Re: [asterisk-users] Touch tone stutter

On 2016-11-27 06:46 AM, Max Grobecker wrote: Hi, you could try switching the DTMF mode of the ATA's SIP peer (and also in the ATA itself) to INBAND transmission. In this mode, the ATA doesn't need to recognise DTMF tones and your Asterisk can interpret it. For this to work, the ATA needs to

Re: [asterisk-users] Asterisk compatibility with SMS services

i'm using gammu[1] with a 3g dongle and my own chip with my preffer provider. It can send over 700 every our and receive to. I don't know if you need asterisk and sms in the same way but with this tool you can make everything. It has python tools to. Best regards. Emiliano. [1]

Re: [asterisk-users] app_queue ringall - 2 agents answer same time problem

hmm. i think customer will not agree this is correct behavior from pcap it looks like there is missing CANCEL to the second device Dne 30/11/2016 v 19:42 Sam Basan napsal(a): Your second call is not without sound, there is simply no call at all. As the first answer the call his channel and

[asterisk-users] inbound T38 to email

I have played around with iaxmodem and hylafax and have a few working installations where PRI's are involved. I have a new customer that will be sending inbound fax calls via a new (SIP) DID provider we are working with (yup, same one from the last message I just sent), and they support T38.

[asterisk-users] new inbound DID provider... no auth?

We are trying to work with a new DID provider and I find myself confused. Their standard integration is to send the call with no authentication. I am expected to whitelist all their possible gateways, and accept their calls I guess with no peer definition. I actually have it working this

[asterisk-users] Specify "name" for Resource in RLS

Is there are way to specify the display name of a resource in a resource list? I have setup a resource list in Asterisk 13 for 1234. All is working on the device, but I want to show "Joe User" instead of "1234". Any thoughts? --

Re: [asterisk-users] Dropped call after 900s: 481 call/transaction does not exist and another anomaly during re-invite in timer - full anonymized trace attached

Derek Bolichowski wrote: HI Michael, You can set this in sip.conf: session-timers=refuse I know of this option - it doesn't help, because the provider ignores it (on some calls) and the call is dropped anyway. Normally, there is no problem with the timers. And the problem which occurred

Re: [asterisk-users] app_queue ringall - 2 agents answer same time problem

Your second call is not without sound, there is simply no call at all. As the first answer the call his channel and the external call channel connected. The second device simply off hook but his channel have no external channel to connect. It's looks like a simple telephony glare. Sam בתאריך 30

Re: [asterisk-users] Dropped call after 900s: 481 call/transaction does not exist and another anomaly during re-invite in timer - full anonymized trace attached

-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michael Maier Sent: Wednesday, November 30, 2016 12:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject:

[asterisk-users] Dropped call after 900s: 481 call/transaction does not exist and another anomaly during re-invite in timer - full anonymized trace attached

Hello all! I can see a strange problem during invite in dialog in the context of timer handling. Given is the following incoming call from provider at 8.195.88.234 (2@2) to my asterisk at 28.19.57.152 (1@1): After 900s suddenly *asterisk* starts the timer reinvite - I would have expected the

[asterisk-users] app_queue ringall - 2 agents answer same time problem

hi, our customer reports problem when 2 agents answer the call in the same time faster operator (device) answer the call, but the second is showed up (on device) and call is without sound asterisk 13.9/app_queue with strategy ringall/operators via Local channel with sip device (chan_sip)

Re: [asterisk-users] _FAX_. extension refuses to work !

On Wednesday 30 Nov 2016, Michele Pinassi wrote: >[stuff deleted] > but on a call directed to, es. FAX_3700 i got: > > [Nov 30 11:38:30] NOTICE[5462][C-0027]: chan_sip.c:26309 > handle_request_invite: Call from 'voip-trunk' (xxx:5060) to extension > 'FAX_3700' rejected because extension not

Re: [asterisk-users] _FAX_. extension refuses to work !

Hello, The letter "X" is reserved for dialplan patterns. You should escape it this way : _FA[X]_ Best regards Jean Aunis Le 30/11/2016 à 11:45, Michele Pinassi a écrit : Hi all, my dialplan is: /;

[asterisk-users] _FAX_. extension refuses to work !

Hi all, my dialplan is: /; ==// //; FROM VOIP// //; ==// // //[from-voip]// //include => default// // //[default] / /; FAXes//

Re: [asterisk-users] Asterisk 14.2 CLI don't show debug/verbose data

Yes, it works ! Thanks :-) Michele On 30/11/2016 10:19, Jonathan H wrote: > I think it might be related to this? > https://issues.asterisk.org/jira/browse/ASTERISK-26391 > > I think I remember having to edit logger.conf - this is what mine > looks like now: > console => notice,warning,error >

Re: [asterisk-users] Asterisk 14.2 CLI don't show debug/verbose data

I think it might be related to this? https://issues.asterisk.org/jira/browse/ASTERISK-26391 I think I remember having to edit logger.conf - this is what mine looks like now: console => notice,warning,error messages => notice,warning,error Try that, restart asterisk and see if it works :) On 30

[asterisk-users] Asterisk 14.2 CLI don't show debug/verbose data

Hi all, after upgrading from 13.7 to 14.2, asterisk cli (asterisk -r) don't show what's happens. I've trying setting debug and verbose to 100 but nothing, no show. All commands works as expected but i can't what's happens on my asterisk server. asterisk*CLI> core show settings PBX Core settings