Hi
thanks to everybody who responded to my earlier post. I have looked at all the material and links provided and tried everything in there, but it simply won't work for me.
My SIP phones register with Asterisk, but they cannot be called (everybody is busy at this time) nor can they call anything (error code 4, whatever that means) not even internal (yes I did give them appropriate context).
Further, Asterisk registers with my VoIP provider via SIP just fine, but I cannot make any calls even from the analog phones.
sip show registry gives me
Host Username Refresh State 63.214.186.6:5060 myusername 120 Registered
sip debug also confirms successful registration.
The command you will find more useful is "sip show peers". If your hosts are "(Unspecified)" then your SIP clients are not registering, and inbound calls will not work if you are using "dynamic=yes" in your sip.conf. Possibly it may be helpful if you would statically register your SIP phones until you get things working better ("host=10.3.2.3" in sip.conf)
I wonder what the syntax is to dial a number via a VoIP provider. This appears to be documented NOWHERE.
I would disagree. A VoIP provider is no different than a SIP phone; they are treated the same. If you are looking for examples, please see http://www.loligo.com/asterisk/ for my sample files, which contain some VoIP provider dial statements.
I tried this:
; International long distance through VoIP service ; exten => _00N.,1,Dial,SIP/${EXTEN:[EMAIL PROTECTED],tr exten => _00N.,2,Congestion
and sip debug tells me that the account doesn't match the one on record, whatever that means.
I tried this:
; International long distance through VoIP service ; exten => _00N.,1,Dial,SIP/[EMAIL PROTECTED]/${EXTEN:2},tr
You may be having at least one error due to syntax. The line above should look like:
exten => _00N.,1,Dial(SIP/[EMAIL PROTECTED]/${EXTEN:2},100,r)
You don't need the "t" unless you want the answering party to be able to transfer your calls in your own system (probably bad) and you need the "100" to tell the Dial statement how long to attempt the dial. If you don't want to specify number of seconds, you'd need to leave that area blank. (i.e.: ...TEN:2},,r) )
exten => _00N.,2,Congestion
and this doesn't even show anything but immediately gives me a busy signal. The fact that there is no debugging output leads me to believe that Asterisk didn't even attempt to try talking to the VoIP server.
Does anybody know how to dial a PSTN number through a VoIP service?
Is this standardised, at least within SIP? Or does it vary from provider to provider?
To the best of my experiences, it does not vary from provider to provider.
JT
any hints appreciated kind regards bk
_______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
_______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
