With the Asterisk Manager Interface Orginate command I manage to set up a call between an external SIP phone number and an internal extension (1234) as follows:
exten => 1234,1,Dial(SIP/local_phone)
exten => 7890,1,Dial(SIP/“an_external_phone_no”@my_proivider2
However, if I replace the internal number (1234) in the Original command by another number that will dial an external SIP phone number (e.g. 7890), there will be NO sound in the established connection (native bridge).
On the other hand:
If I change the Exten parameters as follows:
Exten: s (start entension)
and change my extension.conf as follows:
exten => s, DISA(nopassword|default)
Then, when the channel has been established, a dial tone can be heard and another outbound number can be dialed.
The result is thus 2 external phone are connected to each other AND there is sound between those phones.
Does any one know why there was no sound in the call in the first case ???
My Asterisk is behind a NAT (but no firewall). It is likely a NAT issue …