similar to: Need help on Text entry applicaon

What kinds of problems were you having? I'm on 1.4.0 and chan_gtalk.so simply doesn't load. Of the 146 files in the /usr/lib/asterisk/modules/ directory, asterisk loads 144 of them, omitting only chan_gtalk.so and res_jabber.so. Connected to Asterisk 1.4.1 currently running on monkey (pid = 9371) Verbosity is at least 3 foo*CLI> module load chan_gtalk.so [Mar 7 10:23:07]

hi, i was able to get this working with google talk. i entered myusername@gmail.com using the gtalk2voip.com website's "invite" box, and as a result, saw a request from service@gtalk2voip.com to be added as a buddy in my google talk contact list. i accepted the request. in my asterisk dialplan, i have this entry... exten => 3501, 1,

Hi, We have some 100 Servers (includes both Intel and PPC based systems). Is there any way to easly install OS (different flavours of Linux) on these systems ? We do not have control over the DHCP Server settings so is there any way to do this with out touching the DHCP Server ? We have static IPs for all the systems. Regards, Naveen

Hi, I am getting the following error.... after installing SPANDSP along with app_dtmftotext.c file. and while making Asterisk again. Error follows:: ******************************************************************* [root@ast asterisk-1.4.1]# make Generating input for menuselect ... menuselect/menuselect --check-deps menuselect.makeopts make[1]: Nothing to be done for `all'. make[1]:

Hi, I was looking for a way to pass alphanumeric variables to asterisk via the keypad, found this application app_dtmftotext.c , And I already tried with 'spandsp' application for this. But I am getting errors.[I followed the instructions at http://www.soft-switch.org/installing-spandsp.html] specifically by running this command:patch <apps_makefile.patch I need clarification on

Hi, I was looking for a way to pass alphanumeric variables to asterisk via the keypad, found this application app_dtmftotext.c , And I already tried with 'spandsp' application for this. But I am getting errors.[I followed the instructions at http://www.soft-switch.org/installing-spandsp.html] specifically by running this command:patch <apps_makefile.patch I need clarification on

Hi, I am using Samba 3.0.7. on a Red Hat 9 system. I tried to configure the machine as a PDC by using a configuration file quite similar to what is giving in the chapter 4.Windows NT domains of 'Using Samba' 2nd edition by Orielly. Using testparm shows that the machine has been identified as a PDC of the domain 'hamsateam.' To test the PDC, I tried to make a WinXP

I'm tring to compile fax applications on Debian system. the spandsp library compiles ok, and when i try to patch the make file in apps directory as is said in the instructions it returns errors. I'm using cvs version of asterisk . -------------------------- voipgw:/usr/src/asterisk/apps# patch < Makefile.patch patching file Makefile Hunk #1 FAILED at 35. Hunk #2 FAILED at 68. 2 out of

I'm trying to install the SPANDSP software into Asterisk to support incoming (mainly) Fax. I'm following the info in http://www.voip-info.org/wiki-Asterisk+Fax. I downloaded and installed the spandsp software from ftp://ftp.opencall.org/pub/spandsp/ and followed the directions in several documents listed on the on the Tiki page. I get down to patch < Makefile.patch that fails with

Hi All, Sorry if this is too much silly. Vorbis encoder gives out packets of information. 1. Is it that each packet contain information of one frame/window either 256 / 2048 or any length set by encoder? If yes then the encoded packet will be variable in length. 2. Every packet is divided into segments, how does the encoder decide on number of segments in each packet? It must be dependent upon

Hi, If this is true: "Granule Position Information in Ogg Header is a hint for the decoder and gives some timing and position information." So say if granule position is 10000, it means that 10000 PCM samples are encoded in this page approximately. If this is true we can neglect this information, it will not effect the decoding right(but might effect for streaming)? Ravi

hi everyone, here i have a question: is there any chance in llvm intermediate representation to compare whether a operand (ie a value) is used as a user without iterating over all the instructions in a basic block. i will explain this with an example: i=p+4; j=i+p; in the above example i mean p,4,i,p which are on the RHS are operands and the terms i,j which are on the LHS are users. so what i

hello sir, i want to control the bandwidth when sending traffic between two logicial address (ip address ) on the same interface (eth0) on the same machine. can i do it using tc tool. i am sending mail using sendmail between 2 users on same machine and sniffing packets at receiving side. but i want to control bandwidth of this traffic so that i do not lose packets at capture.

Hi, I am running the dbt2 test kit for postgres-8.0.1.The driver and client are running on Sun Solaris 10 and the database is running on Red Hat 9.I am running the test kit for 100 warehouses with tpw(terminal per warehouse) value of 80. I did truss on driver and found that after executing some transactions an error comes related to EPROTONOSUP. The particular point where error comes looks like

Hi All, Is it possible to Call an external number For X minutes if first attempt resulted In Busy or No Answer? i.e, for example, if I called 335 8698, and it was busy, or may be said it was busy, Asterisk will try calling back that number for 15 minutes, once every 5 minutes. Thanks Jibu ________________________________________________________________________ Yahoo! India Matrimony: Find your

Hi, I need an immediate help from anyone of you. Does asterisk support call waiting in SIP phones? Thanks for advance Jibu ________________________________________________________________________ Yahoo! India Matrimony: Find your life partner online Go to: http://yahoo.shaadi.com/india-matrimony

hello list, i have problem when i am useing wrapuptime with agents.conf my agents.conf looks like this [agents] autologoff=15 musiconhold => default wrapuptime=50000 group=1 agent => 1001,4321,Mark Spencer recordagentcalls=yes my aim is every call needs have wrapuptime of 5000 ms but when ever a call comes its directly connecting not wating any more. your views will be highly

I am using samba-3.0.7 on kernel version 2.6.5-3 (client as well as server). I traced the exact function calling sequence when a "mkdir" command is given at the client side. I got stuck at the server side, source/smbd/vfs.c - vfs_MkDir() - SMB_VFS_MKDIR(conn,name,mode) (line no 357) Could anyone please tell me exactly where is the vfs_ops struct initialised, and which function will

hello list , my set up is like this ip device -->ser ---> asterisk(astcc) --> pstn gatewsy my asterisk version is 1.0.2 iam using the ser as registration and asterisk aa the prepaid one with the help of the astcc. now my problem is the destination people i.e the pstn line s are listening low voice and also the blurr sound quality along with the audio of the ip device at

We have made some changes in samba-3.0.7, as part of our graduation project. Are there any test suites which we could use to check the conformance of the changes, with the original samba specifications? Also, with reference to your statement in document tridge_cifs04_tutorial.pdf , saying that inputs about test documents would be welcome, if the above mentioned suites are not available,then we