I’m doing some testing with a very simple setup. 1 asterisk server and 2 grandstream sip phones.
I have the phones set up so that when I dial 100 on one sip phone, I connect to the other phone - this works well.
If I set up a call file to dial the first phone and then the other, the phones are connected, but I hear loud crackling or clicking (sound does carry through between the phones - I can just make out noises from one phone on the other)
For debug information, I have this on the non working connection initiated by the call file
With the working connection between the phones (where I dial the number), the debug is
As near as I can tell, these should be doing the exact same thing, but when initiated by the call file I am getting an error
WARNING[19076]: channel.c:2612 ast_indicate_data: Unable to handle indication 3 for ‘SIP/phone2-081be0c0’
The call file is
Channel: SIP/phone2
MaxRetries: 1
RetryTime: 60
WaitTime: 30
Context: phones
Extension: 200
Priority: 1
The sip.conf is
[general]
context=default
allowoverlap=no
bindport=5060
bindaddr=10.0.4.1
[phone1]
type=friend
context=phones
host=dynamic
[phone2]
type=friend
context=phones
host=dynamic
My extensions.conf is
[general]
;static=yes
;clearglobalvars=no
autofallthrough=yes
[globals]
[default]
exten => s,1,Verbose(1|Unrouted call handler)
exten => s,n,Answer()
exten => s,n,Wait(1)
exten => s,n,Playback(tt-weasels)
exten => s,n,Hangup()
[incoming]
[internal]
exten => 100,1,Verbose(1|Extension PHONE1)
exten => 100,n,Dial(SIP/phone2,30)
exten => 100,n,Hangup()
exten => 200,1,Verbose(1|Extension PHONE2)
exten => 200,n,Dial(SIP/phone1,30)
exten => 200,n,Hangup()
exten => phone1,1,Dial(SIP/phone1,30)
exten => phone1,n,Hangup()
exten => phone2,1,Dial(SIP/phone2,30)
exten => phone2,n,Hangup()
[phones]
include => internal