Hi Tony, I don't have a sipxbridge.xsd file. And a grep for relay in /etc/sipxpbx/ only produced four results in nattraversalrules.xml, none of which have anything to do with media.
If you no the exact entry to add to sipxbridge.xml I can give that a go later today. Regards, Andrew Radke On 02/12/2010, at 9:23 AM, Tony Graziano <[email protected]> wrote: > Can you see if you have the value in the sipxbridge.xsd file? > ============================ > Tony Graziano, Manager > Telephone: 434.984.8430 > Fax: 434.984.8431 > > Email: [email protected] > > LAN/Telephony/Security and Control Systems Helpdesk: > Telephone: 434.984.8426 > Fax: 434.984.8427 > > Helpdesk Contract Customers: > http://www.myitdepartment.net/gethelp/ > > ----- Original Message ----- > From: [email protected] > <[email protected]> > To: Discussion list for users of sipXecs software > <[email protected]> > Sent: Wed Dec 01 18:06:11 2010 > Subject: Re: [sipx-users] SipX 4.2 media relay > > Hi Tony, > > This has been asked in the past by other people but not by me. I read > through the archives and did indeed find this solution but there is no > reference to relay media in my sipxbridge.xml (sipx 4.2.1). As such I > assumed that this solution was for an older version of sipx and was loath to > play with it without more info. > > There was also another thread that indicated that "Enable NAT Traversal" and > "Server behind NAT" will effect the media relay behaviour so I thought that > this could have superseded the manual sipxbridge.xml modification. > > Is there a wiki document or the like that I have missed that describes > exactly what needs to be changed/added? > > Thanks, > Andrew Radke > > On 01/12/2010, at 10:04 PM, Tony Graziano <[email protected]> > wrote: > >> Manually edit sipxbridge.xml and set always relay media tag to "false" >> instead of true, then restart sipxbridge. >> >> This is considered experimental and should be tested thoroughly, because >> it >> may not work well with every user or itsp (I.e. Hairpinned calls, etc.). >> ============================ >> Tony Graziano, Manager >> Telephone: 434.984.8430 >> Fax: 434.984.8431 >> >> Email: [email protected] >> >> LAN/Telephony/Security and Control Systems Helpdesk: >> Telephone: 434.984.8426 >> Fax: 434.984.8427 >> >> Helpdesk Contract Customers: >> http://www.myitdepartment.net/gethelp/ >> >> ----- Original Message ----- >> From: Tony Graziano <[email protected]> >> To: [email protected] <[email protected]> >> Sent: Wed Dec 01 06:24:56 2010 >> Subject: Re: [sipx-users] SipX 4.2 media relay >> >> There is/was a way to achive this. He has been given the instructions on >> how >> to do this before but he obviously did not make notes of it. >> >> I will look it up when I get in front of my system. >> ============================ >> Tony Graziano, Manager >> Telephone: 434.984.8430 >> Fax: 434.984.8431 >> >> Email: [email protected] >> >> LAN/Telephony/Security and Control Systems Helpdesk: >> Telephone: 434.984.8426 >> Fax: 434.984.8427 >> >> Helpdesk Contract Customers: >> http://www.myitdepartment.net/gethelp/ >> >> ----- Original Message ----- >> From: [email protected] >> <[email protected]> >> To: 'Discussion list for users of sipXecs software' >> <[email protected]> >> Sent: Wed Dec 01 05:54:33 2010 >> Subject: Re: [sipx-users] SipX 4.2 media relay >> >> I see your problem now. >> I dont think that it is possible to turn off media relay when calling >> through sipxbridge. >> Let somebody correct me if i'm mistaken. >> >> I see the only thing you could play with: you could use >> separate/independent >> FS profile as your gateway to itsp. >> That is you create independent FS instance (it may run on the same machine >> as sipx, but listen on some free port). >> Configure in sipx unmanaged gw, pointing to this FS instance. >> FS is capable of doing "proxy authentication" and there is "proxy media" >> option, that controls media stream. >> See >> http://wiki.sipfoundry.org/display/sipXecs/Custom+FreeSWITCH+programming >> as starting point. >> >> Sounds complicated and requires some significant preliminary efforts, but >> may work for you... >> >> Rgds, >> Nikolay. >> >> >> _____ >> >> From: [email protected] >> [mailto:[email protected]] On Behalf Of Andrew Radke >> Sent: Wednesday, December 01, 2010 1:09 PM >> To: Discussion list for users of sipXecs software >> Subject: Re: [sipx-users] SipX 4.2 media relay >> >> >> The main reason is that we have a couple of phones at remote locations >> connected via VPN. Since we are in a rural area with slow ADSL1 as our >> only >> Internet connection option, having the audio from these offices come in >> through the VPN and then back out to the ITSP is less than ideal. With QOS >> on the links quality is okay but latency is terrible and our bandwidth is >> very limited. >> >> Regards, >> Andrew Radke >> >> On 01/12/2010, at 7:40 PM, "Nikolay Kondratyev" <[email protected]> wrote: >> >> >> >> "Proxy authentication" with itsp for outgoing calls is also performed by >> sipxbridge. >> That is, it will not work via 'unmanaged gw'. >> By the way, why don't you like media relay? Is there any real problem >> because of that? >> Just in case it concerns you: >> If there is no NAT between you and ITSP you can configure your siptrunk gw >> to use internal ip address (uncheck "use public address for call setup" >> checkbox, which is checked by default). >> But media relay will be used anyway. >> Rgds, >> Nikolay. >> >> >> >> _____ >> >> From: [email protected] >> [mailto:[email protected]] On Behalf Of Andrew Radke >> Sent: Wednesday, December 01, 2010 12:21 PM >> To: Discussion list for users of sipXecs software >> Subject: Re: [sipx-users] SipX 4.2 media relay >> >> >> At this stage we don't need to register since we don't accept any incoming >> calls from our ITSP, but we do need to authenticate outgoing calls via >> them. >> >> Regards, >> Andrew Radke >> >> On 01/12/2010, at 7:00 PM, "Nikolay Kondratyev" < <mailto:[email protected]> >> [email protected]> wrote: >> >> >> >> You mean register to ITSP? >> No, it can't (afaik). >> Rgds, >> Nikolay. >> >> >> _____ >> >> From: <mailto:[email protected]> >> [email protected] >> [mailto:[email protected]] On Behalf Of Andrew Radke >> Sent: Wednesday, December 01, 2010 11:23 AM >> To: Discussion list for users of sipXecs software >> Subject: Re: [sipx-users] SipX 4.2 media relay >> >> >> Can an unmanaged gateway authenticate with an ITSP though? Sorry, I didn't >> even look in 4.2 and I've left work for the day now. >> >> Regards, >> Andrew Radke >> >> On 01/12/2010, at 4:37 PM, "Nikolay Kondratyev" < <mailto:[email protected]> >> <mailto:[email protected]> [email protected]> wrote: >> >> >> >> Andrew, >> do you use "sip trunk" to connect to ITSP? >> my understanding is that, when you use siptrunk (sipxbridge), you get the >> following: >> 1. Refer messages are converted to re-Invite. >> 2. media relay >> 3. shortened headers (not so many via's). >> And you can not make sipxbriddge not to do it. >> If you really want your media streams go directly from the phones to ITSP, >> just use "unmanaged" gaeway. >> Rgds, >> Nikolay. >> >> >> >> _____ >> >> From: <mailto:[email protected]> >> <mailto:[email protected]> >> [email protected] >> [mailto:[email protected]] On Behalf Of Andrew Radke >> Sent: Wednesday, December 01, 2010 9:07 AM >> To: <mailto:[email protected]> >> <mailto:[email protected]> >> <mailto:[email protected]> [email protected] >> Subject: [sipx-users] SipX 4.2 media relay >> >> >> Hi all, >> >> Something that I have noticed in 4.2 is that calls made via an ITSP have >> their audio routed via the SipX media relay. I have been looking at how to >> turn this off by reading previous discussions on this list but it seems >> that >> since SipX notices that there is a NATing firewall between it and the ITSP >> it will relay the media anyway. >> >> Under Internet Calling I have the correct intranet subnets defined and >> have >> tried all combinations of "Enable NAT Traversal" and "Server behind NAT". >> The notes on the right of this screen states: "Whenever it is detected >> that >> the two endpoints that are starting a media session are separated by one >> or >> more NATs, the server will insert its Media Relay into the media path." >> which seems to define my issue. >> >> Is it possible to disable this behaviour? >> >> Regards, >> >> >> Andrew Radke >> Yuruga Nursery Pty Ltd >> Clonal Solutions Australia Pty Ltd >> PO Box 220 >> Walkamin Qld 4872 >> Phone: (07) 4093 3826 >> Fax: (07) 4093 3869 >> Email: <mailto:[email protected]> [email protected] >> Web: <http://www.yuruga.com.au/> www.yuruga.com.au >> >> >> _______________________________________________ >> sipx-users mailing list >> <mailto:[email protected]> >> <mailto:[email protected]> [email protected] >> List Archive: <http://list.sipfoundry.org/archive/sipx-users/> >> <http://list.sipfoundry.org/archive/sipx-users/> >> <http://list.sipfoundry.org/archive/sipx-users/> >> http://list.sipfoundry.org/archive/sipx-users/ >> >> _______________________________________________ >> sipx-users mailing list >> <mailto:[email protected]> [email protected] >> List Archive: <http://list.sipfoundry.org/archive/sipx-users/> >> <http://list.sipfoundry.org/archive/sipx-users/> >> http://list.sipfoundry.org/archive/sipx-users/ >> >> _______________________________________________ >> sipx-users mailing list >> [email protected] >> List Archive: <http://list.sipfoundry.org/archive/sipx-users/> >> http://list.sipfoundry.org/archive/sipx-users/ >> _______________________________________________ >> sipx-users mailing list >> [email protected] >> List Archive: http://list.sipfoundry.org/archive/sipx-users/ >> > _______________________________________________ > sipx-users mailing list > [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-users/ > _______________________________________________ > sipx-users mailing list > [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-users/ > _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users/
