similar to: Create meetme conference rooms on the flight.

Displaying 20 results from an estimated 3000 matches similar to: "Create meetme conference rooms on the flight."

2007 Mar 05
4
TC400B
Anyone tried the digium TC400B transcoding card? What are your opinions? Thnx
2007 Jul 02
4
Help. Cannot compile version 1.4.6 with the following error
Hi all, I need the zap channels going, but got the following error. What do I need to change in my configuration? Thnx. chan_zap.c: In function `zap_send_keypad_facility_exec': chan_zap.c:2309: warning: implicit declaration of function `pri_keypad_facility' chan_zap.c: In function `pri_dchannel': chan_zap.c:9292: structure has no member named `call' make[1]: *** [chan_zap.o]
2007 Sep 10
5
Asterisk Manager API - Originate command
Hi all, Just ran into some issue with the originate AMI command. It seems that there is a limit of around 120 calls I can place with the originate command simutanously. By that I mean sending Asterisk a lot of originate command very fast. Anyone know if there is a limitation? Thnx. -------------- next part -------------- An HTML attachment was scrubbed... URL:
2007 Sep 18
4
Linux limits
Hi all, Any one know how to increase the Linux limit? I am hiting a wall on 200 calls playing files at the same time. From Asterisk console, I am getting messages like Sip_request_call: Unable to build sip pvt data for "asterisk1/700" Too many open files Is this a limit of my Linux box? I only have 512MB of ram. Will increase it to 2G help or I have to change some configuration in
2007 Sep 20
4
Asterisk 1.2.24 simultaneous call limits.
Hi everyone, I am running into wall today with simultaneous call limits. I have two Asterisk machines (fast 3GHz C2D with 2GB of ram). I tried to create a lot of sip calls from one machine to the other by issuing AMI Originate commands to one machine. The machine that makes calls plays a message (demo-intruct) upon the other machine answer. The machine receives the calls just waits for 40 seconds
2006 Mar 22
6
Can this box handle 8 T1s (PSTN) with Asterisk?
Hi all, I am handed a project to setup *. The requirement is that it can handle 8 T1s. Half of the calls coming into the system will be routed to SIP extensions (with transcoding). The machine we have in our disposal is a new dual Xeon 3.2gHz server with 2g of ram and an dual 1000mb nic. Voice will be coming in from the PSTN (through 2 quad digium cards) in g711ulaw, and most of the time will
2006 Feb 08
6
Connecting to live calls
Hi all, Is there a way to connect two live calls through the manager api without directing them to a meeting room? Currently, I can connect them by sending them to a meeting room. However, I don't know what the overhead is, and I kind of think that if I can connect them or link them up, the overhead would be minimum. -------------- next part -------------- An HTML attachment was scrubbed...
2006 Apr 03
3
Monitor or mixmonitor
Hi all, I am setting up a script to record all the call. There are two app for recording. "Monitor" and "Mixmonitor", one mixing the audio on the fly and one mixing it at the end but also allow a option not to mixing the audio at all. If mixing the audio on the fly is not that taxing on the CPU, I would like to use 'mixmonitor' app. My question is, what is penalty on
2006 Apr 27
12
PRIs from two different telco
My TE411p does not seem to like to have two PRIs from different telcos (span 1 and span 2). I can get one working, but not the other. However, if I use vpmsupport=0 when loading the wct4xxp module, they both work. But here is the problem, vpmsupport=0 disables the on board echo cancellation. Any ideas? BTW, here is zaptel.conf span=1,1,0,esf,b8zs span=2,2,0,esf,b8zs bchan=1-23 dchan=24
2007 Mar 01
1
Test
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2013 Jul 25
2
limitation on number of contexts in extensions.conf
Hello Asterisk version 1.6.2.9. I want to know is there any limitation on number of contexts or including external file (#include <filename>) which can be defined in extensions.conf. When I try to include around 40 external files, my dialplan doen't get reloaded. Regards, Kamlesh -------------- next part -------------- An HTML attachment was scrubbed... URL:
2002 Dec 13
2
Must Samba have DNS server?
Dear all, I have an Linux Server 7.1 which doen't connect to internet and doesn't have a DNS server as well. I use 192.168.10 IP addresses. If I still setup Samba server (ver 2.2), can Windows 2000 make a connection to the Samba server? Many thanks Tan Duong -------------- next part -------------- HTML attachment scrubbed and removed
2003 Aug 15
2
Samba3: PDC and local admins
Hi! I have samba 3 beta2 as PDC. Now I need to make all mebers of the unix grop "users" local admins on their Windows systems, because Wordperferct 8 doesn't run otherwise. As the "domain admin group" setting from smb.conf doen't exist anymore, I don't know, how to do the group mapping correctly. Could someone explain the steps to do it? Thanks in advance for
2009 Oct 23
2
How to generate 183 Session Progress
Hello everybody, I have 2 users connected on the same Asterisk server that are connected with 2 different Asterisk servers. For outgoing calls, one in receiving 183 Session Progress and the other not! Do you have any idea why? Thanks. I have tried to understand by myself and in their INVITE they have almost the same Allow and Supported SIP Headers The one that works: Allow: INVITE, ACK,
2006 Jul 09
2
Wine - almost a laugh but really a cry
After reading about a major Direct3D rewrite in the new wine-0.9.16 version , I gave it a try (again) and installed it on my Fedora Core 5 system (Athlon64 3200+ with a Geforce 6600GT and latest proprietary nvidia driver installed - kernel-2.6.17-1.2145_FC5). To see whether anything has really changed, I dug up my old Diablo2/LoD and installed it. And Lo, things have changed! Whereas previous
2006 Apr 12
4
call center running Asterisk -sound quality-critical!
Except that mixmonitor still has a bug in it. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Kevin P. Fleming Sent: Wednesday, April 12, 2006 11:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] call center running Asterisk -sound quality-critical! Matt Roth
2006 Mar 13
1
Need help implementing call center featuresofAsterisk
It sounds like Naren and company has their own CRM application. They need a predictive dialer that allows third party app integration. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of Matt Florell Sent: Monday, March 13, 2006 8:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re:
2009 Mar 04
3
Converting file system
Hi @all! I tested what happens with a file which is saved at a FAT32 partition and then this partition is converted to NTFS. So first I transfered the file with rsync from the FAT32 partition to my Linux /home folder. Then I converted the FAT32 partition to NTFS. After the convertation I transfered the file again to see what has changed (amount of data). I was surprised!!!! Nothing has changed!!!
2005 Nov 18
3
How to run R in batch mode
Hello to everybody! I want to run R in batch mode but it doen't work (Error: syntax error) I've found this in R help: R CMD BATCH [options] infile [outfile] I have tried differents commands: (I have been working in the same directory I have "test.txt" file and "test2.txt" would be the output file) > R CMD BATCH "test.txt" or > R CMD BATCH
2006 Nov 03
0
*****SPAM***** Meetme Conference Rooms
Software zur Erkennung von "Spam" auf dem Rechner priamus.teamware-gmbh.de hat die eingegangene E-mail als m?gliche "Spam"-Nachricht identifiziert. Die urspr?ngliche Nachricht wurde an diesen Bericht angeh?ngt, so dass Sie sie anschauen k?nnen (falls es doch eine legitime E-Mail ist) oder ?hnliche unerw?nschte Nachrichten in Zukunft markieren k?nnen. Bei Fragen zu diesem