[asterisk-users] Call bridged, but no sound

[Brian Candler]

> I turned on sip debugging and noted folowing differences in the output
> (1st='8904676', 2nd='890467610'):
> 
> 1st: INVITE sip:s at 81.223.241.115 SIP/2.0
> 2nd: INVITE sip:01890467610 at 81.223.241.115 SIP/2.0
> 
> 1st: To: sip:8904676 at p1.voip.inode.at
> 2nd: To: sip:890467610 at p1.voip.inode.at
> 
> 1st: From: sip:0132079780 at p1.voip.inode.at;tag=f6554db4ac48a72
> 2nd: From: sip:0132079780 at p1.voip.inode.at;tag=b9295878fc2630d
> 
> 1st: Looking for s in from-inode (domain 81.223.241.115)
> 2nd: Looking for 01890467610 in from-inode (domain 81.223.241.115)
> 
> From this point on, debug output is completely different because 1st
> answers and 2nd hangs up.
> But why is this so?

>From this, it looks to me like your SIP provider is being very kind and
sending you the full DDI in the INVITE when you dial the longer version of
the number. And so Asterisk is looking for this number in extensions.conf,
and currently fails to match.

Just try matching it in extensions.conf:

[from-inode]
exten => 01890467610,1,Answer()
exten => 01890467610,n,Echo()
exten => 01890467610,n,Hangup()

If that works, then you can use a pattern match to match everything
beginning with 018904676. More likely, you'll want to route to your internal
extensions using this number, or by stripping off the first 9 digits, so
that everyone gets their own DDI for free!

Regards,

Brian.