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[Announce] SAIL Asterisk 2.1.10 Beta Available

Offline SARK devs

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[Announce] SAIL Asterisk 2.1.10 Beta Available
« on: March 06, 2006, 07:14:11 PM »
Hello everyone,

SME Asterisk Integration Layer (SAIL) V2R1.10, Build 113 is now ready for general test.  We put it up to our ftp site at around 18:00 (GMT) today, 6th March 2006.

New Features in 2.1.10 include Queues/Agents support and Areski CDR Stats integration plus better validation routines on non-freeform panels.  We’ve also updated the docs wiki and added a few HowTos for some of the more commonly used procedures.

Late installation notes (please read to avoid tears at bedtime) can be found at: -

http://selintra.com/docs/cgi-bin/view/Main/DownLoadPages

ftp site address can be found at: -

http://selintra.com/docs/cgi-bin/view/Main/DownLoadPages

Quickstart install documentation can be found at: -

http://selintra.com/docs/cgi-bin/view/Main/SysKwik

Full Manual can found on our Wiki at

http://www.selintra.com/docs

/docs is a Wiki so we would suggest you register with the site and update it with your comments/findings/tips.

asterisk-SME7-1.2.3, Build 100 is also ready and should be installed ahead of SAIL.

Please be patient, the rpms are big and the site is very busy.

There are three rpms.  

asterisk-SME7-1.2.3-100.i686.rpm, which includes all of the add-ons, zaptel and libpri drivers.
selintra-sail-2.1.10-113.noarch.rpm
A perl module rpm called TermReadKey which you will need for the Digium board sniffer in SAIL/Asterisk.  You don’t need this rpm if you don’t have any Digium boards.  

You don’t have to register to download the software but it would be nice if you did.  It means we can keep a rough track of you all and how many images we have out there.

WARNING.

asterisk-SME7 WILL NOT RUN WITH ANY KERNEL OTHER THAN

2.6.9-22.0.2.EL

THIS CORRESPONDS TO SME7pre2, SME7pre3 & SME7pre4 ONLY.  
IF IN DOUBT DO uname –r ON YOUR SYSTEM BEFORE ATTEMPTING THE INSTALL.


We will open up our bugzilla image later in the week for general fault reporting.

Selintra

Offline psoren

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Re: [Announce] SAIL Asterisk 2.1.10 Beta Available
« Reply #1 on: March 06, 2006, 11:14:43 PM »
Hi,

This looks very interesting, i will try it as soon as possible, but:

"asterisk-SME7-1.2.3-100.i686.rpm"

Does this mean that i won't run on my Mini-ITX board with onboard C3 CPU? This is detected as an i585 CPU and not i686.
Any chance for me and all the rest that uses these nice powersaving boards?

Per

Offline jester

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[Announce] SAIL Asterisk 2.1.10 Beta Available
« Reply #2 on: March 07, 2006, 11:37:54 AM »
Hi!

Quick question: is there any HFC-card support ?! For my home-office server want to connect my ISDN-line to a cheap HFC BRI card (Digium cards are a bit expensive).

regards,
jester.

Offline SARK devs

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[Announce] SAIL Asterisk 2.1.10 Beta Available
« Reply #3 on: March 07, 2006, 12:22:25 PM »
Quote
Does this mean that i won't run on my Mini-ITX board with onboard C3 CPU? This is detected as an i585 CPU and not i686.
Any chance for me and all the rest that uses these nice powersaving boards?


Hello Per

Asterisk-SME7 will fail if you try to bring it up on the C3.  In theory it should be possible to get Asterisk to run on the C3 by specifying 586 in the Make, but it’s not trivial because it’s also pretty hard work to get Zaptel to run in a CentOS udev environment.  The Makefiles don't work properly and we ended up having to incorporate big chunks of the Zaptel Make into our rpm.

All we can suggest is that you install Asterisk manually from sources, do the Makes with i586 specified  then download our SRPM and rebuild a new i586.rpm from the two halves.   SAIL itself will run without mods on the C3.  

Sorry Per

Selintra

Offline JonB

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[Announce] SAIL Asterisk 2.1.10 Beta Available
« Reply #4 on: March 07, 2006, 12:47:53 PM »
Unfortunately even Asterisk@Home will not run on a Via C3 without installing the .iso and having A@H fail, changing the build enviroment to 586 and rebuilding.

Works sweet once you have done that tho.

Jon
...

Offline Franco

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[Announce] SAIL Asterisk 2.1.10 Beta Available
« Reply #5 on: March 07, 2006, 02:21:24 PM »
Selintra,
On the documentation you mentioned a package for 6.5, but it's not on the FTP. Could you make that available?

Thanks,

Offline psoren

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[Announce] SAIL Asterisk 2.1.10 Beta Available
« Reply #6 on: March 07, 2006, 02:50:27 PM »
Hello Selintra

I was afraid you where going to say that.....

Quote from: "selintra"


All we can suggest is that you install Asterisk manually from sources, do the Makes with i586 specified  then download our SRPM and rebuild a new i586.rpm from the two halves.   SAIL itself will run without mods on the C3.  


I just have NO idea how to do RPM's
But i have before installed Asterisk from source and got it working.
Then i found A@H and installed that on a separate box with another C3 CPU the same way as Jon suggests, works fine.
But i would really like to integrate SME and Asterisk, but i didn't like A@H and SME7 together and was hoping yours would be better.

Thanks
Per

Offline SARK devs

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[Announce] SAIL Asterisk 2.1.10 Beta Available
« Reply #7 on: March 07, 2006, 05:57:15 PM »
Quote
Quick question: is there any HFC-card support ?! For my home-office server want to connect my ISDN-line to a cheap HFC BRI card (Digium cards are a bit expensive).


HFC-cards require Bristuff/zaphfc to be compiled into the system. Unfortunately we don’t have any BRI lines here at Selintra (almost no-one uses them in the UK anymore).  However, if anyone would like to volunteer to modify the asterisk rpm then we’d be more than happy to support you.

Offline SARK devs

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[Announce] SAIL Asterisk 2.1.10 Beta Available
« Reply #8 on: March 07, 2006, 09:26:17 PM »
Quote
Then i found A@H and installed that on a separate box with another C3 CPU the same way as Jon suggests, works fine.
But i would really like to integrate SME and Asterisk, but i didn't like A@H and SME7 together and was hoping yours would be better.


Hello Per

There is a small gift for you on our FTP site.  It would appear that one of our developers also likes the C3 chip.  No promises for the rpm but we've been playing with it this afternoon on a VIA Eden board and it seems to work OK.  :-)


Selintra

Offline JonB

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[Announce] SAIL Asterisk 2.1.10 Beta Available
« Reply #9 on: March 08, 2006, 05:39:00 AM »
Selintra,

How do you want bugs reported.

asterisk .100
selintra-sail .113


CDR Stats is not working. The database is not being populated.
CDR stats is enabled.
asterisk database exists.

Error in Asterisk is

Mar  7 18:05:58 NOTICE[11682] cdr.c: CDR simple logging enabled.
Mar  7 18:05:58 NOTICE[11682] indications.c: Removed default indication country 'nz'
Mar  7 18:05:58 ERROR[11682] res_config_mysql.c: MySQL RealTime: Failed to connect database server  on . Check debug for more info.
Mar  7 18:05:58 WARNING[11682] res_config_mysql.c: MySQL RealTime: Couldn't establish connection. Check debug.

I have also found that I cannot dial *50* or *51* from a Grandstream BudgeTone 100 sip phone. I can dial *5 but as soon as I hit the 0 or 1 the call is disconnected. If I put *50* as the code for the message button on the phone then it works fine and I am prompted for the mailbox password.

All other system keys work fine.

Jon
...

Offline SARK devs

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[Announce] SAIL Asterisk 2.1.10 Beta Available
« Reply #10 on: March 08, 2006, 06:45:14 AM »
Quote
How do you want bugs reported.


We can either open our bugzilla site up or request a category for SAIL on the contribs bugzilla.  We were going to open ours but then it occured to us that maybe it would be more convenient for the community if we used contribs.  Views?

Thanks for the info Jon - we have four or five BT102's but they all have the *50* in the message default so we didn't know about this.  

We'll have a look at the logging today.

Thanks very much Jon.

Selintra  



[/quote]

Offline SARK devs

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[Announce] SAIL Asterisk 2.1.10 Beta Available
« Reply #11 on: March 08, 2006, 10:17:58 AM »
Quote
I have also found that I cannot dial *50* or *51* from a Grandstream BudgeTone 100 sip phone


Hello Jon,

We're pretty sure this is because you have Enable Call Features set to "Yes" on your BT100.  Set it to "No" and it will work fine.   We simulated it on a GS BT102 and a GXS2000 this morning and it's the same for both.  *5xx numbers won't get transmitted to Asterisk if the Enable Call Feature is set to Yes.  We don't know why at this point but *5 is obviously of some significance to the GS software.

Selintra

Offline jester

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[Announce] SAIL Asterisk 2.1.10 Beta Available
« Reply #12 on: March 08, 2006, 12:37:36 PM »
Quote
HFC-cards require Bristuff/zaphfc to be compiled into the system. Unfortunately we don’t have any BRI lines here at Selintra (almost no-one uses them in the UK anymore).  However, if anyone would like to volunteer to modify the asterisk rpm then we’d be more than happy to support you.

I wish i could do it myself, but i'm to much of a newbie! I could compile the zaphfc kernel module for you (using the install-ZAPHFC script supplied with *) but that's about as far as my knowledge goes.

Offline SARK devs

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[Announce] SAIL Asterisk 2.1.10 Beta Available
« Reply #13 on: March 08, 2006, 02:56:48 PM »
Quote
CDR Stats is not working. The database is not being populated.


Hello Jon,

We've tracked this down.  It's a small problem in the regeneration code for one of the .conf files.  Anyway, to cut a long story short, we'll put a new rpm up later today or early tomorrow which will fix this and a few other issues we've found.

Thanks Again

Selintra

Offline NickCritten

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[Announce] SAIL Asterisk 2.1.10 Beta Available
« Reply #14 on: March 08, 2006, 07:02:06 PM »
This is absolutely bloody WONDERFUL!! :-D  :-D

Thankyou for sorting out my major headache with this!!


Your Documentation is brilliant, the installation is flawless, it even detected my X100P clone first try and set it up properly. (Which has been another headache..)
I've tried about 6 different Asterisk Distros & Live CD's and none of them are as good as this, and it integrates it into an SME server too! FAB :-)

One tiny little thing.. I just canot get X-Lite to talk to SME/SAIL.
Its probably something really stupid.. I've got no problems setting up IAX softphones, Menus, voicemails or anything else I've tried, but x-lite just refuses to play:

sip.conf
Code: [Select]

[general]
tos=0x18
localnet=192.168.1.0/255.255.255.0
context=mainmenu
maxexpirey=180
defaultexpirey=160
disallow=all
allow=alaw
allow=ulaw

;Internal IP phones

[5000]
type=friend
username=phone1
secret=5000
host=dynamic
qualify=3000
context=internal
callerid="phone1" <5000>
canreinvite=no
mailbox=5000
pickupgroup=1
callgroup=1
disallow=all
allow=alaw
allow=ulaw

;External Sip lines
<--snip-->


X-Lite Settings:
Code: [Select]

Display Name: phone1
User Name:phone1
Authorization User:<same as user name>
Password:5000
Domain/Realm: 192.168.30.3
SIP Proxy:192.168.30.3

I'm just constanly getting the following coming up under asterisk:

Code: [Select]

Connected to Asterisk 1.2.3 currently running on avatar (pid = 3361)
Mar  8 17:54:02 NOTICE[3424]: chan_sip.c:10851 handle_request_register: Registration from 'phone1 <sip:phone1@192.168.30.3>' failed for '192.168.30.29' - Username/auth name mismatch
Mar  8 17:54:04 NOTICE[3424]: chan_sip.c:10851 handle_request_register: Registration from 'phone1 <sip:phone1@192.168.30.3>' failed for '192.168.30.29' - Username/auth name mismatch
Mar  8 17:54:07 NOTICE[3424]: chan_sip.c:10851 handle_request_register: Registration from 'phone1 <sip:phone1@192.168.30.3>' failed for '192.168.30.29' - Username/auth name mismatch


Like I said it's probably something blaringly obvious, but I've literally been trying this for HOURS, googling, searching, debugging. I'm gonna go nuts soon!

Once again, fantastic work. Thankyou Very Much!
...
Nick

"No good deed goes unpunished." :-x...

Offline SARK devs

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[Announce] SAIL Asterisk 2.1.10 Beta Available
« Reply #15 on: March 08, 2006, 07:53:12 PM »
Quote
I'm gonna go nuts soon!


Hello Nick,

We sincerely hope you don't!

We'll spin up an X-Lite and take a look at for you.

Have a beer with us while you're waiting!

 :pint:

Best

Selintra

Offline Franco

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[Announce] SAIL Asterisk 2.1.10 Beta Available
« Reply #16 on: March 08, 2006, 07:54:54 PM »
@NickCritten
Two things:
[general]
localnet=192.168.30.0/255.255.255.0

For x-lite, add:
[5000]
nat=1

Offline NickCritten

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[Announce] SAIL Asterisk 2.1.10 Beta Available
« Reply #17 on: March 08, 2006, 08:17:44 PM »
Quote from: "selintra"

We'll spin up an X-Lite and take a look at for you.

Have a beer with us while you're waiting!


Cheers guys!   :pint:

Quote from: "stuntshell"
@NickCritten
Two things:
[general]
localnet=192.168.30.0/255.255.255.0

For x-lite, add:
[5000]
nat=1


Thanks Stuntshell

I've updated the sip.conf header so that the sip.conf file now shows the correct localnet (192.168.30.0/255.255.255.0)

But I'm still getting the same error!
I've tried it with nat=1 aswell but no joy.

Any other ideas?
Thanks,
...
Nick

"No good deed goes unpunished." :-x...

Offline SARK devs

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« Reply #18 on: March 08, 2006, 08:24:25 PM »
Hi Nick,

Quote
   -- Registered SIP '5002' at 192.168.1.70 port 5060 expires 180
    -- Saved useragent "X-Lite release 1105x" for peer 5002
    -- Executing AGI("SIP/5002-874a", "selintra|*56*") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/selintra
    -- AGI Script Executing Application: (Playback) Options: (vm-extension)
    -- Playing 'vm-extension' (language 'en')
    -- Playing 'digits/5' (language 'en')
    -- Playing 'digits/0' (language 'en')
    -- Playing 'digits/0' (language 'en')
    -- Playing 'digits/2' (language 'en')
    -- AGI Script selintra completed, returning 0
  == CDR updated on SIP/5002-874a


Er... Ours came straight up and played back its extension quite happily after *56*.

Here's our settings - first from our generated SIP entry in Extensions panel
Quote
type=friend
username=test-xlite
secret=5002
host=dynamic
qualify=3000
context=internal
callerid="test-xlite" <5002>
canreinvite=no
mailbox=5002
pickupgroup=1
callgroup=1
disallow=all
allow=gsm
allow=alaw
allow=ulaw


Here it is in raw sip.conf

Quote
[general]
tos=0x18
localnet=192.168.1.0/255.255.255.0
context=mainmenu
maxexpirey=180
defaultexpirey=160
disallow=all
allow=gsm
allow=alaw
allow=ulaw
.
.
.
.
.
[5000]
type=friend
username=test1
secret=5000
dtmfmode=rfc2833
host=dynamic
qualify=3000
context=internal
callerid="test1" <5000>
canreinvite=no
mailbox=5000
pickupgroup=1
callgroup=1
disallow=all
allow=gsm
allow=alaw
allow=ulaw


Here are the settings in the phone...

Quote

enabled: yes
Display name: Fred
Username: 5002
Authorization User: 5002
Password: 5002
Domain/Realm 192.168.1.214
SIP Proxy: 192.168.1.214
Out Bound Proxy: 192.168.1.214
Use Outbound Proxy: Default
Register: Default
Voicemail: SIP URL
Forward: SIP URL


Only obvious difference is that we're running SAIL in Thruput mode on this test server (see Globals - compression strategy).  We tried it on Fidelity as well and it worked fine.

Kind Regards

Selintra

Offline JonB

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« Reply #19 on: March 08, 2006, 08:36:55 PM »
Nick,

The Username needs to be the same as the Auth name i.e your extension number.

The logs were telling you that.

Jon
...

Offline NickCritten

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« Reply #20 on: March 08, 2006, 09:04:21 PM »
Quote from: "JonB"
Nick,

The Username needs to be the same as the Auth name i.e your extension number.

The logs were telling you that.

Jon


HOLY CRAP its working!!!  :-D

OK so the username field in sip.conf isn't the same as the username field in X-lite :-?  Oh well....

Thanks Everyone!!
Much Beerage to celebrate methinks! :pint:

By the way, Why do I need NAT enabled on the SIP device if it's on my LAN?  I noticed from the SIP debugs that it seems to be trying to communicate via my gateway SME Servers EXTERNAL IP, instead of going straight to the device if I have NAT disabled. Whats up with that?
...
Nick

"No good deed goes unpunished." :-x...

Offline JonB

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[Announce] SAIL Asterisk 2.1.10 Beta Available
« Reply #21 on: March 09, 2006, 11:55:39 AM »
Selintra,

There is still some bugs with the Asterisk port opening.

Brand new install on pre4 today

asterisk-SME7   .100
selitra-sail   .113

Data base properties are

Code: [Select]
asterisk=service|UDPorts|4569,5060,10000:20000|status|enabled

should be

Code: [Select]
asterisk=service|UDPPorts|4569,5060,10000:20000|status|enabled|access|public

note the P is still missing in UDPPorts and |access|public is missing.

However this is all a moot point at the moment as SME7 does not currently support opening multiple ports using UDPPorts or TCPPorts

See Bug 989 that I raised earlier today.

The asterisk UDP ports will need to be opened individually at the moment if people want external access to *

Code: [Select]
config set IAX2 service status enabled access public UDPPort 4569
config set SIP service status enabled access public UDPPort 5060
config set RTP service status enabled access public UDPPort 10000:20000  
signal-event remoteaccess-update


Jon
...

Offline JonB

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« Reply #22 on: March 09, 2006, 12:08:45 PM »
Quote from: "selintra"
We're pretty sure this is because you have Enable Call Features set to "Yes" on your BT100.  Set it to "No" and it will work fine.   We simulated it on a GS BT102 and a GXS2000 this morning and it's the same for both.  *5xx numbers won't get transmitted to Asterisk if the Enable Call Feature is set to Yes.  We don't know why at this point but *5 is obviously of some significance to the GS software.


Cheers, that was the problem. Is there a hardware wiki page where this sort of information can be placed as the Enable Call Features is enabled by default on the GS. Its bound to catch someone else out.

Jon
...

Offline SARK devs

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[Announce] SAIL Asterisk 2.1.10 Beta Available
« Reply #23 on: March 09, 2006, 01:37:43 PM »
Code: [Select]

asterisk=service|UDPorts|4569,5060,10000:20000|status|enabled


Ouch!  Sorry Jon.  Too many late nights here.  We've just put a 117 release up to the ftp site which we really, really promise fixes the ports.  

The lack of port range opening in 7.0 is a worry tho'. In truth for most users you can get away with just having 5060 open as long as your carrier is running session border control.  However it does mean that remote sip phones are a complete no-no unless you want to sit and laboriously open a bunch of ports manually.  IAX, of course, is fine.

We'll put a page up on the wiki for softphone/hardphone settings.

Selintra

       [/quote]

Offline psoren

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« Reply #24 on: March 19, 2006, 11:53:28 PM »
Quote from: "selintra"
Quote
Then i found A@H and installed that on a separate box with another C3 CPU the same way as Jon suggests, works fine.
But i would really like to integrate SME and Asterisk, but i didn't like A@H and SME7 together and was hoping yours would be better.


Hello Per

There is a small gift for you on our FTP site.  It would appear that one of our developers also likes the C3 chip.  No promises for the rpm but we've been playing with it this afternoon on a VIA Eden board and it seems to work OK.  :-)


Selintra


Selintra,

Sorry for the late reply but my job takes a lot of my time off.... :-D
I'm downloading now and will start the "fight" soon, i will let you know how it goes.
Thanks....

Per

Offline CharlieBrady

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Re: [Announce] SAIL Asterisk 2.1.10 Beta Available
« Reply #25 on: March 23, 2006, 07:47:08 PM »
Quote from: "selintra"

WARNING.

asterisk-SME7 WILL NOT RUN WITH ANY KERNEL OTHER THAN

2.6.9-22.0.2.EL


I'd suggest that you package the kernel specific portion of asterix separately. I presume it's one or more kernel modules. Let me know if you'd like help in doing that.

RHEL/CentOS release new kernel updates from time to time, and SME 7 pick them up from time to time. It doesn't make sense for everyone to get a new asterix install every time that happens.

Offline SARK devs

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« Reply #26 on: April 01, 2006, 02:23:09 PM »
Quote
RHEL/CentOS release new kernel updates from time to time, and SME 7 pick them up from time to time. It doesn't make sense for everyone to get a new asterix install every time that happens.


Hi Charlie,

We hadn’t ignored this, we’ve just been giving it some thought.

We agree and thanks for the offer of help. We suggest we do the split as follows…
    Asterisk core mainline + add-ons + sample configs (about 10-12Mb in all)
    Asterisk ZAP (TDM) drivers; Zaptel & libpri  (a few hundred K)
    Asterisk Sounds (about 8Mb)


We'll also remove any SAIL dependencies so that the rpms are of general use to anyone wanting to use asterisk on SME server.  Finally, if you wish, we’ll rename them along the following lines;

smeserver-asterisk-component.1.2.3-release-num.arch.rpm

Suggest we reset the release num to 1. So first set will be

smeserver-asterisk-main.1.2.3-1.i686.rpm
smeserver-asterisk-zappri.1.2.3-1.i686.rpm
smeserver-asterisk-sounds.1.2.3-1.noarch.rpm

 
After we’re done we’ll hand them over to contribs.

Thoughts/comments?

Kind Regards

Selintra

Offline CharlieBrady

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« Reply #27 on: April 01, 2006, 05:00:06 PM »
Quote from: "selintra"

Suggest we reset the release num to 1. So first set will be

smeserver-asterisk-main.1.2.3-1.i686.rpm
smeserver-asterisk-zappri.1.2.3-1.i686.rpm
smeserver-asterisk-sounds.1.2.3-1.noarch.rpm


Please be sure to include one or more src.rpm file(s). That will allow others to contribute to the development work, but will also allow others to comply with their GPL obligations.

Offline SARK devs

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[Announce] SAIL Asterisk 2.1.10 Beta Available
« Reply #28 on: April 01, 2006, 09:10:19 PM »
Quote
Please be sure to include one or more src.rpm file(s). That will allow others to contribute to the development work, but will also allow others to comply with their GPL obligations.


Yes of course.  It was implicit in our offer to do the work. However your point is well taken and we're only too happy to help.  :-)


Kind Regards

Selintra

Offline gregswallow

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[Announce] SAIL Asterisk 2.1.10 Beta Available
« Reply #29 on: April 02, 2006, 07:08:38 PM »
Selintra (Jeff?), did you get my email with the username/password for the FTP directory on contribs.org.  I'll set up a space in the bug tracker here as well in the SME Contribs category called smeserver-asterisk.  Do you have a bugzilla account set up that I can set as the default assignee for bugs?

[EDIT] I see you did have an account set up.

Bugs can now be submitted for smeserver-asterisk (asterisk-SME7) here:
http://bugs.contribs.org/enter_bug.cgi?product=SME%20Contribs&component=smeserver-asterisk

...and for selintra-sail here:
http://bugs.contribs.org/enter_bug.cgi?product=SME%20Contribs&component=selintra-sail

Bugs for both can be seen here:
http://bugs.contribs.org/buglist.cgi?product=SME+Contribs&component=selintra-sail&component=smeserver-asterisk

And probably soon the files will all be here:
http://mirror.contribs.org/smeserver/contribs/selintra/

Thanks, sorry for the delay setitng all that up.

Offline SARK devs

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[Announce] SAIL Asterisk 2.1.10 Beta Available
« Reply #30 on: April 02, 2006, 08:25:53 PM »
Thanks Greg,

We really appreciate the help when you must be up to yours eyes in the 7.0 roll-out.


Thanks again

Jeff@selintra.com

Offline gregswallow

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[Announce] SAIL Asterisk 2.1.10 Beta Available
« Reply #31 on: April 03, 2006, 03:37:18 AM »
No problem.  Any bugs that you'd like the dev team's advice on or help with just add a cc: in the bug report to bugteam-at-lists.contribs.org.