similar to: standardization

I have a text string that contains text within two brackets. e.g. "testdata[3]" "testdata[-4]", "testdata[-4g]", I wish to "extract" the string enclosed in brackets? What is a good way to do this? e.g. fun(testdata[3]) = '3' fun(testdata[-4g]) = '-4g' --------------------------------- Moody friends. Drama queens. Your life?

Hi, I would like to know if one can set the outgoing caller ID within Asterisk when calls are going out through: 1) an analog POTS line (I suppose not) 2) a telco BRI line (I don't think so) 3) a telco PRI line (maybe) 4) a voip provider (surely) Thanks, Vieri ____________________________________________________________________________________ Moody friends. Drama queens. Your

Dear all I have confusion how to asterisk genrate tone and what ringing code use default 180 or 183 i have setup asterisk with mediant 2000 with avaya [asterisk]-----[mediant 2000]--------[Avaya] when i call from avaya side to ---> asterisk i don't got ringback Sound so how to asterisk genrate tone for calling party is there any soution and what is the problem of

Dear all I have setup of asterisk 1.4.11 Now i want soft phone which one support file sharring + video + voice call with asterisk SIP is there any soft phone which support this all feature ?? with asterisk Regards Satish Patel --------------------------------- Moody friends. Drama queens. Your life? Nope! - their life, your story. Play Sims Stories at Yahoo! Games.

Hello, I have a question about error handling. I run simulation studies and often the program stops with an error, for example during maximum likelihood. I would like the program not to stop but to continue and I would like to ask how the error handling can be set up for this (if it can). I tried to look through manuals etc but unfortunately did not get closer to the solution. Below is a

hmm...forgive my ignorance here. icould have explained it wrong. the rtp header has the pt (payload) field as a 7 bit value. i was under the impression speex had a particular value i should set it to. is this so? if no what value should i assign it, whether by convention or otherwise? Note that i'm implementing a simple rtp header and combining it with the speex payload i'm not using

Hi List; How can I configure asterisk to receive a call from SIP end point without being registered at asterisk and its IP address is dynamic, and authentication to be based on the username and password or any other string? I know that if I place the host with static IP then no need to register, but what if the voip gateway was having dynamic IP and I do not need to register on asterisk, but I

Hi, We have a simple bridge setup but the ping (and other network traffic) does not work reliably. After tracing the code, it looks like a software bug. Since bridge software is been running by thousands of people. I guess I am wrong. Anyway, here is the problem. There are 2 boxes and each one has 2 interfaces, 1 ethernet and 1 wifi. STP is enabled for the bridge to avoid the loop. So the box 1

Hi, I was wondering if this is normal or if it something i need to wonder about. I get Monit alerts that my mongrels PID files have changed. Changed Service MONGREL_2 Date: Tue, 09 Oct 2007 14:19:38 +0000 Action: alert Host: sa.greenling.com Description: ''MONGREL_2'' process PID changed to 18368 Your faithful employee, monit

hello, Does our beloved Centos project have a page where one can request that an RPM package be built, especially for C5? How about alerts when those requests have been satisfied? I know about 3rd party repos like rpmforge, epel, and kbextras. I can search those in YUM. Centos seems to have specialized as a server OS in the past, but my experience is that it is an up and coming

Hi List; Why I need zlib1g to do installation for Zaptel? Will zlib1g do compression or it will what extactly do during the installation process? Regards Bilal ____________________________________________________________________________________ Moody friends. Drama queens. Your life? Nope! - their life, your story. Play Sims Stories at Yahoo! Games. http://sims.yahoo.com/

Hi, I have an 8-port Grandstream GXW-4008 V1.2A ATA converter with analog phones connected to it. They work fine except for just one "feature" I would like to modify. Somehow, each time the ATA re-registers the SIP clients or each time the device has to be rebooted for maintenance, the phones ring once. This feature can be useful as it notifies the user of the re-registration.

Dear all I have '*' 1.4.11 and 2E1 port hardware installed on it now i have single lan not nat anywhere ( 10.20.1.x ) all phone in single network domain without NAT now i have configured canreinvite=no so that asterisk work in meddle path of RTP so what is suggestable option of canreinvite=yes or no. I have SNOM phone at my users end.

Hi: I installed A102d sangoma's card successfully but Asterisk doesn't answer to incoming call from pstn and console doesn't show any message of incoming call in the other word when I diall the number of E1 I can't connect to asterisk and dial the number of extension. I'd apreciateany idea. --------------------------------- Moody friends. Drama queens. Your life?

Hi all, i've been having problems for the past little while with finding a app library for my voip application. then looking at speexclient i noticed there are timestamps and sequence numbers included in the implementation. my question is: should i get a third party RTP lib for my application or simply emulate the speex client in its RTP like properties. wouldn't it be easier and

I am installing Oracle Clusterware 10.2.0.1 on Linux 4.5 ES on OCFS2 and ASM. my ocr and oss files are on ocfs2. I got following error while running it. Please let me know what went wrong. -------------------------------------------------------------------------------------------------------------------- [root@linux2 crs]# ./root.sh WARNING: directory '/u01/app/oracle/product' is not

Hi List; I am using Firefly softphone Version 1.9.9 Build 4521 and I select IAX protocol and did the configuration in Network1 (and I checked the Active checkbox) as following: Server: 192.168.8.4 username: iax2user1 password: password In the Asterisk, I did the following configuration on the /etc/asterisk/iax.conf: [iax2user1] type=friend context=internal username=iax2user1 secret=password

first sorry my english Is posible when reinstalling a server ( changing distro too) , make account machine whith out rejoined all win2k/xp to de domain ?? i create before to login the account machine , but win say can't login account machine does exist I try changin the sid to old , but no work Por si hay gente que hable espa?ol en la lista!!!! Es posible luego de reinstalar un server

Hello, once again I have the great pleasure to announce the release of a new Theora version - please welcome libtheora beta 2! Changes for this version: libtheora 1.0beta2 (2007 October 12) - Fix a crash bug on char-is-unsigned architectures (PowerPC) - Fix a buffer sizing issue that caused rare encoder crashes - Fix a buffer alignment issue - Build fixes for MingW32, MSVC - Improved

Hello, once again I have the great pleasure to announce the release of a new Theora version - please welcome libtheora beta 2! Changes for this version: libtheora 1.0beta2 (2007 October 12) - Fix a crash bug on char-is-unsigned architectures (PowerPC) - Fix a buffer sizing issue that caused rare encoder crashes - Fix a buffer alignment issue - Build fixes for MingW32, MSVC - Improved