[asterisk-users] Asterisk Tag Berlin live notes page

[randulo]

Hi,

If you're interested in what's happening in Berlin at Asterisk Tag for
the next few days, you can look here:

 http://www.scribblelive.com/Thread.aspx?Id=815

I will try to keep an event trail going and notes if interviews are
posted. Anyone can post questions and I don't think there's even a
need to create an account or log in;

You can also follow asterisktag on some of these

http://twitter.com/asterisktag
http://twitter.com/randulo
http://twitter.com/wintermeyer

YATS!

IF YOU ARE IN BERLIN please watch this space and contribute:
http://www.scribblelive.com/Thread.aspx?Id=815

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Re: [asterisk-users] [asterisk-biz] New York Asterisk Users

[C F]

I aam in South Jersey would love to participate

On 5/24/08, | dave cantera | <[EMAIL PROTECTED]> wrote:
> dean,
> I am an active member of AUG NYC... you can email me off list for any info
> you need.
>
> also, I am preparing to start a south jersey * UG.  the phila group is
> waning...
>
> thanks,
> daveC
>
> Dean Collins wrote:
>>
>> This is an email to all New York based Asterisk users.
>>
>>
>>
>> For some time it's been bugging me that we don't have a local contact
>> point/user community. If you are involved in Asterisk and in NY/NJ shoot
>> me an email, I'm going to try and revitalize either meetup.com or some
>> other shared environment for Asterisk users in NY.
>>
>>
>>
>> Shoot me an email and once I get an idea of how many Asterisk users there
>> are in NY we'll work out what to do from there.
>>
>>
>>
>>
>>
>>
>>
>> Cheers,
>> Dean
>>
>>
>>
>> 
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>>
>> asterisk-biz mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-biz
>>
>> 
>> No virus found in this incoming message.
>> Checked by AVG.
>> Version: 7.5.524 / Virus Database: 269.23.21 - Release Date: 05/19/2008
>> 12:00 AM
>>
>
>
> --
> My wife's sister is in California.
> I should buy her a Videophone2008!
>
> Truly, The Next Best Thing to Being There!
> --
>
> WorldWideVideoPhones.com
> 856.380.0894
>
>
>

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Re: [asterisk-users] Error after upgrading from 1.2.18 to 1.4.20

[Freddi Hansen]

Hi I better answer my own post.

I went to the code and the issue is in q931.c

/* wait for a RELEASE so that sufficient time has passed
for the inband audio to be heard */
   
  if (c->progressmask & PRI_PROG_INBAND_AVAILABLE)
break;
  
Changing this line to a comment makes the 1.4 work exactly as 1.2 for 
this issue.
I think that this line should only be executed on 'outbound' pri calls, 
not on inbound.

Freddi
 


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[asterisk-users] Error after upgrading from 1.2.18 to 1.4.20

[Freddi Hansen]

scenario is incoming calls on ZAP (TE410p  euro isdn) to SIP (or any 
other channel) and call is answered.

When I hangup on the ISDN side on the 1.2 then the SIP hangs up to 
immidiatly so everything is fine (se short pri debug below).

When I do the same on 1.4.20 then it take more than 30 seconds to 
disconnect when the ISDN calling party hangs up

I have tried to use the use the 1.4 zaptel with the 1.2 installation, 
that works perfect to so to me it looks like a libpri or chan_zap problem.
I do hope though that some has a hint to where I scre... up.

Freddi


OK trace from 1.2

< Protocol Discriminator: Q.931 (8)  len=13
< Call Ref: len= 2 (reference 7603/0x1DB3) (Originator)
< Message type: DISCONNECT (69)
< [08 02 80 90]
< Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0) 0: 0   
Location: User (0)
<  Ext: 1  Cause: Normal Clearing (16), class = Normal 
Event (1) ]
< [1e 02 82 88]
< Progress Indicator (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0) 
0: 0   Location: Public network serving the local user (2)
<   Ext: 1  Progress Description: Inband 
information or appropriate pattern now available. (8) ]
-- Processing IE 8 (cs0, Cause)
-- Processing IE 30 (cs0, Progress Indicator)
-- Channel 0/14, span 1 got hangup request
  == Spawn extension (Pstn-incoming-fwd, 77348855, 2) exited non-zero on 
'Zap/14-1'
NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Disconnect Indication, 
peerstate Disconnect Request
 > Protocol Discriminator: Q.931 (8)  len=9
 > Call Ref: len= 2 (reference 7603/0x1DB3) (Terminator)
 > Message type: RELEASE (77)
 > [08 02 81 90]
 > Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0) 0: 0   
Location: Private network serving the local user (1)
 >  Ext: 1  Cause: Normal Clearing (16), class = Normal 
Event (1) ]
-- Hungup 'Zap/14-1'
< Protocol Discriminator: Q.931 (8)  len=5
< Call Ref: len= 2 (reference 7603/0x1DB3) (Originator)
< Message type: RELEASE COMPLETE (90)
NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null
NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null
fonet2*CLI

BAD TRACE from 1.4  below:

fonet3*CLI>
< Protocol Discriminator: Q.931 (8)  len=13
< Call Ref: len= 2 (reference 7582/0x1D9E) (Originator)
< Message type: DISCONNECT (69)
< [08 02 80 90]
< Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)  Spare: 0  
Location: User (0)
<  Ext: 1  Cause: Normal Clearing (16), class = Normal 
Event (1) ]
< [1e 02 82 88]
< Progress Indicator (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard 
(0)  0: 0  Location: Public network serving the local user (2)
<   Ext: 1  Progress Description: Inband 
information or appropriate pattern now available. (8) ]
-- Processing IE 8 (cs0, Cause)
-- Processing IE 30 (cs0, Progress Indicator)
q931.c:3779 q931_receive: call 7582 on channel 13 enters state 12 
(Disconnect Indication)
fonet3*CLI>

~ 30 SECOND DELAY HERE.

fonet3*CLI>
fonet3*CLI>
< Protocol Discriminator: Q.931 (8)  len=9
< Call Ref: len= 2 (reference 7582/0x1D9E) (Originator)
< Message type: RELEASE (77)
< [08 02 80 90]
< Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)  Spare: 0  
Location: User (0)
<  Ext: 1  Cause: Normal Clearing (16), class = Normal 
Event (1) ]
-- Processing IE 8 (cs0, Cause)
q931.c:3754 q931_receive: call 7582 on channel 13 enters state 0 (Null)
-- Channel 0/13, span 2 got hangup, cause 16
  == Spawn extension (Pstn-incoming-fwd, 77348855, 2) exited non-zero on 
'Zap/44-1'
-- Executing [EMAIL PROTECTED]:1] Hangup("Zap/44-1", "") in new stack
  == Spawn extension (Pstn-incoming-fwd, h, 1) exited non-zero on 'Zap/44-1'
NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Release 
Request
 > Protocol Discriminator: Q.931 (8)  len=9
 > Call Ref: len= 2 (reference 7582/0x1D9E) (Terminator)
 > Message type: RELEASE COMPLETE (90)
 > [08 02 81 90]
 > Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)  Spare: 0  
Location: Private network serving the local user (1)
 >  Ext: 1  Cause: Normal Clearing (16), class = Normal 
Event (1) ]
NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null
NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null
-- Hungup 'Zap/44-1'fonet3*CLI>






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Re: [asterisk-users] Manual Wardialer

[Matthew Gibson]

On Sat, Apr 26, 2008 at 7:13 PM, Brian J. Murrell <[EMAIL PROTECTED]>
wrote:

> On Sat, 2008-04-26 at 18:41 -0400, Andreas van dem Helge wrote:
> > Does anyone have a script for manual wardialer for asterisk? not sure
> >  if "wardialer" is the correct term but basically I want to call X
> >  number say 555- through 555-0050 and be able to listen to each
> >  call and when I hang up or press a key it will dial the next number
> >  for me. I guess sort of like "scanning" an exchange but I want to be
> >  on the line and if possible complete / talk on certain calls.
>

Legal issues aside, have you tried this?

http://www.softwink.com/iwar/

Thanks,
Matt
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Re: [asterisk-users] Manual Wardialer

[Eric Wieling]

I would say anyone in the USA that is on the donotcall.gov list should 
also be excluded -- unless you either qualify as an exemption or like 
paying $10k for each violation.

Brian J. Murrell wrote:
> On Sat, 2008-04-26 at 18:41 -0400, Andreas van dem Helge wrote:
>> Does anyone have a script for manual wardialer for asterisk? not sure
>>  if "wardialer" is the correct term but basically I want to call X
>>  number say 555- through 555-0050 and be able to listen to each
>>  call and when I hang up or press a key it will dial the next number
>>  for me. I guess sort of like "scanning" an exchange but I want to be
>>  on the line and if possible complete / talk on certain calls.
> 
> Hrm.  I wonder what you could possibly want that for?  Do you mind if
> include my area code and exchange in your script as an exception to your
> "scanning"?  Anyone else care to be on that exception list?  Actually,
> it might be more efficient for me to ask who cares to be on the list to
> be scanned.  Anyone?  Anyone at all?  ;-)


-- 
Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, 
T-1, PRI, Frame Relay, Linux, and network design.  Based near 
Birmingham, AL.  Now accepting clients worldwide.

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Re: [asterisk-users] Manual Wardialer

[Brian J. Murrell]

On Sat, 2008-04-26 at 18:41 -0400, Andreas van dem Helge wrote:
> Does anyone have a script for manual wardialer for asterisk? not sure
>  if "wardialer" is the correct term but basically I want to call X
>  number say 555- through 555-0050 and be able to listen to each
>  call and when I hang up or press a key it will dial the next number
>  for me. I guess sort of like "scanning" an exchange but I want to be
>  on the line and if possible complete / talk on certain calls.

Hrm.  I wonder what you could possibly want that for?  Do you mind if
include my area code and exchange in your script as an exception to your
"scanning"?  Anyone else care to be on that exception list?  Actually,
it might be more efficient for me to ask who cares to be on the list to
be scanned.  Anyone?  Anyone at all?  ;-)

b.



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Re: [asterisk-users] [asterisk-biz] New York Asterisk Users

[| dave cantera |]

dean,
I am an active member of AUG NYC... you can email me off list for any
info you need.

also, I am preparing to start a south jersey * UG.  the phila group is
waning...

thanks,
daveC

Dean Collins wrote:

  
  
  
  
  
  
  
  This is an email to all New
York
based Asterisk users.
   
  For some time it’s been
bugging me that we don’t
have a local contact point/user community. If you are involved in
Asterisk and
in NY/NJ shoot me an email, I’m going to try and revitalize either
meetup.com or some other shared environment for Asterisk users in NY.
   
  Shoot me an email and
once I get an idea of how many
Asterisk users there are in NY we’ll work out what to do from there.
   
   
  
  
Cheers,
Dean 
   
  
  

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No virus found in this incoming message.
Checked by AVG. 
Version: 7.5.524 / Virus Database: 269.23.21 - Release Date: 05/19/2008 12:00 AM
  


-- 
My wife's sister is in California.  
I should buy her a Videophone2008!

Truly, The Next Best Thing to Being There!
--

WorldWideVideoPhones.com
856.380.0894






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Re: [asterisk-users] Asterisk Database Handling

[Matthew J. Roth]

Tilghman and Jay,

Thanks for the licensing advice.  If anyone is interested in replicate, 
I'm now ready to distribute it under the GPL.

Regards,

Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer


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Re: [asterisk-users] One way Speech issue - only in PAP2T to SIP device attached to Asterisk but not PAP2T to Voip service provider

[Stefan Schmidt]

Shaun schrieb:
> Hi All, 
> 
> This is puzzling me greatly. 
> 
> The setup: PAP2T over ADSL registers to Asterisk 1.4?using SIP. Attached to 
> Asterisk are SIP clients. Codec throughout G729 (only have 1 license on 
> Asterisk server loaded though). When calling the SIP clients from PAP2T I 
> can't hear them but they can hear me. 
>  
> If I call from PAP2T through Asterisk using?IAX2 to a VOIP provider there is 
> speach in both directions! 
>  
> Any suggestions? 
>  
> Thanks Shaun
> 

check your firewall/nat settings.

If your setup will work for around 5 minutes after you have rebooted the 
pap2t then you have to active the nat keep alive and nat mapping service 
  in the pap2t.

best regards

steve smith

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Re: [asterisk-users] (Newbie)How to reduce security risks inopening IAX & Sip Ports

[Shaun]

Dear Tzafrir,

Thank you for the kind response. Will do further searching, but my initial
findings are that very little is available on the net regarding the actual
ports.

Its on a public ip hosted at an isp and have full control over machine.
There is one other port that needs to be opened to handle an sms application
links and outbound smtp traffic.
Running CentOS 5 .

My main issue is around securing the system after having opened up for SIP
an IAX traffic.

Any help is greatly appreciated.
Thanks and regards
Shaun Wingrin 


Sent via my BlackBerry from Vodacom - let your email find you!


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[asterisk-users] One way Speech issue - only in PAP2T to SIP device attached to Asterisk but not PAP2T to Voip service provider

[Shaun]

Hi All, 

This is puzzling me greatly. 

The setup: PAP2T over ADSL registers to Asterisk 1.4?using SIP. Attached to 
Asterisk are SIP clients. Codec throughout G729 (only have 1 license on 
Asterisk server loaded though). When calling the SIP clients from PAP2T I can't 
hear them but they can hear me. 
 
If I call from PAP2T through Asterisk using?IAX2 to a VOIP provider there is 
speach in both directions! 
 
Any suggestions? 
 
Thanks Shaun

Sent via my BlackBerry from Vodacom - let your email find you!


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Re: [asterisk-users] excessive bounces???

[Mark Hamilton]

I get this once a week myself.

 Original Message 
Subject: Re: [asterisk-users] excessive bounces???
From: "Michael Graves" <[EMAIL PROTECTED]>
Date: Sat, May 24, 2008 11:32 am
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<[EMAIL PROTECTED].com>

I got this also. 

In fact, it happens quite a bit for me. My mail host does not relay
email in foreign languages, which can generate a bounce. When someone
posts to the list in some asian (non roman character set) language I
always get bumped.

Michael

On Sat, 24 May 2008 16:18:29 +0100, Steve Howes wrote:

>I got it too. I wouldn't worry.
>
>-Original Message-
>From: "Doug Lytle" 
>To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
>Sent: 24/05/08 02:52 PM
>Subject: [asterisk-users] excessive bounces???
>
>My membership has been disabled for excessive bounces?  Are we having 
>issues again with the list?  I've check both my primary and secondary 
>MTA and show no issues with mail.
>
>Doug
>
>-- 
>Ben Franklin quote:
>
>"Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety."
>
>
>
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>

--
Michael Graves
mgravesmstvp.com
http://blog.mgraves.org
o713-861-4005
c713-201-1262
sip:mjgraves@pixelpower.onsip.com
skype mjgraves
54245@fwd.pulver.com



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Re: [asterisk-users] excessive bounces???

[Michael Graves]

I got this also. 

In fact, it happens quite a bit for me. My mail host does not relay
email in foreign languages, which can generate a bounce. When someone
posts to the list in some asian (non roman character set) language I
always get bumped.

Michael

On Sat, 24 May 2008 16:18:29 +0100, Steve Howes wrote:

>I got it too. I wouldn't worry.
>
>-Original Message-
>From: "Doug Lytle" <[EMAIL PROTECTED]>
>To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
>
>Sent: 24/05/08 02:52 PM
>Subject: [asterisk-users] excessive bounces???
>
>My membership has been disabled for excessive bounces?  Are we having 
>issues again with the list?  I've check both my primary and secondary 
>MTA and show no issues with mail.
>
>Doug
>
>-- 
>Ben Franklin quote:
>
>"Those who would give up Essential Liberty to purchase a little Temporary 
>Safety, deserve neither Liberty nor Safety."
>
>
>
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>
>asterisk-users mailing list
>To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>

--
Michael Graves
mgravesmstvp.com
http://blog.mgraves.org
o713-861-4005
c713-201-1262
sip:[EMAIL PROTECTED]
skype mjgraves
[EMAIL PROTECTED]



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Re: [asterisk-users] Asterisk not answering calls

[Joseph L. Casale]

>I can make outbound calls, but when I call any of my did's they ring busy.
>A tcpdump at the Asterisk server shows no inbound traffic and neither does sip 
>set debug
>show any activity. I have the providers routing set to sip user, I am using 
>that user in my registration.
>
>Anyone know if there is anything obvious that I missed?
>Thanks!
>jlc


Additionally,
I noticed this in sip set debug:

<--- SIP read from [sip_providers_ip]:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP [sip_providers_ip]:5060;branch=z9hG4bK621e7af8;rport=5060
From: "asterisk" ;tag=as0ca55ebf
To: ;tag=6da5cb3c58ecfc1b91772f44357856fa.a539
Call-ID: [EMAIL PROTECTED]
CSeq: 102 OPTIONS
Accept: */*
Accept-Encoding:
Accept-Language: en
Supported:
Content-Length: 0

Is there any significance to the "From:", could that be the reason the user is 
not registered at the provider right possibly?
Thanks!
jlc

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Re: [asterisk-users] excessive bounces???

[Steve Howes]

I got it too. I wouldn't worry.

-Original Message-
From: "Doug Lytle" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Sent: 24/05/08 02:52 PM
Subject: [asterisk-users] excessive bounces???

My membership has been disabled for excessive bounces?  Are we having 
issues again with the list?  I've check both my primary and secondary 
MTA and show no issues with mail.

Doug

-- 
Ben Franklin quote:

"Those who would give up Essential Liberty to purchase a little Temporary 
Safety, deserve neither Liberty nor Safety."



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[asterisk-users] OT: Re: Dear asterisk-users@lists.digium.com May 80% 0FF

[Philipp Kempgen]

Hey Steve,

Steve Totaro schrieb:
> Darn, it was 87% off just yesterday!

If you're interested I could forward some offers for
Rolex watches, cheap software etc.  :-P

Grüße,
Philipp Kempgen
-- 
Asterisk-Tag.org 2008, 26.-27. Mai   ->  http://www.asterisk-tag.org
Amooma GmbH - Bachstr. 126 - 56566 Neuwied  ->  http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998

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Re: [asterisk-users] Incoming calls not being answered by asterisk

[Roberto Milani]

Ciao Roand

I think you should buy a book and do some reading to build up your  
knowledge.


but in the meantime try something like this in the dialplan  
(extensions.conf)


exten => PSTN,1,Answer() ; Answer inbound calls or internal miss-dials
exten => PSTN,2,Playback(silence/1)
exten => PSTN,3,Background(enter-ext-of-person) ; input an extension
exten => PSTN,n,WaitExten(20) ; Adjust wait, default 5 sec
exten => PSTN,n,Goto(internal,${EXTEN},1) ; Goto the correct extension
exten => PSTN,n,Hangup() ; End the call

where PSTN is your sipura SIP name (1002 i think)

Ciao
Roberto


On May 24, 2008, at 3:09 AM, RoLaNd RoLaNd wrote:


Hello all,

ive got the following setup currently:


   __Sipura 3102-PSTN
  |
Lan |
  |
  |__asterisk

i configured both asterisk and pstn to be able to receive/make calls  
through each other using sip of course..
but the thing is i want asterisk that when it receives an incoming  
call from sipura, to answer it, play msg that i recorded and wait  
for the caller to dial in an extension, where it would transfer the  
caller to that exntension, and in case the extension owner isnt  
available to answer it would direct him to his voicemail(tht i dont  
know how to set yet), and in case the caller didnt dial any  
extension in a certain amount of time, it automaticly directs it to  
a specific extensions i'd specify..


i tried the examples given in lots of forums and so on none of them  
worked, the phone keeps on ringing with every incomign dial plan ive  
specified without asterisk answering it..
the thing i did is that sipura directs incoming calls to 1002, so  
ive set the context of 1002 in sip.conf to a dial plan of [incoming- 
sipura] and ive set the commands i mentioned earlier tht i took out  
of those forums.. but theyre not working!!!


anyone has an example i could go on with ?
any help would be apreciated:)

Discover the new Windows Vista Learn more!  
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Re: [asterisk-users] Dear asterisk-users@lists.digium.com May 80% 0FF

[Rob Hillis]

Steve Totaro wrote:

Darn, it was 87% off just yesterday!


  
But with all that wonderful value-adding spam, it's worth paying more 
for isn't it?


(then again, I guess that very much depends on /what/ you're paying for!)
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Re: [asterisk-users] reload stopping EVERYTHING on CLI and causing havoc.

[Mark Hamilton]

Another case of Extreme Asterrhea.
http://lists.digium.com/pipermail/asterisk-users/2008-May/212427.html


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Al Baker
Sent: May 24, 2008 4:35 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] reload stopping EVERYTHING on CLI and causing
havoc.

Quote "

Oh and also, in my implementation there are no queues. It seems to be
>> not related, I've had it in EVERY version of Asterisk I've used."

 Hmmm- maybe this should be mentioned in the next "is * Really Good Thread
?"


Mark Hamilton wrote:
> Same here.
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Sherwood
> McGowan
> Sent: May 22, 2008 4:16 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] reload stopping EVERYTHING on CLI and
causing
> havoc.
>
> Steve Totaro wrote:
>   
>> On Thu, May 22, 2008 at 3:56 PM, Sherwood McGowan
>> <[EMAIL PROTECTED]> wrote:
>>   
>> 
>>> Steve Totaro wrote:
>>> 
>>>   
 On Thu, May 22, 2008 at 2:02 PM, Sherwood McGowan
 <[EMAIL PROTECTED]> wrote:

   
 
> Mark Hamilton wrote:
>
> 
>   
>> Hi,
>>
>> Yesterday I made a change in queues.conf and so tried doing a reload
>> app_queue.so in the CLI. (Using 1.4.18). It didn't seem to do
>> anything, infact all action on CLI stopped.
>>
>> Then, I did a reload. Same thing.
>>
>> After that there was no other way.. because even stop now wouldn't
>> work, so I did a service asterisk restart
>>
>> And then asterisk kept giving the same thing on prompt "Died
>> successfully" and all that it usually says when you issue a stop now,
>> except it kept showing that on root prompt after doing a service
>> asterisk restart.
>>
>> Did a killall asterisk, and finally it stopped.
>>
>> Then started asterisk service. It was fine.
>>
>> Did a full restart at night, and it was fine.
>>
>> NOW, I wanted to do a reload again today mid-day when in full use,
and
>> it still didn't work, and ALL of the above happened again.
>>
>> --
>>
>> How do I diagnose what's causing this?
>>
>> Thanks,
>>
>> Mark.
>>
>>
>> 
> 
>   
>> ___
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>>   
>> 
> I've had this problem before, haven't debugged it. I definitely look
> forward to hearing what is said about this.
>
> Example from my recent experience, I wanted to restart the server and
>   
> so
>   
> did
>
> pbx0*CLI> restart now
>
> But nothing happened...system continued to allow calls to take place.
> I've found that sometimes exiting and reconnecting to the CLI helps,
>   
> but
>   
> there have been a couple occasions where NOTHING would allow the
server
> to restart save for a reboot. Even killall asterisk didn't kill the
> process
>
> Sherwood McGowan
>
>
> 
>   
 You are using Asterisk 1.2.x?  I have seen this many, many times.

 Sometimes the CLI becomes unresponsive, sometimes queues crap out or
 stops delivering calls to agents, sometimes it just takes a bit and
 then becomes responsive again.

 The rule of thumb is don't reload queues when there are people in
 queue, at least that seems to eliminate the problems I have seen.
 Makes sense too.

 Not sure if it is fixed in 1.4.

 Thanks,
 Steve Totaro

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>>> Oh and also, in my implementation there are no queues. It seems to be
>>> not related, I've had it in EVERY version of Asterisk I've used.
>>>
>>> 
>>>   
>> I have observed it on repeated general reloads on all versions.
>> That's why I don't reload very much, only planned.
>>
>> Thanks,
>> Steve Totaro
>>
>> ___
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>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>   
>> 
> My problem exists even when issuing a restart now or stop 

Re: [asterisk-users] Asterisk semi-hangs

[Mark Hamilton]

Welcome to the club.
http://lists.digium.com/pipermail/asterisk-users/2008-May/212281.html ->
Another similar issue.
http://bugs.digium.com/view.php?id=12709 -> Bug report for it.

Mark.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Carles Pina i
Estany
Sent: May 24, 2008 7:56 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Asterisk semi-hangs


Hello,

(Note: again, I'm asking for experiences/suggestions, because
we have a problem in some environment that it's quite difficult to
debug, test, etc.)

I have seen, during last year, that some Asterisk has hangup up. I mean,
not crashing, we could access to Asterisk console, but phones couldn't
call. Doing "dial [EMAIL PROTECTED]" was not doing/showing anything, even
with verbose setted up as 99. This was in Asterisk 1.2

Now, in a couple of Asterisk 1.4 boxes (I think that Asterisk
1.4.19.2?), this has happened again. Not often, every some months maybe,
etc. One installation is only using VoIP, so I don't think that it's a
Zaptel, Misdn, etc. problem.

Even typing "stop now" nothing is happening, killall asterisk either,
only killall -9 asterisk. 

We use a standard installation (no extra modules, etc.)

Debian distribution, our Kernel.

Has anyone some experience like this? Anything to tune? or something
to see? (logs doesn't say a lot, and hard disk is not full :-) )

We would like to avoid that this is happening again, understand why is
happening, etc.

Thank you,

-- 
Carles Pina i EstanyGPG id: 0x8CBDAE64
http://pinux.info   Manresa - Barcelona

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Re: [asterisk-users] reload stopping EVERYTHING on CLI and causing havoc.

[Mark Hamilton]

Al, 

Yup, it seems so. Unanswered problems.
I've started a bug report, if you're interested:
http://bugs.digium.com/view.php?id=12709

Mark.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Al Baker
Sent: May 24, 2008 4:35 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] reload stopping EVERYTHING on CLI and causing
havoc.

Quote "

Oh and also, in my implementation there are no queues. It seems to be
>> not related, I've had it in EVERY version of Asterisk I've used."

 Hmmm- maybe this should be mentioned in the next "is * Really Good Thread
?"


Mark Hamilton wrote:
> Same here.
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Sherwood
> McGowan
> Sent: May 22, 2008 4:16 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] reload stopping EVERYTHING on CLI and
causing
> havoc.
>
> Steve Totaro wrote:
>   
>> On Thu, May 22, 2008 at 3:56 PM, Sherwood McGowan
>> <[EMAIL PROTECTED]> wrote:
>>   
>> 
>>> Steve Totaro wrote:
>>> 
>>>   
 On Thu, May 22, 2008 at 2:02 PM, Sherwood McGowan
 <[EMAIL PROTECTED]> wrote:

   
 
> Mark Hamilton wrote:
>
> 
>   
>> Hi,
>>
>> Yesterday I made a change in queues.conf and so tried doing a reload
>> app_queue.so in the CLI. (Using 1.4.18). It didn't seem to do
>> anything, infact all action on CLI stopped.
>>
>> Then, I did a reload. Same thing.
>>
>> After that there was no other way.. because even stop now wouldn't
>> work, so I did a service asterisk restart
>>
>> And then asterisk kept giving the same thing on prompt "Died
>> successfully" and all that it usually says when you issue a stop now,
>> except it kept showing that on root prompt after doing a service
>> asterisk restart.
>>
>> Did a killall asterisk, and finally it stopped.
>>
>> Then started asterisk service. It was fine.
>>
>> Did a full restart at night, and it was fine.
>>
>> NOW, I wanted to do a reload again today mid-day when in full use,
and
>> it still didn't work, and ALL of the above happened again.
>>
>> --
>>
>> How do I diagnose what's causing this?
>>
>> Thanks,
>>
>> Mark.
>>
>>
>> 
> 
>   
>> ___
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>>   
>> 
> I've had this problem before, haven't debugged it. I definitely look
> forward to hearing what is said about this.
>
> Example from my recent experience, I wanted to restart the server and
>   
> so
>   
> did
>
> pbx0*CLI> restart now
>
> But nothing happened...system continued to allow calls to take place.
> I've found that sometimes exiting and reconnecting to the CLI helps,
>   
> but
>   
> there have been a couple occasions where NOTHING would allow the
server
> to restart save for a reboot. Even killall asterisk didn't kill the
> process
>
> Sherwood McGowan
>
>
> 
>   
 You are using Asterisk 1.2.x?  I have seen this many, many times.

 Sometimes the CLI becomes unresponsive, sometimes queues crap out or
 stops delivering calls to agents, sometimes it just takes a bit and
 then becomes responsive again.

 The rule of thumb is don't reload queues when there are people in
 queue, at least that seems to eliminate the problems I have seen.
 Makes sense too.

 Not sure if it is fixed in 1.4.

 Thanks,
 Steve Totaro

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>>> Oh and also, in my implementation there are no queues. It seems to be
>>> not related, I've had it in EVERY version of Asterisk I've used.
>>>
>>> 
>>>   
>> I have observed it on repeated general reloads on all versions.
>> That's why I don't reload very much, only planned.
>>
>> Thanks,
>> Steve Totaro
>>
>> ___
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>   
>> 
> My problem exists even

[asterisk-users] Asterisk not answering calls

[Joseph L. Casale]

I can make outbound calls, but when I call any of my did's they ring busy.
A tcpdump at the Asterisk server shows no inbound traffic and neither does sip 
set debug
show any activity. I have the providers routing set to sip user, I am using 
that user in my registration.

Anyone know if there is anything obvious that I missed?
Thanks!
jlc

sip.conf (relevant info)
[general]
context=default
disallow=all
allow=ulaw
allow=alaw
dtmfmode=rfc2833
canreinvite=yes
externip=xx.xx.xx.xx
localnet=192.168.0.0/255.255.255.0

register => user:[EMAIL PROTECTED]:5060
register => user:[EMAIL PROTECTED]:5060

[provider-sw1]
context=incoming
type=friend
host= sip1.provider.com
username=user
secret=pass
canreinvite=no ; if using a nat, do not change
insecure=port,invite ; do NOT remove this
qualify=5000 ; do NOT remove this
dtmfmode=auto
nat=no ; do NOT remove/change this
disallow=all
;allow=g729 ;uncomment if you have purchased a g729 license or can do passthru
allow=ulaw

[provider-sw2]
context=incoming
type=friend
host= sip2.provider.com
username=user
secret=pass
canreinvite=no ; if using a nat, do not change
insecure=port,invite ; do NOT remove this
qualify=5000 ; do NOT remove this
dtmfmode=auto
nat=no ; do NOT remove/change this
disallow=all
;allow=g729 ;uncomment if you have purchased a g729 license or can do passthru
allow=ulaw

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Re: [asterisk-users] excessive bounces???

[Doug Lytle]

Doug Lytle wrote:
> listgolden`
>   


Guess I should have done some clean up.  It was time to change the 
password anyways.

Doug


-- 
Ben Franklin quote:

"Those who would give up Essential Liberty to purchase a little Temporary 
Safety, deserve neither Liberty nor Safety."


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[asterisk-users] excessive bounces???

[Doug Lytle]

My membership has been disabled for excessive bounces?  Are we having 
issues again with the list?  I've check both my primary and secondary 
MTA and show no issues with mail.


Doug

--
Ben Franklin quote:

"Those who would give up Essential Liberty to purchase a little Temporary Safety, 
deserve neither Liberty nor Safety."

--- Begin Message ---
Your membership in the mailing list asterisk-users has been disabled
due to excessive bounces The last bounce received from you was dated
24-May-2008.  You will not get any more messages from this list until
you re-enable your membership.  You will receive 3 more reminders like
this before your membership in the list is deleted.

To re-enable your membership, you can simply respond to this message
(leaving the Subject: line intact), or visit the confirmation page at


http://lists.digium.com/mailman/confirm/asterisk-users/f0c63888a56b25c2444a65b916674c001bcb0ad6


You can also visit your membership page at

http://lists.digium.com/mailman/options/asterisk-users/support%40drdos.info


On your membership page, you can change various delivery options such
as your email address and whether you get digests or not.  As a
reminder, your membership password is

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If you have any questions or problems, you can contact the list owner
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Re: [asterisk-users] install asterisk on linux that uses software raid

[Steve Howes]

On Sat, 24 May 2008 02:29:30 -0700 (PDT), ronald ramos wrote
> hi all,
> 
> we recently bought a clone box, motherboard with ICH7R raid controller (which 
> i thought was a hardware raid controller). but recently i learned that those 
> things are called FRAID( Fake RAID) which is basically a software raid also. 
> so i decide to just use Software RAID (using CentOS 5.1).
> 
> has anyone installed asterisk on such configuration? is there any prob with 
> regards to performance or quality of calls? thank you any info will be 
> appreciated.
> 
> regards,
> ron

Never had any problems with Asterisk and Linux software RAID. I really wouldn't 
worry.

Steve

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Re: [asterisk-users] Incoming calls not being answered by asterisk

[Grey Man]

The first thing to do is type "sip debug" on the console and place the
call from the Sipura. If you get a bunch of SIP messages flashing down
your console you know the call is reaching Asterisk and it's most
likely going to be an issue authenticating the call or a problem in
your dial plan.

If no SIP messages flash up then the call is not reaching your Asterisk server.

Regards,

Greyman.

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Re: [asterisk-users] Asterisk semi-hangs

[Grey Man]

The main thing I have noticed over the years that causes temporary
asterisk hangs is the DNS server asterisk is trying to use becoming
inaccessible. If a network issue causes Asterisk to be unable to
connect to the DNS server you will get the classic freeze on the
console and then when the DNS server is accessible again a flood of
messages. This could be your problem although if the console never
comes back perhaps not.

Hopefully 1.6 will be able to cope better with an unresponsive DNS and
at the very least be able to continue running and carry out any tasks
that aren't related to DNS lookups.

Regards,

Greyman.

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Re: [asterisk-users] Dear asterisk-users@lists.digium.com May 80% 0FF

[Steve Totaro]

Darn, it was 87% off just yesterday!

On Sat, May 24, 2008 at 8:53 AM, VIAGRA (R) Official Site
 wrote:
> About this mailing:
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Re: [asterisk-users] dialplan syntax error: need new eyes

[sean darcy]

Barry Miller wrote:
> On Sat, May 24, 2008 at 12:01:50AM -0400, sean darcy wrote:
>> Barry Miller wrote:
>>> On Fri, May 23, 2008 at 05:08:28PM -0400, sean darcy wrote:
 This doesn't work:

 exten =>_1NXXNXX,n,Set( CALLERID(num) = ${IF ( $[${CALLERID(num)} > 
 140] ? ${MAINSTUB}${CALLERID(num)} : ${MAINNUMBER} )})
>>> Change "IF (" to "IF(".
>> Same result.
> 
> Sorry.  This time I actually tested it.  *After* de-spacing the " = ",
> 
> exten => test,n,Set(CALLERID(num)=${IF( $[${CALLERID(num)} > 140] ? 
> ${MAINSTUB}${CALLERID(num)} : ${MAINNUMBER} )})
> exten => test,n,NoOp(${CALLERID(num)})
> 
> behaved properly.  At least with 1.4.19.1. 

I cut and pasted that, and got the same error. I'm still at 1.4.13. I'm 
also testing with a blank callerid. If you could test with a blank 
callerid, I'd appreciate it, but it looks like I need to upgrade.

  FWIW, every time I try to use
> whitespace to make a dialplan more readable, it jumps up and bites me.
> 
> Again, sorry for jumping in with an untested response.
> 
If you hadn't responded, tested or not, I'd still be going crazy staring 
at this.

Thanks for all your help.

sean


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Re: [asterisk-users] digium cards with sangoma cards

[Steve Totaro]

On Sat, May 24, 2008 at 6:53 AM, wassim darwish <[EMAIL PROTECTED]> wrote:
>
> Hi:
> Iam an Asterisk user and i have a Sangoma A200 with 4 fxo modules  and i want 
> to buy Digium card with 4 fxo modules and insert it on the PCI besides the 
> sangoma card ,so i will have 8 fxo channels on my asterisk box ,Is that right?
> Does Asterisk make errors if there is two different cards ?
>
> Thanks in advance;

You should be fine.

Thanks,
Steve T

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[asterisk-users] Asterisk semi-hangs

[Carles Pina i Estany]

Hello,

(Note: again, I'm asking for experiences/suggestions, because
we have a problem in some environment that it's quite difficult to
debug, test, etc.)

I have seen, during last year, that some Asterisk has hangup up. I mean,
not crashing, we could access to Asterisk console, but phones couldn't
call. Doing "dial [EMAIL PROTECTED]" was not doing/showing anything, even
with verbose setted up as 99. This was in Asterisk 1.2

Now, in a couple of Asterisk 1.4 boxes (I think that Asterisk
1.4.19.2?), this has happened again. Not often, every some months maybe,
etc. One installation is only using VoIP, so I don't think that it's a
Zaptel, Misdn, etc. problem.

Even typing "stop now" nothing is happening, killall asterisk either,
only killall -9 asterisk. 

We use a standard installation (no extra modules, etc.)

Debian distribution, our Kernel.

Has anyone some experience like this? Anything to tune? or something
to see? (logs doesn't say a lot, and hard disk is not full :-) )

We would like to avoid that this is happening again, understand why is
happening, etc.

Thank you,

-- 
Carles Pina i EstanyGPG id: 0x8CBDAE64
http://pinux.info   Manresa - Barcelona

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Re: [asterisk-users] digium cards with sangoma cards

[Thomas Kenyon]

wassim darwish wrote:
> Hi:
> Iam an Asterisk user and i have a Sangoma A200 with 4 fxo modules  and i want 
> to buy Digium card with 4 fxo modules and insert it on the PCI besides the 
> sangoma card ,so i will have 8 fxo channels on my asterisk box ,Is that right?
> Does Asterisk make errors if there is two different cards ?   
> 
> Thanks in advance;

I know this sounds silly, and isn't what you are after. Why don't you 
buy a daughterboard with an additional 4 fxo modules for your existing A200?

I don't know which option is cheaper, but upgrading the A200 has got to 
be better for IO bandwidth on your machine.

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[asterisk-users] digium cards with sangoma cards

[wassim darwish]

Hi:
Iam an Asterisk user and i have a Sangoma A200 with 4 fxo modules  and i want 
to buy Digium card with 4 fxo modules and insert it on the PCI besides the 
sangoma card ,so i will have 8 fxo channels on my asterisk box ,Is that right?
Does Asterisk make errors if there is two different cards ?   

Thanks in advance;
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Re: [asterisk-users] install asterisk on linux that uses software raid

[Sam Tam]

There will be a slight of delay on writing files but not really a performace
issue at all.
You will hardly notice.
Sam 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of ronald ramos
Sent: Saturday, May 24, 2008 5:30 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] install asterisk on linux that uses software raid

hi all,

we recently bought a clone box, motherboard with ICH7R raid controller
(which i thought was a hardware raid controller). but recently i learned
that those things are called FRAID( Fake RAID) which is basically a software
raid also. so i decide to just use Software RAID (using CentOS 5.1).

has anyone installed asterisk on such configuration? is there any prob with
regards to performance or quality of calls? thank you any info will be
appreciated.

regards,
ron





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[asterisk-users] Incoming calls not being answered by asterisk

[RoLaNd RoLaNd]

Hello all,

ive got the following setup currently:

 
   __Sipura 3102-PSTN
  |
Lan | 
  |
  |__asterisk

i configured both asterisk and pstn to be able to receive/make calls through 
each other using sip of course..
but the thing is i want asterisk that when it receives an incoming call from 
sipura, to answer it, play msg that i recorded and wait for the caller to dial 
in an extension, where it would transfer the caller to that exntension, and in 
case the extension owner isnt available to answer it would direct him to his 
voicemail(tht i dont know how to set yet), and in case the caller didnt dial 
any extension in a certain amount of time, it automaticly directs it to a 
specific extensions i'd specify..

i tried the examples given in lots of forums and so on none of them worked, the 
phone keeps on ringing with every incomign dial plan ive specified without 
asterisk answering it..
the thing i did is that sipura directs incoming calls to 1002, so ive set the 
context of 1002 in sip.conf to a dial plan of [incoming-sipura] and ive set the 
commands i mentioned earlier tht i took out of those forums.. but theyre not 
working!!!

anyone has an example i could go on with ? 
any help would be apreciated:)

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[asterisk-users] install asterisk on linux that uses software raid

[ronald ramos]

hi all,

we recently bought a clone box, motherboard with ICH7R raid controller (which i 
thought was a hardware raid controller). but recently i learned that those 
things are called FRAID( Fake RAID) which is basically a software raid also. so 
i decide to just use Software RAID (using CentOS 5.1).

has anyone installed asterisk on such configuration? is there any prob with 
regards to performance or quality of calls? thank you any info will be 
appreciated.

regards,
ron




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Re: [asterisk-users] reload stopping EVERYTHING on CLI and causing havoc.

[Al Baker]

Quote "

Oh and also, in my implementation there are no queues. It seems to be
>> not related, I've had it in EVERY version of Asterisk I've used."

 Hmmm- maybe this should be mentioned in the next "is * Really Good Thread ?"


Mark Hamilton wrote:
> Same here.
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Sherwood
> McGowan
> Sent: May 22, 2008 4:16 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] reload stopping EVERYTHING on CLI and causing
> havoc.
>
> Steve Totaro wrote:
>   
>> On Thu, May 22, 2008 at 3:56 PM, Sherwood McGowan
>> <[EMAIL PROTECTED]> wrote:
>>   
>> 
>>> Steve Totaro wrote:
>>> 
>>>   
 On Thu, May 22, 2008 at 2:02 PM, Sherwood McGowan
 <[EMAIL PROTECTED]> wrote:

   
 
> Mark Hamilton wrote:
>
> 
>   
>> Hi,
>>
>> Yesterday I made a change in queues.conf and so tried doing a reload
>> app_queue.so in the CLI. (Using 1.4.18). It didn't seem to do
>> anything, infact all action on CLI stopped.
>>
>> Then, I did a reload. Same thing.
>>
>> After that there was no other way.. because even stop now wouldn't
>> work, so I did a service asterisk restart
>>
>> And then asterisk kept giving the same thing on prompt "Died
>> successfully" and all that it usually says when you issue a stop now,
>> except it kept showing that on root prompt after doing a service
>> asterisk restart.
>>
>> Did a killall asterisk, and finally it stopped.
>>
>> Then started asterisk service. It was fine.
>>
>> Did a full restart at night, and it was fine.
>>
>> NOW, I wanted to do a reload again today mid-day when in full use, and
>> it still didn't work, and ALL of the above happened again.
>>
>> --
>>
>> How do I diagnose what's causing this?
>>
>> Thanks,
>>
>> Mark.
>>
>>
>> 
> 
>   
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>>
>>   
>> 
> I've had this problem before, haven't debugged it. I definitely look
> forward to hearing what is said about this.
>
> Example from my recent experience, I wanted to restart the server and
>   
> so
>   
> did
>
> pbx0*CLI> restart now
>
> But nothing happened...system continued to allow calls to take place.
> I've found that sometimes exiting and reconnecting to the CLI helps,
>   
> but
>   
> there have been a couple occasions where NOTHING would allow the server
> to restart save for a reboot. Even killall asterisk didn't kill the
> process
>
> Sherwood McGowan
>
>
> 
>   
 You are using Asterisk 1.2.x?  I have seen this many, many times.

 Sometimes the CLI becomes unresponsive, sometimes queues crap out or
 stops delivering calls to agents, sometimes it just takes a bit and
 then becomes responsive again.

 The rule of thumb is don't reload queues when there are people in
 queue, at least that seems to eliminate the problems I have seen.
 Makes sense too.

 Not sure if it is fixed in 1.4.

 Thanks,
 Steve Totaro

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>>> Oh and also, in my implementation there are no queues. It seems to be
>>> not related, I've had it in EVERY version of Asterisk I've used.
>>>
>>> 
>>>   
>> I have observed it on repeated general reloads on all versions.
>> That's why I don't reload very much, only planned.
>>
>> Thanks,
>> Steve Totaro
>>
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>>   
>> 
> My problem exists even when issuing a restart now or stop now command at 
> the CLI.
>
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Re: [asterisk-users] [asterisk-dev] Asterisk 1.6 Realtime Database must use ', ' not '|'

[Al Baker]

quote "And hackers ignoring pleasantries to get right down to the 
technical issues isn't abusive at all"
ABUSIVE - No not at all.
Unnecessarily rude, insensitive, tacky - Yep

Jay R. Ashworth wrote:
> On Fri, May 23, 2008 at 01:25:43PM -0400, Donny Kavanagh wrote:
>   
>> This is getting downright abusive, and is totally uncalled for, this
>> is not a list for personal attacks.
>> 
>
> You thought that Steve suggesting JT step in was abusive?
>
> If that's not what you meant, then you need to either a) be clearer, or
> b) reply to the proper message.
>
> And hackers ignoring pleasantries to get right down to the technical
> issues isn't abusive at all. 
>
> See Jargon File; see also Asperger's Syndrome, How To Ask Good Questions.
>
> Cheers,
> -- jra
>   

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Re: [asterisk-users] dialplan syntax error: need new eyes

[Barry Miller]

On Sat, May 24, 2008 at 12:01:50AM -0400, sean darcy wrote:
> Barry Miller wrote:
> > On Fri, May 23, 2008 at 05:08:28PM -0400, sean darcy wrote:
> >> This doesn't work:
> >>
> >> exten =>_1NXXNXX,n,Set( CALLERID(num) = ${IF ( $[${CALLERID(num)} > 
> >> 140] ? ${MAINSTUB}${CALLERID(num)} : ${MAINNUMBER} )})
> > 
> > Change "IF (" to "IF(".
> 
> Same result.

Sorry.  This time I actually tested it.  *After* de-spacing the " = ",

exten => test,n,Set(CALLERID(num)=${IF( $[${CALLERID(num)} > 140] ? 
${MAINSTUB}${CALLERID(num)} : ${MAINNUMBER} )})
exten => test,n,NoOp(${CALLERID(num)})

behaved properly.  At least with 1.4.19.1.  FWIW, every time I try to use
whitespace to make a dialplan more readable, it jumps up and bites me.

Again, sorry for jumping in with an untested response.

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