[asterisk-users] CISCO 2600 - VWIC 1MFT-E1

Fran Oliveira tech.oliveira at gmail.com
Thu Dec 7 11:25:23 MST 2006


Hi
In dial-peer voice 697617664 voip

your must specify into voip dial peer

session protocol sipv2
and check if session target sip-server is corect doing a ping to  sip-server
.
I think you must configure it with ipv4:ip_addres or map a host entry with
ip host sip-server x.x.x.x in global configuration mode

you have forgotten to configure a pots dial peer for your controler.
put something like this
dial-peer voice 10 pots
 destination-pattern 0T
 fax rate disable
 direct-inward-dial
 port 1/0:15
and try if you can write
authentication username "asterisk-uername" password XXXXXX

this last command should allow dial-peer voice 10 to register within
asterisk

I hope it will help you

best regards
2006/12/7, FaberK <f.faberk at gmail.com>:

> Hi to all,
> I got a Cisco 2651XM wired to an E1 PRI.
> What I want to do is to pass all incoming calls to my asterisk.
> This is my actual conf:
> http://pastebin.ca/270677
> with this I'm able to call my number from outside, but the call stop on
> the 2600, infact I can hear the tone, but I'm not able to forward calls to
> my asterisk.
>
> Anyone got an idea of my errors?
>
> Thanks to all.
> --
> .:FaberK:.
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com <http://easynews.com/>--
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061207/ea46052a/attachment.htm


More information about the asterisk-users mailing list