similar to: res_cepstral for 1.6.2

Hello. I spoke with some of the support staff at Digium regarding the Digium/Cepstral partnership. I was trying to find out when something may be available, and if there is documentation about the project. I was told that the project has already been released in to the CVS head, and is available to us now, but not available yet for the business edition. They had no documentation links

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I have a license for Allison-8kHz and two concurrent port licenses that I purchased from Cepstral at the end of last year. I just got around to installing to my * 1.6.0.10 machine. I've decided that the best way for me to integrate the two would be res_cepstral, which I downloaded and installed. Everything is fine, except the register program,

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

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

When I look at the lastest UniMRCP manual, they only mention as high as Asterisk 13. Does anybody know if I need to do anything to allow it to work on Asterisk 14 and, if so, what that is?

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*

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

Interesting. Are you using the included cbend.php script to terminate conferences? I occasionally get questions about using WMM with Confbridge, 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

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

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?

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 getting cycles of repeated crashes which occur and then stop occurring. Looking at the dumps via gdb shows that something peculiar is happening that looks like memory corruption: Program terminated with signal 6, Aborted. #0 0x0000003686e30285 in raise () from /lib64/libc.so.6 (gdb) up #1 0x0000003686e31d30 in abort () from /lib64/libc.so.6 (gdb) up #2 0x0000003686e6971b in

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?

I'm getting dozens of these at a very high rate: [Jan 20 09:15:27] NOTICE[16958]: chan_sip.c:12687 check_auth: Correct auth, but based on stale nonce received from '"" <sip:121 at gnat.com>;tag=as5f1a9480' [Jan 20 09:15:28] NOTICE[16958]: chan_sip.c:12687 check_auth: Correct auth, but based on stale nonce received from '"" <sip:130 at

It was only when I ran AsteriskLint over my dialplan that I noticed this: https://wiki.asterisk.org/wiki/display/AST/Asterisk+14+Application_Set https://wiki.asterisk.org/wiki/display/AST/Asterisk+14+Function_SET Hmmm, they both seem to do the same thing. Or don't they? Confused!

Set(CALLERID(number)=XXXXXXXXXX) works here. Also check with your VoIP provider what 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

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?

>>> 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 the Linux kernel that have no support under the

I did a usual "svn update", "./configure" and "make" and got [CC] chan_oss.c -> chan_oss.o gcc: @SDL_INCLUDE@: No such file or directory I don't see any changes to chan_oss recently, so don't understand this. What could be going on?

Hi, I know I am using SVN, but I was wondering if anybody ever came across this error. I can't read my voicemails because files seems to be corrupted, for lack of a better word. When I do access my messages, I get those errors: [Dec 2 19:45:05] NOTICE[25993]: app_voicemail.c:7432 open_mailbox: Mailbox: /var/spool/asterisk/voicemail/xxx/709/INBOX, expected 0 but found 3