I have my asterisk Server A registered as a client with another asterisk Server B. When I place a call from Server A to B I get the following: WARNING: chan_sip.c:12673 check_auth: username mismatch, have , digest has NOTICE: chan_sip.c:19..
Hello! Ater several successful SRTP-enabled calls with SRTP set to Mandatory, asterisk starts to give the following warnings in Log: WARNING res_srtp.c: SRTP unprotect: authentication failure (continiously) and client hears no sound. After i rest..
We had 2 asterisk 1.4 connected together in iax, all was fine. One of them was upgraded (server and Asterisk) in 220.127.116.11, the other end is in 1.4.38 When calling to 1.4 to 1.6.2 -remember, its iax- all is good. But calling from 1.6.2 to 1.4 give a ..
Is it possible to use a live audio stream in asterisk I want to call a number and then hear an external audio stream. For example http://www.radioveronica.nl/radioveronicaplayer/radioveronica.asx I thought it was possible to use musiconhold, but I ..
Friends, Do we need to change any Asterisk configuration files (Or any file related to Asterisk for that matter) when we put Asterisk box from one network to another? It is assumed that DB is on the same box. Asterisk box has got Asterisk running..
All; A2Billing is working fine for Asterisk, but in case I need to use Asterisk and Gnugk and I need to manage the accounts and the billing from one Database and one billing system, so I need a prepaid billing that can work with both. Which prepaid bill..
All; In cisco IP Phone, I can assign for a features for the buttons at the Phone (one button to be for call pickup and one button to be for call forward and button to be for bridge and one button to be for another extension). How can I do the same t..
Lots of VoIP, SIP and Asterisk-related discussion and some free phones and Polycom software today, join us at the usual place: http://www.voipusersconference.org Call sip:email@example.comSkype:vuc.me (via Skype for Asterisk and PhonefromHere.c..