# make make -C linux all make[1]: Entering directory

[asterisk-users] Can Asterisk handle in any way an SDP with m=application webrtc-datachannel ?

I have a working telephone project that uses SIP.js 0.7.5 with Asterisk on the server side. Currently it handles both audio and video correctly. The SIP.js webpage has instructions for setting up a datachannel through a SIP call. The online demo uses OpenSIPS. When setting up a SIP call with

Re: [asterisk-users] Getting 'no shared cipher' on call to webrtc endpoint from asterisk-11.24.0

El 26/10/16 a las 13:16, Alex Villací­s Lasso escribió: I am making SIP calls using SIP.js and configuring Asterisk 11.x for websockets calls under CentOS 7. On 11.23.1 and earlier, I had to patch the code to disable auto negociation due to ASTERISK-25659. Now that the bug is supposedly fixed

[asterisk-users] Getting 'no shared cipher' on call to webrtc endpoint from asterisk-11.24.0

I am making SIP calls using SIP.js and configuring Asterisk 11.x for websockets calls under CentOS 7. On 11.23.1 and earlier, I had to patch the code to disable auto negociation due to ASTERISK-25659. Now that the bug is supposedly fixed in commit 8653da4fa228e1e289e09e5d024e11d24da87d94, I

Re: [asterisk-users] Replacement for phpagi?

El 10/08/16 a las 12:06, Carlos Chavez escribió: Anyone know a good replacement for phpagi? Unfortunately development stalled long ago and it has not been updated. What is the best solution for AMI and AGI on PHP? Thanks. In the case of AMI, you could use the AMI client from the

Re: [asterisk-users] Trouble applying regex to dialplan variable that contains double-quotes

, what am I doing wrong? On 08/08/2016 04:31 PM, Alex Villací­s Lasso wrote: I am writing a dialplan context under asterisk 11.21.0 to handle SIP message routing between registered SIP peers using chan_sip. I am having trouble with double-quotes when the source peer uses a display name, which appears

[asterisk-users] Trouble applying regex to dialplan variable that contains double-quotes

I am writing a dialplan context under asterisk 11.21.0 to handle SIP message routing between registered SIP peers using chan_sip. I am having trouble with double-quotes when the source peer uses a display name, which appears in quotes before the SIP URI. I want to extract the SIP URI from

[asterisk-users] Incoming webrtc call succeeds in Firefox but fails in Google Chrome

x=1001@device permit=0.0.0.0/0.0.0.0 callerid=Usuario Alex <1001> callcounter=yes faxdetect=no With this setup, I can make calls using the SIP softphone as usual, both into and out of asterisk. After approving the certificate exceptions, I can also use the webrtc account to generate a call

Re: [asterisk-users] Incoming webrtc call succeeds in Firefox but fails in Google Chrome

El 20/01/16 a las 16:25, Alex Villací­s Lasso escribió: I am having trouble getting Google Chrome to accept a WebRTC call coming from Asterisk, even though Firefox can (now) accept the same call without issue. My setup is as follows: Server: CentOS 7 x86_64 (Elastix 4 RC) with IP: 10.1.0.4

Re: [asterisk-users] Incoming webrtc call succeeds in Firefox but fails in Google Chrome

El 20/01/16 a las 18:33, Alex Villací­s Lasso escribió: El 20/01/16 a las 16:25, Alex Villací­s Lasso escribió: Partial fix: Google Chrome accepts the call if videosupport is set to "no". This is the SDP of the successful INVITE that Chrome accepts: INVITE sip:8cj802p8@192.0.2.240

[asterisk-users] We're one week away from OpenSIPS Week in Austin!

We're almost there everyone! The OpenSIPS Summit in Austin is 1 week away. There's still some room left for all you procrastinators! We also have a few spots left for the OpenSIPS LIVE Bootcamp following the summit. Enjoy a rare opportunity to train with the founder of the OpenSIPS Project

[asterisk-users] Detect queue agent in wrapuptime period through AMI

Is there a way to use AMI to detect whether an agent that appears to be free is in its wrap-up-time period? I am using AMI to query the queue status and its members, in order to generate calls directed to the queue, and I do not want to originate calls if some of them will not be assigned

[asterisk-users] tls forasterisk

hi, is there some tutorials on the you tube how to setup tls for asterisk and how to get ip phones register like thatwhat ip phones can support tls any way?Alex -- _ -- Bandwidth and Colocation Provided by http://www.api

[asterisk-users] Why is CDR(recordingfile) not being written to the database despite being set in the dialplan?

I am using Asterisk 11.17.1 with my program that uses AMI Originate calls to generate a bunch of calls for a callcenter. The PBX configuration is handled by FreePBX 2.11. I want to understand the dialplan behavior in order to figure out why the CDR(recordingfile) is blank on the CDR records

[asterisk-users] Help debugging a possible SIP channel leak in 11.17.0, possible race condition

I am trying to collect enough information about an problem a client is having with its asterisk 11.17.0 x86_64. This issue was observed before in 1.8.20, and we upgraded to 11.15.0 and then to 11.17.0 with no solution. Background: this client is a telemarketing call-center that generates

Re: [asterisk-users] Help debugging a possible SIP channel leak in 11.17.0, possible race condition

El 07/04/15 a las 17:38, Alex Villací­s Lasso escribió: I am trying to collect enough information about an problem a client is having with its asterisk 11.17.0 x86_64. This issue was observed before in 1.8.20, and we upgraded to 11.15.0 and then to 11.17.0 with no solution. Background

Re: [asterisk-users] Showing sip subscriptions in Manager

You can use Command command, and sip show subscriptions as a parameter -- Alex Epshteyn email: a...@thirdlane.com web: www.thirdlane.com phone +1 415.261.6601 - Original Message - From: Leandro Dardini ldard...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion

[asterisk-users] Problems linking asterisk against self-compiled openssl on CentOS 5

I am trying to enable full WebRTC support on asterisk-11.15 for installation on a CentOS 5 machine. Currently the distro cannot be upgraded to any later CentOS series. This CentOS series ships with openssl-0.9.8e, which lacks DTLS-SRTP support required for WebRTC. So I decided to build a parallel

Re: [asterisk-users] Problems linking asterisk against self-compiled openssl on CentOS 5

El 23/12/14 a las 12:19, Alex Villací­s Lasso escribió: I am trying to enable full WebRTC support on asterisk-11.15 for installation on a CentOS 5 machine. Currently the distro cannot be upgraded to any later CentOS series. This CentOS series ships with openssl-0.9.8e, which lacks DTLS-SRTP

Re: [asterisk-users] asterisk and elastix

El 24/11/14 a las 10:59, Salaheddine Elharit escribió: Hello list, i have installed elastix 2.4.0 with call center model and i have created an Outgoing Calls https://192.168.1.251/index.php?menu=outgoing_calls my question i want to know the name of the tbale where the csv file is uploaded in

[asterisk-users] OpenSIPS Summit Oct 21st before Astricon

Hello Everyone! We wanted to let everyone coming to Astricon know that we will be holding an OpenSIPS Summit on Tuesday Oct 21st, 2014 at the Suncoast Casino Spa. Suncoast is about 10 minutes away from Red Rock and we will be provide shuttle service to and from the Summit. For those of you

Re: [asterisk-users] Making sense of SDP for debugging of missing audio in SIP trunk

El 18/06/14 13:44, Alex Villací­s Lasso escribió: I am debugging an intermittent issue of missing audio on calls that come from a SIP provider into our asterisk-11.10 installation. Sometimes, incoming calls from this provider work correctly, with audio streaming in both directions. Other times

[asterisk-users] Making sense of SDP for debugging of missing audio in SIP trunk

I am debugging an intermittent issue of missing audio on calls that come from a SIP provider into our asterisk-11.10 installation. Sometimes, incoming calls from this provider work correctly, with audio streaming in both directions. Other times, with the same setup, the calling party is unable to

Re: [asterisk-users] Login by AMI ok, by AJAM fails

El 16/05/14 16:01, Michael L. Young escribió: - Original Message - From: Michael L. Young myo...@acsacc.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, May 16, 2014 4:55:30 PM Subject: Re: [asterisk-users] Login by AMI ok,

Re: [asterisk-users] Proper way to make Asterisk recognize SIP trunk of incoming INVITE when IP is not available

El 27/04/14 07:47, Barry Flanagan escribió: On 26 April 2014 00:29, Alex Villací­s Lasso a_villa...@palosanto.com mailto:a_villa...@palosanto.com wrote: I am currently preparing a kamailio-asterisk combination. The asterisk installation uses realtime for SIP. The kamailio configuration

Re: [asterisk-users] Proper way to make Asterisk recognize SIP trunk of incoming INVITE when IP is not available

El 02/05/14 10:49, Alex Villací­s Lasso escribió: El 27/04/14 07:47, Barry Flanagan escribió: On 26 April 2014 00:29, Alex Villací­s Lasso a_villa...@palosanto.com mailto:a_villa...@palosanto.com wrote: I am currently preparing a kamailio-asterisk combination. The asterisk installation

[asterisk-users] SOLVED: Re: Proper way to make Asterisk recognize SIP trunk of incoming INVITE when IP is not available

El 02/05/14 11:41, Alex Villací­s Lasso escribió: El 02/05/14 10:49, Alex Villací­s Lasso escribió: El 27/04/14 07:47, Barry Flanagan escribió: On 26 April 2014 00:29, Alex Villací­s Lasso a_villa...@palosanto.com mailto:a_villa...@palosanto.com wrote: I am currently preparing a kamailio

[asterisk-users] Proper way to make Asterisk recognize SIP trunk of incoming INVITE when IP is not available

I am currently preparing a kamailio-asterisk combination. The asterisk installation uses realtime for SIP. The kamailio configuration was based on the reference at http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb but has been heavily modified. Currently asterisk runs on

Re: [asterisk-users] Proper way to make Asterisk recognize SIP trunk of incoming INVITE when IP is not available

El 25/04/14 18:29, Alex Villací­s Lasso escribió: I am currently preparing a kamailio-asterisk combination. The asterisk installation uses realtime for SIP. The kamailio configuration was based on the reference at http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb but has

[asterisk-users] Asterisk SSL support broken with update from openssl-1.0.0 to 1.0.1e, recompiling does *not* help

I am having an issue that prevents WebSockets over SSL/TLS (or any kind of encrypted HTTP traffic to Asterisk) from working after an openssl library update. My setup is CentOS 6 x86_64, and initially, with openssl[-devel]-1.0.0-20.el6_2.5.x86_64 . With this openssl versions, https over TCP

[asterisk-users] SOLVED: Re: Asterisk SSL support broken with update from openssl-1.0.0 to 1.0.1e, recompiling does *not* help

El 27/03/14 11:59, Alex Villací­s Lasso escribió: I am having an issue that prevents WebSockets over SSL/TLS (or any kind of encrypted HTTP traffic to Asterisk) from working after an openssl library update. My setup is CentOS 6 x86_64, and initially, with openssl[-devel]-1.0.0-20.el6_2.5

Re: [asterisk-users] How to configure asterisk to only accept SIP from kamailio@localhost but exchange RTP on all interfaces?

El 25/02/14 08:30, Karsten Wemheuer escribió: Hi Alex, Am Donnerstag, den 20.02.2014, 13:48 -0500 schrieb Alex Villací­s Lasso: I have a setup with asterisk-11.7.0 and kamailio-4.1.1. I am following the setup guide at http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb

[asterisk-users] How to configure asterisk to only accept SIP from kamailio@localhost but exchange RTP on all interfaces?

I have a setup with asterisk-11.7.0 and kamailio-4.1.1. I am following the setup guide at http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb . I want to run asterisk and kamailio on the same server, with SIP realtime configuration (MySQL database) so that kamailio

Re: [asterisk-users] How to configure asterisk to only accept SIP from kamailio@localhost but exchange RTP on all interfaces?

El 20/02/14 15:07, Markus escribió: Am 20.02.2014 19:48, schrieb Alex Villací­s Lasso: My concern is that asterisk is left listening for SIP through all interfaces and with no SIP passwords. I want to secure the setup against directed traffic to the asterisk UDP port (5080), that bypasses

Re: [asterisk-users] Asterisk 11.7.0 not receiving registration from local address

El 22/01/14 08:39, Administrator TOOTAI escribió: Le 22/01/2014 14:01, Administrator TOOTAI a écrit : Hi, I face a problem which look like the same as David with his thread Asterisk not receiving call from VPN address. I had an Asterisk (Elastix) working well in a VM (Debian Wheezy - KVM)

[asterisk-users] Transfer call placed from console (with chan_alsa)

, in principle? Alex -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing

Re: [asterisk-users] Reading DTMF sent by callee during a SIP

the door), rather than just #. Alex -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello

[asterisk-users] Reading DTMF sent by callee during a SIP call

on the same machine as the Asterisk server itself is not possible, because both won't be able to bind to port 5060. My guess is that the solution is to originate a call from the CLI; but I haven't gotten to that part yet. Thank you for your patience, I am looking forward to your feedback, Alex

[asterisk-users] How do I remotely force an *unconfigured* Digium DPMA phone to re-query the network for the DPMA server?

Consider the following scenario: 1) One or more Digium DPMA phones are plugged into the network. I know their IP addresses and MACs. 2) The Asterisk I want to use as the telephony server starts without the DPMA module. Therefore there are no DPMA sessions between the phones and the server. 3)

Re: [asterisk-users] DPMA, check-sync

El 06/09/13 14:44, Malcolm Davenport escribió: Howdy, Please forgive the off-list e-mail. I'm not subscribed to the list, I only peruse the archives. The follow up from George is correct. For phones that have already been attached to DPMA, DPMA disables the enable_check_sync phone setting.

Re: [asterisk-users] LibopenR2 with debug symbols

El 05/09/13 07:02, Rodrigo Montiel escribió: Hi all, Recently I have raised up a bug (related to segmentation fault of asterisk process) due to a core dump generated by Asterisk under the following environment: CentOS release 5.7 (Final) Kernel 2.6.18-238.12.1.el5 #1 SMP Tue May 31 13:23:01

[asterisk-users] Use DPMA to enumerate unconfigured Digium phones in LAN

Is there a way to use DPMA to enumerate the Digium phones that are plugged in and visible in the local network, but not (yet) configured through the DPMA configuration files in Asterisk. I would like to write a frontend that lists the DPMA capable phones, presents a GUI to specify the various

[asterisk-users] Need to figure out DAHDI logical group from CDR record

I have a bunch of CDR records in the mysql database asteriskcdrdb on a FreePBX system. There is a DAHDI trunk defined in FreePBX which uses the gN identifier to make calls. So in this setup the trunk is roughly equivalent to a DAHDI logical group. I want to know, given a CDR, which logical group

[asterisk-users] Asterisk 11, SIP. OK to BYE goes to wrong ip/port combination

(or from From') and OK gets lost. I'm just trying to adjust sip configuration that used to work for simple call scenarios (in 1.4, for example) for Asterisk 11. Your input is appreciated. Thank you. Alex Zarubin In sip.conf [general] nat = no outboundproxy=PROXYipaddress:PROXYport [CARRIER

Re: [asterisk-users] how to launch a URl when dialing a number

El 31/05/13 09:21, Salaheddine Elharit escribió: thanks justin i try to do this but the issue still the same.this link is stored in my server 192.168.5.109 .but what i want to receive this link when i call this number in my pc ip adresse of my pc 192.168.5.131 ip adresse of server when the

Re: [asterisk-users] Not able to build the chan_sip.c module

El 27/05/13 01:56, upendra escribió: Hi, i am trying to install asterisk newer version on the Elastix machine, but while installing the chan_sip,c module is not building while make. when i see in make menuselect options it showing XXX -- extended , please let me know how to enable it and make

Re: [asterisk-users] How to allow AMI access to Originate yet deny Application: System

El 15/05/13 10:10, Alex Villací­s Lasso escribió: While doing a security audit on a system I maintain, I stumbled upon an unvalidated use of a variable to compose an Originate request to the local Asterisk instance via AMI. Taking as an example an earlier exploit for FreePBX, I realized

[asterisk-users] How to allow AMI access to Originate yet deny Application: System

While doing a security audit on a system I maintain, I stumbled upon an unvalidated use of a variable to compose an Originate request to the local Asterisk instance via AMI. Taking as an example an earlier exploit for FreePBX, I realized that this, combined with Application: System as an injected

Re: [asterisk-users] Error to install Asterisk

El 06/03/13 11:52, Carlos Alvarez escribió: I'm going to make an observation here that may upset you, and I don't mean it to, but it's fact. If you are so unfamiliar with Linux, you will have a bad time managing Asterisk servers. You really need to know how to use the OS before you can learn to

Re: [asterisk-users] Exiting the queue doesn't work

in the dialplan. After extensively googling the issue, I've found everything (also bug related), accept my answer. What am I missing here? It's Asterisk 1.8 on a Debianbox. Thanks! Gertjan -- Look at context= in queues.conf. Alex

Re: [asterisk-users] Exiting the queue doesn't work

to leave us a message. You can also play the message before entering the queue (only once and caller may forget what key to press). This way the caller looses their position in the queue only if they choose to leave a message. Alex

Re: [asterisk-users] Exiting the queue doesn't work

On 3/4/2013 8:00 AM, Gertjan Baarda wrote: This will only work with the n option in the queue command and retry=0 in queue.conf. Is it not? On Mon, Mar 4, 2013 at 2:55 PM, Alex Kauffmann akauf...@prodigy.net.mx mailto:akauf...@prodigy.net.mx wrote: On 3/4/2013 7:27 AM, Gertjan Baarda wrote

Re: [asterisk-users] Point a Digium phone to a configuration URL using mDNS without DPMA or DHCP option 66

El 28/02/13 08:40, A J Stiles escribió: On Wednesday 27 February 2013, Alex Villací­s Lasso wrote: I have the following scenario. A small network has DHCP but does not publish option 66. An Asterisk server is on the network, but the Asterisk version does not support DPMA and it is hard

[asterisk-users] Point a Digium phone to a configuration URL using mDNS without DPMA or DHCP option 66

I have the following scenario. A small network has DHCP but does not publish option 66. An Asterisk server is on the network, but the Asterisk version does not support DPMA and it is hard to switch the version. However, there is a possibility to have a web server and an mDNS (Avahi) server. I

Re: [asterisk-users] Asterisk AMI - Create a daemon (background process)

El 24/02/13 07:30, Shahid H escribió: I wanted to create a daemon (background process) in PHP. A daemon will use socket to connect with Asterisk AMI to send events and listen the actions. A daemon will also listen the commands from agents via HTTP, for example: A agent pressed a hang up

Re: [asterisk-users] PRI can receive calls but cannot dial out

there be something wrong with my /etc/dahdi/system.conf or chan_dahdi.conf (see above)? Thanks, Vieri The first port has channels defined in group 2, but the port is down. Have you tried dialing out with G2 as opposed to g2? Alex

Re: [asterisk-users] Change phone display from queue calls

(number) before we dial the queue. The new value for callerid(name) will show on the agent's screen. Setting Callerid(number) will work as well. alex -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com

[asterisk-users] Query list of defined channel variables via AMI

Is there a way to list the names of the channel variables that are currently defined on a given channel via AMI? I know of GetVar and SetVar, but GetVar needs the name of the variable to get. -- _ -- Bandwidth and Colocation

Re: [asterisk-users] leading ghost 0

On 11/21/2012 10:53 AM, gincantalupo wrote: Alex, I had already tried itreloading chan_dahdi.so module is enough...I saw Asterisk was behaving differently after reload. To tell the truth, setting pridialplan=unknown causes Asterisk to stop reading following channels configuration...it says

Re: [asterisk-users] leading ghost 0

seen no changes. Did you try: pridialplan=unknown Did you restart dahdi and asterisk after the changes? Alex -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live

Re: [asterisk-users] DAHDI and Tiger320 Chip

El 25/10/12 07:46, Antonio Modesto escribió: Hi, I've got a ISDN Interface: /Tiger Jet/ Network Inc. Tiger3XX Modem/ISDN interface, I'm trying to use it with DAHDI 2.6 but it doesn't work, I'm thinking that dahdi doesn't support this device, I've loaded all of available dahdi drivers and none

Re: [asterisk-users] How to tie orders taken to specific CDR records

?) Is there some combination of fields in the CDR that might uniquely identify a specific call? Open to any and all ideas. Try looking at the queue_log. Configure your system to log to mysql and you should be able to get everything you need in realtime. Alex

Re: [asterisk-users] Agents in more than one queue at once

with this, but not without at least a decent chance that the work will be integrated into mainline (assuming it doesn't suck, of course :) Alex Forster -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New

[asterisk-users] Tell apart between network disruption and asterisk restart via AMI

I have a program that connects to the Asterisk Manager Interface through port 5038 on a remote machine. Suppose I get a TCP disconnection on my program. The program will then attempt to reconnect to the AMI and will eventually succeed. Is there a way to check whether the disconnection was caused

Re: [asterisk-users] Agents in more than one queue at once

Are there any developers that are familiar with the Queue() app implementation and how it distributes calls? Alex Forster -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us

[asterisk-users] Agents in more than one queue at once

-eb1e (wait: 0:45, prio: 0) My question is: when Alice gets off the phone, which call will she get? My expectation is that she will get the call which has been waiting longer, but I'm not sure that's actually the case. Alex Forster

Re: [asterisk-users] How to log caller IP address in the CDR?

)}, SIP FROM: ${SIPCHANINFO(from)}) Alex 2012/10/5 Ishfaq Malik i...@pack-net.co.uk On Fri, 2012-10-05 at 14:10 +0200, Benoit Panizzon wrote: Hello We had this situation: Some bot-net did try to guess SIP logins and finally succeeded. The Asterisk Server was abused to call a large number

Re: [asterisk-users] So long, and thanks for all the fish!

No Kevin, we thank you for the fish, for sharing the knowledge and for having the patience... Good luck on your new journey! P.S. As a token of our appreciation, just a word and we'll make the life of the new Director of Software Technologies miserable! 2012/7/31 Kevin P. Fleming

Re: [asterisk-users] Flowroute: howto set outbound callerid (ast 1.4)?

SIPAddHeader() comes to mind. :-)  -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http

Re: [asterisk-users] Elastix 2.3.0.1

El 05/07/12 02:19, Satria Anamarta escribió: Greetings, I know this is not a Elastix mailing list, but could anybody please tell where I can download Elastix 2.3.0.1 (the latest version) ? There is only version 2.3.0 (April 2012) on Elastix website, not the 2.3.0.1 (May 2012), but the

Re: [asterisk-users] 2GB Elastix memory limit

El 28/06/12 03:58, resea...@businesstz.com escribió: I have sevaral elastix installed but all of them show the physical memory is 2GB while the server has 4GB and some has 8GB. I've upgraded to PAE kernel but yet i cant see mem beyond 2GB. How can i configure the centos kernel to use more memory

Re: [asterisk-users] Need queue name in CDR

Hi there, You can do that by making a context in your dial-plan with the name of your queue. Then in your queue config put : context="context you write in your dial-plan". That way the asterisk will save the name of your queue in the 'dcontext' column

Re: [asterisk-users] Replacing PBX with Asterisk, need feedback on my new architecture.

was painstakingly thumbed out on my mobile, so apologies for brevity and errors. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Atlanta, GA 30030 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/, http://www.alexbalashov.com On May 11

Re: [asterisk-users] Flashphoner

messages for some monthly price. Is it interested for you? -- Thanks, Pavel Ismailov skype: pavel.ismailov www.flashphoner.com -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http

Re: [asterisk-users] Flashphoner

On 04/27/2012 01:24 PM, shayne.al...@gmail.com wrote: congratulations @};- It's a match made in Heaven. I have spare signature space to sell, and Pavel wants signature space to rent! At a low introductory rate of US$1800/word, he and I are going to make this happen... -- Alex Balashov

Re: [asterisk-users] Flashphoner

Only the premium dailing minties. The regular flashphoner ones are indebted to a complex vanilla ice cream + pork belly + cardboard mixture... -- This message was painstakingly thumbed out on my mobile, so apologies for brevity and errors. Alex Balashov - Principal Evariste Systems LLC 235 E

Re: [asterisk-users] Monitoring voice-quality with sip/rtp/rtcp

OpenVZ is not really virtualisation, though for some reason people insist on throwing it into the same discursive space as Xen, VMware, HyperV, etc. -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Fax: +1-404-961

Re: [asterisk-users] Asterisk 1.8 and DeadAGI

Look up the definition of NoOp. A moral and practical ambivalence inheres in that definition. It is neither more nor less beneficial to use or not use it, for it is a NoOp. -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678

Re: [asterisk-users] Rate sheet normalization

We solve this problem for our customers all the time, in various situationally-specific ways. But yes, we are not really in a position to genericise it and give it away. It's not because we are greedy. The time and resources just aren't there. -- Alex Balashov - Principal Evariste Systems

Re: [asterisk-users] Rate sheet normalization

On 03/28/2012 03:15 PM, Raj Mathur (राज माथुर) wrote: Times change -- the way to deal with that is to adapt I don't think you'll get any serious disagreement on that from anyone here. -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Atlanta, GA 30030 Tel

Re: [asterisk-users] Routing premature media to the calling channel

media is going. -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Atlanta, GA 30030 Tel: +1-678-954-0671 Web: http://www.evaristesys.com/, http://www.alexbalashov.com Leandro Dardini ldard...@gmail.com wrote: Hello, I have a problem with premature media and inband

Re: [asterisk-users] Routing premature media to the calling channel

not logically imply or mandate backward early media, though 183+SDP is generally used as a convention to indicate that it is about to be sent. -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Atlanta, GA 30030 Tel: +1-678-954-0671 Web: http://www.evaristesys.com/, http

Re: [asterisk-users] Routing premature media to the calling channel

I think I may have misunderstood your initial question, sorry. You are looking for Asterisk to directly pass through the early media from upstream? Why would it do that? -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Atlanta, GA 30030 Tel: +1-678-954-0671

Re: [asterisk-users] Routing premature media to the calling channel

As far as I know, this is not the general tendency of any B2BUA that generates such media independently. However, I could be mistaken. -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Atlanta, GA 30030 Tel: +1-678-954-0671 Web: http://www.evaristesys.com

Re: [asterisk-users] Rate sheet normalization

Our system just rolls over until it finds a carrier that will take it. Up to 30 different routes are supported, and rollover is pretty instantaneous in most cases. -- This message was painstakingly thumbed out on my mobile, so apologies for brevity and errors. Alex Balashov - Principal

Re: [asterisk-users] DAHDISendCallreroutingFacility

El 10/03/12 12:05, Karsten Wemheuer escribió: Hi, Am Samstag, den 10.03.2012, 08:42 -0800 schrieb Mehdi Shirazi: Hi I installed Asterisk 1.8.7 with CD ISO(Elastix 2.2) I want to use DAHDISendCallreroutingFacility Application on a PRI link(LIBPRI Already installed). according to

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