Thx for the lowdown
From: Tony Graziano [mailto:tgrazi...@myitdepartment.net] Sent: Friday, October 01, 2010 3:09 AM To: Ujjval Karihaloo Cc: Discussion list for users of sipXecs software Subject: Re: [sipx-users] SIP Trunk --> AA--> SIP trunk call flow I'm not sure "any" build will fix this at this time. I have a feeling this is related to failing "attndened transfers" when the media server is involved. Ultimately, this will be fixed post 4.3 when the specific work is done in sips to fix "how" these two calls get bridged. sipx also needs to be running FS1.07. At least that's my theory. On Tue, Sep 28, 2010 at 1:54 PM, Ujjval Karihaloo <ujj...@simplesignal.com<mailto:ujj...@simplesignal.com>> wrote: I probably need to be on a higher load that: sipXconfig (4.2.1-018930 2010-06-04T15:25:27 build34) for this setting to take effect..? From: sipx-users-boun...@list.sipfoundry.org<mailto:sipx-users-boun...@list.sipfoundry.org> [mailto:sipx-users-boun...@list.sipfoundry.org<mailto:sipx-users-boun...@list.sipfoundry.org>] On Behalf Of Ujjval Karihaloo Sent: Tuesday, September 28, 2010 11:48 AM To: Discussion list for users of sipXecs software; Tony Graziano Subject: Re: [sipx-users] SIP Trunk --> AA--> SIP trunk call flow I added the below entry and it did not work From: sipx-users-boun...@list.sipfoundry.org<mailto:sipx-users-boun...@list.sipfoundry.org> [mailto:sipx-users-boun...@list.sipfoundry.org<mailto:sipx-users-boun...@list.sipfoundry.org>] On Behalf Of Ujjval Karihaloo Sent: Tuesday, September 28, 2010 11:07 AM To: Tony Graziano Cc: sipx-users@list.sipfoundry.org<mailto:sipx-users@list.sipfoundry.org> Subject: Re: [sipx-users] SIP Trunk --> AA--> SIP trunk call flow I see, I can try it..Will let everyone know From: Tony Graziano [mailto:tgrazi...@myitdepartment.net<mailto:tgrazi...@myitdepartment.net>] Sent: Tuesday, September 28, 2010 10:34 AM To: Ujjval Karihaloo Cc: mar...@ezuce.com<mailto:mar...@ezuce.com>; dhub...@ezuce.com<mailto:dhub...@ezuce.com>; sipx-users@list.sipfoundry.org<mailto:sipx-users@list.sipfoundry.org> Subject: Re: [sipx-users] SIP Trunk --> AA--> SIP trunk call flow It's not implemented, as the JIRA says. It's experimental, and I'm not sure if it "should" be exposed, as it could do more harm than good by exposing it. You would do this by manually editing the file and restarting sipxbridge service. On Tue, Sep 28, 2010 at 12:16 PM, Ujjval Karihaloo <ujj...@simplesignal.com<mailto:ujj...@simplesignal.com>> wrote: I do not see the check box in the ITSP account mentioned here: http://track.sipfoundry.org/browse/XX-7461 starting from r17753, hairpinned media relays (i.e. call originating from ITSP side relayed back to ITSP ) are removed after call setup. This makes our product better suited for cloud deployments where bandwidth is not for free. TO enable this behavior a new tag is defined and must be present in the <itsp-account> element for the ITSP in sipxbridge.xml. Setting <always-relay-media>false<...> enables this behavior. (The default is to set it to true) i.e. for example, in sipxbridge.xml : <itsp-account> <itsp-proxy-domain>ot.bandwidth.com<http://ot.bandwidth.com></itsp-proxy-domain> <register-on-initialization>false</register-on-initialization> <use-global-addressing>true</use-global-addressing> <strip-private-headers>false</strip-private-headers> <default-asserted-identity>true</default-asserted-identity> <is-user-phone>true</is-user-phone> <loose-route-invite>true</loose-route-invite> <registration-interval>600</registration-interval> <sip-session-timer-interval-seconds>1800</sip-session-timer-interval-seconds> <sip-keepalive-method>CR-LF</sip-keepalive-method> <rtp-keepalive-method>NONE</rtp-keepalive-method> <always-relay-media>false</always-relay-media> </itsp-account> Ujjval Karihaloo VP Voice Engineering IP Phone: +13032428610 E-Fax: +17202391690 SimpleSignal Inc. 88 Inverness Circle East Suite K105 Englewood, CO 80112 [cid:image001.jpg@01CB63CD.A286A650]<http://www.simplesignal.com/> From: Tony Graziano [mailto:tgrazi...@myitdepartment.net<mailto:tgrazi...@myitdepartment.net>] Sent: Friday, September 24, 2010 2:27 PM To: Ujjval Karihaloo Cc: mar...@ezuce.com<mailto:mar...@ezuce.com>; dhub...@ezuce.com<mailto:dhub...@ezuce.com>; sipx-users@list.sipfoundry.org<mailto:sipx-users@list.sipfoundry.org> Subject: Re: [sipx-users] SIP Trunk --> AA--> SIP trunk call flow So I am not sure this is "exactly" supported at this time... http://track.sipfoundry.org/browse/XX-7362 as a result you might consider trying this: http://track.sipfoundry.org/browse/XX-7461 On Fri, Sep 24, 2010 at 4:04 PM, Ujjval Karihaloo <ujj...@simplesignal.com<mailto:ujj...@simplesignal.com>> wrote: I found out if I disable NAT traversal (for Endpoints) this call flows works and I get 2 way a audio. This would mean I can forward call off of an AA and get 2 way audio, but endpoints behind NAT wont be able to register to SIPx... Any ideas? -----Original Message----- From: Tony Graziano [mailto:tgrazi...@myitdepartment.net<mailto:tgrazi...@myitdepartment.net>] Sent: Sunday, August 22, 2010 7:53 AM To: mar...@ezuce.com<mailto:mar...@ezuce.com>; dhub...@ezuce.com<mailto:dhub...@ezuce.com>; Ujjval Karihaloo Cc: sipx-users@list.sipfoundry.org<mailto:sipx-users@list.sipfoundry.org> Subject: Re: [sipx-users] SIP Trunk --> AA--> SIP trunk call flow If it is available, I assume one could edit the xml file directly and restart the service (no config changes from sipxconfig,so projection would not happen) and try it? Or edit the vm "to on" by default? ============================ Tony Graziano, Manager Telephone: 434.984.8430 Fax: 434.984.8431 Email: tgrazi...@myitdepartment.net<mailto:tgrazi...@myitdepartment.net> 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: sipx-users-boun...@list.sipfoundry.org<mailto:sipx-users-boun...@list.sipfoundry.org> <sipx-users-boun...@list.sipfoundry.org<mailto:sipx-users-boun...@list.sipfoundry.org>> To: 'Douglas Hubler' <dhub...@ezuce.com<mailto:dhub...@ezuce.com>>; 'Ujjval Karihaloo' <ujj...@simplesignal.com<mailto:ujj...@simplesignal.com>> Cc: sipx-users@list.sipfoundry.org<mailto:sipx-users@list.sipfoundry.org> <sipx-users@list.sipfoundry.org<mailto:sipx-users@list.sipfoundry.org>> Sent: Sun Aug 22 09:37:36 2010 Subject: Re: [sipx-users] SIP Trunk --> AA--> SIP trunk call flow > > On Fri, Aug 20, 2010 at 3:52 PM, Ujjval Karihaloo > <ujj...@simplesignal.com<mailto:ujj...@simplesignal.com>> wrote: > > Guys: > > > > > > Looking for some help on this....has anyone tried thisL I get no Audio > in > > each direction. Tony tried it with 2 different ITSPs and it works...but > I > > cannot get it to work with only one ITSP that I have to test. > > > > Any other suggestions from group as to where to look for a solution > to this > > issue: > > Sounds like this issue that i'm still waiting for a toplink test > account to test with > http://track.sipfoundry.org/browse/XX-8663 > > Because the call seems to be successful, it's just that the audio is > missing, I'd look closer at the RTP message in both the SDP portion of > the SIP messages the wireshark of RTP source and destination. > > In a successful call, i'm not sure if the final RTP path should flow > thru sipxbridge at all, in theory in shouldn't have to AFAIK. I think this question relates to: http://track.sipfoundry.org/browse/XX-7362. While this was implemented, the setting in sipXconfig has not: http://track.sipfoundry.org/browse/XX-7461 The call flows through sipXrelay, which is part of the proxy, and not sipXbridge. --martin _______________________________________________ sipx-users mailing list sipx-users@list.sipfoundry.org<mailto:sipx-users@list.sipfoundry.org> List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- ====================== Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net<mailto:tgrazi...@voice.myitdepartment.net> Fax: 434.984.8431 Email: tgrazi...@myitdepartment.net<mailto:tgrazi...@myitdepartment.net> LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net<mailto:helpd...@voice.myitdepartment.net> Fax: 434.984.8427 Helpdesk Contract Customers: http://www.myitdepartment.net/gethelp/ Why do mathematicians always confuse Halloween and Christmas? Because 31 Oct = 25 Dec. -- ====================== Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net<mailto:tgrazi...@voice.myitdepartment.net> Fax: 434.984.8431 Email: tgrazi...@myitdepartment.net<mailto:tgrazi...@myitdepartment.net> LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net<mailto:helpd...@voice.myitdepartment.net> Fax: 434.984.8427 Helpdesk Contract Customers: http://www.myitdepartment.net/gethelp/ Why do mathematicians always confuse Halloween and Christmas? Because 31 Oct = 25 Dec. -- ====================== Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net<mailto:tgrazi...@voice.myitdepartment.net> Fax: 434.984.8431 Email: tgrazi...@myitdepartment.net<mailto:tgrazi...@myitdepartment.net> LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net<mailto:helpd...@voice.myitdepartment.net> Fax: 434.984.8427 Helpdesk Contract Customers: http://www.myitdepartment.net/gethelp/ Why do mathematicians always confuse Halloween and Christmas? Because 31 Oct = 25 Dec.
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