Hello, Am new to Asterisk and am trying to set up a sip trunk with multiple lines.
My sip provider offers multiple numbers which end with 0800…00 - 0800…99.
Am able to make calls using only the number ending with 00, unless i declare the sip-trunk to use any other number in the from-user section. if i don’t declare any number it produces an error Temporarily unavailable when making a campaign with multiple outgoing calls, some calls do not go through because the trunk is busy or in use
Below are my current configurations
exten => _0XXXXXXXX.,1,AGI(agi://127.0.0.1:4577/call_log);
exten => _0XXXXXXXX.,n,Dial(SIP/trunk1/${EXTEN},,tTo);
exten => _0XXXXXXXX.,n,Hangup();
I also tried the below dialplan but there was no difference
exten => _0XXXXXXXX.,1,AGI(agi://127.0.0.1:4577/call_log);
exten => _0XXXXXXXX.,n,Dial(SIP/trunk1/${EXTEN},,tTo);
same => n, Verbose(2, D0 DIALSTATUS => ${DIALSTATUS})
same => n, GotoIf($[ $['${DIALSTATUS}'='BUSY'] | $['${DIALSTATUS}'='NOANSWER'] ]?next_trunk)
same => n(next_trunk),Dial(SIP/trunk2/${EXTEN},,tTo);
same => n,Hangup();
Sip.conf
[trunk1]
disallow=all
allow=ulaw
allow=alaw
type=friend
host=IP
dtmfmode=RFC2833
qualify=yes
fromuser=0800544400
fromdomain=IP
context=from-pstn
insecure=port,invite
[trunk2]
disallow=all
allow=ulaw
allow=alaw
type=friend
host=IP
dtmfmode=RFC2833
qualify=yes
fromuser=0800544401
fromdomain=IP
context=from-pstn
insecure=port,invite
Oh, and am running asterisk 1.8