similar to: Astersik can not detect DTMF key

Hi all, When a user make a call to Asterisk, and when user hang up the call at any point of the conversation,? Asterisk will stop Diaplan intermediately. At this situation,? Are there any way to make? Asterisk continue execute the Diaplan ?, so Asterisk can do something like that delete temporary file, .. etc. Thanks in advance, Giang -------------- next part -------------- An HTML

Hi List, I'm finding a solution to provide failover and load balancing features to my IVR system. Anyone suggest me what is the best solution please?. what the hardware I should use ?. I heard about RedFone, but someone on the mail list said that it is not good because TDMoE module in asterisk is not so stable and TDMoE is stale. And It seems that RedFone doesn't not support load

Hi List, How can I define an array of sip number in sip.conf ? I want to define an array of sip number from 1000 to 2000, so I can make a performance test on Asterisk using sipp. Thanks in Advance, Giangnh -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100318/858dcd1d/attachment.htm

Dear List, Are there any way of configuring of Asterisk so it'll cache sound files in memory, and when Asterisk receive a call, instead of loading sound files from the disk, it will load from the memory and so Asterisk can process much more call at a time than with faster speed it is not caching. Thanks, -------------- next part -------------- An HTML attachment was

Hi all, I'm being requested to deploy an?IVR service?using SS7. I've deployed Asterisk before using ISDN connection, but never with SS7. Can anyone explain me the different between using ISDN and SS7 ? What need I do now to change to use SS7 ?. Many thanks, Giang -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, For my work, I want to use LLVM to compile SPEC 2k for Alpha. Since Alpha support has been dropped, I tried using version 2.8, but it is quite buggy, probably because the Alpha backend has not been maintained. I was wondering if there is an earlier version where the Alpha backend is stable enough to compile SPEC 2k? For my purpose, I do not need the most advanced optimizations, so an

Hi- I am having a weird problem with setting dom0 kernel output to the serial console with Xen 3.0.4-1, below is my grub setting. With this setting if I don''t enter the grub menu and have the default boot to the first image, everything works fine and I can get output/input to the serial console. The problem is when I enter the grub menu and select the image to boot from, if I

Hi Giang, Given that the community deprecated the Alpha backend, I'm doubtful anyone would be able to point you in the right direction. Have you iteratively tried the difference versions of LLVM (i.e., 2.9, 2.8, 2.7 on down the line)? Chad On Feb 3, 2012, at 12:34 PM, Giang Hoang <ghoang84 at gmail.com> wrote: > Hi, > > For my work, I want to use LLVM to compile SPEC 2k

HI; Thanks for your reply. The reason for why I am going through asterisk in such case is just "using asterisk voicemail service" I mean: ATA1 calls ATA2, suppose ATA2 is unreachable or he is not at the office, then the call reroute (my GK is able to reroute calls if the first route is not valid) to atersik for voicemail service. Do you think I can handle it with asterisk native

Slash-dot is pointing to this article on Asterisk and Pingtel. http://www.theregister.co.uk/2005/05/22/pingtel_voip/ Paul Paul Mahler www.signate.com

Hello, I am writing a program based on Astersik Manager which needs to put calls on hold and to redirect them to others extensions. I haven't funded any action able to do this. Is there a way to place calls on hold using Asterisk Manager Actions? Amaury -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello All: Does the Asterisk support to insert an off the board transcoder for a call? Thanks, Charles ____________________________________________________________________________________ Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping -------------- next part -------------- An

Hi to all!! I'm successful to connect Asterisk to PostgreSQL database... If it's possible, can anyone learn me how to store sip user in PostgreSQL database and how to configure voicemail?? Thanks for all!!!

Dear Sirs, I had compiled PWlib and OpenH323 correctly in my Fedora Core 2. But when I try to compile asterisk-oh323 I get the following error: `__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__' How can I solve it? Thank you for your help. Juanjo

Hi, I am new to the Asterisk world. I don't know much about the architecture, but I am involved in installing and configuring the VoIP system. My requirement is to build a VoIP system using the 4 input lines (ISDN up0 telephone lines), it must be possible to receive calls from outside through the 4 ISDN up0 input lines, and also possible for outgoing calls, conferencing .etc. I

We are planning to install a voip system based on asterisk for 2000-3000 retail locations and up to 6000-8000 sip accounts/users. Instead of setting up a new, centralized PSTN gateway, we are intend to use a CISCO gateway/router of an existing CISCO voip solution in the headquarter and we must able to call all CISCO based voip phones in the headquarter running together with a CCM. SIP-Phones

Hi everyone, I want to call from one Asterisk to another Asterisk via SIP, but i dn't know how. I have found out something in these links: http://www.voip-info.org/wiki/index.php?page=Asterisk+config+sip.conf http://www.voip-info.org/tiki-index.php?page=Asterisk%20SIP%20Channels but I don't understand them very well. At first, I tried simply doing this: In SIP Client:

Thanks Andrew. I would like to clarify what I tried to do. I want to use llvm-gcc on x86 linux to compile C programs into Alpha binary. Giang On Fri, Oct 15, 2010 at 5:41 PM, Andrew Lenharth <andrewl at lenharth.org>wrote: > llvm-gcc doesn't not compile *on* alpha (128bit fp and int issues). I > haven't tried it as a cross compiler. > > Andrew > > On Fri, Oct

Hi, I wonder if anyone has been able to successfully build llvm-gcc as an Alpha cross compiler? I have tried many different combinations of flags and gcc compiler, but have not been able to build successfully. Currently, this is the command that I used to build on Ubuntu 10.10: ../llvm-gcc-4.2-2.8.source/configure --enable-llvm=/home/ghoang/research/llvm/llvm-objects --enable-languages=c,c++

In my Asterisk server following things have been done to detect answering machines before the answered call connects to the agents in queue. In extension_additional.conf ============================== [ext-queues] include => ext-queues-custom exten => 5000,20,Macro(user-callerid,) ; changed the priority to 20 ............... ============================== In extension_custom.conf