Don

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.

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.

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

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.

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?

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

Where can I get a copy of the "newer" build? Thanks for fixing this bug.

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.

GMAIL requires TLS (Transport Layer Security) so that is not an option. Can you tell me the procedure to import the certificate?

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

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

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.

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

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

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.

Thanks for the excellent reply. 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. Additionally, we found another use for this as well. Since we forward our support phone to the on call engineer during off hours they had to answer every call with our branding. We set up a separate trunk group just for call to the support line and set that parameter to NONE. The engineer then can see the dialed number rather than the calling number in the display and knows to answer appropriately. Thanks again for the help in this matter. It is fully resolved. Don

We have been running version 1.X of PBXNSIP for some time and have been using Call Redirection to redirect support and trouble calls to our on-call engineer. The Caller ID received in these calls was always the ANI (Calling Number) of the original caller. This seems to be correct based on telephony standards. Clearly this is the way the local phone company handles call forwarded calls. When we recently upgraded several systems to 2.1.4 we noticed a change in the Caller ID value being delivered. We now received the Dialed number (Called Number) being delivered to the engineer. I have read all of the applicable release notes and searched the forums and cannot find any reference to a change or anyone else seeing this problem. Which is a little surprising. Is there some parameter that we could have inadvertently changed or a setting that could be causing this change in operation? Any help in this matter is appreciated. Best Regards, Don

The IAD can only associate one called number with one extension. To be able to change the ringing cadence there must be some indication in the ALERT-INFO field. In RFC3960 this is mentioned in Section 5. The entire document is at: http://www.ietf.org/rfc/rfc3960.txt The pertinent section reads: 5. Alert-Info Header Field The Alert-Info header field allows specifying an alternative ringing content, such as ringing tone, to the UAC. This header field tells the UAC which tone should be played in case local ringing is generated, but it does not tell the UAC when to generate local ringing. A UAC should follow the rules described above for ringing tone generation in both models. If, after following those rules, the UAC decides to play local ringing, it can then use the Alert-Info header field to generate it. One example of how this field could be coded is described in : http://sofia-sip.sourceforge.net/refdocs/s...lert__info.html Just to be clear. The PBX would need to indicate in this field that for a given call that a different ringtone is required thus telling the IAD to send distinctive ringing to the extension. Based on this exchange and further reading I am fairly sure this is not supported by PBXNSIP. If I am missing something please let me know. FYI, we are using the Audiocodes MP-124D which supports distinctive ringing. Note that the datasheet on this IAD is available at : http://www.audiocodes.com/objects/30010_DS...D.pdf Best Regards, Don

I understand the hunt group situation you describe. In our application we supply phone service to multi-tenant buildings using PBXNSIP as our PBX. Many customers have only one physical line (or extension) in their home. The ability to use distinctive ringing to differentiate between services is widely used in the US. To quote Wikipedia in the ringtones article: "A service akin to party line ringing is making a comeback in some small office and home office situations allowing facsimile machines and telephones to share the same line but have different telephone numbers; this CLASS feature is usually called distinctive ringing generically, though carriers assign it trademarked names such as "Smart Ring", "Duet", "Multiple Number" and "Ringmaster." This feature is also used for a second phone number assigned to the same physical line for roommates or teenagers, in which case it is sometimes marketed under the name teen line." Accordingly we have a need to provide distinctive ringing on a given IAD terminal based upon the dialed number. We would like to be able to use a different alias for a given extension and then have the PBX supply a distinctive ring to allow the services mentioned above. Again, if you can support this please provide specifics on how it can be implemented. If not then please consider this as a potential feature for a future release. Clearly this feature has significant merit for the CLEC market place that we arr serving using PBXNSIP. Thank you. Don

Is there a way to produce distinctive ringing from a unique called party number? Distinctive Ringing is a rather common telco feature that allows the user to place a fax machine (or other device that can detect DR) on a shared extension with a phone. The IAD that we use supports it if the PBX sends the appropriate code. Do you support multiple distinctive ringing patterns in the Alert-Info Header in the Invite message? If so, how is this invoked? Please be as specific as possible. Thanks for any assistance in this matter. Don