similar to: Park function and billsec

---------------------------------------- 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

I have been trying to get the dynamic parking working. For some reason when I park a call using this method the console says it is using the default parking context not the one I am trying to specidfy. It also is playing the parked extension to the caller. I am transfering the call to an extension that is doing a goto to the context below. Any ideas or examples on how to make this work.

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

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

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

Does anyone have any examples of using realtime database driven call parking lots. I am on version 1.8.x My goal is to be able to do database driven multi-tenant parking lots with out adding sperate entries into Features.conf for each lot. I also need to be able to use the same parking extension pool for each tenant but sand box them into sperate lots. We have been able to do this for every

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

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

I want to add an entry to a database every time a brute force registration attempt is done. from this database we are updating cisco routers with our ban list so our entire network is protected. The database side of things is working and has been for some time. I really would like to add the file2ban side of it to protect our asterisk system better. How would I best go about doing this

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:

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.

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:

How can we set the music on hold class using the Dynamic Parking lots? The variables set the PARKINGLOT, PARKINGDYNAMIC, PARKINGDYNPOS,PARKINGEXT,PARKINGDYNCONTEXT I can't find a PARKINGMOH variable. This is becoming a big issue. We are using the current release 11. version We have to be able to set the MOH dynamically I just can't find the mechanism. Any ideas? Thanks

Hey all Please disregard my question. I was looking for the word Verbose to show up. I was just being dense. There was no real issue it is working just different than what I was expecting. Thanks Bryant ---------------------------------------- From: "Bryant Zimmerman" <BryantZ at zktech.com> Sent: Friday, June 27, 2014 11:25 AM I am working on an AGI script and

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,

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

I have deployed several 1.8.3.2 systems as upgrades of customers systems and now I am seeing random crashes. For some reason the builds lock up and stop taking sip connections. Existing calls stay on but when the user hangs up no new calls or reg attempts work. In most cases a "core restart now" cleans things up. Some times I have to kill the asterisk process. The stability of 1.8.2

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 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

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: