Hi, I'm trying to connect Asterisk 1.4 and Cisco CallManager 5 using SIP Trunk without MTP (media termination point). Howerver, Cisco 79xx phones do not support RFC2833, they always notify CCM5 via SKINNY channel no matter where they send RTP to. For non-MTP trunk there's Out-of-band DTMF support in CCM5 called "kpml". I wonder if Asterisk can support it. I found an intertnet-draft for kpml: http://tools.ietf.org/id/draft-ietf-sipping-kpml-07.txt, but it seems to be very old - "Expires June 25, 2005". I know that using MTP in SIP Trunk at CCM5 makes DTMF work in RFC2833, but MTP resource is very limited and I don't want to proxy RTP via CCM5. Please, do not offer to use H.323. Thanks in advance. Grigoriy.
KPML is now an RFC -- http://www.ietf.org/rfc/rfc4730.txt Asterisk doesn't support KPML today. That doesn't mean it can not be developed if there is sufiicient interest. The true value of adding KPML support in Asterisk is when it is acting as a 'softswitch' (call controller without media hairpin) such that it can install a digit map and collect digits from end-points over the signaling path. Raj On 4/19/07, Grigoriy Puzankin <gpuzankin@gmail.com> wrote:> > Hi, > > I'm trying to connect Asterisk 1.4 and Cisco CallManager 5 using SIP > Trunk without MTP (media termination point). Howerver, Cisco 79xx phones > do not support RFC2833, they always notify CCM5 via SKINNY channel no > matter where they send RTP to. > > For non-MTP trunk there's Out-of-band DTMF support in CCM5 called > "kpml". I wonder if Asterisk can support it. > > I found an intertnet-draft for kpml: > http://tools.ietf.org/id/draft-ietf-sipping-kpml-07.txt, but it seems to > be very old - "Expires June 25, 2005". > > I know that using MTP in SIP Trunk at CCM5 makes DTMF work in RFC2833, > but MTP resource is very limited and I don't want to proxy RTP via CCM5. > Please, do not offer to use H.323. > > Thanks in advance. > Grigoriy. > > > > _______________________________________________ > --Bandwidth and Colocation provided by 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/20070419/a5b69fce/attachment.htm
Grigoriy wrote:> I'm trying to connect Asterisk 1.4 and Cisco CallManager 5 > using SIP Trunk without MTP (media termination point). > Howerver, Cisco 79xx phones do not support RFC2833, they > always notify CCM5 via SKINNY channel no matter where they > send RTP to.If you are running the phone loads that shipped with CCM5, then your skinny phones have 'support' for RFC2833. CCM figures out during the call if the call will traverse a SIP trunk and instruct the phone to use RFC2833 for DTMF I have a CCM5<->Asterisk trunk setup for MeetMe conferencing with NO MTP and DTMF works fine.> For non-MTP trunk there's Out-of-band DTMF support in CCM5 > called "kpml". I wonder if Asterisk can support it.Interesting, will look it up...> I found an intertnet-draft for kpml: > http://tools.ietf.org/id/draft-ietf-sipping-kpml-07.txt, but > it seems to be very old - "Expires June 25, 2005".> I know that using MTP in SIP Trunk at CCM5 makes DTMF work > in RFC2833, but MTP resource is very limited and I don't want > to proxy RTP via CCM5.I don't blame you, nut again as of CCM5 you are no longer required to use an MTP for SIP trunks.> Please, do not offer to use H.323.OK, not an offer, but I have found that even as of the latest CCM5 release, the SIP stack is 'quirky'. I also maintain a H323 trunk between the same CCM cluster and Asterisk and in general it is much better behaved (using chan_ooh323). Either will work.... Dan