[Asterisk-Users] Please help -- Syntax for dialing VoIP provider

Paul Cheng
Sun, 06 Jul 2003 23:59:19 -0000

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/${EXTEN:2}@calamar0.nikotel.com,tr
> exten => _00N.,2,Congestion
>
> and sip debug tells me that the account doesn't match the one on
> record, whatever that means.
>
> I tried this:
>
> ; International long distance through VoIP service
> ;
> exten => _00N.,1,Dial,SIP/myusername@calamar0.nikotel.com/${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
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>


---
Paul Cheng
M?ty?s kir?ly ut 10
H-1121 Budapest HUNGARY
paul.cheng@alum.mit.edu
mobile: +36 30 381-9311