Archives : February-2014
Home » 2014 (paged 44)
Greetings to all. I am not sure of this is a user question or a business so apologies in advance if it should be asked in the business list.A client of mine has a UK branch that is served by a provider that uses the Broadsoft solution. I want to cre..
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I am having trouble compiling dahdi-linux-complete-2.9.0+2.9.0.1 on a Raspbien 3.10.25+ kernel. I get the following error – /usr/src/dahdi-linux-complete-2.9.0+2.9.0.1/linux/drivers/dahdi/dahdi-base.c:570:2: error: implicit declaration of function kzal..
Im running a test system with Ast 11.7 and DAHDI 2.9.0. I loaded a TE205Pcard. Then I enter asterisk console, and once I do a pri show channels, the console no longer works correctly. Theres no output for any command after that. If I type pri show ..
This is evenily.My one Asterisk server and event listen App was working well for several month. Yesterday My event listen app stop work suddenly. I telnet to AMI via localhost 5038 to check the events, and I find Asterisk does not push the events un..
Hey all,Ive been fighting with this all morning, and I feel like this should be a relatively simple task, but I just cant get it to work.I currently have a very basic asterisk v11.6 setup with a single extension (a Bria softphone) and a single sip tr..
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..