search for: kenner

Is this patch correct? The "&&" doesn't make logical sense to me. I think it should be "||" and making this change fixes the problem I have with SIP phones in MeetMe conferences. If it's correct, is there someplace more formal that I should submit it to? *** app_meetme.c.old 2009-10-11 17:56:44.000000000 -0400 --- app_meetme.c 2009-10-17

I'm having a wierd problem. Somewhere around 1-2% of the time, the first DTMF digit dialed gets dropped. This is occurring during a SpeechBackground application call. If the caller reenters the digits when given a second chance, all is OK. Any suggestions how to debug this intermittent problem?

...tQueue2: [testQueue2] music=default strategy=ringall timeout=15 retry=5 wrapuptime=0 maxlen = 0 announce-frequency = 0 announce-holdtime = no member => Agent/6002 servicelevel = 60 ----------------------------------------------------------------------------- Thanks a lot, Thomas -- Thomas Kenner

> CALLERID is a read only variable. That's not correct. I set it all over the place in my dialplan.

I'm confused exactly what's supported with LDAP and Asterisk. What I want to do is to have SIP peer information read directly (in realtime) from LDAP. Can this be done? If so, with what Asterisk versions?

It appears that there are no transcoders from g723 to anything else in Asterisk 10.7.1. Does anybody know how to fix that?

>>> On Aug 2, 2017, at 6:45 AM, Richard Kenner kenner at gnat.com wrote: >>> I wouldn't believe it either. I'd be quite surprised if something won't >>> work with any ESXI version. *Perhaps* there's a configuration issue, but >>> I'd be surprised about that too. There are certain versions of th...

...bridge, and to date I have not had an answer . If you can provide details, even vague ones, about how you did it, I can update the WMM package. Dan -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Richard Kenner Sent: Friday, October 13, 2017 2:14 PM To: asterisk-users at lists.digium.com Subject: Re: [asterisk-users] Confbridge GUI? > I have a very old server that is used only for conferences on > Meetme. To manage the conference rooms we use Web Meetme. Now it is > time to upgrade ever...

It's kind of low for me. How does one control that volume?

Which speed recognition products will also recognize DTMF? In other words, I want to say "Please speak or dial the conference number". Does Vestec allow that? LumenVox? Any other way?

I'm confused. What does Asterisk do when it gets a 302 with a new number to forward to? Is there anything I have to do in the dialplan to make this work? I can't find any clear documentation on this issue.

I recently tried to update my Aastra 57i to version 3.2 and ran into a problem. It won't properly register and says "contact mismatch". I added "sip contact matching: 2" to aastra.cfg, but that didn't help. When I look at the SIP trace, but I see is the Aastra sending a REGISTER and Asterisk replying with the 401. The phone then sends the REGISTER again, this time

I'm trying to interface Asterisk with an Alcatel PABX and trying to find a code that works well. It says it doesn't support ulaw, though it doesn't reject it. It supports G.729, and that works fine, but we'd prefer not to use compression. When I use alaw, the path from Asterisk to the Alcatel is completely clean, but the other way has a set of clicks that kind of sound like

On Thu, 18 Jun 2015 13:45:10 EDT kenner at gnat.com (Richard Kenner) wrote: > > CALLERID is a read only variable. > > That's not correct. I set it all over the place in my dialplan. Then someone needs to fix this page. https://wiki.asterisk.org/wiki/display/AST/Asterisk+Standard+Channel+Variables -- D'Arcy J....

...format they want for the number. (I believe) most accept a 10-digit number, but I seem to remember reading about the odd provider that wanted a leading "1". On Thu, Jun 18, 2015 at 11:47 AM, D'Arcy J.M. Cain <darcy at vex.net> wrote: > On Thu, 18 Jun 2015 13:45:10 EDT > kenner at gnat.com (Richard Kenner) wrote: > > > > CALLERID is a read only variable. > > > > That's not correct. I set it all over the place in my dialplan. > > Then someone needs to fix this page. > > > https://wiki.asterisk.org/wiki/display/AST/Asterisk+Standa...

> A Siren codec is not currently available and the one for 12 will not > work. I have no timeframe for when this might change. So the only option is to build one from the Polycom sources? I'm already doing this for Siren14 (I forget why).

On 2017-08-01 15:48, Doug Lytle wrote: >>>> I am having a very tough time trying to replace an Elastix 2.X >>>> install running as a virtual machine on ESXI 4 > > Licensed or free ESXI? > > I want to say your version is too old. I'm currently running ESXI 6.0 > update 3 at home and Asterisk in a VM under debian without issue. > > Doug The

I sent a query about this before, but have some further information and am hoping somebody has a suggestion as to what to try next to debug this. I'm using an Asterisk box primarily for MeetMe conferencing. There are two sources: TDM via two Q.SIG T1's and SIP phones. Conferencing works fine between TDM channels. But when a SIP phone calls the conference, there's no voice path *to*

I am running asterisk 1.6.2.6 and have configured hints for our extensions and have a couple of Aastra 6755i test phones. The phones register fine but 'core show hints' shows the lines as idle even if they are in use. I read the wiki and see mention about needing to set call-limit in asterisk 1.4 but that has been depreciated in 1.6 so what is the way it should be done in 1.6?

I'm now getting these errors: [Jan 25 09:19:01] WARNING[29877]: abstract_jb.c:284 ast_jb_put: DAHDI/i1/2128518396-ba7 received frame with invalid timing info: has_timing_info=1, len=0, ts=426891164, src=RTP [Jan 25 09:19:01] WARNING[29877]: abstract_jb.c:284 ast_jb_put: DAHDI/i1/2128518396-ba7 received frame with invalid timing info: has_timing_info=1, len=0, ts=426891174, src=RTP even