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Fred Gaston

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Everything posted by Fred Gaston

  1. Thanks very much Hirosh: We have a different subnet on the 2nd NIC that is connected to the Intertex router which in turn is conncected straight to the internet. Should virtual ip address for openvpn be changed from in your sample config to one from within that subnet? Can you explain the 2 IP addressed in the config file? Thanks, Fred
  2. Kristan: I'm not sure we ever got sip multicast to work properly but attached works for us and gets the phones provisioned with pbxnsip. I'm sure there are several ways to do this, and maybe some are simpler. Hope some or all of this helps. Regards, Fred Provisioning_Snom_360_Phones_2.doc snom360.htm
  3. Can you speak to the setup please? My understanding of the ports on home firewall that need to be opened are: TCP: 5060-5061 UDP: 5060,49152-64512 Does TLS pass through these ports, does the firewall have to support TLS? I could only get connection by setting ;transport=TLS switch via port 5061 in outbound proxy field. Should it be different setup if connecting from outside the domain? Thanks
  4. Thank you very much for the clarification. On another note, I'm trying to hook-up Snom 370's from employee's homes. I would like to use VPN feature, but the Snom directions seem geared more to Linux setup & are difficult to follow. I'm able to connect to pbxnsip (without phone VPN) with Intertex SIP router connected that that server however still have to jump through all the SIP hoops at home & if home router doesn't support SIP, it connects but no audio. Anyway with or without phone VPN to beat the home firewall configuration (ICE, STUN?) that you know of, and if via phone VPN any advice on configuration? Thanks, Fred
  5. Your reply makes no sense to me. You stated in August: "Well, with the snom we are in the middle of finishing something really nice. The 7.2 version will support "buttons", where the PBX can take full control over the LED. it will also support XML-based directory, where the phones pull the address book on the fly from the PBX. Unfortunately, there is no usable 7.2 version available yet..." I am simply following up on your historical post. Why would you indicate there is no need for 7.2 when you claimed in August it was being worked on? Familiarizing yourself with all prior posts before responding would be appreciated.
  6. Any update please on timeframe for v.7.20? Thanks, Fred
  7. Valerio: Thanks for the update. Did you force registration of ocs number in pbxnsip for each user? Can you provide example of dialplan which would enable the call forking?Assume no luck on presence from phone, correct? Regards, Fred
  8. Per MS: Qualified IP-PBXs for Microsoft Office Communicator 2007 Our IP-PBX Partners are currently developing solutions for integrating Office Communications Server 2007 with Direct SIP, Dual Forking, and Dual Forking with Remote Call Control. This section will be updated to reflect IP-PBX and firmware combinations that have been independently qualified for a given configuration. More information coming soon Perhaps you can contact them to investigate what's necessary for integration & let us know if it's doable and something you'll persue. I also found 3rd party CSTA server that is fully compliant with OCS. Depending on cost, it might be easier to use than you guys incorporating CSTA, if it would take months of development (I got impression from your 1st reply it didn't look that hard). It is: http://www.unigone.com/en/products/TelServer/description Thanks, Fred
  9. Charl: Let me clarify. OCS supports call forking but to invoke the pbx needs to support CSTA (please see post below), which does not sound like it's coming anytime soon. I searched for 3rd party solutions and found none. Perhaps TAPI will work. My tech did say in the absence of a presence/sip server like Cisco employs, you'd need to set static routes for each user in OCS. Perhaps this combined with setting static registration in pbxnsip for OC uri & it might work... Keep me in the loop on what you uncover. Thx, Fred http://forum.pbxnsip.com/index.php?showtopic=350
  10. Can you please advise if there is current workaround with 3rd party solution and/or your timeframe for implementing.
  11. OCS client doesn't register with pbxnsip. Is it possible to force a manual registration or denote the SIP address of the OCS client in pbxnsip, to permit forking?
  12. Interesting you ask this. I posted this question & received following answer from Admin. "Forking calls was supported since version 1.0. Just register an extension twice and they will ring at the same time. I don't know why some people have so many problems with this trivial feature." So while pbxnsip supports forking, UM/OCS apparently does not. Per my OCS expert, OCS will not register with pbxnsip, as endpoint, & you can't have both simultaneously ring. We reasoned that if someone wants to dial a hard phone they dial extension; if they want to connect to another OC client they just choose name. Still exploring PBX Integration field, but he is saying this is used with Cisco to light up MWI lamp on phone. I am exploring SIP server that would broadcast presense of phones via the SIP url to OC. Is your understanding on any of the above different? Regards, Fred
  13. Per the 2.1 release notes there is new feature in trunk: Feature: When a trunk initiates a redirect, there was a problem that there was no user available that could be used for charging (and for the dial plan). This was e.g. a problem when using Microsoft Exchange. This problem is now solved by a new setting that explicitly tells the PBX what accounts to charge for such redirected calls. Can someone explain the use of this please and what extension to designate. Thx
  14. Valerio: Couple of questions: What extension did use for Assume that call comes from user? Are you using Exchange UM Auto Attendant, and if so do you route from your external gateway through pbxnsip Can you provide contact information so I can bounce my configuration off you? Thanks, Fred
  15. I have reviewed that document. Here's my scenario 1. I can dial UM prefix + UM AA = 7222 internally & get to UM AA just fine. 2. But when I have an incoming call via another trunk it can resolve a pbxnsip AA extension (500) but not 7222. 3. I imagine there must be a code that will redirect it via another trunk to Exchange & ext 222. Basically I want to send all incoming calls to 7222, regardless of the trunk, but the ITSP doesn't know what to do with 7222. Thanks
  16. What should I put in "Send Call to Extension" for an ITSP trunk to route all calls to Exchange UM auto-attendant extension? Thanks, Fred
  17. I have followed the directions above but all I get when I call to Audiocodes once connected is new dial-tone and one of the ports lights blue (handset offhook), which apparently indicates there is no active RTP session. Questions: 1. Is it necessary to complete Gateway Name under SIP Parameters, and if so should it also be IP of pbxnsip server? 2. Is it necessary (or useful) to change anything under Advanced Parameters? 3. Is it necessary to use user name & password for the trunk in pbxnsip & audiocodes proxy setup? 4. Is there a log in the Audiocodes to help troubleshoot? 5. I assume routing is unnecessary if using proxy? 6. In general how do I get it to speak pass the call to pbxnsip? 7. Any additional config suggestions would be a big help for a 114 or 118 FXO. Thanks, Fred
  18. Thanks. Can you comment on the Trunk to Trunk feature in 2.1 and how that would get around the need for adding numbers per Jan's post. Is there any configuration necessary for this feature, if OCS is sending in E.164 format? Thanks, Fred
  19. Jan: 1. Presume you are referring to v2.1x as next version from your July post? Have you confirmed dialing externally from OC client works in 2.1? 2. Can you elaborate please on how & where to setup (same) E.164 numbers in pbxnsip? Do you mean extensions or outside numbers, or both? Is this obviated by trunk to trunk in 2.1 & now unnecessary? 3. Did you have to synch dial plans in pbxnsip & OCS and can you provide examples? 4. Do you have integrated with Exchange UM and if so were you able to do without SP1, which MS doesn't advocate applying to production server. Thanks, Fred
  20. Will pbxnsip support Call Control and manage CSTA-SIP protocol?
  21. My understanding is pbxnsip is committed to eventually integrating with MS OCS 2007. In order to get the OC softphone and a desk phone to ring simultaneously, the pbx needs to support "forking" on inbound calls. This would require the ability to register two endpoints in pbxnsip: the desk phone and the OC client as the softphone. [http://forums.microsoft.com/Ocs2007/ShowPost.aspx?PostID=1954002&SiteID=57]. Is this a future possibility, as it's my understanding you can only register one device now? Thanks, Fred
  22. "you need to put the various *.ld files and the sip.ver files into the tftp directory, but *not* all these other template files (maybe the SoundPointIPWelcome.wav if you like)." Where is the documentation for this? Is it definitely working for http on latest 2.1? Where are these files you're referring to? Thanks, Fred
  23. Is there any update on the Sangoma PRI cards & an API that will work with pbxnsip? Thanks, Fred
  24. Is it necessary to put up an RTM server like Openfire or SER to get presence & IM working with pbxnsip & X-Lite? Put another way: will pbxnsip support without the need for additional software? Also has anyone been able to get video working with pbxnsip & X-Lite. It sort of works in 2.1 but then craps out. Thanks in advance, Fred
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