How do I set up a dial plan for outgoing SIP URI dialing?

I want to be able to dial directly to any publicly accessible SIP URI from a softphone.
I have found I can express them explicitly in the dial plan, ie:

exten => 1,1,Dial(SIP/echo@iptel.org)

Then if i dial 1 it works fine.

However, if I call "echo@iptel.org" from a softphone then what i see is

  1. an invite from the UA to asterisk to SIP:echo@iptel.org
  2. then a DNS SRV query by the asterisk server for _sip.udp.echo
  3. DNS query response ‘no such name’
  4. an invite going out to a public IP address that i do not recognize to SIP:echo.

Is there i way that I can look for “@” in the dial plan, and when found send the entire string including the FQDN out the sip channel? Before it is stripped? Something like

exten => _.[@].,1,Dial(SIP/${ORIGINAL_ADDRESS})

Thanks

Yes you can allow sip Uri dialing check google for that If i recall there is popular macro.

This is not really supported, Asterisk can be made to select the context based on the domain, but accessing the whole URI is probably not possible. If it is possible look through the list of standard functions and built-in channel variables.

Are you sure you don’t want a SIP proxy?

Hi, Please suggest how to setup sip trunk between Asterisk server and Avaya CM 5.2.1 ??

Kindly share the required configuration of extensions.conf & sip.conf file… it would be great if it covers all the basic configuration from Avaya CM as well.
Thanks in advance!!

[quote=“anilksi”]Hi, Please suggest how to setup sip trunk between Asterisk server and Avaya CM 5.2.1 ??

Kindly share the required configuration of extensions.conf & sip.conf file… it would be great if it covers all the basic configuration from Avaya CM as well.
Thanks in advance!![/quote]

Is this a SPAM? It’s not related to this topic.

I think it is just someone who hasn’t read:

catb.org/~esr/faqs/smart-que … bespecific

catb.org/~esr/faqs/smart-que … l#explicit

and particularly:

catb.org/~esr/faqs/smart-que … #noprivate