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

[John Todd]

The human going by "beekay" on IRC and BK on this list said that he 
had solved the problem with Nikotel by using the "fromuser=" 
parameter in sip.conf under his nikotel peer to make his outbound 
calls match his registrations.

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/${EXTEN:2}@calamar0.nikotel.com,tr
>>exten => _00N.,2,Congestion
>>
>>and sip debug tells me that the account doesn't match the one on 
>>record, whatever that means.
>>
>>I tried this:
>>
>>; International long distance through VoIP service
>>;
>>exten => _00N.,1,Dial,SIP/myusername at calamar0.nikotel.com/${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
>>
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>>
>
>
>---
>Paul Cheng
>Mátyás király ut 10
>H-1121 Budapest HUNGARY
>paul.cheng at alum.mit.edu
>mobile: +36 30 381-9311
>
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