Im testing a few IAX trunk scenarios in a controlled lab.From server2 extension 5000 (Server IP Address 172.16.200.212) I dial 6001 which goes across the IAX trunk to server 1 (IP address 172.16.200.210).Instead of ringing the 6001 phone, it plays tt-weas..
I am having a problem with SendDTMF – it is not sending the numbers properly during the phone call.. I want the numbers key to to be pressed/sent automatically after 3 seconds during a phone call. I use software phone to test it… when I dialed 5..
This is a really simple problem that I just cant get to work. There are two Asterisk servers with the following sip user and peer. When a call is attempted, Asterisk is not sending authentication details in response to the 401. Note, if the secret..
I have been trying to get the dynamic parking working. For some reason when I park a call using this method the console says it is using the default parking context not the one I am trying to specidfy. It also is playing the parked extension to the call..
all, I am using to Xlite to save video voice mail. when i retreive it, then only video show , no voice is here out. Plz tell me where ,i am wrong , and how i can able to see video plus here audio in voice mail box. I did following configuration In Sip.c..
all In sip.conf i take as [general] videosupport=yes [phone1] type=friend host=dynamic context= employees disallow=all allow=ilbc allow=g729 allow=gsm allow=g723 allow=ulaw allow=alaw allow=adpcm allow=h2..
all In sip.conf i take as [general] videosupport=yes ; then UDPTL will flow to the remote device [phone1] type=friend host=dynamic context= employees disallow=all allow=ilbc allow=g729 allow=..
I am making confrence application. In sip.conf [phone1] type=friend host=dynamic Takes an alphanumeric string. context= employees [phone2] type=friend host=dynamic context= employees [phone3] type=friend host=dynamic context= employees In extension.c..
, Im still having trouble using asterisk with the grandstream gxw4108, in the gxw4108 Im using 1 stage dialing in the profile1 I already type my asterisk server address 192.168.14.80 and my grandstream IP is 192.168.101.184 heres my asterisk config fi..
Hello Have a setup of asterisk with realtime SIP devices. Trying to organise monitoring of my SIP devices. Once device registered, its state becomes NOT_INUSE (result of DEVICE_STATE(SIP/device) function). Simulating of device breakage – powerdown ..