similar to: ODBC crashing asterisk

Hi all, I'm trying to configure a few conference bridges. I've started with the very basic: [general] [default_bridge] type=bridge [default_user] type=user [default_bridge] type=bridge [5340] type=bridge However: confbridge list Conference Bridge Name Users Marked Locked? ================================ ====== ====== ======== *CLI> It doesn't seem to be

Hi all, I've got a device that seems to become unreachable for about 2 minutes, every hour. From what I can tell, it isn't due to network or server issues. Any ideas? TIA. -- Mike Diehl Diehlnet Communications, LLC. Voice: (505) 903-5700 Fax: (505) 903-5701

I am trying to get cdr via odbc to work on Asterisk 13.7.2 but I keep getting this error: [Feb 9 16:21:43] WARNING[2088]: cdr_odbc.c:160 execute_cb: cdr_odbc: Error in ExecDirect: -1, query is: INSERT INTO cdr (calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode,uniqueid,userfield,peeraccount,linkedid,sequence) VALUES ({ts

I use realtime on my asterisk installation. I have always used mysql for my realtime connection but as mysql seems to be on the "soon to be deprecated" list of asterisk features I am trying to move to ODBC (still using MariaDB/Mysql on backend). I find ODBC support in Asterisk very unstable. Just today my asterisk server (a test server and a production server) has crashed often

Thank you for your time. I've put my replies to your questions in-line, below. On Wednesday, June 07, 2017 10:19:41 AM Antony Stone wrote: > On Tuesday 06 June 2017 17:54:59 Mike Diehl wrote: > > > Hi all, > > > > I'm upgrading to Asterisk 13.14.0 x86_64. During my beta testing, I've > > discovered that my server crashes as soon as I leave a

Hi all, I'm upgrading to Asterisk 13.14.0 x86_64. During my beta testing, I've discovered that my server crashes as soon as I leave a voicemail message. I'm using odbc voicemail storage as well as mysql dynamic configuration. I'm using unixODBC 2.3.2-r2 with myodbc 5.2.7-r1 I suspect that the odbc drivers are the problem. Is ther an alternative drive that I should be using?

Hi all. I'm in the process of configuring a phone system for my family and friends. I'm wondering if I should try to implement the "Vertical Services" (http://www.nanpa.com/number_resource_info/vsc_assign) in the Asterisk dialplan, or if I should delegate those functions to the various ATA's. For example, the Sipura SPA 2002 can handle*69 internally. On the other

Hey everyone! This is a friendly reminder that Asterisk 12 will be entering security fix only mode soon. As a Standard release of Asterisk, Asterisk 12 received one year of maintenance fixes, and will receive one year of security fixes. Asterisk 12 was first released on 2013-12-20 - the one year anniversary of which is just around the corner! After 2014-12-20, additional releases of Asterisk 12

Hi all, I've upgraded to Asterisk 13.14.0 and now I'm seeing that Asterisk is using 100% CPU. I have one AMI agent connected that is acting rationally. I've got a hand full of SIP (RT) registrations. There is no other call activity. I've tried to unload various modules; nothing resolved the issue. Any suggestions? -- Mike Diehl

Hi all, While trying to track down a T.38 issue, I came across a series of log entries like this: ============================================================================ [Jan 26 10:23:31] WARNING[32508]: udptl.c:948 ast_udptl_new_with_bindaddr: Unable to allocate socket: Too many open files [Jan 26 10:23:31] ERROR[32508]: acl.c:488 ast_ouraddrfor: Cannot create socket

Hi all, I've trying to fix a problem. If I'm in a call and I press the * key, the call goes silent but doesn't hang up. I need to be able to send the * key for various IVR's that I interact with. Since I thought this was related to the features.conf file, you can view it at: http://www.diehlnet.com/features.conf Any ideas are welcome. TIA, -- Mike Diehl

Hi all, I seem to be having a strange problem with a sip trunk. On a fairly frequent basis, I'll make a call, ore receive a call, and there will be NO sound. The strange part is that both endpoints are public IP addresses so NAT isn't in play and a sniffer trace reveals that the packets simply aren't being sent. It only seems to happen on a particular trunk. The same phone

Hi all, I'm installing Hylafax on my Asterisk system. From what I've read, I can either use IAXModem or T38Modem to provide the virtual fax device. So at the risk of starting a religious war, which one should I use? I don't mind running IAX if I have to. I want as much flexibility and stability as I can get. So, what are your recommendations? Mike. -------------- next part

Hi all, I'm trying to tighten things up a bit and I seem be be running into something that doesn't make sense to me. I've got 2 contexts, one for customers, and one for guests, that I include into [customers] and [default], in extensions.conf, as below: ============================================================= [default] include = dial_GUEST [customers] include = parkedcalls

Hi all, Lately, I've been seeing more and more instances where I get a flood of warning messages like this: [Apr 26 14:09:50] WARNING[21054] udptl.c: No UDPTL ports remaining The next thing I know, my server is dropping calls and starting to misbehave. I use fax via T.38, so I can't just turn udptl off. I could expand the port range, but I suspect that will just mask the situation.

Hi all, I've got a user who wants to receive voicemail notifications at two different email addresses. I could probably setup an alias in /etc/aliases, but then I'd have to manage that across multiple servers, which I don't want to do. Is there a way I can tell Asterisk to send to multiple addresses? Mike -------------- next part -------------- An HTML attachment was scrubbed...

Hi all, I've got a problem where many times, there is silence at the first 1-2 seconds of a call. Then it clears up and it's crystal clear. I've not put a sniffer on it, yet, but I suspect that the media channel is still being set up. The server shouldn't be too overloaded. Can anyone give me some advise on how to solve/mitigate this problem? Mike.

Hi all, I have a user who would like to stream their favorite radio station from iHeart radio for their music on hold. It this TECHNICALLY possible? If so, any pointers would be appreciated. Is this LEGAL in the US? Thanks in advance, Mike. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, Is it possible to send a SIP header to a PAP2T or SPAxxxx and cause the device to automatically answer? I can do this with my Polycom phones and would like to do it with my ATA's. Any ideas? -- Take care and have fun, Mike Diehl.

Hi all, I need to be able to delete a voicemail message using a program. Is is sufficient to simply delete the .wav and .txt files in the spool directory? Or do I need to also renumber the remaining files? For example, let say a given mailbox has 20 messages in it and I want to delete message number 5. Can I just delete the 2 files and expect that asterisk will renumber them? Or do I