similar to: Using patch -p0 <meetme-diff-cbmysql_1.txtproduces 'malformed patch' message

Displaying 20 results from an estimated 2000 matches similar to: "Using patch -p0 <meetme-diff-cbmysql_1.txtproduces 'malformed patch' message"

2005 May 22
0
Using patch -p0 <meetme-diff-cbmysql_1.txt produces 'malformed patch' message
I have googled for several hours and have read several threads but I have not found an answer yet. I have downloaded asterisk-1.0.7 and WebMeetMe-Gui. I have tried to use the diff file 'meetme-diff-cbmysql_1.txt' to add the changes needed for WebMeetMe-Gui. Using 'patch -p0 <meetme-diff-cbmysql_1.txt' in the apps directory returns: patching file app_meetme.c patch: ****
2010 Jan 11
1
MeetMe Conferencing - Announce your own join/leave to yourself and other conference members
Hi all, I'm trying to get the MeetMe system to take a caller and announce to them they've joined the conference in addition to the other members of the conference assuming previous members of the conference >= 1. I can see where the meetme.c app actually processes it using the ast_pthread_create_background(&conf->announcethread, NULL, announce_thread, conf); function. The
2008 Dec 11
1
Meetme realtime table structure
Hi guys, Sorry if I'll be very very stupid but really I write to this conference first. I have problems with configuration of app_meetme in realtime environment. I use last stable release of asterisk 1.6.0.3 Now situation is following. I create database and table in it. Th table is CREATE TABLE IF NOT EXISTS `booking` ( `bookId` int(10) unsigned NOT NULL auto_increment, `clientId` int(10)
2009 Mar 18
1
Video phone crashing meetme on asterisk 1.4.
Hello, I am running asterisk 1.4. For argument's sake I have 4 telephones. 2 support video, 2 do not. Calls between phones work fine and codecs are properly negociated. I have videosupport=yes in sip.conf and when the two video phones communicate I have video. I call meet me with this command EXEC MEETME 1234|d SIP looks like this : -- AGI Script Executing Application: (MeetMe)
2005 Feb 09
2
Problem with meetMe
I try to use meetme app after reading manual i compile and install zaptel with ztdummy when i make lsmod i have ztdummy 2532 0 (unused) wcusb 20064 0 (unused) zaptel 179168 4 [ztdummy wcusb] usb-uhci 26348 0 [ztdummy] usbcore 51616 0 [wcusb usb-uhci] after it i recompile asterisk and after it i have
2004 Nov 27
1
Meetme Help !!!!
Hello , I am new to Asterisk. Trying to use Meetme for Audio Conferencing. Got Zaptel card etc. and i could see app_meetme.so nicely loaded. Now : 1. how to start a conference ? 2. how to add a user ? 3. How can a user join a conference ? After looking at certain links on Net I tried to
2006 Apr 20
1
MeetMe: lots of buffer overruns/underruns when connecting over IAX
Hello, Situation: I've got two asterisk 1.2.4 servers, connected to each other over the internet with IAX2 with about 20msec delay. One of the servers is hosting MeetMe. It's working fine as long as only SIP phones connected to the meetme server participate in the conference. As soon as a participant using IAX2 is connecting, lots and lots of buffer overruns and underruns are
2005 Jan 27
1
Moh in meetme doesn't work if I transfer to meetme
Hi, if I dial meetme from extension 200 directly it works ok - I get moh as only user (first trace). If I dial to other local extension and trasfer from there I get second trace... Apparent difference between those two is warning : Jan 27 11:06:33 WARNING[6133]: res_musiconhold.c:466 moh_alloc: No class: random What this could mean ? Direct Call log-----------------------------------------:
2010 May 22
1
Manual Web-meetme
Hi anyone, I need to install an application to organize conference in Asterisk, and to this I wanna use webmeetme, but I don't get find a good manual, anyone have or know where I can find a good manual to this application? Thank you very much. Renato -------------- next part -------------- An HTML attachment was scrubbed... URL:
2006 Mar 02
1
IAX Video and Meetme
Hi I'm browsing around the internet looking for signs that the IAX client library and app_meetme support video. I stumbled across this post by SteveK on the 27th of Feb 2006. "My company is looking to hire a full-time developer, who will be working about 25-50% of the time on iaxclient; in particular to finally integrate, build, polish and enhance video in iaxclient, add video
2007 Sep 16
0
Problem with asterisk 1.4.11 and playing files to meetme conference
I am using asterisk Version: 1:1.4.11~dfsg-1 as found in Debian. I'm using a call file to connect a meetme conference to an AGI script which plays files using the stream_file method. I have four files which should play in sequence, though only the first two files actually play. I get these errors in the CLI: [Sep 16 06:20:43] NOTICE[18424]: app_meetme.c:1911 conf_run: Audio bytes: 276 Buffer
2006 Mar 07
1
MeetMe 'i' option not working correctly?
I'm running 2.4.5 and app_meetme never plays conf-hasleft or conf-hasjoined with user names. I looked at app_meetme.c, but couldn't determine the cause. Any suggestions are greatly appreciated. exten => 600,1,MeetMe(600|i) I get the following: -- Executing MeetMe("SIP/jon-21f8", "600|aciMps") in new stack == Parsing '/etc/asterisk/meetme.conf': Found
2006 Oct 13
1
Asterisk (meetme) and SMP/HT OK?
In the past, there have been reports of problems with Asterisk with multiple processors and/or HyperThreading. I'm having a !@#$ of a problem with an HPDL380 with 2 3.4gHz Xeon processors, 2 gb RAM -- if I got 24 hours I'd think I had died and gone to heaven :) Am I missing something obvious like "Asterisk is single CPU, single core?" I can't access the ILO so I
2008 Feb 05
0
meetme with ztxen - WARNING[10725]: app_meetme.c:772 build_conf: Unable to open pseudo device
Hi, I have asterisk installed in the xen virtual server. I installed zaptel 1.4.2.1 and patched it to have ztxen module. I loaded ztxen module but when I try to invoke or call to my meetme application I get the following warning and negative result of connecting to conference: [Feb 5 17:46:13] WARNING[10725]: app_meetme.c:772 build_conf: Unable to open pseudo device [Feb 5 17:46:13] --
2006 Apr 05
1
WebMeetme Problem Please help!!!
I am running Feodra, I have downloaded the WebMeetMe Program, untar it to /var/www/html/WebMeetMe. I can access teh web page as of now. I cannot for the life of me figure out where defines.conf is. The install tells me it is in /var/www/html/WebMeetMe/lib/ however a complete search of the computer cannot find it anywhere. The /lib/ subdirectory does not exist in the untar'ed folder either. I
2007 Apr 18
2
MeetMe Error
Hi! i have an error using the meetme aplication, and just dont work.. my meetme.conf is: [rooms] conf = 700 i calling from a sip phone, the extension number is 600. there is the error: Executing [700@numberplan-custom-1:1] MeetMe("SIP/600-09111e58", "700|MI") in new stack WARNING[20055]: channel.c:3024 ast_request: No channel type registered for 'zap' WARNING[20055]:
2004 Jun 16
1
error loading meetme module
Hi all, I'm configuring meetme in asterisk but no luck. I dont have a digium card, so based on the wiki, i have to install the zaptel driver and enable the ztdummy on it. I compiled, install the module and enable it on the module.conf. When i tried to start the asterisk, asterisk wont start because of the error in loading the meetme module. Jun 16 15:36:56 WARNING[-1084989312]:
2007 Apr 24
0
3 way calls and meetme problem
Hello, I have a problem with the meetme application, but I'm not sure if it's a bug or just a misuse. I'm trying to get a 3 way call system working as follow : A calls C B calls C C who's speaking with A or B, presses one keypad (only one) and the 2 incoming SIP (A, B) and C are redirected into a conference room. Therefore, I created an entry in the applicationmap
2004 Jul 01
2
Grandstream HT286 1.0.4.63 & Meetme
Good day! Have a weird problem with HT-286 and Conference room. I use Asterisk CVS-HEAD-06/04/04. Here it is: When HT-286 get into the conference room first and nobody in that room everything seems ok (with any codec HT286 allowed), but when HT-286 get into conference room when somebody already there, have got different HT behavior: 1. When HT use GSM codec => it connects to conference room,
2007 Jul 30
1
MeetMe through DeadAGI has changed to return -1 on Hangup
I have a "support call" AGI script that has been working flawlessly for a couple of years now. It dumps the customer into a MeetMe conference room, then dials a bunch of support engineers, and connects anyone who accepts the call into the conference room. The conference room is recorded. After the support call is over, the recording is emailed to a list for quality control, etc. It