George, Thanks for the reply. I have seen that and read it before. However, they way it reads, and the way I'm still understanding it is this:
If the call flow was say, from CME as SIP and rtp-nte to CCM via the IPIPGW, then from IPIPGW to CCM as H323 and h245-alphanumeric, then I would absolutely use dtmf-relay rtp-nte digit drop on the incoming SIP dial peer on the IPIPGW. Then on the outbound dial peer to CCM have dtmf-relay h245-alphanumeric. However, on this specific question, where the call flow is H323 to SIP, I don't see the need to drop rtp-nte. But to add to my confusion, from the link provided is this: The following example configures DTMF relay with the digit-drop keyword to avoid both in-band and out-of band tones being sent to the outgoing leg on H.323 to H.323 or H.323 to SIP calls: dial-peer voice 1 voip session protocol sipv2 dtmf-relay h245-alphanumeric rtp-nte digit-drop But SIP doesn't support h245-alphanumeric, so why is it here? Is this example supposed to be an H323 dial-peer instead of a SIP dialpeer? Thanks! Date: Wed, 26 Mar 2008 15:25:40 -0400 From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [OSL | CCIE_Voice] RFC2833 strip CC: [EMAIL PROTECTED]; [email protected] 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. Get news, entertainment and everything you care about at Live.com. Check it out! _________________________________________________________________ Watch “Cause Effect,” a show about real people making a real difference. Learn more. http://im.live.com/Messenger/IM/MTV/?source=text_watchcause
