Archives : March-2016
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Ive read SIP Connect 2.0 draft lately.It mentions specific use if either of the following values is present in the From: field of an INVITE message. The values are:sip:unavailable@unkown.invalid sip:anonymous@anonymous.invalidIm using Asterisk 13 ..
I start asterisk 13.7.2 and it dies before I can rasterisk into it. Ive tried getting a coredump, but it doesnt coredump.I know there are a lot of errors in the log below, but most of those just say itll not load a module, and no big deal.When launch..
Thank you all for pointing me in the right direction.Now I learned I have to care about MTU.Best regards2016-03-03 21:27 GMT+01:00 Toufic Khreish..
I have an Asterisk 13 installation with an E1 card and I thought that DAHDI would be the default timing source for the system: pbxcore*CLI> module show like timing Module DescriptionUse CountStatus Support Level res_timing_dahdi.soDAHDI Timing Interf..
Im remotely managing an asterisk setup using an OpenVPN client on this Asterisk box, connecting to an OpenVPN server of mine).This box is mainly connected to PSTN. It is also connected to the Internet, only for remote management.The former ADSL l..
Im currently evaluating if it would possible/not too difficult to build and maintain an automatic attendant application.More precisely, my requirements are:- must work with Asterisk- should be installable on debian or CentOS- works this way :. cal..
I am having trouble with RTP and NAT :Below is a SIP SDP invite from a remote endpoint which is trying to call extension 420 which is the ECHO application .As you can see, the public IP is where the request comes in from,but the SDP contains the priva..
everyone!I tried to install chan_dongle for Asterisk 11 in a Ubuntu 14.04, but my Huawei E153 is not working in my Asterisk.I fallow this rules http://blog.denisbondar.com/post/asterisk11-chan_dongle_e1550-ubuntu14But not successes.Thanks in..
I am wondering what the best solution is for initiating a call from Outlook Contacts. I imagine something that would start the call from the outlook card (or similar) and then dial the users extension and the contacts phone number and place them i..
HelloI am trying to use the functions SHARED and IMPORT to share variables across SIP-channels.During my use I encounter 2 problems/questions.Question 1. only 1 shared variable per channel ??When I set 2 shared variables on a channel, and I read t..