similar to: AGI script VERBOSE cmd

I am working on an AGI script and all is going well except I can not get any of my "VERBOSE" commands to display. Is there any undocumented reason for this to occur? I am able to set variables, call other commands ect.. I am sending my verbose command in the following format (VERBOSE "Message to send" 4) Any ideas what I might be doing incorrect? Thanks

---------------------------------------- From: "Andrew Latham" <lathama at gmail.com> Sent: Thursday, January 17, 2013 3:04 PM To: bryantz at zktech.com, "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com> Subject: Re: [asterisk-users] Conf Bridge On Thu, Jan 17, 2013 at 3:02 PM, Bryant Zimmerman <BryantZ at

Update When no greeting is recorded the default you have reached ext # greeting is cropped. When there is a greeting it is just ignored and not played at all. Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 ---------------------------------------- From: "Bryant Zimmerman" <BryantZ at zktech.com> Sent: Saturday, November 23, 2013 8:32 AM To: asterisk-users at

The Cisco/Linksys SPA devices are also able to be provisioned automatically. On Wed, Apr 15, 2015 at 3:20 PM, Bryant Zimmerman <BryantZ at zktech.com> wrote: > Alejandro > > All of the Grandstream devices can be remote provisioned if you know what > you are doing. > > Bryant > > ------------------------------ > *From*: "Alejandro" <cdgraff at

Hi Scott, thanks for the answer, can share some link or documentation about how setup this in SPA3102? I try to get something about this using google, but found comments but nothing useful. Alejandro 2015-04-15 19:28 GMT-03:00 Scott Griepentrog <sgriepentrog at digium.com>: > The Cisco/Linksys SPA devices are also able to be provisioned > automatically. > > On Wed, Apr 15,

Did you open a Jira issue for this yet? I can actually work on this this week. On Fri, Oct 16, 2015 at 9:44 AM, George Joseph <george.joseph at fairview5.com> wrote: > On Fri, Oct 16, 2015 at 4:00 AM, Bryant Zimmerman <BryantZ at zktech.com> > wrote: > >> Is there a way to limit the items returned by pjsip show [type] using like >> > > There isn't but

I am testing out inbound faxing using res_fax and res_fax_spandsp.so My system answers the call but then sets there on the ReseiveFax line then comes back with an error that it exceeded the maximum retries. How would I go about debugging this? Below is my very simple dialplan code I am using, and the fax show version gives the following as well. FAX For Asterisk Components:

I have a project that is requiring the use of MS SQL from asterisk. I get an error when the pjsip contact tries to update the contact table. [Oct 15 21:34:55] WARNING[3033]: res_odbc.c:649 ast_odbc_prepare_and_execute: SQL Execute returned an error -1: 22018: [FreeTDS][SQL Server]Conversion failed when converting the varchar value '3.000000' to data type int. (101) The datatype

Hey all I have been beating on this all weekend long. Any feed back would be appreciated. We stood up a 11.6 system. We tested everything we could think of. We moved over to it and all seemed to be working good than a customer told us that they were not hearing our vociemail greetings. When we call into the system and it drops to voicemail we just get a beep no greeting played. We checked

Where can I get the latest stable version of spandsp. That will work with res_fax_spandsp.so. The link listed on the voip-info website is dead. Any other location for download? http://www.soft-switch.org/ Thanks Bryant Zimmerman -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hey all For some reason the mailing list is sending all messages from the sending party. This makes it less than ideal when responding; as selecting reply goes to the person and not the list. Can we have it set back to the old way please? Thanks Andrew for pointing this out to me. Bryant -------------- next part -------------- An HTML attachment was scrubbed... URL:

I am having issues with Blind Transfer on asterisk 1.8 If I call from one Grandstream phone to another and us the transfer key to do a blind transfer everything works fine. When calling in on a sip trunk and then trying to use the transfer key to transfer from Grandstream phone to Grandstream phone the call just hangs up. It did not do this on Asterisk 1.4.x or 1.6.2.x . If we use

Hey all I am having instances where voicemail boxes will have a 00001 message and no 00000 message this causes the user to be told that they have a message that they can't get at. If I renumber the messages manually to start with the 00000 numbering then the user can get their messages. What could be causing this and how can I get it out of the system. Is there a patch I can apply to the

Is there a way to limit the items returned by pjsip show [type] using like chan_sip allowed for sip show peers like xxxx, but I can't seem to figure out how to lookup or limit my returns with pjsip Thanks Bryant -------------- next part -------------- An HTML attachment was scrubbed... URL:

We are having an issue where on the latest version of asterisk when configuration pjsip via realtime. we do a pjsip list endpoints it shows our endpoints but lists them as invalid. When we do the pjsip list endpoints again it shows no objects. This applies to pjsip list aors as well. We did not have this issue on our older asterisk 13 installs. My guess is something has changed

We have written some monitoring and stat collection scripts that use asterisk -rx "command" The script runs once a min and logs data and posts any critical notifications. Everything is working well with this method but we get the -- Remote UNIX connection / disconnect message once a min and we would like to suppress it. Is it possible without reducing the verbose logging level.

Bryant, I have the same problem with dynamic public IPs and PJSIP. What is your idea to solve the problem? My suggestion would be to write a script that monitors the change, pjsip.transports.conf updated and Asterisk restarts? Daniel > Am 26.01.2016 um 14:21 schrieb Joshua Colp <jcolp at digium.com>: > > Bryant Zimmerman wrote: >> Joshua >> So once a transport is

I did an upgrade to the SVN trunk on the 12/9 and when I looked in my mysql table for CDR's today there are no entries since the update. I have rebuilt and re-installed and re-started asterisk still no CDR's flowing to mysql. I did not change any configs. I checked to make sure that the cdr_mysql option was selected under the make menu options. The module shows it is there when I do a

I have the following senerio. Call file calls 1st party. When connected give called party option to connect to second party. Issue Dial to second party. Caller answers and the two are bridged together. My issue is that 4 out of 5 calls fail to bridge the audio. Am I missing something or is there some kind of bug? Here is my test dialplan ;Dialer Base Code Files. ;Variables

Hey all. RE: Conf Bridge. I am looking into a project that would need 8 to 10 thousand parties in a single conference. Most would be on mute but 5 to 6 would be presenters. Is the new conf bridge solid enough to handle this kind of load? Any ideas on hardware projections? If not 8 to 10 thousand how many would be realistic? If not asterisk any other suggestions. Thanks for any input.