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). 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 And on the connection to the pbx I have canreinvite=yes On the pbx I have the trunk connection set to canreinvite=yes. 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? Thanks David -- -- www.ringfree.biz 828-575-0030
Hi, Can you check if there is any transcoding involved with these calls, or maybe some NAT handling needs to be done by asterisk so it's not stepping out of the media-path !? Regards, Sammy On Mon, May 21, 2012 at 5:03 PM, David Wessell <david at ringfree.biz> 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). > > 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 > > And on the connection to the pbx I have canreinvite=yes > > On the pbx I have the trunk connection set to canreinvite=yes. > > 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? > > Thanks > David > -- > -- > 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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120521/ed848330/attachment.htm>
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=yesPlease 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=yesWhy '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: kfleming at digium.com | SIP: kpfleming at digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com & www.asterisk.org
Hi Jared & Kevin, Thanks for taking the time to answer my questions. I wonder if I could just be reading the tcpdump incorrectly? I'm still seeing rtp streams (and Jared, I have modified the dial string to remove the L).. Here's a screenshot of what I'm seeing in wireshark. I really appreciate the suggestions. Screenshot: http://dl.dropbox.com/u/4156401/Screenshot%20from%202012-05-23%2007%3A39%3A51.png pcap: http://dl.dropbox.com/u/4156401/trace3000.pcap Thanks David On Wed, May 23, 2012 at 7:41 AM, David Wessell <david at ringfree.biz> wrote:> Hi Jared & Kevin, > > Thanks for taking the time to answer my questions. I wonder if I could > just be reading the tcpdump incorrectly? I'm still seeing rtp streams (and > Jared, I have modified the dial string to remove the L).. > > Here's a screenshot of what I'm seeing in wireshark. I really appreciate > the suggestions. > > Thanks > David > > > > > On Mon, May 21, 2012 at 6:08 PM, Jared Geiger <jared at compuwizz.net> wrote: > > A2billing usually stays in the media path due to the dialstring > > parameters that it uses to cut a call off when the balance would reach > > $0. To get Asterisk to step out of the media path, I had to change > > dialcommand_param_sipiax_friend and dialcommand_param to |60|S(14400) > > which lets all calls go to 14400 seconds. The default uses the L > > parameter. You need to use the S parameter instead. However the S > > parameter doesn't like very large numbers in Asterisk 1.4 so I've just > > hard set mine. > > > > ~Jared > > > > On Mon, May 21, 2012 at 5:18 PM, Kevin P. Fleming <kpfleming at digium.com> > wrote: > >> On 05/21/2012 03:45 PM, David Wessell wrote: > >>> > >>> More specific on sip.conf > >>> > >>> In sip.conf I have a trunk specified for the SIP provider, and a trunk > >>> specified for the PBX itself. > >>> > >>> Do I need to specify directmedia=yes on both sides? > >> > >> > >> Yes, it has to be set on both peers involved in the bridged call. > >> > >> > >> -- > >> Kevin P. Fleming > >> Digium, Inc. | Director of Software Technologies > >> Jabber: kfleming at digium.com | SIP: kpfleming at digium.com | 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 > > > > -- > > _____________________________________________________________________ > > -- 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 >-- -- www.ringfree.biz 828-575-0030 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120523/28da98a9/attachment.htm>