similar to: outbound caller ID

I setup two asterisk servers with identical settings (same extensions, same queues, etc). Each one is connected to the same amount of incoming/outgoing links (1 PRI, 4 BRI, 1 IAX friend, etc, on each box). Most extensions are sip and they register via DNS SRV and other methods so that the two servers are load balanced. Incoming PSTN calls (BRI) reach 50% each server so that's load balanced

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.

Hi, Is Asterisk "fully QSIG-compliant"? I currently have an Alcatel 4400 connected to Asterisk 1.2 and 1.4. Zaptel versions are 1.2.26 and 1.4.11. I am using switchtype=euroisdn and all works fine. However, it seems that Alcatel's latest firmware has dropped support for euroisdn which is really despicable. So now I need to see if I can migrate to QSIG which is supported by

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 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

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

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

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?

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

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

Hi, My initial setup for receiving faxes worked as follows: fax call arrives on ISDN BRI connected to a BOSCH PBX, signal sent to ALCATEL PBX via PRI QSIG then finally sent to ASTERISK via PRI EUROISDN. The Asterisk server then forwarded the call to a iaxmodem and HylaFax received the data. All worked fine. Now I got rid of both BOSCH and ALCATEL in the "fax path" and it's as

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, I'm trying to set a rule to dial out through multiple Zap groups so that, say, g0 is the cheaper POTS lines group and must be used first. However, if g0 is busy or disconnected then try dialing out g1. My g0 group is made up of 4 analog lines connected to a 4-FXO card. I disconnected the RJ-11 wires from the FXO card to simulate a line disconnection. So theoretically all calls should

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.

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

You're quite right. We'll need to see your misdn.conf file to check the settings. -->> -----Original Message----- -->> From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users- -->> bounces at lists.digium.com] On Behalf Of Vieri -->> Sent: 06 February 2009 13:49 -->> To: asterisk-users at lists.digium.com -->> Subject:

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]]

Use Set(CALLERID(num)=9999999999) instead of using CALLERID(all). Remember to set this BEFORE you Dial. -->> -----Original Message----- -->> From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users- -->> bounces at lists.digium.com] On Behalf Of Vieri -->> Sent: 06 February 2009 12:36 -->> To: asterisk-users at lists.digium.com -->>