search for: kantharuban

...On Thu, Sep 3, 2015 at 7:20 PM, Vinicius Fontes <vinicius at aittelecom.com.br> wrote: > You might want to use the Originate() application instead. Check its usage > by issuing the command 'core show application originate' on Asterisk CLI. > > 2015-09-03 9:09 GMT-03:00 Kantharuban Ruban <kanth.ruban at gmail.com>: > >> Hello Group, >> >> I have a requirement to dialout some external number, once >> the call is answered the same has to be forwarded to an Internal Queue. >> >> Please help me. >> >> I have tri...

Hello Group, I have a requirement to dialout some external number, once the call is answered the same has to be forwarded to an Internal Queue. Please help me. I have tried calling with two SIP end point forwarding , even that is not working, My dial plan line is , Dial(SIP/19201/19202,300) -- *Best regards,* *Ruban.S* -------------- next part -------------- An HTML attachment

Hi List, I am trying to setup a Asterisk setup in AWS instance Centos6.5 . I have installed Asterisk 13.4 with srtp,pjproject. I have configured two numbers for webRTC clients, when i try to call from a client (sipml5) to another client (sipml5) it throws the following error: "chan_sip.c:5851 dialog_initialize_dtls_srtp: Attempted to set an invalid DTLS-SRTP configuration on RTP

...ed such issues or any solutions in latest versions of Asterisk to handle such situation. Best regards, Ruban.S Siemens Limited, IC BT SSP ES R&D IPI 1 SP Infocity, Block B, 2nd Floor, #40, MGR Salai, Kandanchavadi, Perungudi, Chennai - 600 096 Tel: +91 44 33524336 Mobile: +91 9500084833 mailto:kantharuban.s at siemens.com -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140401/7f699671/attachment.html> -------------- next part -------------- An embedded and charset-unspecified text was scrubbed......

Hi List, The below error caused the Asterisk to crash, if anyone have idea on this please reply,(Asterisk version :1.8.9) [Sep 2 15:59:53] WARNING[24418] channel.c: Codec mismatch on channel SIP/18202-0002d011 setting write format to ilbc from ulaw native formats 0x4 (ulaw) [Sep 2 15:59:53] WARNING[24418] channel.c: Unable to find a codec translation path from 0x4 (ulaw) to

Hello List, Good day, We have an application, where we redirect a channel to meet me. Sometimes the channel is getting hanged up by Asterisk, and we get an hang-up event. Please reply back, if any one faced such issue. Here is the hangup event info, -HANGUP {calleridname=<unknown>, connectedlinename=<unknown>, uniqueid=1384413814.79523, cause=0,