Archives : January-2014
Home » 2014 (paged 57)
everyone.Having experimented a but with a prototype of a system I described in an earlier thread (Reading DTMF sent by callee during a SIP call), Idecided to implement my requirement by transferring the call to another extension. This way, the cal..
I have an asterisk box with a peer configured with nat=force_rport,comedia, but asterisk keeps sending the audio to the private IP address and ignoring the client peer nat settings.If I check the sip show peer extension, I see both symmetric RTP ..
My target system is :PSTNSip ProviderRouter with fw/NATAsteriskSIP PhonesAsterisk is configured to keep NAT connection with SIP provider open (with qualifyfreq) so I dont have any problem (yet) with either casual incoming or outgoing calls.To work aro..
Im having this issue on my pbx, it appears that asterisk is refusing the codecs that my providers is proposing. My trunk configur..
there,Running patlooptest following instructions on Digium KB, using dahdi 2.5.1with a TE205P. Im getting this kind of output (not full output). Note, Im using a standard cat5 ethernet cable (30cm), with a Digium T10i crossover.Error 4985 (loop 198..
everyone!Asterisk is a large project and is used all over the world. As a result, the Asterisk project is lucky to have a large and thriving user community that communicates in a variety of places: IRC channels, forums, mailing lists, social media, ..
I asked this on the list over the weekend, and likely missed a few people inboxes. Im having a problem pulling data from RTPAUDIOQOS.For testing purposes I have asterisk sending QOS data to the console. It seems I get QOS data only if the caller ha..
I have a fresh install of asterisk 11.7.0when I run it and do a dialplan showThe only thing I see is:[ Context parkedcalls created by features ]700 =>1. Park() [features]-= 1 extension (1 priority) in 1 context. =-extensions.conf has the MANY conte..
Everyone,Calls that are private name private number have the following TO header:From: Unavailable ;tag=as120a1079.Dont tell anyone, but we are trying to put on a Were big enough to own the pricey softswitch look. Even though I would pick a OpenS..
all,Im looking into adding the ability to call me at me@mydomain.org on my Asterisk 11 box. Does anyone have any tips or dialplan snippets to allow this kind of access as securely as possible?Thank..