similar to: Meetme delay issue

When a call is placed out the Zap interface there is a long pause followed by an error message from the telco that the call can not be placed as dialed. We have a tdm2413e with 11 1FB (POTS) lines. The number being dialed is a working local number, all dialed numbers get the same error. What should we be looking for ? Best regards, Duane Pudenz Network Infrastructure Manager Shasta

I get the following errors when starting Asterisk. == Parsing '/etc/asterisk/zapata.conf': Found Feb 19 21:14:35 WARNING[10440]: chan_zap.c:920 zt_open: Unable to specify channel 1: No such device Feb 19 21:14:35 ERROR[10440]: chan_zap.c:6860 mkintf: Unable to open channel 1: No such device here = 0, tmp->channel = 1, channel = 1 Feb 19 21:14:35

Hi, I`ve done a few tests with comparing a two personne conf call (both phones next to each other) and a simple call from one phone to the other. In the simple call, there is virtually no delay when I am speaking. In the meetme call, on an underutilized server, with a transcoder card for G729, with one two participants, I have a relatively large delay (1/3 of a sec maybe?). There is an

I've just noticed a strange behaviour with a MeetMe conference. I have a pair of phones (GS BT102) on my desk, and dialled both of them into a conference on speakerphone. If I spoke or made a sound, I heard it replayed from both speakers together a split second later, as expected. I went away for about 15 minutes, leaving the conference running. When I came back any sound I made came back

I recently installed 1.4.5 and I've noticed a recurrence of a problem that I thought was solved long ago, namely a very long (2-4 seconds) delay on meetme calls. That means with two people in the conference room, it takes 2-4 seconds for what one person says to reach the other person. Is anyone else having this problem, and if so, is there a fix or solution? TIA Bruce Komito High Sierra

I have Asterisk running on a Linux 2.4.x box with ztdummy. Once I did a modprobe on ztdummy I was able to enter into a conference room using my softphone clients. I'm using SJphone and Firefly. I have noticed a significant delay (1 to 3 seconds) while talking within the conference room. I have tried both clients, SIP and IAX protocols and various codecs. I have also tried it from different

Hi !! I know that a conference room can be made infinitely. but, I think that there is actually a limit. For example, how many conference rooms can be made from CPU 866 [MHz] and RAM 256 [MB]? Is there any person who tried someone? I am studying MeetMe now. Please tell me a hint!!

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:

I've installed icecast2 and ices (from CVS 3 days ago) on a Redhat9 box (Celeron 2Ghz, 256Mb ram, SoundBlaster Live 5.1 sound card). I'm streaming live from soundcard at 44.100 with a 128kbit stream and 2 channels. The problem is that with 2 channel, there is a small delay between the channels; if I use only 1 channel, all is ok, but I need stereo signal with optimal quality. The cpu

Dear, I have asterisk 1.6.2.13 in a small hardware PBX (512 GB RAM and Celeron), and last days when I call from one extension to another of the same PBX after I dial the number the rings sound after 20 seconds. In the CLI log, when I debug the AGI, I see always goes good until dialparties.agi, and after that there are 20 seconds without any log, and so the ring sound. I've read

Hi List, We have a major issue while broadcasting DTMF using meetme application. We are sending DTMF to asterisk using SIP INFO message with duration 160. INFO sip:xxx at xxx SIP/2.0 Via: SIP/2.0/UDP xxx:5060 From: <sip:xxx at xxx>;tag=43 To: <sip:xxx at xxx>;tag=9753.0207 Call-ID: xxx at xxx CSeq: 25634 INFO Content-Length: 26 Content-Type: application/dtmf-relay Signal= 2\r\n

My boss is complaining that the delay between speaking and hearing in a meetme conference is noticeable and doesn't want to roll out our system until I can eliminate the delay. Personally, I don't think the delay is significant, but I don't sign his check. The system consist of 3 1u's, each with a single quad t1 card. Each card has 2 t1's running NFAS. The "t1

== What is Family Connection? == Family Connection is an easy-to-setup online hub for your family that includes a Family News section with Photos, and an Address Book. For more information, visit http://familyconnect.rubyforge.org/ == Downloads == Latest: svn checkout svn://rubyforge.org/var/svn/familyconnect/trunk Version 1.1.2: svn checkout

Hi, Is it possible to create a meet me room on the go through dial plan? I am looking to use AMI Originate to drop a call into meetme room and once it's proved that party is joined, play him an announcement, grab few numbers from them, and then dial a second number and drop into the same meetme room. The reason to use this is to be able to know when the channels connected because both

Hello All, Asterisk: 1.8.13.0 Dahdi : 2.6.2 Kernel : 2.6.32-431.3.1.el6.i686 #1 SMP Fri Jan 3 18:53:30 UTC 2014 i686 i686 i386 GNU/Linux OS : CentOS 6.4 When I show meetme room details using "meetme list" command it shows Minus in activity column. Any Idea. >meetme list Conf Num Parties Marked Activity Creation Locked 54682 0002 N/A

On 29/03/2020 14:31, Patrick DERWAEL wrote: > Hi Phil > > Here it is: > > > [root at totorbex ~]# lspci -nn > .00:00.0 Host bridge [0600]: Intel Corporation Atom/Celeron/Pentium > Processor x5-E8000/J3xxx/N3xxx Series SoC Transaction Register [8086:2280] > (rev 21) > 00:02.0 VGA compatible controller [0300]: Intel Corporation > Atom/Celeron/Pentium Processor

I want to have a "personal meetme conference", so when on a call I can transfer the other party to my personal conference with "#7". (I can then make other calls, and dump them into the conference using "#7" as well, then join myself by dialing "7"). Using: exten => 7,1,MeetMe(${CALLERIDNUM}|Mpd) this works as long as I originate the call. However,

Is it possible to play and audio file into a meetme conference for both parties to hear? I thought I remembered reading something about it, but I can't find it now. Any help would be greatly appreciated. Paul

We are testing our Asterisk server prior to deployment. The server has a 4 port T1 Digium adapter (WCT4XXP) with 3 Long Distance (LD) T1s and one PRI for local calls. We are using sipp from two different stations routing a test number out the LD lines and another test number out the PRI line. We can not get more then 23 total active calls to connect to the test numbers, the test numbers

Hi there I am using IAX2 softphones dialing into meetme conferences. I also have jitterbuffer=yes, with typical jitterbuffer settings. The problem I am having is that as soon as there is a delay from a participant, then the delay continues until the participant hangs up and dials in again. When dialing in again the delay seems to go. It seems to me as though as soon as the server registers