JT
Hi,
To dial a PSTN number through Nikotel used to work from Asterisk, but they had a very serious security issue (you could make calls anytime anywhere and their billing wouldn't charge it) and after I informed them of this, they changed their authentication mechanism and since then I have not gotten it to work (they didn't even thank me!).
Their tech people said it should work with a slight change: "yes, we changed it yesterday. Now the user part of the From: address has to be the same as the username in the Proxy-Authentication line. I don't know if the Asterisk can do that. The ATA186 does it b[y] default."
This CAN be done if you edit chan_sip.c, but when I did this, it billed me a few times for unconnected calls and I gave up trying to debug and switched to iConnect. iConnect is worse quality, but it is very easy to connect to.
I had much better quality with calls via Nikotel than iConnect, but their support is non-existent/bad at best. I sent them 3-4 e-mails about their security issue before they even responded.
FYI. Registering with Nikotel was futile anyways, because I never figured out how anyone could call into me. iConnect provides a PSTN-SIP dial in as an option, but I haven't tried it. Outbound calls do not require registering.
I can provide examples of iConnect connection scripts if you contact me offline.
On Saturday, July 5, 2003, at 07:42 PM, BK [address only for mailing lists] wrote:
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.
I wonder what the syntax is to dial a number via a VoIP provider. This appears to be documented NOWHERE.
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 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?
any hints appreciated kind regards bk
_______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
--- Paul Cheng M�ty�s kir�ly ut 10 H-1121 Budapest HUNGARY [EMAIL PROTECTED] mobile: +36 30 381-9311
_______________________________________________ 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
