Hey Eric the following link to the doc CD has a pretty good write on this: http://www.cisco.com/en/US/docs/ios/voice/command/reference/vr_d2.html#wp1449003
Here is the write up under usage guidelines: To avoid sending both in-band and out-of band tones to the outgoing leg when sending IP-to-IP gateway calls in-band (rtp-nte) to out-of band (h245-alphanumeric), configure the *dtmf-relay *command using the *rtp-nte *and *digit-drop* keywords on the incoming SIP dial peer. On the H.323 side, and for H.323 to SIP calls, configure this command using either the * h245-alphanumeric* or *h245-signal* keyword. hth, George On 3/26/08, Jose Linero Welcker <[EMAIL PROTECTED]> wrote: > > > Hi Erin: > > I think you are wright, if the call are coming from H323 to SIP you don´t > need to strip the DTMF tones, this is just for calls from H323 to SIP where > you strip the DTMF tones in the SIP incoming dial-peer. > > Regards, > > Jose > > ------------------------------ > From: [EMAIL PROTECTED] > To: [email protected] > Date: Wed, 26 Mar 2008 12:55:39 -0500 > Subject: [OSL | CCIE_Voice] RFC2833 strip > > I'm confused about a question that is in almost every multiprotocol lab. > > 'Configure the HQ-RTR as an IPIPGW. Calls coming into it from CCM via H323 > using G711ulaw and routed out to BR2 CME via SIP using G729. Also, calls > coming from CME via H323 using G729 and routed to CCM via SIP using > G711ulaw' > > My question comes at this part... > 'Ensure when calls are coming from SIP to H323 that RFC2833 is properly > stripped. Ensure that if calls are coming from H323 to SIP that RFC2833 is > used for the SIP side.' > > So, it's my understanding that in this call flow, calls are only H323 to > SIP. I don't see a SIP to H323 call. Am I correct? So then, why would I need > the following? > > dial-peer voice 11 voip > destination-pattern 3... > session protocol sipv2 > session target ipv4:10.1.202.1 > dtmf-relay rtp-nte digit-drop h245-alphanumeric <--- Why the digit drop? > and h245-alphanumeric? > > From what I've read, the digit drop will drop the in band dtmf tones from > the rtp-nte relay, allowing the out of band to pass through. > But in this call flow, the call is coming from CCM to IPIPGW as H323, and > leaving IPIPGW as SIP. > > It's my opinion that I would only need the following: > dial-peer voice 11 voip > destination-pattern 3... > session protocol sipbv2 > session target ipv4:10.1.202.1 > dtmf-relay rtp-nte > > Am I missing something? > Thanks, > Erin > > ------------------------------ > Watch "Cause Effect," a show about real people making a real difference. Learn > more. <http://im.live.com/Messenger/IM/MTV/?source=text_watchcause> > > > ------------------------------ > Get news, entertainment and everything you care about at > Live.com<http://live.com/>. > Check it out! <http://www.live.com/getstarted.aspx> >
