similar to: PRI voice optimization

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:

Hi, Can anyone suggest which is the best scripting language for Asterisk or any telecom device? Thanks in advance. -- Thank you with regards, Gopalakrishnan A.N. VoIP call - sip:saigop at gtalk2voip.com -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110401/051f68d3/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, 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

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

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, 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

Anyone here who has configured zaptel/dahdi for a singtel E1 line? What are the settings for coding, framing, line type and switchtype? -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110210/4288ed84/attachment.htm>

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

hello All, i have one issue with Asterisk Meetme Application i am recording through Meetme channels through option *'r'* and format for recording a file is '*wav*' lines used is DAHDI version 2.1.0.4. and asterisk version 1.6.0.5. i have very strange problem of meetme_recording , once conference starts recording file having a *recording is 2x faster *than normal recording .

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>

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 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

Hello All, after so long time i posted a new question regarding billing, hope anyone have some solution. I have situation in that i want to do billing of more than 1 call in real time below are scenario and explanation. Scenario: A customer called my DID number and after that from here i dial few number let say 5 number. once number are placed into DIAL i will put this customer into

Hi, our Asterisk is connected to an E1 port. So we are using the DAHDI-Driver. Please , how do I tell the driver/Asterisk to wait for overlap digits for in-calls? I found the option "overlapdial=yes" but I did not try yet. Is that "my" option? Is there any option for setting an timeout? Thorsten

Dear All, I have Asterisk and Kamailio Configuration. everything works fine, now the situation is like i have following Dial pattern in Dialplan. exten => s,n, Dial(SIP/1002 at glbvoice.com,20,m) now in my /etc/hosts i have following entry 192.168.1.30 glbvoice.com then call get forwarded to kamailio and everything is working fine now question is if i want add one more domain like

Hi, I am trying to setup a GSM-Card for Asterisk. I currently use a "vgsm I" from voismart (http://www.voismart.it/) but the driver is very bad (compile-problems and no echo cancellation). Is there anybody out there who can recommend me another piece of hardware (pci card)? I need 1 or better 4 gsm-ports. Should be stable and have an echo cancelltaion feature. And of course it