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Don

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About Don

  • Birthday 03/28/1956

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    Plano, Tx.

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  1. I should have mentioned that the link above was for Windows 32 bit only. I don't know if the MAC version has the same bug but the PBXnSIP Appliance did so you will probably need to get a MAC version of PBXCNTRL that is at least the version as the one that is linked above from the PBXnSIP people.
  2. Did you load the GMail Certificate mentioned in the above messages? You need to go to the top menu Settings -> Certificates -> Main CA Server and load the entire text including the begin and end comments. Then you should see the certificate loaded. Once you do that then the setup is pretty straight forward. Also be sure that you are using the PBXnSIP load that is mentioned int he above text. All you do is replace the PBXCNTRL.EXE file in the PBX directory with the one that you get from that link. Good luck.
  3. This download fixed the issue. I had to remove the final period for it to be downloadable. http://pbxnsip.com/cs410/update-4.2.0.3963.tgz
  4. It runs out that we had to build the trunk as an incoming only trunk in each of the other domains for the calls to be delivered there even though they are routed to that specific domain from the M600 (Audiocodes) directly. This is apparently a change from 3.X. FYI to all.
  5. We just converted a PBX that has 3 Domains from 3.X to 4.X. This PBX has several incoming Trunks loaded in the LOCALHOST domain. Previously we simply routed the calls to the appropriate domain by setting the destination IP address in the M600 (Audiocodes) based on the incoming number. So for example: 850xxxxxxx (x = valid DID number) is sent to ip address "resortsands" and we have a domain built that = "resortsands" * (wildcard for all not matched numbers) is sent to ip address 10.10.4.5 (PBXnSIP private IP address) As stated this works fine in 3.4.0.3201 but when we went to 4.0.1.3499 all incoming calls are getting a 404 not found. It seems that the PBX is not looking in the resortsands domain for a match even though that is the IP address that is being delivered. Is there a new datafill or parameter requirement in 4.X that we are missing?
  6. After we upgraded a PBX from 3.X to 4.X all (almost all) Account extensions have an exclamation point in a yellow triangle under status. There does not seem to be any explanation of what this icon means in the documentation (as pertaining to an extension). As far as I can tell the extension seems to be working fine. It is registered and can make calls. What could it mean? Is there a way to clear it? Thanks, Don
  7. Where can I get a copy of the "newer" build? Thanks for fixing this bug.
  8. Did you have any luck figuring out why this does not work? GMail is a very popular mail service and I would like to use it to send out notices.
  9. GMAIL requires TLS (Transport Layer Security) so that is not an option. Can you tell me the procedure to import the certificate?
  10. I upgraded my PBXnSIP appliance to the latest version of PBXnSIP (4.2.0.3958) and now I cannot figure out how to configure the settings for GMAIL. I tried the same settings that worked in 3.X but the messages always fail and the pbx reports it but it is not specific in why. Can someone provide an example of a working GMAIL configuration for the appliance in 4.2.0.3958? My old working config looked like this: "From" Address (e.g., "PBX" <pbx@domain.com>): MASTER <someone@gmail.com> Account (e.g., pbx): someone@gmail.com Password (e.g., secret): Password (repeat): SMTP Server (e.g., smtp.domain.com): smtp.gmail.com:587 Thanks in advance. Don
  11. I upgraded my CS410 to the new (4.0.1.3499) release from 3.4.0.3201. When I did all calls that come to the auto attendant and transfer out to a public number fail. Calls to internal extensions are OK. I even tried to set an internal extension that redirects to an external number and that fails as well. Note that the call is extended and even rings once at the distant end but then the call fails. This same configuration works perfectly in the 3.4 release. Any thoughts? I can provide more detail and SIP traces upon demand. BR, Don
  12. Yes they are identical. I edited out caller specific data in the exanple below. As I stated we get two of these on every missed call unless the caller leaves a voicemail, then we get one. Thanks, Don -------------------------------- Missed Call from WIRELESS CALLER (+1850xxxxxxx) Reply | seychelles@ipacketnet.com ✆ to support show details 7:01 PM (11 hours ago) Missed Call from WIRELESS CALLER (+1850xxxxxxx) You missed the following call: From: WIRELESS CALLER (+1850xxxxxxx) (click +1850xxxxxxxx to call back) To: IPN Support Day Mailbox (998) Time: 2009 12 19 19:01:36 This email was sent because your account settings have "send missed call" turned on. Please note that some Email clients may suppress the click-to-dial link. Do not reply to this Email. It was sent automatically.
  13. We use a Hunt Group to send calls to our technical support group. We ring a number of extensions and if no one answers then we go to a dedicated extension to leave a message. This extension has the email flag set to, "Send email on missed calls:". The problem is that when a user calls and goes to the voicemail prompt and then abandons before leaving a message we get two emails sent to the email address listed. If the caller leaves a message then we get 1 missed call email and the voicemail in an email with attachment. I would expect to get the one missed call email on calls where the caller leaves no voicemail and no missed emails on calls where the caller leaves a voicemail. Any I missing something here? Any help is gratefully accepted. Thanks, Don IPacket Networks
  14. In our operating environment we have multiple PBX instances at resorts that are nearby each other. Each building has it's own IP service via fiber optic entrance. We have a high speed microwave IP route between the buildings for backup. This allows us to route to the alternative path on a fiber failure. This works fine for video and Internet restoral. We would like to be able to be able to send calls from the PBX in one building to the other so that on a failure of that building fiber systems calls could still be completed via the link to the redundant IP entrance. Is there any way to tandem SIP calls? In other words can a call coming into a SIP trunk be routed to another SIP trunk in a tandem arrangement? This would allow us to have DID traffic sent to the backup SIP trunk and then completed on the appropriate PBX. Kevin Moroz suggested that in 3.0 using the tandeming capability that was added for MS Exchange Server calls may be a potential answer. Any help is appreciated. Don Heckman IPacket Networks
  15. We ran into this same problem and found it was an error in the trunk statement. After we moved from Version 1.X of PBXNSIP to 2.1.4 for some reason the value in the Trunk statement for the P-Asserted field was set to none. We tried several settings and found that setting the value to "Remote-Party-ID" matched the results of typical in most US carriers. Now redirected calls show the original calling parties ANI and not the called party number. Hope this helps.
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