Vodia PBX

The version that you have probably has the NTP time server set (in the admin/ports page). Just clear it and then it should use the default time server again.

The handling of calls coming from a trunk were designed to send them to internal destinations. If we are starting to send them to external destinations we get into new areas. The problem occured for Microsoft Exchange and OCS; so we introduced some ad'ons to make this possible. But I would not call that a "proper way" to route incoming calls.

Aha. I think the problem must be some keep-alive traffic. ... can we have a Wireshark :blush: ?

At the moment I would say it is an application that can be easily and safely run as a external application.

Well, I think "free" might be tricky... You might get only trash for free. Check out http://forum.pbxnsip.com/index.php?showtopic=1358.

Oh so you mean routing everything through one extension? Might be tricky, but could actually work... I guess you have to tell the domain to keep the From/To headers unchanged. Then in the trunk you could just use a pattern like "!123456[0-9]{2}!123!t!" that would send all calls to 123456xx to extension 123, then there you can use a static registration with something like "<sip:exchange@192.168.1.2;transport=tcp>". I did not try this, bug it might get "closer".

Maybe that was a misunderstanding... Do you mean the general login or the login into a specifc queue (putting an agent on the list)? Which button does not night? The light of the queue or the login button?

Whow, there are still too many pending packets. Maybe after a restart and waiting for two minutes, enter the following onto the wen interface: http://pbx/reg_status.htm?save=save&ex...pending_packets. Then you should see what packets are actually pending and this should give us a hint where all that memory goes to!

The problem with that setting is that it applies to all incoming numbers then and not only to a range...

the PBX is a bad protocol converter....

So the latest and greatest are to be found here: http://pbxnsip.com/download/pbxctrl-3.1.1.3101.exe (Windows, manual upgrade) http://pbxnsip.com/download/pbx3.1.1.3101.exe (full Windows installer) http://pbxnsip.com/download/pbxctrl-freebsd7.0-3.1.1.3101 (FreeBSD7.0 32-bit) http://pbxnsip.com/download/pbxctrl-rhes4-3.1.1.3101 (RedHat 4) http://pbxnsip.com/download/pbxctrl-centos5-3.1.1.3101 (CentOS 5) http://pbxnsip.com/download/pbxctrl-suse10-3.1.1.3101 (SuSE10) http://pbxnsip.com/download/pbxctrl-debian4.0-3.1.1.3101 (Debian 4) http://pbxnsip.com/cs410/update-3.1.1.3101.tgz (CS410) We have to address the attended transfer into IVR after this 3.1.1 version. This will mess up a lot of things, and we first want to have a stable release before we can mess things up again...

Hmm, yea we have seen that before in another location. Is there any chance to get a SIP log from that? 2106 sounds like the timestamp is zero; maybe we should just discard such calls (ok that is just a fix).

See http://wiki.pbxnsip.com/index.php/Release_Notes_3.1.

BTW the latest & greatest is now: http://pbxnsip.com/download/pbxctrl-3.1.1.3100.exe (Windows, manual upgrade) http://pbxnsip.com/download/pbx3.1.1.3100.exe (full Windows installer) http://pbxnsip.com/download/pbxctrl-freebsd7.0-3.1.1.3100 (FreeBSD7.0 32-bit) http://pbxnsip.com/download/pbxctrl-rhes4-3.1.1.3100 (RedHat 4) http://pbxnsip.com/download/pbxctrl-centos5-3.1.1.3100 (CentOS 5) http://pbxnsip.com/download/pbxctrl-suse10-3.1.1.3100 (SuSE10) http://pbxnsip.com/download/pbxctrl-debian4.0-3.1.1.3100 (Debian 4) http://pbxnsip.com/cs410/update-3.1.1.3100.tgz (CS410) Guys, make a backup before you upgrade, this is still a beta image!

I would do that with the static registration. See http://wiki.pbxnsip.com/index.php/Extension#Registrations. Otherwise you will have trouble with the "charge for redirect" setting in the trunk. The limitation is that you need an extension account for every redirection that you want to program.

As for the registrations between the PBXes - that problem should disappear with your next upgrade, as both sides should support "outbound". The 45 seconds are probably half of 90 seconds. What time do you see in the contact header of the registration reply? It is probably 90 seconds? Then you will find that number in the settings of the admin mode.

Ops, seems the "Request Timeout" was not covered in the trunk settings - check out the updated http://wiki.pbxnsip.com/index.php/Trunk_Settings.

Right, there is a setting on domain level. If the other person does not pick up, the caller will hear ringback for some time (e.g. 20 seconds). When the other user is busy (no call waiting) then the PBX will offer the camp on immediately. The tricky case is when the other use has call waiting enabled on the phone. Then the only way to make that case working is that like Bills said the number of "lines" to that user is set to "1".

Whow, did they say what the problem was?

Do you still have a high number of pending_packets?

I believe that RFC3265_4235 would be the best match. Do you see that the phone sends a SUBSCRIBE with the Event: dialog? That would be a good first step. Then the PBX should send XML notifications to that phone.

You should see the SIP messages embedded in the PBX log like the following example: [5] 2008/12/04 09:53:23: Identify trunk (domain name match) 17 [7] 2008/12/04 09:53:23: SIP Tx tcp:215.111.36.226:2096: SIP/2.0 100 Trying Via: SIP/2.0/TCP 192.168.11.188:2096;branch=z9hG4bK-cpeym97f03ql;rport=2096;received=215.111.36.226 From: <sip:524@pbxnsip.com>;tag=t1qkfe13b2 To: <sip:*97@pbxnsip.com;user=phone>;tag=0b3eebce61 Call-ID: 3c2690d4ed8e-y9flwusywaav CSeq: 1 INVITE Content-Length: 0 Check out http://wiki.pbxnsip.com/index.php/Log_Setup on how to turn SIP packet logging on. Having the SIP packets with the other log messages of the PBX makes it a lot easier to figure out why and what is gonig on. Maybe you can give that a quick shot. It you cannot get a SIP trace, send me a PM and we'll try to register from here.

Aastra is also already included in the PBX. If you have a Astra 57i then it should also work right out of the box. The PnP page should not be blank... There should be a list of parameters. What is the content of your html directory? Did you put your own pnp.xml there? Don't do that, at least not now (maybe later).

Maybe it is a misunderstanding... The PBX generates files e.g. for Polycom on the fly. You don't have to put anything into the tftp directory. Just give it a try! If the PBX generates files, it will put them into a special directory "generated" - so that you can review the result of the automatic provisioning. If you are using Polycom, you should check out http://wiki.pbxnsip.com/index.php/Polycom. Polycom is well supported with the PBX, so maybe you give that a try first. Grandstream is not so well supported, but maybe this is the opportunity to update the provsioning process for Grandstream phones.

Make a backup of your working directory and try upgrading to this: http://pbxnsip.com/download/pbxctrl-3.1.1.3100.exe.