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Fred Gaston

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Posts 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. Hi all,


    Regarding the firmware upgrade options in the pnp for the snoms, what's the process for this? I have 10 360's on 6.5.8 I'd like to auto provision on a brand new PBX and thought great, I'll just let the PBX do it. Nope, when the phone subscribes to the sip mutlicast network it gets a user agent not supported back. Ok I thought, fair enough, going from 6->7 is probably a bit much to hope for, so upgraded the phone to 7.1.6 manually, reset it... and I can see it talks to the PBX via SSL, but not what happens. And the phone doesn't get upgraded any further, and it doesn't get auto-provisioned, despite me setting the MAC in the registration tab.


    What have I done wrong? ;)


    Also, I wanted to put 7.1.24 on, so changed the pnp config setting for the 360 in the admin menu to the 7.1.24 file, and put the firmware file in the tftp folder. Is this the right place?







    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



  3. VPN on a phone is a difficult topic... It is very useful if your SIP infrastructure does not support application layer security.


    The good thing about pbxnsip is that TLS and SRTP already keep your voice pretty private. I think it is much easier to go this way.

    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. Okay, sorry if the response was not very clear. The 7.2 version had a feature called buttons (see http://wiki.pbxnsip.com/index.php/Assigning_Buttons), and because 7.2 is still not out snom put the support for buttons also in version 7.1. Therefore, there is no more need to wait until 7.2 is out. 7.1 is also fine now.

    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. At the moment I don't see a need for that - 7.1.x seems to do everything that we can dream of...

    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. Version 7.1.23 also does that, and this version is available as beta from the snom web server. If you are using then PnP should be working smoothly.


    For example, you find the snom firmware here (it is not so easy to find on the snom Wiki):







    If you need to upgrade from version 6, see the descriptions on http://wiki.pbxnsip.com/index.php/Snom.

    Any update please on timeframe for v.7.20? Thanks, Fred

  7. if you assign to the pbxnsip extension the same number as in ocs they will ring simultaneously.


    I have the following config:


    PSTN <--> PBXNSIP <---> Mediation <---> OCS <--> Exchange UM


    it works if you have DID and also if you dont (tried both)......if you dont have DID just let a pbxnsip AA answer and the route calls based on extesnions in ocs create a fake E164 numbering scheme and create dialplan accordingly


    if your ocs number is +14255454300 then you ext is 300 so use 300 as extension number in pbxnsip and put as alias your E164 number plus another fake ext. like 400.


    if you play with the dialplans in ocs and pbxnsip you will get simultaneous ring on incoming calls from pstn and will be able to call the pbxnsip extensions by using the fake ext #.


    the only problem I have is when calling from a OC client to another OC client I cannot get simultaneous ring of the corresponding pbxnsip ext.


    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. I guess we have to look at OCS in more detail. Just adding a feature here and there is not a good strategy. CSTA is not trivial, and it will take months to get it stable. Therefore I would not count on it. TAPI could be a workaround, there is a lot of TAPI stuff available. But I have no overview if there is soemthing that could convert CSTA into TAPI!

    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. Mm this was not the answer I hoping for.

    OCS is supporting forking that is the hole purpose of the Server URI with you can set on a per user bases.

    Ill wil try to find a solution and if I find something ill post IT here.

    Thanks for the help so far


    Regards, 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



  10. 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.


    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?

  11. There is no connection problem I can call someone from the PBXnSIP to OCS and from OCS to PBXnSIP.

    But what I want is the when someone calls extension 900 (example) the normal Voip Phone is ringing and the OCS client with extension 900

    Ill believe this is known as forking but how can I do that.

    I?ll know I?ll have to set the Server URI of the user but what should ill type there?

    my address is pbxnsipforum<at>spam.cpels.com

    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

  12. 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

  13. Hi,

    in the ocs trunk check that "Assume that call comes from user" has a valid extension, in this way the pbx knows who to charge for the call.


    If you have problems with ocs let me know I have integrated it in full with pbxnsip and I very happy with it.


    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

  14. Did you see the stuff on the Wiki? http://wiki.pbxnsip.com/index.php/Microsoft_Exchange Did it touch that topic.


    2.1 has a new setting called "Assume that call comes from user" that is not covered by the Wiki page. It tells the PBX what account to charge if Exchange wants to initiate an outbound call from the trunk.


    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.


  15. The first step is to configure the endpoints to make sure that they are enabled so you can put any number in there or use the defaults. I usually just set them to 401 port 1, 402 port 2, 403 port 3 etc.. Then go into proxy and registration and set the proxy IP address to the pbxnsip server. For fxo you usually don't have to have them register unless you wanted to connect to that particular port and grab a dial tone (two stage). Make sure the dialing mode is set to one stage in the fxo parameters. If you want to use this as a regular gateway you would set the dialing mode to single stage, and the channel select mode (important) to ascending so it grabs the first available port. Then in pbxnsip create a Gateway trunk and put the IP address of the audiocodes in there. In the advanced parameters I would set the debug level to 6 and then go into status and diagnostics and look in the message log and send a call to it.


    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.


    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

  16. Yes there was someone using it with the speech server and it worked. The setup was prettys much the same as with Exchange - I guess they are using the same underlying components.

    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

  17. Hi @all,


    its possible. Place pbxnsip in front of the OCS mediation server. It talks SIP over TCP and pbxnsip can this too, like with Exchange 2007:

    No authentification, no encryption - its the same like the gateways from AudioCodes, Dialogic, Vegastreams,... do. Only the communication between OCS and OCS Mediation Server is SIP over MTLS. You need to use a Windows Public Key Infrastructure or commercial certificates, dont use selfsigned certificates!


    Create a Trunk in pbxnsip: SIP-Gateway


    Domain: your-ocs-mediation-server-ip-adress

    no password + accept redirect


    You need to place the same numbers in pbxnsip, you like to call from within OCS. Its important to use E.164 numbers, because OCS will only accept this and it will only call outside with E.164 numbers.


    To call any number from the real world in Office Communicator, please wait for the next pbxnsip-version. It will support Trunk to Trunk calls. So you can use a trunk to your prefered VoIP-Provider and call from OCS with unknown numbers will be transfered.


    For now, give the test-version a try with OCS.


    best regards, :rolleyes:





    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

  18. 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

  19. "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

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