Hello At this point, I have an Asterisk 1.4 + PC running XLite behind a NAT set up with a VOSP trunk that I can use to make/receive calls to/from the PSTN. Now, I'd like to be able to call any number on the Net that is advertised as "sip:user at domain.com", such as those: www.voip-info.org/wiki/view/Phone+Numbers Do I need to register a second trunk (FWD, etc.) through which those calls will be made? Can't my VOSP perform both tasks (landline + Internet calls)? Can I just let my Asterisk server connect to the remote SIP server through the SRV DNS record and have it dial the extension? Any example appreciated, thank you.
Just add something like this to your dialplan:
exten=>1234,1,Dial(SIP/user at domain.com)
Then, when you dial 1234 on your XLite, it will connect you to user at
domain.com.
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces
at lists.digium.com] On Behalf Of Gilles
Sent: Thursday, December 16, 2010 5:55 AM
To: asterisk-users at lists.digium.com
Subject: [asterisk-users] Call sip:user at domain.com?
Hello
At this point, I have an Asterisk 1.4 + PC running XLite behind a NAT
set up with a VOSP trunk that I can use to make/receive calls to/from
the PSTN.
Now, I'd like to be able to call any number on the Net that is
advertised as "sip:user at domain.com", such as those:
www.voip-info.org/wiki/view/Phone+Numbers
Do I need to register a second trunk (FWD, etc.) through which those
calls will be made? Can't my VOSP perform both tasks (landline +
Internet calls)? Can I just let my Asterisk server connect to the
remote SIP server through the SRV DNS record and have it dial the
extension?
Any example appreciated, thank you.
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On Thu, 16 Dec 2010 11:54:31 +0100, Gilles <codecomplete at free.fr> wrote:>Now, I'd like to be able to call any number on the Net that is >advertised as "sip:user at domain.com", such as those:I mean: Do I really have to first create a section in sip.conf each time a user needs to call a number on a new SIP server? http://wiki.ekiga.org/index.php/Connecting_Asterisk_to_ekiga.net
On Fri, 17 Dec 2010 14:36:58 +0100, Administrator TOOTAI <admin at tootai.net> wrote:>Domain part disappear. > >exten=>_9*.,1,Dial(SIP/${EXTEN:1}@ekiga.net) > >In Xlite call 9*031600Thanks for the tip but I wanted to be able to call _any_ SIP number, not just Ekiga, so needed a destination-agnostic solution.
On Fri, Dec 17, 2010 at 04:52:32PM +0100, Gilles wrote:>Thanks for the tip but I wanted to be able to call _any_ SIP number, >not just Ekiga, so needed a destination-agnostic solution.How would you _expect_ to be able to specify a destination server from a telephone keypad?
Le 17/12/2010 16:52, Gilles a ?crit :> On Fri, 17 Dec 2010 14:36:58 +0100, Administrator TOOTAI > <admin at tootai.net> wrote: > >> Domain part disappear. >> >> exten=>_9*.,1,Dial(SIP/${EXTEN:1}@ekiga.net) >> >> In Xlite call 9*031600 >> > Thanks for the tip but I wanted to be able to call _any_ SIP number, > not just Ekiga, so needed a destination-agnostic solution. >You can use SipBroker. http://www.sipbroker.com/sipbroker/action/providerWhitePages -- Daniel