Archives : March-2014
Unfortunately, afterhttp://git.asterisk.org/gitweb/?p=dahdi/linux.git;a=commitdiff;h=6cebc1c1fb12cc0661f3810ef47ad33206b2e398I am unable to build DAHDI-Linux in a mock chroot for packaging purposes.I believe this is related to the Makefile calling install_firmw..
Ive setup TLS/SRTP with Asterisk 11.8.1 and wonder if there is a CLI command to see if SRTP is active on a channel/call. I went through sip show … and core show channel… and did not see any mentioning of SRTP while there is an SRTP call active.Than..
I (canadian) store has a deal on for the vera lite controller:http://www.tigerdirect.ca/applications/searchtools/item-Details.asp?EdpNo=8930107&sku=VEP-STARTER1but this looks different than the vera lite green & white:http://www.amazon.com/Mi-Casa-Verde-VeraLite-Controller/dp/B007005364/ref=cm_cr_pr_product_top?..
The Asterisk Development Team has announced the releases of:DAHDI-Linux-v2.9.1DAHDI-Tools-v2.9.1dahdi-linux-complete-2.9.1+2.9.1This release is available for immediate download at:http://downloads.asterisk.org/pub/telephony/dahdi-linux http://downloads.asterisk.org/pub/telephony/dahdi-to..
all,Ive got some PHP code that opens an AMI socket and does a ConfBridgeList for a specific bridge (8888).This all works just fine but I need to filter the information displayed to only CallerIDName so I can see a complete list of names of participants.Af..
Asterisk 11.1.0 running on Ubuntu 12.04 64 bit Dahdi Digium T1 cardOccasionally, I will find an inbound call that just seems to be stuck, usually in our after-hours menu portion of the dial plan.This morning I had this onecore show channels concise DAHDI/i1/5184097869-1baf2!MainMenu!s!20!Up!BackGround!custom/aa-night-hello&custom/hours_8:0-17:0_0:0-0:0_0:0-0:0&custom/aa-night-instructions!5184097869!!!3!9393!(None)!sip1-1396004671.285644wh..
Experts, I want to know if there is any way to modify welcome banner on asterisk console when I connect using asterisk -rTha..
I would like to use AMD on outgoing calls using analog line. I tested with SPA3102 and cisco2811 as gw and asterisk 1.8.26.1 as well as 11.8.1 Other end is analog number behind another cisco/asterisk, also tested calling a mobile number with the s..
again,Ive got a user whos using a bunch of Grandstream GXP2xxxs.For the most part, they work, except for when they try to do a phone-based call transfer.Heres what it looks like is happening:Phone A is on a call with phone B. (B could be another pho..