[asterisk-users] Help - DTMF relay in meetme is not reliable
Hello List, I am facing some issue while passing DTMF (RFC2833 set globally in sip.conf) in meetme (asterisk 1.8). The issue I have observed that - if two users tries to pass DTMF simultaneously at the same time from their phones only one DTMF is detected in asterisk and broadcasted to other users. Other DTMF lost somewhere. We have tested only with sip phones. Can someone help me with this, or is there any configuration option that can resolve this problem? I want asterisk receive the DTMFs send at the same time and to pass those either by queuing them or by some other means. We can not use confbridge at this moment as we have developed the application on meetme. Please help! Regards, -- Rajib Deka Sr. Programmer Siemens Ltd. Chennai, India -- _ -- 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 list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Make phone ring through webserver using Asterisk
What is the easiest way? And how can it be implemented? I thought to something like: 1. I request a page to the webserver 2. Perl sends to asterisk a number to dial (Perl and asterisk are running in the same machine) 3. Asterisk calls the phone or 1. A Perl sip client registers to remote asterisk server 2. Perl sip client sends to asterisk the number to dial 3. Phone rings i don't care if i can hear something, it's enough that it rings -- _ -- 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 list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Make phone ring through webserver using Asterisk
I thought to something like: [...] or [...] Or make the script place a call file [0]. -nik [0]: http://www.voip-info.org/wiki/view/Asterisk+Call+Files -- burny Ein Jabber-Account, sie alle zu finden; ins Dunkel zu treiben und ewig zu binden; im NaturalNet, wo die Schatten droh'n ;)! PGP-Fingerprint: 3C9D 54A4 7575 C026 FB17 FD26 B79A 3C16 A0C4 F296 signature.asc Description: Digital signature -- _ -- 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 list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fraud detection
Check out transnexus We use their product... Seems to work well On Oct 18, 2013 2:09 AM, binary dreamer binary.vor...@gmail.com wrote: hello everyone. i am concerned about security to the PBX and i would like to discuss different fraud detection methods. Apart from making everything to secure the PBX (latest patches, iptables, firewalls, no outside users, strongs passwds,...) i would like to find out if there are any fraud detection techniques. As for my setup i do have a PBX running asterisk 11.4 and it has 3 sip trunks (over internet) -- _ -- 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 list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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 list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] calendar.conf include
On 13-11-13 10:20 AM, Jonas Kellens wrote: Hello, can I use include-statements in the calendar.conf configuration file ? You _should_ be able to use it will every .conf file, otherwise it is a bug. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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 list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Make phone ring through webserver using Asterisk
What do you want to happen once the call is made? You can choose to fire the call off using the originate command with the Asterisk Manager Interface from a PHP page or some other similar language. No need for Perl on the Asterisk box at all really unless you need it for something else. http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+Originate Date: Sat, 16 Nov 2013 16:53:59 +0530 From: omakhileshch...@gmail.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] Make phone ring through webserver using Asterisk What is the easiest way? And how can it be implemented? I thought to something like: I request a page to the webserverPerl sends to asterisk a number to dial (Perl and asterisk are running in the same machine) Asterisk calls the phoneor A Perl sip client registers to remote asterisk serverPerl sip client sends to asterisk the number to dialPhone ringsi don't care if i can hear something, it's enough that it rings -- _ -- 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 list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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 list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Bulk forwarding to another Asterisk
I want to be able to pass any number (variable length) to a context and then
forward that to another asterisk server for processing by that servers dial
plan. I have the two talking IAX2 so that part is done. I can also dial a
number from the sending to the server asterisk. The problem is I don't want to
have to create (duplicate) dial plans at originating Asterisk to equal those at
the server. I want to send the raw extension that enters a context and then
send it to the other Asterisk in entirety for processing. Is there a way to do
this?
What I want to do is like early dial on a SIP phone but between two Asterisk
systems. As soon as a match is made the call is processed.
This works for a three digit extension but I want to send an any length
extension that hits this context in entirety. If I use a _x! it just stops at
the first character. Extensions I want to send to the far end are a combination
of 2,3,4 and 10 digit numbers.
[pbx_server]
exten = _xxx,1,Answer
exten = _xxx,n,Dial(IAX2/pbx/${EXTEN})
exten = _xxx,n,Playback(vm-nobodyavail)
exten = _xxx,n,Hangup
Doug--
_
-- 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 list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bulk forwarding to another Asterisk
On Sat, 16 Nov 2013, Doug wrote: I want to be able to pass any number (variable length) to a context and then forward that to another asterisk server for processing by that servers dial plan. If I use a _x! it just stops at the first character. How about '_x.' or '_!.' Note that '!' will match any alphanumeric -- including T, h, i, s, t, etc. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000-- _ -- 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 list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bulk forwarding to another Asterisk
If using IAX then I would recommend setting up DUNDi or Switch statement in dialplan. Mitul On Nov 17, 2013 12:50 PM, Steve Edwards asterisk@sedwards.com wrote: On Sat, 16 Nov 2013, Doug wrote: I want to be able to pass any number (variable length) to a context and then forward that to another asterisk server for processing by that servers dial plan. If I use a _x! it just stops at the first character. How about '_x.' or '_!.' Note that '!' will match any alphanumeric -- including T, h, i, s, t, etc. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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 list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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 list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
