similar to: missing CDR records in cdr while kick from meetme

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 List, I have seen from the following link that, for SIP channels there is no audio communication possible in MeetMe with AGI_BACKGROUND. http://www.voip-info.org/wiki/view/Asterisk+cmd+MeetMe Currently we are using asterisk-1.6.2 and the problem still persists. Is there any solution available to overcome this problem? According to our requirement, we have to run an AGI script in MeetMe.

Hello - I am fairly new to Asterisk, but we have a fully operational system with very few hiccups. Much of that is because of this list. Thanks. My question is this - We have assigned MeetMe conference IDs to all of our employees. We then setup a TN to accommodate the MeetMe() app. Everything works fine. In fact, it works great. However, I can't seem to figure out a good way to log

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

hello List, i am in a situation where i cannot get cdr records for call originated from CLI , i am not able to get when i used application or extension. is there any solution regarding this ,i working since last 3 days onto this. regards Dhaval -------------- next part -------------- An HTML attachment was scrubbed... URL:

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 |

Hey Users, I am using record application to record MeetMe conf. but look like its creating individual files for every channel. What applucation is best to record MeetMe conf ? ~ # ls -l /var/spool/asterisk/monitor/ total 489220 -rw-r--r-- 1 asterisk asterisk 44 Feb 16 08:42 8881-conf-20110216-084224.wav -rw-r--r-- 1 asterisk asterisk 1858284 Feb 16 13:05 8881-conf-20110216-130321.wav

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

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

Dear All, i have following CLI error while try to run this command from Dialplan *TrySystem("DAHDI/45-1", "asterisk -rx "dialplan add extension 1234111,1,Goto(incomingdundi,s,1) into dundilookup"") in new stack WARNING[32626]: app_system.c:81 system_exec_helper: Unable to execute 'asterisk -rx "dialplan add extension 1234111,1,Goto(incomingdundi,s,1) into

Hi, I have Asterisk 1.4.4 on my linux box. Whenever i try to kick a participant in conference say "59681446" using following command meetme kick 59681446 1 where "1" is the participant number, following are the actions that asterisk takes * IVR "You have been kicked from this conference" is played. * Participant is taken out from that conference

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

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:

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 I am using asterisk version 1.6.0.5 I have build up one utility that will fire Originate Action on Manager... In which, i have define number to call eg. 919912312345 (MobileNumber) How can i know that this number format is true for Indian Number... In originate action, user can enter any international number.. How can I came to know this number format is right for that country...?? IS there

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

Please keep the ?me to? emails off the list. Regards; JV From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Magno Guimar?es Sent: Monday, June 22, 2015 3:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Product CDR/Queue/Meetme Hello, I am interested, too. Att,

Alex, Thank you so much for your response. I've been so consumed with other business that I only just now getting back to this issue. We have implemented your suggestion which is perfect. Thank you again. I've never asked a question of the community before and I'm extremely happy with the rapid response I received. Somewhat related to this initial problem I have an additional

Hey Helvio, Would like to check it out as well. Do email me, Mitul On 22-Jun-2015 9:05 AM, "Helvio Junior" <helvio.listas at gmail.com> wrote: > Gentleman, > > Moderators, i don't know if this topic if OFF-Topic, if yes, please tell > me. > > I had some difficult looking for a Asterisk software that provide me some > functions (For exemple: CDR, Queue