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As discussed over the last few weeks, there is a new 2.1.6 available. The release notes are as usual available on the Wiki at http://wiki.pbxnsip.com/index.php/Release_Notes_2.1.6. The primary goal of this version was to increase stability and usuability, for example we removed some mouse traps like including tftp in the backup.

 

The following versions are available:

 

http://www.pbxnsip.com/download/pbx2.1.6.2446.exe (complete InstallShield)

http://www.pbxnsip.com/download/pbxctrl-2.1.6.2446.exe (bare executable)

http://www.pbxnsip.com/download/pbxctrl-cs410-2.1.6.2446 (CS410)

http://www.pbxnsip.com/download/pbxctrl-debian4.0-2.1.6.2446 (Debian 4.0)

http://www.pbxnsip.com/download/pbxctrl-rhes4-2.1.6.2446 (RedHat ES4)

http://www.pbxnsip.com/download/pbxctrl-suse10-2.1.6.2446 (SuSE10)

 

Unfortunately, we had to move from Debian 3.1 to Debian 4.0, but people said that 3.1 was quite old anyway and it seems that the new image also runs on the old platform. However, of course we recommend to upgrade to Debian 4.0 as well.

 

Please let us know if there is any problem with the new build.

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on a CS410 I let this:

 

Starting the FXO gateway:

/pbx/driver /pbx

insmod: error inserting './csmencaps.ko': -1 Invalid module format

/pbx

Check reset flag:

TEPI> /dev/fxo/ch0 open error

[0] 192047: Setup TDM side codec to U law

TEPI> /dev/fxo/ch0 open error

[0] 192047: Cannot open /dev/fxo/ch0

Starting the PBX:

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Guest Paul McCabe

There appears to be a bug in V2.1.6.2446 involving the license key. If the number of domains licensed in the key is set to 1 (one), you will see the following error in the logfile:

 

[0] 2008/02/13 20:30:43: License suspended: There are too many domains

On the license page you will also see: Current license state: No license

 

If the # of domains is set to 'unlimited' or greater than 1, things appear to work fine with this software version.

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There appears to be a bug in V2.1.6.2446 involving the license key. If the number of domains licensed in the key is set to 1 (one), you will see the following error in the logfile:

 

[0] 2008/02/13 20:30:43: License suspended: There are too many domains

On the license page you will also see: Current license state: No license

 

If the # of domains is set to 'unlimited' or greater than 1, things appear to work fine with this software version.

 

Hi Paul,

 

I have to add that even with the domains set to unlimited you still get the other failure during startup but as u said the license is valid again:

 

[4] 2008/02/13 17:22:03: Translation item dom_ext2.htm#default#sp not found

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There appears to be a bug in V2.1.6.2446 involving the license key. If the number of domains licensed in the key is set to 1 (one), ...

 

Oh oh. You are right, here comes the fix:

 

http://www.pbxnsip.com/download/pbx2.1.6.2447.exe (complete InstallShield)

http://www.pbxnsip.com/download/pbxctrl-2.1.6.2447.exe (bare executable)

http://www.pbxnsip.com/download/pbxctrl-cs410-2.1.6.24467 (CS410)

http://www.pbxnsip.com/download/pbxctrl-debian4.0-2.1.6.2447 (Debian 4.0)

http://www.pbxnsip.com/download/pbxctrl-rhes4-2.1.6.2447 (Redhat ES4)

http://www.pbxnsip.com/download/pbxctrl-suse10-2.1.6.2447 (SuSE10)

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PBXnSIP Community - I was running 2.1.5.2357 for about 25 days straight with little to no problems on 3 separate servers.

 

On Monday - one of them stopped the services unexpectedly - with no indication in the logs (log level 3)

Yesterday - one of them stopped the services unexpectedly - with no indication in the logs (log level 6)

Yesterday - the same server stopped AGAIN unexpectedly - an hour later.

 

I'm contemplating proactively starting and stopping the services on the 3rd server today - so I don't have to explain the outage to customers when it decides that it is going to restart during business hours.

 

My question to the community - are we ever going to get a stable release - like 2.0.3.1715 so that I can safely deploy these to my customer base? I'm getting hammered daily by the instability of each release since 2.1.2

 

Is 2.1.6 - latest version being run in a production environment by anyone? And how long do I need to wait to see if problems are going to occur... I'm trying to quit using my customer base as a testing ground for new releases.

 

Any help is appreciated.

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My question to the community - are we ever going to get a stable release - like 2.0.3.1715 so that I can safely deploy these to my customer base? I'm getting hammered daily by the instability of each release since 2.1.2

 

Is 2.1.6 - latest version being run in a production environment by anyone? And how long do I need to wait to see if problems are going to occur... I'm trying to quit using my customer base as a testing ground for new releases.

 

Any help is appreciated.

 

Well, I am running always the latest releases in three locations in the US and Mexico and so far I have not expirienced any crashes or major bugs that would prevent me from using the latest version. But I am the mean administrator and don't have to please customers (hehehe).

 

Overall I am very happy with pbxnsip - no major issues and always quick responses and fixes to little problems I discovered so far.

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Is 2.1.6 - latest version being run in a production environment by anyone? And how long do I need to wait to see if problems are going to occur... I'm trying to quit using my customer base as a testing ground for new releases.

 

2.0.3.1715 was a great release.

 

Yes, we are running 2.1.6.2448 in several locations in real life. For us it works, of course. But that is not the problem.

 

We created a 2.2 branch for new features. 2.1 will be only used for fixes to make sure that we have a sock-solid build again. We did run millions of calls through the 2.1 branch, but the environment is not changing much during the tests. The real life is different, and we will see in the next couple of days and weeks if there is anything else that we need to address.

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I have a PBXnSIP UM problem that seems to be caused by Exchange SP1... but it might be a 2.1.6 thing.

 

The problem seems to be in the REFER coming back from the Exchange Server.

 

Scenario: Extension (2412 & 5000, respectively) calls Exchange Auto Attendant (8564), presses 1, transfers back to PBXnSIP extension (2504 and 5001, respectively).

 

Right now we have two systems that we are using to figure out what exactly is going on:

1. PBXnSIP 2.1.6.2446 with a trunk to Exchange SP1 server (testing)

2. PBXnSIP 2.1.2.2292 with a trunk to Exchange without SP1 (production)

 

On the SP1 Server, the REFER-TO line is:

REFER-TO: <sip:5001;phone-context=PBXnSIP-exchangesp1.ourdomain.com@192.168.11.4;user=phone>

 

On the Exchange Server (no SP1), the REFER-TO line is:

REFER-TO: <sip:2504@192.168.11.4:3848;transport=tcp;user=phone>

 

Where the 'PBXnSIP-exchangesp1.ourdomain.com' is the Exchange dialplan associated with the PBXnSIP 2.1.6 gateway and '192.168.11.4' is our mailbox server. If you look at the UM properties of a user in Exchange, both users 5001 and 2504 have the phone-context listed.

 

Is anyone else aware of this issue or what might be done to fix it?

 

--- Call logs below ---

 

--- PBXnSIP 2.1.2 to Exchange w/o SP1

REFER sip:2412@192.168.11.4:3848;transport=tcp SIP/2.0

FROM: <sip:8564@192.168.11.7;user=phone>;epid=25-50-99-AE-EF;tag=741759d2a0

TO: <sip:2412@192.168.11.7>;tag=22099

CSEQ: 1 REFER

CALL-ID: 0a45efff@pbx

MAX-FORWARDS: 70

VIA: SIP/2.0/TCP 192.168.11.7:5065;branch=z9hG4bK879ddf92

CONTACT: <sip:exchange.ourdomain.com:5065;transport=Tcp;maddr=192.168.11.7>

CONTENT-LENGTH: 0

REFER-TO: <sip:2504@192.168.11.4:3848;transport=tcp;user=phone>

USER-AGENT: RTCC/2.0.6017.0

Referred-By: sip:8564@192.168.11.7;user=phone

 

[9] 2008/02/20 14:56:36: Resolve destination 144704: tcp 192.168.11.7 5065

[7] 2008/02/20 14:56:36: SIP Tx tcp:192.168.11.7:5065:

SIP/2.0 202 Accepted

Via: SIP/2.0/TCP 192.168.11.7:5065;branch=z9hG4bK879ddf92

From: <sip:8564@192.168.11.7;user=phone>;epid=25-50-99-AE-EF;tag=741759d2a0

To: <sip:2412@192.168.11.7>;tag=22099

Call-ID: 0a45efff@pbx

CSeq: 1 REFER

Contact: <sip:2412@192.168.11.4:3848;transport=tcp>

User-Agent: pbxnsip-PBX/2.1.2.2292

Content-Length: 0

 

[5] 2008/02/20 14:56:36: Redirecting call to 2504

[7] 2008/02/20 14:56:36: Calling extension 2504

---

 

--- PBXnSIP 2.1.6 to Exchange SP1

REFER sip:5000@192.168.11.13:4510;transport=tcp SIP/2.0

FROM: <sip:8564@192.168.11.4;user=phone>;epid=4BA968830A;tag=da77a4c889

TO: <sip:5000@192.168.11.4>;tag=50275

CSEQ: 1 REFER

CALL-ID: 4a853a28@pbx

MAX-FORWARDS: 70

VIA: SIP/2.0/TCP 192.168.11.4:5065;branch=z9hG4bK16cbdb5f

CONTACT: <sip:exchangesp1.ourdomain.com:5065;transport=Tcp;maddr=192.168.11.4;ms-opaque=fd2ca9a11a23831e>;automata

CONTENT-LENGTH: 0

REFER-TO: <sip:5001;phone-context=PBXnSIP-exchangesp1.ourdomain.com@192.168.11.4;user=phone>

REFERRED-BY: <sip:8564@192.168.11.4;user=phone>

USER-AGENT: RTCC/3.0.0.0

 

[9] 2008/02/20 15:02:51: Resolve 36489: tcp 192.168.11.4 5065

[7] 2008/02/20 15:02:51: SIP Tx tcp:192.168.11.4:5065:

SIP/2.0 202 Accepted

Via: SIP/2.0/TCP 192.168.11.4:5065;branch=z9hG4bK16cbdb5f

From: <sip:8564@192.168.11.4;user=phone>;epid=4BA968830A;tag=da77a4c889

To: <sip:5000@192.168.11.4>;tag=50275

Call-ID: 4a853a28@pbx

CSeq: 1 REFER

Contact: <sip:5000@192.168.11.13:4510;transport=tcp>

User-Agent: pbxnsip-PBX/2.1.6.2446

Content-Length: 0

 

[5] 2008/02/20 15:02:51: Redirecting call to 5001;phone-context=PBXnSIP-exchangesp1.ourdomain.com

[9] 2008/02/20 15:02:51: Dialplan: Evaluating !^5([0-9]*)@.*!5*!i against 5001;phone-context=PBXnSIP-exchangesp1.ourdomain.com@192.168.11.4

[9] 2008/02/20 15:02:51: Resolve 36490: aaaa tcp 192.168.11.4 5065

[9] 2008/02/20 15:02:51: Resolve 36490: a tcp 192.168.11.4 5065

[9] 2008/02/20 15:02:51: Resolve 36490: tcp 192.168.11.4 5065

[7] 2008/02/20 15:02:51: SIP Tx tcp:192.168.11.4:5065:

BYE sip:exchangesp1.ourdomain.com:5065;transport=Tcp;maddr=192.168.11.4 SIP/2.0

Via: SIP/2.0/TCP 192.168.11.13:4510;branch=z9hG4bK-d05437562f4762e6c579cb1bbd0c5a45;rport

From: "Extension 5000" <sip:5000@192.168.11.4>;tag=50275

To: <sip:8564@192.168.11.4;user=phone>;tag=da77a4c889

Call-ID: 4a853a28@pbx

CSeq: 10137 BYE

Max-Forwards: 70

Contact: <sip:5000@192.168.11.13:4510;transport=tcp>

RTP-RxStat: Dur=8,Pkt=187,Oct=32164,Underun=0

RTP-TxStat: Dur=7,Pkt=342,Oct=57108

P-Asserted-Identity: "Extension 5000" <sip:5000@pbxnsip216.ourdomain.com>

Content-Length: 0

 

[9] 2008/02/20 15:02:51: Dialplan: Evaluating !^7([0-9]*)@.*!sip:\1@\r;user=phone!i against 5001;phone-context=PBXnSIP-exchangesp1.ourdomain.com@192.168.11.4

[8] 2008/02/20 15:02:51: Play audio_en/ex_permission.wav

---

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Exchange really asks the PBX to call "5001;phone-context=PBXnSIP-exchangesp1.ourdomain.com", no kidding (no idea why the number must include all those strange parameters, but it is legal according to the RFC). The dial plan now must match also those strange parameters!

 

My idea for ERE in the dial plan:

 

([0-9]*);phone-context=.*

 

That should really fish out only requests that were initiated by the Exchange. Maybe a feature!

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The dial plan in the last two releases is all mucked up. For example, if we have the following dial plan:

 

Trunk Pattern

carrier 1 011([0-9]*)@.*

Not Allowed 1900([0-9]*)@.*

carrier 2 ([0-9][0-9][0-9][0-9][0-9][0-9][0-9][0-9][0-9][0-9])@.*

Deny 1900([0-9]*)@.*

 

If a person dials 5196530114 that the call will route into carrier 2 trunk. That is not the case. It seems that it matches the 011 in the strings and attempts carrier 1. This was not the case before.

 

If someone dials 1604519002, the 1900 is detected and the call is denied.

 

This is causing us major headaches as we have upgraded to the latest builds. Please fix the situation ASAP or recommend a path. We are using the Debian and Suse latest builds.

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Thanks, pbxnsip.

 

Adding the following line to the dialplan allowed Exchange to call the extension:

Call Extension - (your ERE) - \1

 

However, the call from the AA to the extension rings forever instead of being transferred to voicemail. Hitting reject on the Polycom phone should force the caller to voicemail (as it does on an extension to extension call, and as it does on our other test server coming from the AA), but instead gives the caller the audio_en/ex_wrong_id.wav 'this number could not be found'.

 

Is there more to this internally than can be fixed via the dialplan? The log files show the DECLINE from 5000 to 78564, the ACK from 5000 to 78564, then the number not found message -- no dialplan matching attempts.

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However, the call from the AA to the extension rings forever instead of being transferred to voicemail. Hitting reject on the Polycom phone should force the caller to voicemail (as it does on an extension to extension call, and as it does on our other test server coming from the AA), but instead gives the caller the audio_en/ex_wrong_id.wav 'this number could not be found'.

 

I think the problem is that the Polycom sends a 6xx code, which is a very high priority code?

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Guest Paul McCabe

A possible 2.1.6 bug...

I just noticed this issue on incoming calls from Anonymous callers to my extension. The caller is prompted for their name and the call forks to my cell phone even with 'Don't call cell phone' set. It appears only Anonymous calls ring the cell phone this way.

 

Redirection is set up on my phone as follows:

 

Do not disturb: off

Agent Logged In: on

Incoming anonymous calls: Ask for name

Hot Desking at:

Call forward all calls to:

Call forward calls when busy to:

Call forward on no answer to:

Call forward no answer timeout: Default value

--------------------------------------------------------------------------------

Cell phone number: 617899xxxx

Include the cell phone in calls to extension: Don't call cell phone

Service Flag: No specific time excluded

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I think the problem is that the Polycom sends a 6xx code, which is a very high priority code?

Hmm... looking at the logs, sorry, but I don't understand what 6xx code it is that you are referring to. That shouldn't have changed unless the phones were provisioned differently, though, correct? They are not. In case my wording was confusing, let me reword what's happening:

 

On 2.1.6 and Exch sp1:

- If I call 5001 from 5000 directly, it goes to voicemail after 5 secs as config'd. Good.

- If I call 5001 from 5000 directly and press reject on 5001 before the auto timeout to voicemail, it plays 'extension is busy, press 2 to leave a message'. Also Good.

- If I call 78564 (Exch AA), press 1 to transfer to 5001, it transfers to 5001 (with the above ERE in the dial plan), but rings forever instead of going to voicemail after 5 secs. Bad

- If (while ringing forever), I press reject on 5001 to force the call to voicemail, the 'this number could not be found' plays. Also Bad.

 

On 2.1.2 and Exch (no sp1):

- Good stuff above works as expected.

- If I call 78564 (Exch AA), press 1 to transfer to 2504, it transfers to 2504 (without ERE), rings the correct number of times, then times out to voicemail. Good!

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I have installed the 2.1.6.2447, but the license was not activated.

 

I use a Wireless NIC in order to provide the server with the MAC address, but the license was not activated.

 

The 2.0.3.1715 version, the last I am using activates the license with the wireless NIC.

 

What can I do? I want to use the 2.1.6.2447 because the new features it has.

 

Anybody can give some idea?

 

Thanks

 

Juan Acevedo

acevedo1@une.net.co

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Guest Paul McCabe

Juan,

 

I just tried a key mapped to a wireless MAC and the system did not like the key. I pasted it in a second time, hit SAVE, and it seemed to like it the second time.

I am running 2.1.6.2446 on my laptop. I did not restart the pbxctrl service...

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PbxnSIP, do you have any additional info regarding the Exchange integration issue above? Paul, I left you a voice message earlier this week to find out whether or not you had exchange sp1 working on your test boxes but haven't heard back. Can you call, e-mail, or post here with a reply?

Thanks.

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Guest Paul McCabe
PbxnSIP, do you have any additional info regarding the Exchange integration issue above? Paul, I left you a voice message earlier this week to find out whether or not you had exchange sp1 working on your test boxes but haven't heard back. Can you call, e-mail, or post here with a reply?

Thanks.

 

Hi Sean, sorry about not getting back to you. I did just play your voice mail and my answer is 'Yes' I am having trouble since we upgraded to Exchange SP1. I don't want to totally blame MS just yet as the pbxnsip software has gone through many iterations since I first started testing Exchange trunking. As an example, in the past I could make a simple call to my Exchange mailbox directly (Dial 7512) and that does not work ('Forbidden'). I can get to my mailbox if I set up a Pilot Identifer associated with the UM gateway. Lately, I have been testing to see if I send send a fax from a fax enabled softphone to my UM mailbox, but that doesn't work either. Is it SP1 or the PBX, I just don't know right now.

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Juan,

 

I just tried a key mapped to a wireless MAC and the system did not like the key. I pasted it in a second time, hit SAVE, and it seemed to like it the second time.

I am running 2.1.6.2446 on my laptop. I did not restart the pbxctrl service...

 

Paul:

 

Can you test with 2.1.6.2447? I have pasted twice and the license is not recognize

 

Thanks

Juan Acevedo

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Thanks, Paul, that's good info to have. SP1 has definitely put us in a pickle; wish it were uninstallable. Thanks for the reply, as you find out more, let me know, our Exchange Gold trial server that we have been using to get by expires in 35 days. If we don't know more soon, my only hope is that the Gold server takes the license key for our already-in-place server without complaining. Otherwise we'll have to try something else out (a second temporary x64 server with another trial?)

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