[cisco-voip] incoming dial peers
Dane Newman
dane.newman at gmail.com
Fri Oct 16 02:13:13 EDT 2009
Hello Nick
Thanks for your help
I created the the following config
voice translation-rule 1
rule 1 /*^/ /190/
voice translation-profile aa
translate called 1
dial-peer voice 1000 pots
description incoming Call
preference 1
incoming called-number 6784663444
progress_ind setup enable 3
progress_ind progress enable 8
dtmf-relay rtp-nte
no vad
translation-profile incoming aa
This should match the incoming called number 6784663444 and translate it
to my auto attendant hunt pilot of 190 correct?
I had read below that Digit translation rules are not supported for inbound
session initiation protocol (SIP) calls.
http://www.cisco.com/en/US/docs/ios/12_2/voice/configuration/guide/vvfpeers.html
I guess this has been updated in the latest code to work right?
On Thu, Oct 15, 2009 at 6:07 PM, Nick Matthews <matthnick at gmail.com> wrote:
> This may help:
>
> http://www.cisco.com/en/US/partner/tech/tk652/tk90/technologies_tech_note09186a008010fed1.shtml
>
> On your first example you don't have a codec set. It will be g729r8
> by default. As well, no DTMF-relay so you will not have that.
>
> You probably want to add something like this:
>
> codec g711ulaw
> no vad
> dtmf-relay rtp-nte
>
>
> To 'plar' the number you would use a translation pattern.
>
>
> Regarding your other problem - there isn't enough information. Your
> dial peer looks like a standard dial peer, but without a better idea
> of the call flow it's hard to tell. There's nothing specific in that
> dial peer that would make it an incoming dial peer, but rather an
> outgoing dial peer.
>
> -nick
>
> On Thu, Oct 15, 2009 at 3:46 PM, Dane Newman <dane.newman at gmail.com>
> wrote:
> > Hello All,
> >
> > I have a question about how to exactly configure incoming dial peers. I
> > have a sip trunk to an ITSP for my incoming telco connection. The number
> is
> > 6784663444. There are four channels in the trunk but only one number.
> Am I
> > matching the called number properly? How can I plar it to the unity
> > connection auto attendant of hunt pilot 190?
> >
> > dial-peer voice 1000 voip
> > description incoming Call
> > preference 1
> > incoming called-number 6784663444
> > progress_ind setup enable 3
> > progress_ind progress enable 8
> > !
> >
> > dial-peer voice 100 voip
> > description 190 AA
> > preference 1
> > destination-pattern 190
> > progress_ind setup enable 3
> > progress_ind progress enable 8
> > voice-class h323 50
> > session target ipv4:10.1.80.6
> > dtmf-relay h245-alphanumeric
> > codec g711ulaw
> > no vad
> >
> >
> >
> > Also....
> >
> >
> > If I have another gateway across a WAN sending calls to extentions
> 100-500
> > Will these incoming dial peers match the incoming calls and route them to
> my
> > cucm 7 properly?
> >
> > dial-peer voice 100 voip
> > description 1-5xx extension to PUBLISHER
> > preference 1
> > incoming called-number [1-5]..
> > progress_ind setup enable 3
> > progress_ind progress enable 8
> > voice-class h323 50
> > session target ipv4:10.1.80.6
> > dtmf-relay h245-alphanumeric
> > codec g711ulaw
> > no vad
> >
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> >
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <https://puck.nether.net/pipermail/cisco-voip/attachments/20091016/3784ae44/attachment.html>
More information about the cisco-voip
mailing list