similar to: Mixing SIP/TDM in MeetMe

Displaying 20 results from an estimated 10000 matches similar to: "Mixing SIP/TDM in MeetMe"

2009 Oct 08
1
MeetMe option question
We've started to use Asterisk for conferencing and have been getting some complaints. Our configuration is that some people call in from home, but we have a physical conference room with a Polycom. When somebody was giving a presentation in the physical conference room, we were told that the remote people kept hearing him cut in and our. To me, this sounds like the talking optimization was
2010 Nov 15
2
Volume on meetme recording
It's kind of low for me. How does one control that volume?
2007 Feb 09
1
Conferencing Phones ...
> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > Gordon Henderson > Sent: Friday, February 09, 2007 9:47 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [asterisk-users] Conferencing Phones ... > > > Anyone got any experiences of good quality
2005 May 28
0
TDM zap channel Exception on 15, channel 1
Hello everybody. I have an customer asterisk 1.0.5 running well since 3 monthes, 2 TDM cards 4 FXO, 4 FXS. Since one week, unable to pass call between Zap and Sip getting the "exception on 15, channel 1" The * box is connected to an eads PBX and it seems that failure started when they make some changes on the PBX. Have someone an idea and what is causisng this failure? Here are the
2009 Oct 17
3
Possible bug in app_meetme.c
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
2010 May 10
2
Speech/DTMF mix?
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?
2003 Dec 03
3
Echo problem on conferencing....no analog interfaces
Okay...here's one for all of you.... 3 party meet-me conference: Call 1: Comes in to MyAsterisk on an E1 PRI into the system. All TDM, no VoIP at all involved. No echo at all. Call 2: Call comes in via IAX....(TDM -> Asterisk_1 -> IAX/GSM -> MyAsterisk. Caller immediately hears his own echo Call 3: Call comes in via IAX....(TDM -> Asterisk_1 -> IAX/GSM -> MyAsterisk.
2003 Dec 14
1
Two Stage Dialing for MF CAMA trunk
Hi all, I am trying to setup a ZAP interface to do MF signaling for a handoff to a 911 tandem. The signaling I need to perform on the T1 is this: 9-1-1 Tandem: Wink CLEC end office: KP (Keypulse) NPA ST (Start) 9-1-1 Tandem: Wink CLEC end office: KP I (Info Digit) NXX XXXX ST As I'm not as familiar with the Zaptel configurabliity, I'm not really sure how to do this. Do I dial twice
2010 Sep 07
3
Losing first DTMF digit (with ASR)
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?
2003 Nov 07
2
Softswitch
Pardon my ignorance, but I was hoping someone could clear up something for me. - For a few POTS lines, digium has a single port card for that, or a T1 card to a channel bank. - For 10 or more lines, digium has a T1 or E1 card for that too based on PRI channels - For 100's to 1000's of lines, I suspect a soft-switch is in order??? A traditional phone company will sell: - POTS lines for
2013 Jan 24
2
g723 transcoding
It appears that there are no transcoders from g723 to anything else in Asterisk 10.7.1. Does anybody know how to fix that?
2010 Dec 25
2
sip.conf, realtime, and LDAP
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?
2015 Jun 18
3
setting outbound caller ID
> CALLERID is a read only variable. That's not correct. I set it all over the place in my dialplan.
2006 Jun 13
7
delay in MeetMe
Hi All! I am facing some delay in conferencing. Using DIAX for Voip calls ,no hardware used yet I am using MeetMe to achieve conferencing and am having a lot of delays. Can anyone tell me how to reduce the delay Regards, Amna Saleem -------------- next part -------------- An HTML attachment was scrubbed... URL:
2005 Mar 21
3
Asterisk/Zaptel on Mac G5 or Xserve
I am considering a G5/XServe for a conferencing system. I need to put a TDM card in the machine for timing. Is anyone out there using Asterisk on Mac with zaptel drivers? I am looking at using Linux for the OS. If so what is your experience? Have a great day! Geoff
2017 Oct 16
2
Confbridge GUI?
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
2003 Nov 07
0
Re: Asterisk-Users digest, Vol 1 #1835 - 12 msgs
Thanks Brian, and thanks again for the included definitions <grin> - that helped too. Your comments are really helping clear many questions. I suppose our intensions are to become an IXC. So if my local carrier is sporting old technology, they'll provide TDM services. So if I understood you correctly, the "in-band signaling" is typically SS7, and the alternative is typically
2004 Aug 16
1
Is "Meetme" a generic term?
Just a trivial question: was the term "Meetme" invented for Asterisk as something like a brand name for its conferencing? Or was it an existing generic term for dial-in conferencing? Cheers Tony -- Tony Mountifield Work: tony@softins.co.uk - http://www.softins.co.uk Play: tony@mountifield.org - http://tony.mountifield.org
2010 Jan 20
2
Odd message: "correct auth, but ..."
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
2011 Nov 28
2
Mixing of compressed streams
Is it possible to mix (sum) two or more compressed audio streams with either CELT or Opus? If not, could you add this functionality? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/opus/attachments/20111128/72223804/attachment-0002.htm