I have a hunch that you may need to look at the source files.
From: Irena Dolovčak [mailto:irena.dolov...@gmail.com] Sent: Monday, May 31, 2010 10:18 AM To: Tony Graziano Cc: Paul Herron; sipx-users@list.sipfoundry.org Subject: Re: [sipx-users] Remote office problem hmm.. so is there a way to turn off the <always-relay-media> ? or was that just an option before? 2010/5/31 Tony Graziano <tgrazi...@myitdepartment.net> All the xsd files are in: /usr/share/sipxecs/schema However, there is no sipxbridge.xsd file there. 2010/5/31 Irena Dolovčak <irena.dolov...@gmail.com>: > Hi Paul, thanks for your answer. > > I have read the post that you have send me, but I cannot find the files.. > Where are they? at least the sipxbridge.xsd I can't find. > And in the file sipxbridge.xml, I can't see anything about > <always-relay-media>.. > Want somebody give me a hint? > > > It is called <always-relay-media> Look in the sipxbridge.xsd file for > a description. By default that flag is true and it is hidden. You can > set it to false by editing sipxbridge.xml and if you do so, then for > hairpinned forwarded or blind transferred calls, the media relay will > be removed from the media path AFTER transfer. > > 2010/5/28 <p...@sagecraftsmen.com> >> >> See this post: >> >> >> >> >> http://forum.sipfoundry.org/index.php?t=msg&goto=44870&S=dbad4bee44cb3fe a8a13cda1d1fea305&srch=sipxbridge+media+relay#msg_44870 >> >> >> >> From: Paul Herron >> Sent: Friday, May 28, 2010 12:01 PM >> To: Irena Dolovčak; Tony Graziano >> >> Cc: sipx-users@list.sipfoundry.org >> Subject: RE: [sipx-users] Remote office problem >> >> >> >> I'm sure someone will correct me here if I am wrong but, it is my >> understanding that the scenario described with the ITSP below isn't possible >> at this time. I believe that three issues are at work: NAT translation, >> authentication, and ITSP support for re-invite (this was discussed in the >> past, search the archives) >> >> >> >> The second scenario you describe is certainly possible if all parties >> (both remote users and SipX) have public IPs. If you're using NAT, your >> only options are the ones Tony described. >> >> >> >> From: Irena Dolovčak [mailto:irena.dolov...@gmail.com] >> Sent: Friday, May 28, 2010 6:57 AM >> To: Tony Graziano >> Cc: sipx-users@list.sipfoundry.org >> Subject: Re: [sipx-users] Remote office problem >> >> >> >> thanks, >> I'll try it out an let you know. >> >> It matters if you have (theoretical speaking) the remote user trying to >> call an outside number, then the media goes like this : >> >> remote user ---- internet ----- sipx ------ internet ------ voip provider >> >> and i want the media to go directly to provider (because of the call >> quality) >> >> and in case where are two remote users: >> >> remote user ------ internet ----- sipx ------- remote user >> >> On Fri, May 28, 2010 at 12:44 PM, Tony Graziano >> <tgrazi...@myitdepartment.net> wrote: >> >> The only thing you can try is aggressive mode in sip to try to see if the >> media will establish as peer-to-peer. >> >> Remote user utilizes media relay, has 2 modes (conservative & aggressive). >> It requires nat be setup at your router properly for either to work. >> >> I don't understand why it matters, if the phone on one leg of the call is >> beHind sipx, it is using the same bandwidth to the remote user whether >> anchored or not. >> >> ============================ >> Tony Graziano, Manager >> Telephone: 434.984.8430 >> Fax: 434.984.8431 >> >> Email: 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: Irena Dolovčak <irena.dolov...@gmail.com> >> To: Tony Graziano <tgrazi...@myitdepartment.net> >> >> Cc: sipx-users@list.sipfoundry.org <sipx-users@list.sipfoundry.org> >> Sent: Fri May 28 06:35:31 2010 >> Subject: Re: [sipx-users] Remote office problem >> >> ok ok.. i think we don't understand each other completely.. >> >> I understand that 3cx and sipx are different. I have put 3cx just as an >> example to describe to you what I want to do.. I don't even know if that >> can >> be done with sipx. That's what I'm trying to find out. And that's why I >> have >> asked you all that stuff. >> >> I know how to configure remote user, and I tried the aggressive mode.. but >> that isn't just that what I want to accomplish.. >> I don't want the sipx to act as a B2BUA.. >> >> how is it exactly done in sipx? can i get a step by step report what is >> happening in sipx and what sipx servers and services are involved? I'm >> just >> trying to understand what is happening and what can be done and what >> can't.. >> >> >> The only thing I am trying to accomplish is to get the phones to make >> calls >> peer-to-peer; the sip signaling should still be servers job.. >> >> >> >> On Thu, May 27, 2010 at 11:49 AM, Tony Graziano < >> tgrazi...@myitdepartment.net> wrote: >> >> > Consider setting up for remote users (assuming sip trunking is not >> > needed). >> > >> > System>Internet Calling>Nat Traversal. >> > >> > Server>(yours)>Media Relay>advanced)>aggressive. >> > >> > Server>(yours)>NAT>setting the static public IP >> > >> > >> > >> > On Thu, May 27, 2010 at 5:40 AM, Tony Graziano < >> > tgrazi...@myitdepartment.net> wrote: >> > >> >> You miss the point in that the 3cx handles nat traversal in a very >> >> different way than sipx. >> >> >> >> With sipx, that same procedure requires use the media relay, which you >> >> might adjust to be in "aggressive" mode, but there is not guarantee it >> >> would >> >> work in every instance, hence the default mode is "conservative". >> >> >> >> >> >> On Thu, May 27, 2010 at 3:49 AM, Irena Dolovčak >> >> <irena.dolov...@gmail.com >> >> > wrote: >> >> >> >>> Hi, >> >>> >> >>> I don't think anchoring is the only way.. >> >>> >> >>> and there are more ways to bypass NAT, not just the SBC that handles >> >>> it. >> >>> >> >>> >> >>> I have done this with 3CX PBX and mikrotik routers on both ends. >> >>> >> >>> I tried to reproduce the same thing in 3CX PBX, and here is the >> >>> picture >> >>> with steps. >> >>> I'm not sure if all is 100% OK, I tried to attach the important >> >>> things. >> >>> >> >>> I want to do the same thing in sipx to.. what are the differences >> >>> between >> >>> them? (the 3cx and sipx) in the structure of the pbx.. >> >>> >> >>> PhoneA - user >> >>> PhoneB - user2 >> >>> >> >>> 1 -src. Address/port 192.168.0.245:5062 dst address/port >> >>> 82.210.15.45:5060 >> >>> phone A sends an INVITE message >> >>> (sip:u...@82.210.15.45:5062 to us...@82.210.15.45) >> >>> >> >>> 2 - the NAT translates the src address to his public address >> >>> 89.201.135.15, the src port stays the same >> >>> -The router sends the packet to the internet >> >>> The router gives his public ip address to eht phone. No STUN is >> >>> involved. >> >>> >> >>> 3 - the packet comes to the second router; he sees the dst. Address >> >>> 82.210.15.45 and the dst. port 5060 - he knows he must forward the >> >>> packet >> >>> with the dst port 5060 to the internal IP address 192.168.1.15 >> >>> >> >>> 4 - the 3cx PBX gets the packet on port 5060 from the ip address >> >>> 89.201.135.15 >> >>> It examines the packet and sees that user is trying to call user2 >> >>> >> >>> 5 - The PBX send the user an information packet telling him that it is >> >>> trying to contact user2; >> >>> The message goes back to dst address 89.201.135.15:5062 >> >>> >> >>> 6 - the PBX sends an INVITE message to user2 >> >>> (sip:u...@82.210.15.45:5062 to us...@82.210.15.45) where the PBX tells >> >>> user2 how to contact user in SDP message which is sends together with >> >>> the >> >>> SIP message >> >>> In the SDP message, user has send information on which port should the >> >>> media packages be send to. The PBX forwarded the message to user2 >> >>> The message goes from src address 192.168.1.15 on port 5060 to dst >> >>> address/port 192.168.1.20:5060 >> >>> >> >>> 7 - user2 starts to ring and sends a RINGING message back to the PBX >> >>> so >> >>> it knows that he is ringing; >> >>> It also sends the SDP information where on which port to send the >> >>> media >> >>> packets >> >>> Src address/port 192.168.1.20:5060 dst address/port 192.168.1.15:5060 >> >>> >> >>> 8 - the PBX gets the message from user2 and forwards it back to user >> >>> so >> >>> he allso knows where to send media packets; it also sends the ring >> >>> alert >> >>> to >> >>> user so the user knows user2 is ringing >> >>> The packet is send to the router with Src address/port >> >> >>> 192.168.1.15:5060dst address/port >> >> >>> 89.201.135.15:5062 >> >>> >> >>> 9 - the router gets the packet form the PBX and sees the dst >> >>> address/port >> >>> 89.201.135.15:5062; to be able to send the packet, he traverses the >> >>> src >> >>> address 192.168.1.15 to 82.210.15.45 and sends it to the internet >> >>> >> >>> 10 - the second router gets the message now with src address >> >>> 82.210.15.45:5060 and dst address 89.201.135.15:5062; he knows he has >> >>> to >> >>> forward the message back to private ip address 192.168.0.245:5062 >> >>> >> >>> 11 - when the user2 answers the call, the phone sends an OK message >> >>> that >> >>> it has picked up the phone. (The process is the same as at step 10) >> >>> >> >>> 12 - the phoneB sends the RTP packet direckt to the phoneA as it has >> >>> all >> >>> necessary information as the Ip address and port to which he should >> >>> send >> >>> it. >> >>> Src address: 192.168.1.20:9000 dst address 89.201.135.15:11000 >> >>> the router gets the packet, translates the private IP to 82.210.15.45 >> >>> and send it to 89.201.135.15:11000 >> >>> The second router gets the packet with src address/port >> >>> 82.210.15.45:9000 and dst address/port 89.201.135.15:11000 and it >> >>> knows >> >>> that it must forward it to the private ip address 192.168.0.245:11000 >> >>> >> >>> 13 - when user hang up, the phone sends a BYE message to the PBX to >> >>> alert >> >>> user2 >> >>> Src address/port 192.168.0.245:5062 dst address/port 82.210.15.45:5060 >> >>> >> >>> 14 - the router gets the message, it outs a new src address >> >>> (89.201.135.15) and send it to the dst address 82.210.15.45 >> >>> >> >>> 15 - the router gets the packet with dst address 82.210.15.45:5060 and >> >>> it forwards it to internal ip 192.168.1.15 becouse it is tols to send >> >>> the >> >>> message with dst port 5060 to that ip address >> >>> >> >>> 16 - the PBX gets the message, and sends the message to user2 on ip >> >>> address 192.168.1.20 >> >>> >> >>> 17- user2 sends an ACK message to the PBX. The call ends >> >>> >> >>> >> >>> >> >>> >> >>> On Fri, May 21, 2010 at 1:55 PM, Tony Graziano < >> >>> tgrazi...@myitdepartment.net> wrote: >> >>> >> >>>> Yes, vpn the remote site and it will take the media peer-to-peer. >> >>>> >> >>>> You seem to be stuck on the fact the sipx must anchor the media and >> >>>> fpr >> >>>> whatever reason don't want to do that. It ONLY ANCHORS the media for >> >>>> the >> >>>> calls that go through NAT, it won't anchor the media for you >> >>>> lan-to-lan >> >>>> calls. >> >>>> >> >>>> In SIP, anytime you cross NAT something needs to anchor the media. >> >>>> That >> >>>> device is called a SBC. >> >>>> >> >>>> I think if you want it to work some other way using a SIP (not sipx) >> >>>> based >> >>>> system, you need to invent your own technology. >> >>>> >> >>>> Good luck. >> >>>> ============================ >> >>>> Tony Graziano, Manager >> >>>> Telephone: 434.984.8430 >> >>>> Fax: 434.984.8431 >> >>>> >> >>>> Email: 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: Irena Dolovčak <irena.dolov...@gmail.com> >> >>>> To: Tony Graziano <tgrazi...@myitdepartment.net> >> >>>> Sent: Fri May 21 07:40:57 2010 >> >>>> Subject: Re: [sipx-users] Remote office problem >> >>>> >> >>>> I'm not sure am I following you. >> >>>> >> >>>> Are you saying that the sipx can only be configured to anchor the >> >>>> media >> >>>> (as >> >>>> a B2BUA)? Is there no other solution? >> >>>> I understand that the sipxbridge anchor the media, that's why I >> >>>> removed >> >>>> it. >> >>>> >> >>>> Is there any solution that I could use to make the remote worker to >> >>>> connect >> >>>> in a by-pass mode and not trunked? >> >>>> >> >>>> >> >>>> > >> >>>> > I thbik it you have remote workers enabled and server behind nat >> >>>> > configured >> >>>> > "it should work". Right now sipx is being told to route to ports it >> >>>> > is >> >>>> not >> >>>> > employing because the sipxbridge component is not being used not is >> >>>> the >> >>>> > remote workers aspect. If you support either, you define an SBC, >> >>>> > you >> >>>> have >> >>>> > none. That's what sipxbridge does. Ingate makes a nice one too. >> >>>> > >> >>>> > >> >>>> What exactly do you mean by "it should work"? >> >>>> >> >>>> -- >> >>>> Irena Dolovčak >> >>>> >> >>> >> >>> >> >>> >> >>> -- >> >>> Irena Dolovčak >> >>> >> >> >> >> >> >> >> >> -- >> >> ====================== >> >> Tony Graziano, Manager >> >> Telephone: 434.984.8430 >> >> sip: tgrazi...@voice.myitdepartment.net >> >> Fax: 434.984.8431 >> >> >> >> Email: tgrazi...@myitdepartment.net >> >> >> >> LAN/Telephony/Security and Control Systems Helpdesk: >> >> Telephone: 434.984.8426 >> >> sip: 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 >> > Fax: 434.984.8431 >> > >> > Email: tgrazi...@myitdepartment.net >> > >> > LAN/Telephony/Security and Control Systems Helpdesk: >> > Telephone: 434.984.8426 >> > sip: 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. >> > >> > >> >> >> -- >> Irena Dolovčak >> >> >> -- >> Irena Dolovčak > > > -- > Irena Dolovčak > > _______________________________________________ > sipx-users mailing list sipx-users@list.sipfoundry.org > List Archive: http://list.sipfoundry.org/archive/sipx-users > Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users > sipXecs IP PBX -- http://www.sipfoundry.org/ > -- ====================== Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net Fax: 434.984.8431 Email: tgrazi...@myitdepartment.net LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: 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. -- Irena Dolovčak
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