http://pastebin.ca/271763
Hi to all, To Fran:
As I understand your configuration , dial-peer voice 697617664 voip, only forward the pattern 697617664( destination-pattern 697617664) to XXX.XXX.XXX. 115:5060 ( session target ipv4:XXX.XXX.XXX.115:5060) that I think is your Asterisk box.
you are right, XXX.XXX.XXX.115:5060 is my * box where I've created a "friend" called 697617664 An incoming call in your E1 must much a destination pattern, your only
destination pattern is 697617664. Usually an E1 has several DID associated it in a consecutive range, 91 5344XXX for example.
here too, you are right, but I'm trying to receive at leat 1 call to 697617664, then for all the others will be not a problem. But first i need to let it works...!!! otherwise, for outgoing calls you must configure a pots dial peer ,you can
put a randon name to the dial peer. You can configure asterisk , without user registration with the sip.confinsecure option when I copied dial-peer voice 10 pots destination-pattern 0T should be .T it tells cisco 26xx router what patterns can be reached throught E1 I´ll take a look into the cisco web site for sip user authentication, I have a configuration done, but with FXS interfaces and worsk fine.
For outgoing calls, at this moment I'm not interested. On the new configuration, I've also changed the codecs, leaving the g711 only. Unfortunately always the same: calling my number, the call reach the 2600(infact I hear the tone), but is not forwarded to the sip-server. To Pavel: thanks for your suggestion regarding MGCP, but the fact is that I got all sip, and never worked with mgcp. Thanks to all Best Regards F.
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