Re: [asterisk-dev] Asterisk 13.17.1 Crash on ConfBridge - NetGen ATA

2017-12-19 Thread Bryant Zimmerman
was occurring? Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 From: "George Joseph" <gjos...@digium.com> Sent: Tuesday, December 19, 2017 2:59 PM To: brya...@zktech.com, "Asterisk Developers Mailing List" <asterisk-

Re: [asterisk-dev] One sip stack to rule them all....

2017-10-08 Thread Bryant Zimmerman
I would agree with this. We have tried to deploy pjsip several times over the last year with limited success. We have had nothing but issues with database real-time deployments. Tables not working from one 13.x release to another. Table builders sorcery failing out. Issues when there are

Re: [asterisk-dev] Park patch to silence slot number

2017-10-03 Thread Bryant Zimmerman
calls at same extension and same context asterisk unable to do it, because unable to insert second time same extension in dialplan. I see no problem in using different extensions range for all parking lots in same context of using different contexts for all users" On 3 October 2017 at 16:

Re: [asterisk-dev] Park patch to silence slot number

2017-10-02 Thread Bryant Zimmerman
You don't need a patch this is possible with the current tools if you keep track of parks pickups and ring backs. We use the current system to do dynamic parks all of the time in multi tenant environment . We create dynamic lots per tenant and address them per sub account. This allows for

Re: [asterisk-dev] sip Messages - No acces to variables set on the peer.- Solved

2017-03-10 Thread Bryant Zimmerman
I will just use sippeer to read the values I need. Thanks Bryant I am working on an application that uses sip messages to send sms. I am bumping into what I think may be a bug. The message is being sent to the asterisk server

Re: [asterisk-dev] sip Messages - No acces to variables set on the peer.

2017-03-10 Thread Bryant Zimmerman
I am working on an application that uses sip messages to send sms. I am bumping into what I think may be a bug. The message is being sent to the asterisk server from the extension, but I have no access to the setvar variables or any variables that would normally be on a standard channel

Re: [asterisk-dev] ARI versioning in 13 and 14

2016-11-17 Thread Bryant Zimmerman
+1 to option 2. Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit

Re: [asterisk-dev] Issue AGI Get Full Variable

2014-09-02 Thread Bryant Zimmerman
Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com

Re: [asterisk-dev] Issue AGI Get Full Variable - Solution

2014-09-02 Thread Bryant Zimmerman
From: Bryant Zimmerman brya...@zktech.com Sent: Tuesday, September 2, 2014 3:10 AM To: Asterisk Developers Mailing List asterisk-dev@lists.digium.com Subject: Re: [asterisk-dev] Issue AGI Get Full Variable I am calling the GET FULL VARIABLE agi command. I am passing in the variable

Re: [asterisk-dev] Asterisk and video conferencing

2014-03-31 Thread Bryant Zimmerman
Sangoma has trans coding solutions that allow for use with virtual machines. They had video codecs in their road map. I am not sure if they have them yet? Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 From: Dan Cropp d