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i am asked to implement a call center of 50 seats for my company , and i was wondering if Asterisk can fit this as a relaibale and low price system is it mature enough for this task?? best regards Gers -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090712/b5bab36d/attachment.htm

does Asterisk suppoet CSTA protocol for CTI applications? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090722/a2149cc5/attachment.htm

Hi Rhishi, PQevalaudio is very unreliable and buggy. I have compared to PEAQ and - as a result - now I am not using it anymore. With best regards, Christian Hoene Von: opus-bounces at xiph.org [mailto:opus-bounces at xiph.org] Im Auftrag von Rhishikesh Agashe Gesendet: Freitag, 4. Oktober 2013 12:35 An: opus at xiph.org Cc: Rasmi Mishra Betreff: [opus] ODG (Objective Difference

does anyone use OpenSBC , or know if it is mature stable opensource for a production enviroment http://rpm.pbone.net/index.php3/stat/4/idpl/10795970/com/opensbc-1.1.4-3.el5.pp.i386.rpm.html -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091201/67c95aac/attachment.htm

Hi, I checked the ODG (Objective Difference Grade) scores for a few reference vectors using the PQEvalAudio Tool and found that some of them show ODG scores as high as -3.5 If we look at the range as described in the link below, it looks unacceptable. http://www-mmsp.ece.mcgill.ca/documents/Software/Packages/AFsp/PQevalAudio.html Am I missing something or are these scores valid? Thanks and

could anyone help explaining what does this error mean? i get this error when make a video/ audio call from X-lite to Bria prof. phone rtp.c:1739 ast_rtp_read: Unknown RTP codec 126 received from '10.0.0.26' Gres -------------- next part -------------- An HTML attachment was scrubbed... URL:

In that case, can you please suggest a reliable tool? Thanks, Rhishi From: Christian Hoene [mailto:christian.hoene at symonics.com] Sent: Friday, October 04, 2013 17:30 To: Rhishikesh Agashe; opus at xiph.org Cc: Rasmi Mishra Subject: AW: [opus] ODG (Objective Difference Grade) scores for Opus Encoder using PQEvalAudio Tool Hi Rhishi, PQevalaudio is very unreliable and buggy. I have compared

Hi all, I want to use a cell phone as my FXO line to Asterisk Box ,did anyone try this and configured it and how to physically connect it to Asterisk server? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080506/6fafcfba/attachment.htm

Does anybody know if i can make (chan_mobile) module to be installed and work with Asterisk 1.4.19 ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080513/57510b8a/attachment.htm

can i use MP3 files as an IVR prompts directly without converting to .gsm format? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091010/67205e07/attachment.htm

Is there a way to ensure that the source IP address from witch the SIP user register is not tampred with , is there a feild in the SIP register message header can be used to achive this security ? i have an asterisk server in witch SIP users register through an SBC(session border controller) , i wanna make sure that those users are really registering from the IP they are claimming they are

Hi, While testing the Opus Decoder we came across the following: If the decoder encounters an 'Invalid Payload Length' the decoding of the stream is stopped. Also, when the decoder encounters 'Range coder state mismatch', the decoding of the stream is stopped. I believe that it should reject the erroneous payload and start decoding the next payload for it to work properly in a

i am trying to make a video call on asterisk 1.6 , my configuration is an - asterisk 1.6 on Centos on virtual machine VmWare - Xlite softphone one windows xp (the Host operating system) - X-lite client on another windows XP (the Guest operating system ) i put the paramter videosupport=yes under the general section in sip.conf i allowed the video codecs for each client in sip.conf for