pbx support

Check the IVR tab under the AA. There are couple of radio buttons at the bottom of the page.

That means the extension was already created and that's why you see that extension

Sure. Please download http://pbxnsip.com/protect/pbxctrl-3.3.2.3183.exe and follow http://wiki.pbxnsip.com/index.php/Microsof...er_for_Exchange.

Is it possible to try something newer, just to verify whether this is the version issue? Please make a complete backup of the PBX before you try 3.3.x version - http://pbxnsip.com/protect/pbxctrl-3.3.2.3183.exe

Transcoding feature has been successfully added and tested in couple of beta sites already. Yes, it will be in the new 3.x release.

Did you try deleting the 'pnp_parms' folder?

Did you see this http://forum.pbxnsip.com/index.php?showtopic=984&hl= already?

Version 3.3.x is capable of writing the CDR to MySQL (no MS-SQL support yet). Please PM support@pbxnsip.com for the documentation on how to use it.

What is the version you are running?

We have plans to do the transfer using PAC. No firm dates yet. But plans to release it along with 4.0 PBX.

There was some issue in saving the PSTN related configuration on CS410. Probably, that's why you could not save it. This version should take care of the saving issue http://pbxnsip.com/cs410/update-3.3.2.3181.tgz

Do you have any custom pnp file? Are these phones on the same LAN as the pbx? If yes, then you do not have to do too much work. All you have to put the MAC address of the phone on the "registration" tab and 'reset' the phone. Otherwise, please take a look at http://www.blogpbxnsip.com/2009/04/wan-pnp...hones-http.html

Thw default is root123

Looking at the pcap trace file, RTP was flowing on both directions till the end of the call. Even the ports seem to be proper.

We have a windows version (not GA yet) that takes care of the recording related issues. You can download this from http://pbxnsip.com/protect/pbxctrl-3.3.2.3183.exe

Looks like mediation is not passing any traffic to pbx. You may need to check the mediation dial plans/routing related settings (i am no big expert on it)

Can you set the "Log Other messages (e.g. INVITE)" to "yes" and take the log. Thanks

Have you set the log level to 9 and taken the log? You should see PSTN logs when you receive/make PSTN calls

It is not that we have locked it up. We do give access to customers as and when the request. Please PM support@pbxnsip.com with the account details, if you need one.

Check out the dial plan on the PBX. If the call works with 21XXXXXXX, then by having the correct dial plan on the PBX, you should be able to make the calls. A SIP log on the PBX would help to debug the issue (http://wiki.pbxnsip.com/index.php/Admin_settings#How_to_get_the_SIP_logging)

Thanks. I added a line to the comparison chart a while ago. http://en.wikipedia.org/wiki/Comparison_of_VoIP_software. Seems like it has been removed. Is there anything special you had to do, other than being a member?

In an effort make it easier for the users, we introduced use of country code in 3.3. Under the Domain->Settings, there's a Country Code field. Set that to 1. Also, you can set the area code if you like. Then on the trunk settings, there is field "Rewrite Global Numbers", you can set the NANPA 11 digits. These 2 settings should take care of the problem.

I am assuming you are not trying to black list an extension. Are you?

Does this happen only on the conference calls or even a call from snom360 to other phones? I am assuming you are pressing mute on the snom 360 and not the other party. Tested it here on the same version (7.3.14) and the problem did not happen. Maybe a wireshark trace will tell us how is disconnecting the call.

We had a similar complaint sometime back and an instance of PAC running against the PBX was causing the issue. After they closed and restarted the PAC, things were fine.