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[Announce] SAIL-2.1.11-161 Beta 2.

Offline SARK devs

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[Announce] SAIL-2.1.11-161 Beta 2.
« on: April 05, 2006, 06:36:09 PM »
Hi all,

Just to let you know we’ve just put SAIL-2.1.11-162 up onto our ftp server.  Broadly speaking, this is a maintenance release but there are some new features; -

Slightly updated look-and-feel
New feature;  Ring Groups (just create a speed-dial with multiple targets separated by whitespace).
Limited KDE iconography in the screen panels
Selintra logo in page footer is active and links to Docs pages.
Cleaned up entity delete
Added template for Aus carrier Koala
Corrected the Engin Template
Added template for Netcomm V85 IP Phone
Enhanced pop-ups on queues
Should now be correctly opening UDP Ports (4569, 5060, 10000-20000)

General bug clean-up (up to yesterday).


You can just rpm –Uvh over your existing install.   You SHOULD NOT need to run autosniff unless you are physically changing boards.

Alternatively, -ivh if you’re new to all this telephony stuff.  

Kind Regards

Selintra

Offline Tib

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[Announce] SAIL-2.1.11-161 Beta 2.
« Reply #1 on: April 07, 2006, 11:07:14 AM »
selintra

Looks nice ... I like the new icons etc.

Question ... I have a problem where sometimes when ppl on the outside line hang up the pstn line does not.

Looking at it through fop you can still see the pstn line open ... all extentions are closed as well as the voip line.

Any idea where I should look for this one ... I in Australia ... not sure if that makes a difference in setup etc.

Regards,

Tib

Offline Tib

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[Announce] SAIL-2.1.11-161 Beta 2.
« Reply #2 on: April 07, 2006, 11:28:02 AM »
this is strange

adding above ... I just revieved a call over the pstn line.

I cannot make a call out through the pstn line though ... so it is still active ... might be recieving calls still because I have call waiting on the line.

Also what is the comand to kill the service and re-start in in this case so I don't have to re-boot the whole server to free up the line again.

I tried dissconecting the phone lines and even turning off the sipura3000 but that didn't help ... This has happened about 3 times now in 2 weeks.

[edit]
Ok this is getting more strange ... just as I finished writing this post the line came back to normal.

One other thing I just noticed as well .... since the update my speed dial numbers don't work :(
Interneal Extentions work but not external numbers.

Bug report entered ... No 1218
[edit]

Regards,

Tib.

Offline Tib

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[Announce] SAIL-2.1.11-161 Beta 2.
« Reply #3 on: April 07, 2006, 11:51:08 AM »
I just noticed as well ... I fixed the conference rooms qty on last ver ... since the update I only have one conf room again.

Regards,

Tib

Offline Tib

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[Announce] SAIL-2.1.11-161 Beta 2.
« Reply #4 on: April 07, 2006, 12:47:48 PM »
selintra

This is the outpu when I try to call a speed dial no ...

Apr  7 20:41:29 WARNING[11348]: channel.c:2535 ast_request: No channel type regi
stered for ''
Apr  7 20:41:29 NOTICE[11348]: app_dial.c:1011 dial_exec_full: Unable to create
channel of type '' (cause 66 - Channel not implemented)
Apr  7 20:41:39 WARNING[3514]: channel.c:787 channel_find_locked: Avoided initia
l deadlock for '0x8876ee0', 10 retries!

And why does this popup all the time and how do I fix it ?

Apr  7 20:44:48 NOTICE[3521]: chan_iax2.c:5676 update_registry: Restricting registration for peer '5003' to 60 seconds (requested 1200)

This is poping up every 45 secs.

Extention 5003 if an extention I'm trying through to my work using cubix as the soft phone.

Regards,

Tib

Offline SARK devs

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[Announce] SAIL-2.1.11-161 Beta 2.
« Reply #5 on: April 08, 2006, 09:56:59 PM »
Hello Tib,

Glad you like 162.

Quote
I have a problem where sometimes when ppl on the outside line hang up the pstn line does not


This is not a SAIL problem per se.  It is an asterisk problem (ish).  What you are seeing is asterisk not recognising what is called a "far-end disconnect".  In other words, when the far end phone goes on-hook, asterisk is not being notified.  Now, different carriers notify far end disconnect in different ways.  Some old fashioned carriers will put a polarity reversal onto the line (very rare nowadays), more commonly they will use CPC (Calling Party Control) which is characterised by a momentary drop in line voltage (usually between 9 and 20 milliseconds).    Some others will give a busy DTMF tone others will give a montone and others, give nothing at all.  

Now, I believe you are using a Sipura 3000 to connect to your PSTN and it is here that your problem lies.  The spa3K is, for whatever reason, not detecting the far end hang-up.   You will probably need to look up  the correct disconnect settings for your carrier (there are lots of Forums dedicated to the spa3K and it's many foibles).  It may be that you may have to ask your carrier to supply you with disconnect supervision (CPC), some charge for the privilege.

Quote
I fixed the conference rooms qty on last ver ... since the update I only have one conf room again


Yeah, we rolled the new realease with one of the PC's uncommitted to SVN. It should be OK in 165.


Quote
Apr 7 20:44:48 NOTICE[3521]: chan_iax2.c:5676 update_registry: Restricting registration for peer '5003' to 60 seconds (requested 1200)


Asterisk won't allow an IAX user to register for more than 60 seconds (don't know why).  The message is benign.  However if you want it to go away set maxexpiry=60 in the friend definition for the phone.

Quote
speed-dial


This is a fault Tib, we're working on it.

Kind Regards

Selintra