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2.1.0 RC2 image available


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what is the:Upload picture: for?

I can't see that. Maybe a temporary thing while we were putting the content there?

 

 

does this version support Aastra tftp plug and play?

Yes. Although it is quite basic. I think it is time to start a forum on Aastra and put the template there. Maybe there are improvements that can be shared on the forum.

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Now there is a beta2 image available at http://www.pbxnsip.com/download/pbxctrl-2.1.0.2090.exe (see http://wiki.pbxnsip.com/index.php/Installi...Manual_Upgrade) on howto upgrade to this image.

 

Things that have been fixed were a memory leak in the subscriptions subsystem and a general review of the memory management, maybe fixing some (extremly rare) race conditions in the memory managament. There is also a new feature (*00) which calls the extensions cell phone directly without ringing the extension and without respecting the fork delay and day of time settings (so that you don't have to remember an extension's cell phone number any more).

 

We will also provide an update for the CS410 in a seperate posting. We would like to get further feedback from the Windows platform users before we turn our attention to the Linux platforms.

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Okay, here we go with 2.1.0.2093:

 

http://www.pbxnsip.com/download/pbxctrl-2.1.0.2093.exe (raw Windows executable)

http://www.pbxnsip.com/download/pbx2.1.0.2093.exe (InstallShield)

http://www.pbxnsip.com/download/pbxctrl-cs410-2.1.0.2093 (CS410 appliance)

http://www.pbxnsip.com/download/pbxctrl-debian3.1-2.1.0.2093 (Debian 3.1)

http://www.pbxnsip.com/download/download/p...se10-2.1.0.2093 (SuSE10/32-bit)

 

The InstallShield contains this time the files neccessary for plug and play of the various vendors. We feel that it is useful to provide these files outside of the image, so that you can easily make changes to it - making the provisioning process easier.

 

We have currently some technical difficulties with our RedHat server... Will add that build later.

 

This version seems to be quite stable so far and we want to give it the name "RC1" (release candidate 1).

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Okay, here we go with 2.1.0.2093:

 

http://www.pbxnsip.com/download/pbxctrl-2.1.0.2093.exe (raw Windows executable)

http://www.pbxnsip.com/download/pbx2.1.0.2093.exe (InstallShield)

http://www.pbxnsip.com/download/pbxctrl-cs410-2.1.0.2093 (CS410 appliance)

http://www.pbxnsip.com/download/pbxctrl-debian3.1-2.1.0.2093 (Debian 3.1)

http://www.pbxnsip.com/download/download/p...se10-2.1.0.2093 (SuSE10/32-bit)

 

Hey guys can someone please fix the SUSE10 link seems every other link works but not the one I want.

Cheers

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Here we go with the latest:

 

http://www.pbxnsip.com/download/pbxctrl-2.1.0.2097.exe (raw Windows executable)

http://www.pbxnsip.com/download/pbxctrl-cs410-2.1.0.2097 (CS410 appliance)

http://www.pbxnsip.com/download/pbxctrl-debian3.1-2.1.0.2097 (Debian 3.1)

http://www.pbxnsip.com/download/pbxctrl-suse10-2.1.0.2097 (SuSE10/32-bit)

http://www.pbxnsip.com/download/pbxctrl-rhes4-2.1.0.2097 (RedHat ES4/32-bit)

 

There were still problems with one-way audio and unwanted Re-INVITEs.

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Here we go with the latest:

 

http://www.pbxnsip.com/download/pbxctrl-2.1.0.2097.exe (raw Windows executable)

http://www.pbxnsip.com/download/pbxctrl-cs410-2.1.0.2097 (CS410 appliance)

http://www.pbxnsip.com/download/pbxctrl-debian3.1-2.1.0.2097 (Debian 3.1)

http://www.pbxnsip.com/download/pbxctrl-suse10-2.1.0.2097 (SuSE10/32-bit)

http://www.pbxnsip.com/download/pbxctrl-rhes4-2.1.0.2097 (RedHat ES4/32-bit)

 

There were still problems with one-way audio and unwanted Re-INVITEs.

 

Thanks SUSE10 link now works cheers

 

Steve

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re: 2.1.0.2097 one-way audio issue. Have been using RC2 since 9/8 and have found that anytime there are more than 2 registrations for an account, the thrid registraiton can receive audio but does not send audio. This only occurs on in-bound calls from a trunk. Internal calls work fine on as many reigstrations as needed. The sets we use are Grandstream GXP-2020's and the gateway is a Grandstream GXW-4108. I've tested many configurations and the one-way audio issue always moves to the third phone registered. ANy suggestions on a work-around until this is fixed?

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Ouch. That was an "old friend". The problem is that the internal table for mixing audio is quite short. Thanks for the tip!

 

We made a quick hot-fix image at http://www.pbxnsip.com/download/pbxctrl-2.1.0.2098.exe. Please do a quick verification (we could not do it here yet, so make sure you can move back easily).

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I tested release 2.1.0.2098 and it did fix the multi-regitration one-way audio issue. Thanks for getting this done so quick!

 

However... there still seems to be one more issue. When paging to any number of registered phones, all phones go on speaker and are listening, but the source phone continues to ring and is not released to be able to speak. When the source phone hangs up, all phones hang up like they are suposed to. For a brief period of time, I did have audio out with a low volume ring in the source phone, however, I could not duplicate this nor tell what had changed. This issue was in the 2093 and 2097 release - I just did not have a chance to test yet.

 

I've tried various settings on the Grandstream phones as well as using a SIP soft phone but have consistent results. Restarting pbxctrl does not help, nor does deleting and creating a new paging account. Any ideas?

 

(By the way: any way you can add a wild card into the paging destination field (i.e. 2* ) just like the source and dialog permissions fields? We use extensions in specific locations that allow easy grouping of paging, however I don't want to have an issue when a new extension is created but is forgotten to be added to the paging group

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Well, usually you should get a call waiting if the caller is also part of the paging group. There are some phones that audomatically put a call on hold once that they receive an INVITE with an auto-answer request (which is IMHO very problematic, think about you most important customer on the phone and someone paging).

 

If you have such a phone, try setting the "lines" parameter in the registration and set it to "1". Then the PBX will not call that extension.

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Okay, here we go with 2.1.0.2100: http://www.pbxnsip.com/download/pbxctrl-2.1.0.2100.exe.

 

Notes: There were issues with one way audio associated with the new RTP-pass through mode. The last bug was when the ROC in SRTP kicked in again, resulting in a kind of one-way audio. The version also now includes a JavaScript password checker that keeps users from entering too simple passwords.

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The Outbound Trunk ANI Facility is not working.

 

We add $2 in the Explicit Remote-Party-ID field.

 

PBXNSIP does this

 

INVITE sip:telephone_number@domain.com;user=phone SIP/2.0

Via: SIP/2.0/UDP 192.168.0.12:5060;branch=z9hG4bK-e570ebb78b789357fbb229d06f25958d;rport

From: "10002 10002" <sip:$2@trunk_domain.com>;tag=10450

 

This stops the CLI from being presented.

 

ALSO

 

You have changed the way trunks use the Remote Party/Privacy Indication:

You have removed the straight RFC3325 option.

 

In the example above the name "10002 10002" would be shown as the ANI and NOT the actual ANI which is stored in $2.

 

Did it show "10002 10002" because of the $2 bug?

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