asterisk users - Jul 2003 - Please help -- Syntax for dialing VoIP provider

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

>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/${EXTEN:2}@calamar0.nikotel.com,tr
>exten => _00N.,2,Congestion
>
>and sip debug tells me that the account doesn't match the one on 
>record, whatever that means.
>
>I tried this:
>
>; International long distance through VoIP service
>;
>exten => _00N.,1,Dial,SIP/myusername@calamar0.nikotel.com/${EXTEN:2},tr
You may be having at least one error due to syntax.  The line above 
should look like:

exten => _00N.,1,Dial(SIP/myusername@calamar0.nikotel.com/${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
>Asterisk-Users@lists.digium.com
>http://lists.digium.com/mailman/listinfo/asterisk-users

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