Archives : February-2014
all,I have a user who is having trouble transferring calls, using a Grandstream GXP2xxx.Heres the use case that Ive seen:I call the user from phone A and he answers on phone B.Then, he hits the transfer button on his phone and dials an extension t..
When deny/permit is used in sip peer, does this only make sense when host=dynamic is used? What happens if host=ip is set?And if insecure=invite is set, does this override all above settings?Whats the relation of those ..
Here is my scenario. I have a SIP call between two SIP endpoints. A calls B. During the ringing, B disconnects (network cable is unplugged).But A continue ringing forever (until the dial timeout) even if asterisk detects that B is disconnected with ..
I have a setup with asterisk-11.7.0 and kamailio-4.1.1. I am following the setup guide at http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb . I want to run asterisk and kamailio on the same server, with SIP realtime configurat..
I havent been able to find the answer online, and am not currently able to conduct an experiment to find the answer…I understand that in a SIP call where G729 has been negotiated as the preferred codec, a G.729 licence is not consumed until there..
all We have an Asterisk s..
Guys,I am using Asterisk 1.8.20.0 built by mockbuild @ buildvm-24.phx2.fedoraproject.org on a x86_64 running Linux on 2013-01-18 19:52:25 UTCHow can I set variable in one context and then Redirect a channel to another context and use variable the..
,I have a fresh install of Asterisk 12.0.0 and Im going to use it only as a client. Im trying to SIP REGISTER with a remote SIP provider.The situation is that Asterisk is running in a VMware VM with a RFC IP address (192.168.1.2). The provider of ..
I am using Realtime on Asterisk 11.5 with a SQLite3 backend. While everything seems to be working fine I keep getting this error on my log files:[2014-02-17 19:55:18] WARNING[20569] res_config_sqlite3.c: Could not execute UPDATE sip_buddies SET ipa..