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Kepress While On Queue 27 Aug 2013 | 07:30 pm
Will Keypress option will work when am in the queue and hearing MoH?Lets say a caller is waiting in queue and while he is hearing MoH, can he key in some DTMF and go to some other queue? is that possi...
ISDN Outgoing Caller Id 27 Aug 2013 | 06:05 pm
is anybody out there who can set the outgoing caller id on ISDN (CAPIor misdn) channels? Ive tryed everything what I found in forums, os voip-info.com but no luck. I use a fritz card with CAPI in my f...
Need Input On Scalable System Design… 27 Aug 2013 | 05:16 pm
Hey All,Growing call center. Currently at about 200 call center staff, running about 1000 calls per hour. Gearing up to double that. Not too sure that a single server will support that growth. So, Im ...
Channel Driver Development In Ireland 27 Aug 2013 | 03:52 pm
Guys.I wanted to know from you, if you know: 1) any companies in Ireland that develop software and hardware for Asterisk integration with legacy interfaces (E1, T1, FXO, FXS...) . 2) any companies in ...
Introducing Sippy Cup: SIPp Load Testing Made Easy 27 Aug 2013 | 02:34 pm
--Apple-Mail=_58FC9642-DA44-4009-8D2B-1932FB21563CContent-Transfer-Encoding: quoted-printable Content-Type: text/plain; charset=us-asciieveryone,Recently weve been focusing quite heavily on making Adh...
Dahdi Gains 27 Aug 2013 | 08:08 am
Im trying to find the differences between the two CLI gain parameters of DAHDI : dahdi set swgain and dahdi set hwgain. When I change one of these parameters the output of :asterisk -rx dahdi show cha...
Getting Asterisk 11.5 To Use TURN 27 Aug 2013 | 05:12 am
Ive configured TURN in rtp.conf in Asterisk 11.5.The credentials are correct because I can get Chrome to get relay candidates and attach them to the SDP, but Asterisk doesnt want to play ball.Theres l...
Interconnect Radio Units Using Asterisk 27 Aug 2013 | 12:12 am
all,This my first mail in the community. My name is Manolo. Im new in Asterisk. My objective is to control some radio using Asterisk via web. In google I could see that the application app_rpt had thi...
Asterisk 11.5 Not Honoring RTP Port Change In RE-INVITE 26 Aug 2013 | 10:08 pm
I have an Asterisk 11.5 system, using SIP Realtime and operating as a ITSP.One of my customers endpoints is a NetVanta 7100 PBX system that has a SIP trunk connection to my Asterisk box.The NV 7100 ha...
How To Get The Original SIP Result Code 22 Aug 2013 | 02:43 pm
B.H. im using AMI Originate action (with async=true) to send outgoing calls to a SIP trunk (using asterisk-java library to connect to AMI).The problem is that in case of failed originate, OriginateRes...
