similar to: 11.6 voicemail message cropped off?

Hey all I am running 11.6 and when a caller is sent to vociemail the greeting is cropped off and the beep occurs quickly. Incoming calls are g729 and this occurs where it is using the default greeting or a user provided greeting. I really want to go production with this are there any ideas what could cause an issue like this we have never seen it in 1.4 - 1.8 Bryant -------------- next part

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

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

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

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,

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:

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

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

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

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

forgot to mentioned I am running Asterisk 1.8.22.0 on CentOS. Thanks, On 04/27/2015 02:38 PM, Motty Cruz wrote: > here is what I have: > > exten => _9XXXXXXX,1,Set(l_HomeAreaCode=381) > > exten => _9XXXXXXX,n,Set(dialnumber=${l_HomeAreaCode}${EXTEN:-1}) > > exten => _9XXXXXXX,n,Dial(SIP/SIP-Provider/${dialnumber},80) > > not having success; > >

here is what I have: exten => _9XXXXXXX,1,Set(l_HomeAreaCode=381) exten => _9XXXXXXX,n,Set(dialnumber=${l_HomeAreaCode}${EXTEN:-1}) exten => _9XXXXXXX,n,Dial(SIP/SIP-Provider/${dialnumber},80) not having success; "Got SIP reponse 503" Service Unavailable" On 04/27/2015 02:19 PM, Bryant Zimmerman wrote: > Motty > Yes > From your dial plan accept 9 + 7 digits

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:

I am running 1.8.3 and my BLF lights have stopped working. The hints appear to be intact when I use core show hints. But none of the phones are getting the BLF updates. This has happend in the past and I have had to restart my server. What could be causing this to occur. It did not do this with the 1.6.x builds. Is there a way to reload the hints or force a refresh without re-starting

It appears that each time a call is parked that the CDR billing seconds are lost and they start again when the parked call is picked back up. The call duration is correct. What is the best way to address this issue to get proper bill seconds? Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 -------------- next part -------------- An HTML attachment was scrubbed... URL:

I am doing dynamic conference bridges using confbridge in asterisk 11. Is there a way to toggle off an on recording of an ongoing conference call I have figured out how to record a conference if it is turned on when someone enters. Also I have noticed that when setting music_on_hold_class dynamically it does not override what is set on the channel. exten =>

D'Arcy J.M. Cain If the device is registering and then dropping there are several usual items. The router may be closing the ports on the device. The router may have a AGL SIP helper that is causing issues. Make sure that the device is sending out keep alive packets. Shut down any AGL helpers on the router. Make sure that the site is not double NATing Try using a stun

SIPAddHeader(Alert-Info:\;info=ring3) In the phone config add the value "ring3" and select Account # / Call Settings / Match Incoming Caller ID (Matching Rule) In the first rule place the word ring3 and select your ring tone. This will cause the selected ringtone to be used when calls with the info value of ring3 is matched Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext.