Hi everyone,
I would like to connect two phone numbers using a web page, asterisk and voipbuster. The web page has two fields: the caller’s extension and the callee’s number. When the caller hits the submit button, the web page calls a java program (asterisk-java) that places the call through asterisk. The caller’s phone rings, he picks up, then the callee’s phone rings and the callee picks up.
When both phones are pstn numbers, there no sound between the two phones.
When the caller’s phone is a sip one (I use X-lite), the sound flows very well between the two phones (exten=>4321,1,DIAL(SIP/mike).
Why there no sound in the first case ?
Thank you in advance
here is my extension.conf
[outgoing]
exten=>1234,1,DIAL(SIP/0015145046024@voipbuster)
exten=>4321,1,Dial(SIP/mike)
here is my sip.conf
[paul]
type = friend
username = paul
secret = talvi
host = dynamic
context = outgoing
[mike]
type=friend
username=mike
secret=enemes
host=dynamic
context=outgoing
nat=yes
[voipbuster]
type=peer
host=sip1.voipbuster.com
username=myusername
fromuser=myusername
secret=mysecret
canreinvite = no
dtmfmode = inband
dtmf=inband
insecure=very
context=from-voipbuster
qualify=yes
here is my manager.cong
[general]
enabled = yes
port = 5038
bindaddr = xx.xx.xxx.xxx
[manager]
secret=thesecret
permit=0.0.0.0/0.0.0.0
read=system,call,log,verbose,agent,command,user
write=system,call,log,verbose,agent,command,user