search for: diehlnet

...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 hand, doing it in the Asterisk dialplan gives me centralized control. I'm torn. Anyone want to weigh in on this? -- Mike gpg key: http://diehlnet.com/~mdiehl/mdiehl.asc 83AD D927 758D 4BFC A800 0277 4B26 75A4 F0D1 C7EB

...======== *CLI> It doesn't seem to be creating any bridges and I'm sure I've left something basic out. What am I missing? Finally, once I've got this working, can all of this be put into a database? If so, what table structure do I use? Thanks in advance, -- Mike Diehl Diehlnet Communications, LLC. Voice: (505) 903-5700 Fax: (505) 903-5701

...39; at line 1 I'm using ODBC for sip and voice mail configuration. I'm running Asterisk 11.20.0-rc3. I've been told that there is a particular version of odbc that is stable. In the mean time, I'm trying to run unixODBC 2.3.2. What version SHOULD I use? TIA, -- Mike Diehl Diehlnet Communications, LLC. Voice: (505) 903-5700 Fax: (505) 903-5701

...#39;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'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

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

...is that I often telecommute and sometimes I do some side business; these practices violate many provider's acceptable use policies. So, I need a provider who doesn't care how I use the phone, and one that works well with Asterisk. Any comments welcome. Thanx, -- Mike gpg key: http://diehlnet.com/~mdiehl/mdiehl.asc 83AD D927 758D 4BFC A800 0277 4B26 75A4 F0D1 C7EB

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'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'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, Since 21 june skype is available to be used on Linux, with a static binary, which includes QT, of 8 meg its big. http://www.skype.com/help_linux_faq.html I presume, with some hacking, there could be a possibility to use the Skype program as a Channel. (Eq. Skype is started, and with a visual scripting thing a connection is made and Asterisk connects via OSS (or the alsa emulation

...do if [ -d ${BASEDIR}${ext} ];then for msgdir in $(ls -d ${BASEDIR}${ext}/*); do ProcessDir ${msgdir} done else echo "${BASEDIR}${ext} is not a valid directory" fi echo "Processed extension $ext" done On Mon, Oct 9, 2023 at 3:06 PM Mike Diehl <mdiehl at diehlnet.com> wrote: > 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 mai...

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

...om the previous configuration was to convert to odbc voicemail storage. > The more info you give us, the more likely it is we can suggest something > useful. Ya, I understand; I was just tired... and frustrated. Thanks again for your time. -- Mike Diehl Diehlnet Communications, LLC. Sales: (800) 254-6105 Support: (505) 903-5700 Fax: (505) 903-5701

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

I am wondering if anyone has sample XML config for the Sipura 2100 ATA. We have been autoprovisioning our 2002s with success and the 2100's take the same XML that we have come up with, but I am not sure of the syntax for specific things that I need these boxes to do, such as turning T.38 on. If anyone is willing to share their xml for autoprovisioning Sipura 2100's, it would be much

...icemail at work using phones on my Asterisk system. I have to press the * key during the voicemail login process. When I do, it seems that Asterisk "eats" it and doesn't send it along. I suspect it has something to do with the features.conf file, which you can look at at: http://diehlnet.com/features.conf Otherwise, any advise would be most welcome. Mike Diehl.

Hi all, I'm looking into being able to send/receive SMS messages with my asterisk box in the US. I've seen the SMS command as well as the Kannel program. I'd prefer to do it from Asterisk. I've tried something like: exten => 999,n,sms(15551234567,s,"This is a test") in my dialplan, but when this runs, it dials the phone number and then nothing. What am I

Hi all, I'm trying to get moh working on * version 1.4.4. I've setup a test extension that answers the call and runs the musiconhold command with the appropriate class name. All I get on the phone is silence. The console tells me that moh started and immediately stopped, but it complains that there is "No class: moh0" *CLI> [Feb 22 12:17:36] WARNING[31142]: