similar to: Digium Echo cancellation.

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, 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

"Security only fixes: 2011-04-21" So in six days, no more bugfix patches will committed into 1.4-branch :( Is a prolongation possible? Because 1.4 is so reliable now. It would be a great loss. And no, 1.8 is not (yet) a replacement. Kristijan

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 |

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

Hello, short question: is there a possibility to use asterisk as an outbound proxy? iam open for any suggestions, use asterisk trunk, dirty patches, ugly workarounds, everything. What is want to build is: SIP Phone -> via TLS/SRTP -> Asterisk as outbound proxy -> via UDP/RTP -> VoIP-Provider So Asterisk should just forward any incoming SIP messages (INVITE, REGISTER) to the

Hi, I would appreciate some advice on the following: how does one use a single BRI port HFC chipset based ISDN cards with the latest DAHDI, libpri and Asterisk 1.8? For Asterisk 1.4 I would first install mISDN & mISDNuser and then build Asterisk 1.4. Now I noticed that Digium's HFC chipset based B410P is supported by the latest DAHDI & libpri but reading the source of the

Hi, Maybe I missed it while checking it, but which spandsp version is recommended to play with Asterisk 10 and T.38/T.30 gatewaying ? I can see both spandsp-0.0.6pre17.tgz and spandsp-0.0.6pre18.tgz here (http://www.soft-switch.org/downloads/spandsp/) but I couldn't find a changelog documenting differences between them. So I prefer to double check ask for recommendations. Regards

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

Hello, what about: "This feature means you can restart Asterisk after a failure (or asterisk restart itself with safe_asterisk), and keep existing calls up with only a few seconds of audio dropped." That would be a feature! there is a "other" pbx that has this feature... Anyone else would like to see that feature? just want to start some brainstorming, Kristijan

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

I am about to setup a new machine and based on a thread in the freetel-oslec list, I came across the idea of compiling Intel optimizations in when using oslec w/ dahdi. So I edit dahdi-linux-complete-2.1.0.4+2.1.0.2/linux/drivers/dahdi/dahdi_config.h to #define CONFIG_DAHDI_MMX which on its own wont compile. It was suggested to edit dahdi-linux-complete-2.1.0.4+2.1.0.2/linux/drivers/dahdi/Kbuild

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

Hi list, I install dahdi-linux successfully with the module of oslec for the echo, but when I specify it in the system.conf the echo canceller oslec it shows me errors: DAHDI_ATTACH_ECHOCAN failed on channel 4: Invalid argument (22) I see that the echo cancellers is supported: mg2, kb1, sec2, and sec because oslec is not supported?, but he has support to compile it with dahdi_linux! best

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 all, Now i want to check a number (channel) online, offline or unreachable on asterisk but i don`t know to do. Can anyone help me to solve this issue. Thanks and best regard! -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110109/c193b48d/attachment.html>

Hi All, i would like to know if anyone has done or having idea regarding PRI terminations in asterisk. i have a requirement where i need to support 80 PRI in one machine i have found a machine which have 10 PCI slots available now i am thinking of arranging 8port sangoma card in this pci slots so i can arrenge 10 card in that. is it possible to run system like that ? is it good idea , can

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>