Hi Kevin, Thank you. Here's the requested information.
1) The Trunk is running 1.6.2.9. Also it's running a2billing. 2) The PBX is running asterisk 1.8.12.0 along with FreePBX. 3) I did directmedia on the trunk and canreinvite on the pbx since they were different versions. Thansk David On Mon, May 21, 2012 at 11:22 AM, Kevin P. Fleming <[email protected]> wrote: > On 05/21/2012 07:03 AM, David Wessell wrote: >> >> I am attempting to get an asterisk server to step out of the media >> path, but am running into a brick wall. Can someone assist? Here's my >> setup.. >> >> Ultimate SIP Provider ---> LCR Trunk (Asterisk 1.6) ----> PBX (Asterisk >> 1.8). > > > In order to be able to know whether any known bugs are interfering with what > you are trying to do, we need more specific version numbers. > > >> >> I am attempting to get the trunk to step out of the media stream. >> There is no NAT involved, all machines have a public IP. >> >> In the trunk's sip.conf I have: >> >> directmedia=yes >> directrtpsetup=yes > > > Please turn off directrtpsetup; it's experimental and doesn't always work as > you'd expect. In theory it is exactly what you want in this scenario, > though. If you are using Asterisk 1.6.0.x or 1.6.1.x, 'directmedia' won't be > recognized either. > > >> >> And on the connection to the pbx I have canreinvite=yes > > > Why 'directmedia' on one side and 'canreinvite' on the other? They are > synonyms, you should use the same name on both sides. > > >> >> On the pbx I have the trunk connection set to canreinvite=yes. > > > This is unnecessary, unless the devices on the other side of the PBX are > also on public IPs and you want the PBX to drop out of the media path as > well. > > >> >> In the CLI on the LCR trunk I see: >> >> -- SIP/blahblah-0000000b answered SIP/1722291028-0000000a >> -- Native bridging SIP/1722291028-0000000a and SIP/siproutes-0000000b >> >> Which would make me think that the lcr trunk is stepping out of the >> media stream. However when I pull up a tcpdump in wireshark I still >> see a RTP connection? Can someone point me in the right direction? > > > No, native bridging just means that the media stream will be bridged at the > RTP layer instead of in the Asterisk core. Whether that is done using a > Packet2Packet bridge in the RTP stack itself, or pushed out to the endpoints > (directmedia), it's still a native bridge. However, the fact that you are > seeing this message means you don't have any of the large number of reasons > that would impede native bridging (transcoding, recording, etc.). > > It seems like you have the configuration set up (mostly) properly, so in > order to know what is going on you're going to have to post a more complete > log snippet, including 'sip debug' output. > > -- > Kevin P. Fleming > Digium, Inc. | Director of Software Technologies > Jabber: [email protected] | SIP: [email protected] | Skype: kpfleming > 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA > Check us out at www.digium.com & www.asterisk.org > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- -- www.ringfree.biz 828-575-0030 -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
