similar to: PBX_DUNDI question

Hi, Reviewing the archives I saw /2004-October/070314.html from Tim Lewis. His error is almost identical to mine i.e. when "make clean; make install" in asterisk sub dir, I get the following: pbx_dundi.c:54:18: zlib.h: No such file or directory pbx_dundi.c: In function `update_key': pbx_dundi.c:1315: warning: implicit declaration of function `crc32' pbx_dundi.c: In function

Anyone seen this...? Is it a known issue? I'd file a bug, but we're on 1.2.9.13, and every time I file a bug and it isn't against the latest code I get given crap for it. Given that most of the time you don't know HOW to reproduce a problem on the latest code anyway, not accepting bugs from older versions does the community no service, because potential bugs are never accepted for

Hi All, Found an issue with DUNDILOOKUP function in Asterisk 1.6.0.5. I was using DUNDIQUERY (Set(ID=${DUNDIQUERY(${MNUM},priv,b)})) for dundilookup and it was working fine. But when I tried to use DUNDILOOKUP function (Set(DL=${DUNDILOOKUP(${MNUM},priv,b)})), it didn't retuen me a result. Moreover, the cli command 'dundi lookup 12345 at priv' returned me the result at the same time!

On Tue, Sep 16, 2008 at 2:21 AM, Michiel van Baak <michiel at vanbaak.info> wrote: > On 22:46, Mon 15 Sep 08, hugolivude wrote: >> I have two Asterisk servers running on the same LAN. One starts fine, >> but when I start the other I get: >> >> pbx_dundi.c:4582 load_module: Unable to bind to 0.0.0.0 port 4520: >> Address already in use >> >>

Parse the TIMEOUT keyword from the config file and pass it to gfxboot core. Signed-off-by: Sebastian Herbszt <herbszt at gmx.de> Index: syslinux-3.82-440-g6483c80/modules/gfxboot.asm =================================================================== --- syslinux-3.82-440-g6483c80.orig/modules/gfxboot.asm +++ syslinux-3.82-440-g6483c80/modules/gfxboot.asm @@ -657,8 +657,7 @@ gfx_input:

Hi, Is it possible to record say 30 seconds of audio and then have LumenVox convert to text ? or any available tool open source for speech to text . Regards Dhaval -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100914/b56c3d9c/attachment.htm

Hello Friends, I am trying to Installl dahdi on amazon EC2 which have Open-SUSE-11.1 X86 version. and here is snap of uname- a command *Linux ip-10-160-86-41 2.6.32.19-0.3-ec2 #1 SMP 2010-09-17 20:28:21 +0200 x86_64 x86_64 x86_64 GNU/Linux* when I try to run DAHDI distribution dahdi-linux-2.1.0.4 I am getting following error *echo "You do not appear to have the sources for the

Hi all, I have n no. of extensions in my dialer. from 456 to 556 extensions. I was created 2 other extensions 667 and 668 I need to allow only STD calls to go from this extensions. These all extensions are same context . I need to define the STD dialplan for only this 2 extensions. how I can ? Best Regards, Mahesh Katta *BUZZ**WORKS* Business Services Private Limited BANGALORE | CHENNAI |

Hi all, The register allocator that I implemented is failing in the LLVM cvs version, but not in LLVM 1.1. The generated code fails a check in the x86 peephole optimizer: llc: PeepholeOptimizer.cpp:128: bool <unnamed>::PH::PeepholeOptimize(llvm::Machi neBasicBlock&, llvm::ilist_iterator<llvm::MachineInstr>&): Assertion `MI->getNum Operands() == 2 && "These

Hello All, can anybody tell me how can i integrate asterisk and skype users so that skype users can dial my asterisk number or dial internal dialplan form skype regars Dhaval -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090708/cccd4587/attachment.htm

Dear All, How can we know the On board supports echo cancellation I have *Digium, Inc. Wildcard TE410P quad-span T1/E1/J1 card 3.3V (rev 02)*board all working fine but sometimes i got echo when user are calling a PRI. is there any way to know on board echo cancellation . regards Dhaval -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi All, This posting regarding PRI voice optimization, on dahdi 2.1.0.4. we have more than 4 machine running on 4 port PRI card with echo cancellation hardware based. i have enabled echo cancel from chan_dahdi.conf using echocancel=yes, now more than 70% of call get good voice but some of calls having issue for callquality and other voice related issues. now my question is that is there any

Hi, I am newbie started learning ruby on rails i had an application which used to run pretty well latter i upgraded from webbrick to mongrel 1.1.5 then I am getting the foloowing errors and a default application runs on port 3000 i have attached its screen shot as an attachement and error which i get in terminal below => Booting Mongrel (use ''script/server webrick'' to force

hi all, can anybody help me to give Opensource SIP client information which can be modified as per our requirment regards Dhaval -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090518/802cc3ac/attachment.htm

Thanks a lot Tony and Dhaval for your much appreciable suggestions. Regards, Rajib Rajib Deka SIEMENS Ltd. Robert V Chandran Tower, First Floor, West Wing, #149, Velechery Tambaram Main Road, Pallikaranai, Chennai-100, INDIA. www.siemens.com Mob: +91-9176780669 | E-Mail: rajib.deka at siemens.com Date: Wed, 20 Apr 2011 13:55:25 +0530 From: DHAVAL INDRODIYA <dhaval.it01034 at gmail.com>

hi all, any one know, about echo cancellation with digium card, is it actually needed or it okay if we dont purchase because it increase price which half of new card, regards Dhaval -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090827/8d6c680a/attachment.htm

Hi All, I want packets [request/response] capture for ISUP packets , i have E1 line terminated to my digium card i just want a packets flow between my machine and teleco side, is any tool or utility [command] availabele for observation this packets and data. any help appericiated Thanks Dhaval -------------- next part -------------- An HTML attachment was scrubbed... URL:

hello All I am using festival as an application but it default voice is not good to hear anybody have solution about better voice in Festival regards Dhaval -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090618/b1cca678/attachment.htm

Dear All, Last Week i tried and goggling more on how to call RESTful webservice from Dialplan? i found *CURL* function but while i tried to use it ,it 's not supported HTTPS request and we cannot set headers while send a request. also without HTTPS . i get result it will return a string means whole xml,json request is represented in string format, how can i parse that request? my

Hi, I am newbie started learning ruby on rails i had an application which used to run pretty well latter i upgraded from webbrick to mongrel 1.1.5 then I am getting the foloowing errors and a default application runs on port 3000 by name iphone configuration utility i am unable to attach the screen shot as it is saying file to large. Error which i get in terminal is listed below => Booting