similar to: extracting text contained in brackets ("[ ... ]") from a character string?

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

Hi I have dataframe which contain 5 columns and 1000 records. I want standard each cell. I want range each column between 0 and 1 . I think i must use loop? could you help me? --------------------------------- Moody friends. Drama queens. Your life? Nope! - their life, your story. [[alternative HTML version deleted]]

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, 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 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

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

I'm currently implementing A rtp header info and using speex for audio. can someone tell me what the rtp payload type value is for speex. i've checked the submitted speex draft and it says its beyond its scope and i can't find anything in RFC3550 that might suggest what it is. any help would be appreciated thank you, Greg --------------------------------- Boardwalk for $500?

Hi List; What the following mean: CONSOLE=Phone/phone0 CONSOLE=Console/dsp TRUNK=Zap/g2 I know SIP/John and Zap/1 but I do not know above (I do not know also the difference between Zap/2 and Zap/g2)? Any advise? Regards Bilal ____________________________________________________________________________________ Got a little couch potato? Check out fun summer activities for kids.

Dev-Cpp is a IDE that can be used with MinGW and GCC but its causing problems. I installed MinGW since I am used to ./configure type builds but im unsure if this is a tinc issue or OpenSSL issue. I installed openssl 0.9.8e and I get to the following, $./configure ... checking openssl/evp.h usability... yes ... checking for openssl/engine.h... yes checking for SHA1_version in -lcrypto... no

Hello Asteriskers, I'm confused about why Asterisk is not a SIP proxy and why exactly this can affect the performance of a large Asterisk system. I know that Asterisk acts as a useragent endpoint, but my doubt is why exactly Asterisk could overload the call flow if the RTP voice stream goes from the caller to the called party. Does someone know how many calls or pencentaje that could handle

Hi List; What is the difference between WaitExten function and TIMEOUT (response)? As I see that both are used to determine the allowed time to enter the digits, any one can advise? Regards Bilal ____________________________________________________________________________________ Shape Yahoo! in your own image. Join our Network Research Panel today!

Been running rock solid servers on Centos 3 and 4 for years but equipment is starting to die so it is time. I was thinking along the lines of an Intel Q6600 on an Asus P5K-VM mobo with the new G33 chipset utilizing SATA. Is anyone having success with this board or any of the G33 chipsets where I am not going to run into all kinds of install problems? I see these other threads of install

Hello everybody, Just was wondering if somebody can help for G711 fax passthrough w/ asterisk. The issue I have is regarding codec upspeed when the call is already connected using G729 for example. The setup is fax---ATA---asterisk---Cisco---fax When codec upspeed should happen, ATA or Cisco will send a G711 reINVITE causing the codec to be switched over, but asterisk does NOT

On Sun, Apr 8, 2018 at 1:06 PM Nicolas Kovacs <info at microlinux.fr> wrote: > Le 08/04/2018 ? 13:54, Nicolas Kovacs a ?crit : > > As far as I can tell, there would be several solutions to this problem. > > > > 1. Ask the EPEL maintainers to keep the application up-to-date. > > > > 2. Patch the darn thing so I don't get the annoying popup. > > >

Hi... With samba4 I sometimes feel like a bloody beginner even I use samba since ages. Miracle of the day: Users connecting using samba4 cannot write to group writeable folders even they should be able to. I upgraded one 3.6 fileserver to 4.2.3. PDC is always a samba 4.2.3 instance. I am using samba4 in classic PDC mode for a couple of reasons. On my fileserver I have a folder called