date:20060531

Regarding my earlier post about labels and the 'n' priority:
The TFOT book covers the use of these.  See the box on page 81 entitled
Unnumbered Priorities.

http://www.nufone.net/downloads/asteriskdocs/AsteriskTFOT.zip

-MC
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[Asterisk-Users] Asterisk receiving call from Panasonic TDA extension issue

2006-05-31 Thread Asterisk Mailing List


Sorry someone screwing with permissions on my server bounced the 2 days
worth of email after I posted this, any and all those lovely people who
replied with suggestions from my post could you sent them again :-)

James 

-Original Message-
From: James Bean On Behalf Of Asterisk Mailing List
Sent: Tuesday, 30 May 2006 12:54 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Asterisk receiving call from Panasonic TDA extension issue

Asterisk, Zap and Libpri version from Asterisk SVN-branch-1.2-r27093

Error:-
-- Accepting overlap call from '123' to '6' on channel 0/31, span 1
-- Starting simple switch on 'Zap/31-1'
-- Hungup 'Zap/31-1'

Primary Rate E1 30 trunks connecting between Asterisk and TDA200
Pansonic TDA200 has 1XX extensions Asterisk is setup with 6XX extensions

If Asterisk calls a 1XX its not an issue, when 1XX calls Asterisk it
looks like the phone system is dialing the digitals individually instead
of at once so Asterisk is receiving the first 6 going I don't know 6
before it receives the rest of the digits from the TDA.

Any clues as to if its possible to have asterisk wait for the rest of
the digits, a wait of sorts, or I have to figure out how to make the TDA
do it?

James


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[Asterisk-Users] Nokia E60 , experience as SIP client

2006-05-31 Thread John Joseph

Hi 
   I want to  check out from the members , about their
experience with Nokia E60 phone as SIP client , I was
able to register the phone , but  my  voice gets
broken during the calls . My other  Wi-Fi VoIP   SIP
phone  are working fine 
I also like to check out  is there any other mobile
manufacture who have SIP supported porducts like Nokia
e-60
   
  Thanks 
 Joseph John 






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Re: [Asterisk-Users] INFO: TFOT book- n priorities and labels

2006-05-31 Thread trixter aka Bret McDanel

On Wed, 2006-05-31 at 02:01 -0700, Michael Collins wrote:
 Regarding my earlier post about labels and the 'n' priority:
 The TFOT book covers the use of these.  See the box on page 81 entitled
 Unnumbered Priorities.
 
 http://www.nufone.net/downloads/asteriskdocs/AsteriskTFOT.zip
 

And one of the authors of that book (Jim Van Meggelen) will be speaking
at ClueCon (on asterisk topics I believe) in august if you want to talk
to him in person :)  for more info see http://www.cluecon.com/





-- 
Trixter http://www.0xdecafbad.com Bret McDanel
Belfast IE +44 28 9099 6461DE +49 801 777 555 3402
Utrecht NL +31 306 553058  US WA +1 360 207 0479
US NY +1 516 687 5200  FreeWorldDialup: 635378
http://www.sacaug.org/ Sacramento Asterisk Users Group



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Re: [Asterisk-Users] Nokia E60 , experience as SIP client

2006-05-31 Thread Antonio Rabena


try using g711 ulaw codec.

At 03:51 PM 5/31/2006, you wrote:

Hi
   I want to  check out from the members , about their
experience with Nokia E60 phone as SIP client , I was
able to register the phone , but  my  voice gets
broken during the calls . My other  Wi-Fi VoIP   SIP
phone  are working fine
I also like to check out  is there any other mobile
manufacture who have SIP supported porducts like Nokia
e-60

  Thanks
 Joseph John



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Re: [Asterisk-Users] Asterisk::AGI and DIALEDTIME

2006-05-31 Thread Jean-Michel Hiver


JP Carballo a écrit :


Jean-Michel Hiver wrote:


Hi List,

In one of my AGIs (using DeadAGI) I grab the answered time using:

   my $res = $agi-exec (DIAL $dialstring);
   my $answeredtime = $agi-get_variable (ANSWEREDTIME);

However this information differs from what's written in the 
Master.csv file (which happens to be the correct value!)


Any ideas why?

On my system, answeredtime returns the time elapsed since the call was 
answered by the destination.
The time elapsed stored in Master.csv is from the time the current 
incoming call (channel) was answered.


There are two values in Master.csv: the first one is the total time of 
the call (including the ringing bits and everything), the second one 
being the time which has been effectively answered (billable time). 
This second value is the correct one and differs from what I expect.


Ah well, no worries. I've setup cdr_pgsql.conf and will process the CDRs 
every minute or so with a cron job. It's a bit patchy but what can you do :)


Cheers,
Jean-Michel.

--
Jean-Michel Hiver - http://ykoz.net/
Découvrez la Réunion des Technologies IP  Telecom
TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE

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[Asterisk-Users] Fax to Email issue with Spandsp tif not correctly sized

2006-05-31 Thread Nicolas FINETIN

Hello, I've google to trouble shoot my issue but i was not able to find a solution, I've install asterisk all the libraries to receive faxes:Spandsp 0.0.2 pre 25 tiff lib.I'make it work i can receive faxes on an extension but the issue is that i m in France an we are sending faxes in A4 format so i set that up in the perl script /var/lib/asterisk/bin/fax-
process.pl by passing the good option to the tiff2pdf.still not good output then i check the .tif  and here i could see that the tif itself is not well formatted.It is looking like larger is bigger and height is smaller.
How can i set the rx_fax.c that i well receiving the Fax's tif.Thank you Regards,-- -= Nicolas Finetin =-[EMAIL PROTECTED]
+33 689 20 90 72
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[Asterisk-Users] Upgrading

2006-05-31 Thread Chris Blunt









Hi List, 



I was wondering what is the best way to upgrade an Asterisk
system to the latest version.



I know there is the patch method, but if I am jumping 3 or 4
versions is a re-install the best way?



Should I just make the files then manually copy them in?
Does this avoid overwriting any modified sound files etc? Should I delete the
current files or move / make a copy to a different location first?



I know this is a lot of questions but I am hoping for a best
practice idea etc



Regards



Chris



--



Chris Blunt

Entropy IT Ltd








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Re: [Asterisk-Users] Handset recommendations

2006-05-31 Thread Andrew Kohlsmith

On Tuesday 30 May 2006 23:13, George A. Roberts IV wrote:
 Any other thoughts on good reasonably priced handsets?  This is for just a
 couple of people who work from home offices and will be connecting to an
 Asterisk server hosted in a datacenter.

I am a *huge* fan of the Polycom ip501.  The 301 works just as well, but the 
display is significantly crappier.  If you've got the cash, go 601.

I have never used Cisco, but I've used the cheaper phones enough to know that 
this is one place were spending a little more is WELL worth it.  And I know 
from personal experience that the Polycom phones have *zero* issues with 
being behind NAT and talking to a public-IP Asterisk box.  No firewall 
configuration, no screwing around whatsoever.  They Just Work.

-A.
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Re: [Asterisk-Users] mpg123 or asterisk

2006-05-31 Thread Steve Totaro


Matt Roth wrote:

Steve Totaro wrote:

Please let us know your results.  I cannot really test this in 
production system since it is a $16,000/hr call center.  I was using 
madplay but it was crashing and creating zombie processes, I figured 
native was not the way to go since all of the different audio 
streams.  Mpg123 works perfectly for me under a load of sixty 
channels, I can confirm that for sure.


Thanks,
Steve


Steve,

mpg123 has the same problem with zombie processes as you were 
experiencing with MadPlay.  For a scalable system, native MOH is the 
way to go.  As per Kevin Fleming, it only introduces a slight memory 
overhead.  mpg123 consumes CPU cycles to decompress the mp3s and in my 
experience, a large scale Asterisk system is much hungrier for CPU 
cycles than memory.


The different audio streams used by native MOH are not really a 
problem for the following reasons:


1) The native MOH files are likely to be cached, so they are probably 
being read from memory.

2) The native MOH files do not require decompression or transcoding.
3) The MOH is handled in the same thread as the call itself, so there 
is very little CPU overhead.


As always, I believe that the information I'm sharing is accurate but 
welcome any corrections or additions.


Thank you,

Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer
___

Thanks for your clarification, I will try native but most likely wind up 
streaming MOH from another box.


Thanks,
Steve

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Re: [Asterisk-Users] Handset recommendations

2006-05-31 Thread Andrew Kohlsmith

On Tuesday 30 May 2006 23:56, Bruce Reeves wrote:
 the Linksys. Actually if your budget supports it get one of each and try
 them out. I tried out several from the Cisco 7960, Polycoms and then a
 UTstarcom F1000. My big concern on the Cisco is that there is a license

Avoid the UTStarCom F1000G.  (802.11g version of F1000).  The buttons are far 
too close and tiny, the phone is too quiet, the ringers are subpar and the 
phone tends to lose its wifi for no particular reason.  When this happens it 
just gives up and does so without warning.  It's really quite random as to 
when it occurs.  Perhaps when the next firmware version is out it'll be 
better.  The manufacturer forums are active enough, but I was pretty 
underwhelmed with these phones.  

They *are* inexpensive, though.  I will grant them that.

-A.
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Re: [Asterisk-Users] Nokia E60 , experience as SIP client

2006-05-31 Thread Florian Overkamp


John Joseph wrote:
Hi 
   I want to  check out from the members , about their

experience with Nokia E60 phone as SIP client , I was
able to register the phone , but  my  voice gets
broken during the calls . My other  Wi-Fi VoIP   SIP
phone  are working fine 
I also like to check out  is there any other mobile

manufacture who have SIP supported porducts like Nokia
e-60


We use them with Alaw/Ulaw and it works pretty well. I do think there 
are some bugs in the firmware, SIP accounts do not get reregistered 
automatically if other applications used the WLAN network, or when 
roaming between different WLAN networks.


I'm also not entirely happy about the battery time when using WLAN :-)

Great phone, though!

Florian
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Re: [Asterisk-Users] I guess my server capacity is ok

2006-05-31 Thread Steve Totaro

All I know is that it is very processor intensive and either not using 
it or just passing it through is your best bet.  I will be working alot 
with G729 in the near future and will post my findings but until then I 
am just relying on the dimensioning page on the wiki.


Thanks,
Steve Totaro

Goke Aruna wrote:

Steve,

Can you please give me an insight on how g729 problem could solved?

goksie

On 5/30/06, *Steve Totaro*  [EMAIL PROTECTED] 
mailto:[EMAIL PROTECTED] wrote:


G729 is your problem.

Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com


 -Original Message-
 From: Lachek Butalek [mailto:[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]]
 Sent: Tuesday, May 30, 2006 10:10 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] I guess my server capacity is ok

 What process is taking up 100% CPU? Is it Asterisk processes or
 something else? Also, is the load spread out over multiple
processes,
 or do you have one or two processes taking up 90% or more of your
 total?

 You also have dual CPUs (and hyperthreading, which to FC3 should
look
 like 4 CPUs if I'm not mistaken) - is the 100% CPU usage across all
 two (or four) processors, or is it only CPU1 that peaks at 100%?
Have
 a look at Last Used CPU in top. What load are the other CPUs at?

 I don't have personal experience running that large of an
 installation, but I imagine your system specs would allow you to
 handle more simultaneous calls than 50, even though you're doing
some
 transcoding.

 On 5/30/06, Goke Aruna [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
  can someone overthere help?
 
  the server specs are as follows
   HP DL380G4 Dual Intel Xeon 3.2GHz processor with 4GB RAM,
   running fedora core 3
   asterisk-1.2.5
   ss7-0.8.3d.
   using sip as advised to receive calls from another gateway in
US.
   using g729 in transcoding way.
 
   however, I noticed the call hit the 51 active calls which is
 102channels, I
  run top to check the system resources usage and i discovered
that
the
 cpu
  is 100% used. asterisk, sip, ss7  never crashed throughout.
 
   however, since transcoding takes alot of system resources..
how can
I
 use
  g729 in passthru mode.
 
   and I guess disabling hyperthreading will save me more system
resouces.
 
   I will be glad, if you can give me more info on system management
cos i
  think with that system, it should able to handle at least five
E1's.
 
  I say thank you for finding time to reply my mail.
 
   goksie
 
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[Asterisk-Users] Need help with Junghanns Quadbri

2006-05-31 Thread Jean-Louis curty

Hi everybody

I hope that somebody can help me with the following 

I have
2 quadbri cards
2 - 1t0 cards
1 pabx alcatel 4200

I would like to connect my asterisk to the alcatel ,

I installed bristuff 0.3.0-1p ,
loaded the zaphfc driver in NT mode
configured zaptel and zapata , it works great.


then I removed the 1 t0 card,
added the quadbri
loaded qozap : insmod qozap.ko ports=15 ( 4 ports in NT )

adjusted the zaptel zapata, specified the right signalling, right context
ran ztcfg -vv ( 12 channels configured ) 
started asterisk,
I get layer1 down message on the 4 ports,
leds remain red
what ever I do in my conf , I am not able to get a reaction from the card ( I tried with my two quadbri, on 2 different pc's ) 


what can I check ?
thanks
jl
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RE: [Asterisk-Users] Need help with Junghanns Quadbri

2006-05-31 Thread Henk









Try to do ztcfg s before
you run ztcfg -vv



Henk











From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jean-Louis curty
Sent: woensdag 31 mei 2006 12:52
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: [Asterisk-Users] Need
help with Junghanns Quadbri







Hi everybody











I hope that somebody can help me with the following 











I have





2 quadbri cards





2 - 1t0 cards





1 pabx alcatel 4200











I would like to connect my asterisk to the alcatel ,











I installed bristuff 0.3.0-1p ,





loaded the zaphfc driver in NT mode





configured zaptel and zapata , it works great.

















then I removed the 1 t0 card,





added the quadbri





loaded qozap : insmod qozap.ko ports=15 ( 4 ports in NT )











adjusted the zaptel zapata, specified the right signalling, right
context





ran ztcfg -vv ( 12 channels configured ) 





started asterisk,





I get layer1 down message on the 4 ports,





leds remain red





what ever I do in my conf , I am not able to get a reaction from the
card ( I tried with my two quadbri, on 2 different pc's ) 

















what can I check ?





thanks





jl








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Re: [Asterisk-Users] Upgrading

2006-05-31 Thread Steven Ringwald


Chris Blunt wrote:


Hi List,

I was wondering what is the best way to upgrade an Asterisk system to 
the latest version.


I know there is the patch method, but if I am jumping 3 or 4 versions 
is a re-install the best way?


Should I just make the files then manually copy them in? Does this 
avoid overwriting any modified sound files etc? Should I delete the 
current files or move / make a copy to a different location first?


I know this is a lot of questions but I am hoping for a best practice 
idea etc…




I believe that make upgrade installs just the applications, and does 
not touch config files (which are only installed with make setup, BTW) 
and the sound files.


Hope this helps.
Steve

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[Asterisk-Users] extra parameter for DB read function

2006-05-31 Thread Julian Lyndon-Smith

There are often times that I want to read a DB value from the dialplan, 
and if this family/key pair does not exist, set it to some default value.


for example:

1234,1 = Set(EMAILADDR=${DB(x/y)}
1234,2 = GotoIf($[${EMAILADDR} = ]?3:4)
1234,3 = Set([EMAIL PROTECTED])
1234,4 = NoOp(${EMAILADDR})
1234,5 = Hangup()

I have modified the db function to take an extra parameter to set if the 
key does not exist. So, the dialplan would now look like:


1234,1 = Set(EMAILADDR=${DB(x/y/[EMAIL PROTECTED])}
1234,2 = NoOp(${EMAILADDR})
1234,3 = Hangup()

It's just a shortcut to acheive the same goal, but with 2 less lines in 
the dialplan.


Now, I am *not* a C programmer, so I may have made some horrendous 
mistake or potential segfault, so is there someone who would look at the 
changes before I make a fool of myself and post it to the -dev list or 
mantis ?


Much appreciated :)

Julian
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[Asterisk-Users] Centos cause Asterisk crash

2006-05-31 Thread chan \(Alpha Trilogies Networks\)

Hi,
Can some one who experience that does those file necessary for the CentOS
and Asterisk installation
/etc/cron.daily/00-makewhatis.cron
/etc/cron.daily/slocate.cron
/etc/cron.daily/prelink
/etc/cron.daily/rpm
/etc/cron.weekly/00-makewhatis.cron

I experience that those file cause my Asterisk Server crash.
Can I just disable them and run the Asterisk stable? 


Any reply will be appreciated.

Thank you in advance.



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[Asterisk-Users] Zap channels ringing too loudly

2006-05-31 Thread Nick Burch


Hi All

I've got an asterisk system, using a couple of Xorcom Astribanks to 
provide FXS ports. (I'm using the zaptel 1.2 branch, if that matters)


I've noticed that the ringing volume is a lot louder than on our old phone 
system, and people are starting to complain it's too loud. (This is the 
noise the phone makes when it rings, not the noise in your handset when 
you ring someone else)



Having had a look through the code, I think that Asterisk passes the 
responsibility for ringing the phones to Zaptel, which drives the 
astribank to make them ring. Is this correct?


Despite looking through the zaptel source code, I couldn't find anywhere 
that screamed I'm the volume your phones ring at. Just a lot of scary 
numbers in zonedata.c, and cryptic comments in tone_zone.h



Could someone suggest how I'd go about making the zap ring volume quieter?

Thanks
Nick
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[Asterisk-Users] Asterisk Bootcamp in Europe :: June 12-16 and the Asterisk SIP Masterclass in Chicago, July 2006

2006-05-31 Thread Olle E Johansson


** Asterisk Bootcamp in Stockholm, Sweden

The next Asterisk Training is the Edvina.net Asterisk Bootcamp - the  
class we have been giving for over a year under the brand name  
Astricon Training. The same teacher, the same material and a new name.


All students have a PC and will install a fully working Asterisk PBX.  
During the week, we will build a business PBX configuration as well  
as more advanced configurations using E1/PRI, SIP and IAX2 protocols.


Stockholm in June is a wonderful city with lots of activity, lots of  
sunshine and Asterisk techies :-)


Facts about this training:

Teacher: Olle E. Johansson, Asterisk developer and trainer.
Material: Training slides (over 300 pages), The Asterisk Quick  
Reference Guide

Dates: June 12-16 (starting 10 AM Monday, ending noon friday)
Options: dCAP exam friday afternoon, June 16th
Price: 2.500 Euro (ex VAT). 200 Euro (ex VAT) for dCAP.
All trainings are pre-paid. Register by e-mail to [EMAIL PROTECTED] today.

For more information, please visit our web site.

** The Asterisk SIP Masterclass :: Building SIP infrastructures with  
Asterisk


The Asterisk SIP Masterclass is a new class we're launching in July.  
It requires knowledge of
Asterisk and starts on a higher level than the bootcamp. The class is  
held by


* Olle E. Johansson, Asterisk SIP developer
* Terry Wilson, a consultant with experience from provider SIP networks

The class agenda is being worked on now, but will include:

* Asterisk basics - a recap
* SIP - an introduction to the protocol
* SIP proxys and network infrastructure
* The Asterisk SIP channel - introduction
* Traversing firewalls and NAT devices
* Key system functionality
* SIP phones - audio and video
* Building a SIP network with Asterisk and SIP proxys
* SIP test tools

As the bootcamp, this class will involve a lot of labs.

At this point, we're opening up for early bird registration on this  
class. Since we have not finalized
the product sheets, you are taking some chance, but will get a lower  
price. For registrations before June 15th,
refering to this mail, you will get the class for 2950 USD (plus VAT  
in Europe). The regular price is
3500 USD. Find out more about this class on our web site, http:// 
edvina.net/training/


See you on the trainings!


Best regards,
/Olle

---
* Olle E. Johansson - [EMAIL PROTECTED]
* Asterisk Training http://edvina.net/training/



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Re: [Asterisk-Users] Compilation issues with s390

2006-05-31 Thread Frank Pani


Hi (Kevin),

I'm on another mail client - so hopefully this will be better. 
Unfortunately, still not a go.   I'm copying the entire error message, 
thinking I may have left something out.   I see the changes you made in 
the gsm Makefile for the addition of s390.   Perhaps there is another 
spot?  I'm also wondering if I have the right gcc - I'm using gcc-3.2.2 
which comes with the distribution for SuSE.


Thanks

make[1]: Entering directory `/usr/src/asterisk-1.2/codecs'
make -C gsm lib/libgsm.a
make[2]: Entering directory `/usr/src/asterisk-1.2/codecs/gsm'
as   -o src/k6opt.o src/k6opt.s
src/k6opt.s: Assembler messages:
src/k6opt.s:9: Error: unknown pseudo-op: `.value'
src/k6opt.s:10: Error: unknown pseudo-op: `.value'
src/k6opt.s:11: Error: unknown pseudo-op: `.value'
src/k6opt.s:12: Error: unknown pseudo-op: `.value'
src/k6opt.s:13: Error: unknown pseudo-op: `.value'
src/k6opt.s:14: Error: unknown pseudo-op: `.value'
src/k6opt.s:15: Error: unknown pseudo-op: `.value'
src/k6opt.s:16: Error: unknown pseudo-op: `.value'
src/k6opt.s:17: Error: unknown pseudo-op: `.value'
src/k6opt.s:18: Error: unknown pseudo-op: `.value'
src/k6opt.s:19: Error: unknown pseudo-op: `.value'
src/k6opt.s:20: Error: unknown pseudo-op: `.value'
src/k6opt.s:27: Error: Unrecognized opcode: `pushl'
src/k6opt.s:28: Error: Unrecognized opcode: `movl'
src/k6opt.s:29: Error: Unrecognized opcode: `pushl'
src/k6opt.s:30: Error: Unrecognized opcode: `pushl'
src/k6opt.s:31: Error: Unrecognized opcode: `pushl'
src/k6opt.s:32: Error: Unrecognized opcode: `movl'
src/k6opt.s:33: Error: Unrecognized opcode: `movl'
src/k6opt.s:34: Error: Unrecognized opcode: `addl'
src/k6opt.s:35: Error: Unrecognized opcode: `emms'
src/k6opt.s:36: Error: Unrecognized opcode: `movl'
src/k6opt.s:36: Error: Unrecognized opcode: `movd'
src/k6opt.s:37: Error: Unrecognized opcode: `movq'
src/k6opt.s:38: Error: Unrecognized opcode: `movq'
src/k6opt.s:39: Error: Unrecognized opcode: `movq'
src/k6opt.s:40: Error: Unrecognized opcode: `xorl'
src/k6opt.s:43: Error: Unrecognized opcode: `movq'
src/k6opt.s:44: Error: Unrecognized opcode: `pmaddwd'
src/k6opt.s:46: Error: Unrecognized opcode: `movq'
src/k6opt.s:47: Error: Unrecognized opcode: `pmaddwd'
src/k6opt.s:48: Error: Unrecognized opcode: `paddd'
src/k6opt.s:50: Error: Unrecognized opcode: `movq'
src/k6opt.s:51: Error: Unrecognized opcode: `pmaddwd'
src/k6opt.s:52: Error: Unrecognized opcode: `paddd'
src/k6opt.s:54: Error: Unrecognized opcode: `movq'
src/k6opt.s:55: Error: Unrecognized opcode: `punpckhdq'
src/k6opt.s:56: Error: Unrecognized opcode: `paddd'
src/k6opt.s:58: Error: Unrecognized opcode: `paddd'
src/k6opt.s:59: Error: Unrecognized opcode: `psrad'
src/k6opt.s:60: Error: Unrecognized opcode: `packssdw'
src/k6opt.s:61: Error: Unrecognized opcode: `movd'
src/k6opt.s:62: Error: Unrecognized opcode: `movw'
src/k6opt.s:63: Error: Unrecognized opcode: `incl'
src/k6opt.s:64: Error: Unrecognized opcode: `cmpl'
src/k6opt.s:66: Error: Unrecognized opcode: `emms'
src/k6opt.s:67: Error: Unrecognized opcode: `popl'
src/k6opt.s:68: Error: Unrecognized opcode: `popl'
src/k6opt.s:69: Error: Unrecognized opcode: `popl'
src/k6opt.s:70: Error: Unrecognized opcode: `leave'
src/k6opt.s:71: Error: Unrecognized opcode: `ret'
src/k6opt.s:92: Error: Unrecognized opcode: `pushl'
src/k6opt.s:93: Error: Unrecognized opcode: `movl'
src/k6opt.s:94: Error: Unrecognized opcode: `pushl'
src/k6opt.s:95: Error: Unrecognized opcode: `pushl'
src/k6opt.s:96: Error: Unrecognized opcode: `pushl'
src/k6opt.s:97: Error: Unrecognized opcode: `emms'
src/k6opt.s:98: Error: Unrecognized opcode: `movl'
src/k6opt.s:99: Error: Unrecognized opcode: `movl'
src/k6opt.s:100: Error: Unrecognized opcode: `movl'
src/k6opt.s:101: Error: Unrecognized opcode: `movl'
src/k6opt.s:102: Error: Unrecognized opcode: `movl'
src/k6opt.s:103: Error: Unrecognized opcode: `subl'
src/k6opt.s:106: Error: Unrecognized opcode: `movq'
src/k6opt.s:107: Error: Unrecognized opcode: `movq'
src/k6opt.s:108: Error: Unrecognized opcode: `pmaddwd'
src/k6opt.s:109: Error: Unrecognized opcode: `movq'
src/k6opt.s:109: Error: Unrecognized opcode: `movq'
src/k6opt.s:109: Error: Unrecognized opcode: `pmaddwd'
src/k6opt.s:109: Error: Unrecognized opcode: `paddd'
src/k6opt.s:110: Error: Unrecognized opcode: `movq'
src/k6opt.s:110: Error: Unrecognized opcode: `movq'
src/k6opt.s:110: Error: Unrecognized opcode: `pmaddwd'
src/k6opt.s:110: Error: Unrecognized opcode: `paddd'
src/k6opt.s:111: Error: Unrecognized opcode: `movq'
src/k6opt.s:111: Error: Unrecognized opcode: `movq'
src/k6opt.s:111: Error: Unrecognized opcode: `pmaddwd'
src/k6opt.s:111: Error: Unrecognized opcode: `paddd'
src/k6opt.s:112: Error: Unrecognized opcode: `movq'
src/k6opt.s:112: Error: Unrecognized opcode: `movq'
src/k6opt.s:112: Error: Unrecognized opcode: `pmaddwd'
src/k6opt.s:112: Error: Unrecognized opcode: `paddd'
src/k6opt.s:113: Error: Unrecognized opcode: `movq'
src/k6opt.s:113: Error: Unrecognized opcode: `movq'

Re: [Asterisk-Users] Zap channels ringing too loudly

2006-05-31 Thread BJ Weschke


On 5/31/06, Nick Burch [EMAIL PROTECTED] wrote:

Hi All

I've got an asterisk system, using a couple of Xorcom Astribanks to
provide FXS ports. (I'm using the zaptel 1.2 branch, if that matters)

I've noticed that the ringing volume is a lot louder than on our old phone
system, and people are starting to complain it's too loud. (This is the
noise the phone makes when it rings, not the noise in your handset when
you ring someone else)


Having had a look through the code, I think that Asterisk passes the
responsibility for ringing the phones to Zaptel, which drives the
astribank to make them ring. Is this correct?

Despite looking through the zaptel source code, I couldn't find anywhere
that screamed I'm the volume your phones ring at. Just a lot of scary
numbers in zonedata.c, and cryptic comments in tone_zone.h


Could someone suggest how I'd go about making the zap ring volume quieter?



I could be way off here, but I thought FXS ringing was signaled only
by a change in voltage on the pair, so I'm not sure how zaptel could
instruct the hardware device to send a different voltage? I think its
only capability with FXS is to fluctuate the voltage to support
distinctive rings.

--
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http://www.btwtech.com/
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Re: [Asterisk-Users] Got SIP response 405 Method not acceptablebackfrom xxx.xxx.xxx.xxx

2006-05-31 Thread Erick Perez


Is the case of this provider, when he receives a sip communications,
he validates the origination IP, if it's one of the authorized, then
it is accepted.
no further auth required.


On 5/30/06, William Piper [EMAIL PROTECTED] wrote:

How exactly do you authenticate then, if it is IP authentication?

I always understood the words IP authentication to mean that the carrier
has the IP address of your server set in their sip.conf  you just send the
call with no registration over to them.  Anything that comes from your IP
address will be accepted.

bp

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kevin P.
Fleming
Sent: Tuesday, May 30, 2006 10:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Got SIP response 405 Method not
acceptablebackfrom xxx.xxx.xxx.xxx

William Piper wrote:
 If you don't need to authenticate, then you don't need a register command.
 Get rid of register = whatever and just have:
 exten = _X.,1,dial,SIP/[EMAIL PROTECTED]

Registration and authentication are not the same thing. Registration is
required for the carrier to be able to deliver calls _to_ his system. It
is possible though that if he is using a static IP, then the carrier
saying they 'authenticate by IP' may very well mean that you are
correct, that registration is not needed _either_ and they are rejecting
the REGISTER request completely.

To the original poster: please post a 'sip debug'/'set debug 10' console
trace of this failing registration attempt (but nothing else), so we can
see what is actually happening.
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--

---
Erick Perez
Linux User 376588
http://counter.li.org/  (Get counted!!!)
Panama, Republic of Panama
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Re: [Asterisk-Users] Compiling Asterisk-addons

Armin Schindler wrote:

 Actually, the error is not in the addons. The Asterisk-trunk installation
 produces incomplete/misconfigured headers, which prevents building of 
 external modules.
 Or is it digiums intention to make life more difficult for external modules?

Are you serious? How could that possibly be the case?

I just don't understand why you persist in making statements like this,
when it has been clearly stated that trunk is a _development_ area and
we will fix these problems before any beta releases are made.
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Re: [Asterisk-Users] Compilation issues with s390

Frank Pani wrote:

 I'm on another mail client - so hopefully this will be better.
 Unfortunately, still not a go.   I'm copying the entire error message,
 thinking I may have left something out.   I see the changes you made in
 the gsm Makefile for the addition of s390.   Perhaps there is another
 spot?  I'm also wondering if I have the right gcc - I'm using gcc-3.2.2
 which comes with the distribution for SuSE.

This is not an appropriate place for this discussion; the best thing to
do at this point is to either open a bug in the tracker at
bugs.digium.com or find a bug marshal on the #asterisk channel on IRC
(in fact, getting a bug marshal remote access to your system is the
fastest way to get this fixed, as they can run the build process
repeatedly until these problems are solved).
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[Asterisk-Users] Global variables - collision?

2006-05-31 Thread Koen Van Impe

If I edit the value of a global variable in my dialplan, could there be a risk of collision between calls?

More in details: could a global var be used to build a counter that will be incremented by every call that passes.
I think when 2 calls come in almost sumiltaneously, they could both be incrementing and saving the same value... which is bad!

Anybody knows how asterisk handles this?

K
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Re: [Asterisk-Users] Handset recommendations

2006-05-31 Thread Bruce Reeves

I had the same opinions of the F1000, not impressive, but cheap. On 5/31/06, Andrew Kohlsmith [EMAIL PROTECTED]
 wrote:On Tuesday 30 May 2006 23:56, Bruce Reeves wrote: the Linksys. Actually if your budget supports it get one of each and try
 them out. I tried out several from the Cisco 7960, Polycoms and then a UTstarcom F1000. My big concern on the Cisco is that there is a licenseAvoid the UTStarCom F1000G.(802.11g version of F1000).The buttons are far
too close and tiny, the phone is too quiet, the ringers are subpar and thephone tends to lose its wifi for no particular reason.When this happens itjust gives up and does so without warning.It's really quite random as to
when it occurs.Perhaps when the next firmware version is out it'll bebetter.The manufacturer forums are active enough, but I was prettyunderwhelmed with these phones.They *are* inexpensive, though.I will grant them that.
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Re: [Asterisk-Users] Global variables - collision?

2006-05-31 Thread Filip Drągowski





Each variable is specyfied by name and callid 
Call number 1. Executing Set("SIP/X-2749", "DL=0") in new stack
Call number 2. Executing Set("SIP/X-9100", "DL=0") in new stack
X - sip provider login and there is -number (i think that this number
is in HEX)
so every "local" variable have diffrent identity

As You can see Asterisk uses stack so there should be:
1. Executing Set(global_VAR) in new stack 
2. Executing Set(global_VAR) in new stack
No.1 resolves then next ... ther is no simultaneous operation
it's my opinion. 
Try it and see what is shown in * console.

-FD

  If I edit the value of a global variable in my dialplan, could
there be a risk of collision between calls?
   
  More in details: could a global var be used to build a counter
that will be incremented by every call that passes.
  I think when 2 calls come in almost sumiltaneously, they could
both be incrementing and saving the same value... which is bad!
   
  Anybody knows how asterisk handles this?
   
  K
  

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[Asterisk-Users] Zap Channels , for round-robin search and call

2006-05-31 Thread John Joseph

Hi 
   I am using a 4FXO , TDM400P card 
   I am able to call outside , after modifiying
extensions.conf 
  with 

   exten = _9X.,1,Dial(ZAP/1/${EXTEN:1})

 using this , I can only dial through one of the
port , Actually I want to  dial outside using round -
robin  search 
   After reading the manuals , I have plans to
modified the above line as 
 
exten =
_9X.,1,Dial(ZAP/1/${EXTEN:1}|ZAP/2/${EXTEN:1}|ZAP/3/${EXTEN:1}|ZAP/4/${EXTEN:1})

   Please let me know wheter the above line ,  is
correct to use 
 I think , it will dial any one of the four
channel which is available 
  Please  give your comments on the  putting
the line 

exten =
_9X.,1,Dial(ZAP/1/${EXTEN:1}|ZAP/2/${EXTEN:1}|ZAP/3/${EXTEN:1}|ZAP/4/${EXTEN:1})
 
  Thanks 
 Joseph John 



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[Asterisk-Users] Bristuff PickUp and call transfers - can it be done?

2006-05-31 Thread Nick Burch


Hi All

I'm using the PickUp application from Bristuff to allow me to pick up 
channel groups across Zap and Sip. The only snag is that having picked up 
a call with it, you can't then transfer it on.


Taking a dive into app_dial, it looks like when you specify the T option, 
it does:

ast_set_flag((config.features_caller), AST_FEATURE_REDIRECT);

Since there's nothing like that in app_pickup, I guess that's why you 
can't do the transfer (nothing has enabled the flag)



Is there any easy way to allow transfers on calls picked up using PickUp?

Failing that, is there a way to call ast_set_flag on the caller features 
from within the Dialplan, or am I going to have to start hacking 
app_pickup.c to add in the ast_flag_set?


Thanks
Nick
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Re: [Asterisk-Users] Zap Channels , for round-robin search and call

2006-05-31 Thread Gareth Blades

Why not just define a group and use :-
exten = _9X.,1,Dial(ZAP/g1/${EXTEN:1})

On Wed, 2006-05-31 at 13:08, John Joseph wrote:
 Hi 
I am using a 4FXO , TDM400P card 
I am able to call outside , after modifiying
 extensions.conf 
   with 
 
exten = _9X.,1,Dial(ZAP/1/${EXTEN:1})
 
  using this , I can only dial through one of the
 port , Actually I want to  dial outside using round -
 robin  search 
After reading the manuals , I have plans to
 modified the above line as 
  
 exten =
 _9X.,1,Dial(ZAP/1/${EXTEN:1}|ZAP/2/${EXTEN:1}|ZAP/3/${EXTEN:1}|ZAP/4/${EXTEN:1})
 
Please let me know wheter the above line ,  is
 correct to use 
  I think , it will dial any one of the four
 channel which is available 
   Please  give your comments on the  putting
 the line 
 
 exten =
 _9X.,1,Dial(ZAP/1/${EXTEN:1}|ZAP/2/${EXTEN:1}|ZAP/3/${EXTEN:1}|ZAP/4/${EXTEN:1})
  
   Thanks 
  Joseph John 
 
 
   
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[Asterisk-Users] Multiple processes

2006-05-31 Thread Lee Archer

Title: Multiple processes






Can someone shed any light on the following. I have 2 identical systems, 1 of which seems to spawn multiple processes which have to be killed manually. It recently kicked up 2 so I ran gdb on them and this is the thread output. I current use FreePBX with these systems.

1st extra process


(gdb) info thread

 1 Thread 1095261104 (LWP 14213) 0xe410 in __kernel_vsyscall ()

(gdb) thread apply all bt


Thread 1 (Thread 1095261104 (LWP 14213)):

#0 0xe410 in __kernel_vsyscall ()

#1 0x4004f13e in __lll_mutex_lock_wait () from /lib/tls/libpthread.so.0

#2 0x4004be41 in _L_mutex_lock_191 () from /lib/tls/libpthread.so.0

#3 0x0001 in ?? ()

#4 0x40e16778 in ?? () from /usr/lib/asterisk/modules/cdr_odbc.so

#5 0x40e16818 in __dso_handle () from /usr/lib/asterisk/modules/cdr_odbc.so

#6 0x0002 in ?? ()

#7 0x in ?? ()

#8 0x080a20b7 in ast_cdr_unregister (name=0x40e1455c ODBC) at lock.h:592

#9 0x40e13299 in odbc_unload_module () at cdr_odbc.c:240

#10 0x40e13978 in reload () at cdr_odbc.c:465

#11 0x0805be32 in ast_module_reload (name=0x0) at loader.c:257

#12 0x080b4623 in hup_handler (num=-4) at asterisk.c:754

#13 signal handler called

#14 0xe410 in __kernel_vsyscall ()

#15 0x401afd99 in sched_setscheduler () from /lib/tls/libc.so.6

#16 0x080b4743 in ast_set_priority (pri=0) at asterisk.c:803

#17 0x40445ee8 in agi_exec_full (chan=0x82782b0, data="" optimized out, enhanced=0, dead=0) at res_agi.c:300

#18 0x0808d521 in pbx_extension_helper (c=0x82782b0, con=value optimized out, context=0x8278400 macro-record-enable,

 exten=0x82784f4 s, priority=4, label=0x0, callerid=0x8159f38 0163861, action="" at pbx.c:553

#19 0x40c44851 in macro_exec (chan=0x82782b0, data="" at app_macro.c:210

#20 0x0808d521 in pbx_extension_helper (c=0x82782b0, con=value optimized out, context=0x8278400 macro-record-enable,

 exten=0x82784f4 s, priority=7, label=0x0, callerid=0x8159f38 0163861, action="" at pbx.c:553

#21 0x40c44851 in macro_exec (chan=0x82782b0, data="" at app_macro.c:210

#22 0x0808d521 in pbx_extension_helper (c=0x82782b0, con=value optimized out, context=0x8278400 macro-record-enable,

 exten=0x82784f4 s, priority=1, label=0x0, callerid=0x8159f38 0163861, action="" at pbx.c:553

#23 0x0808ead4 in __ast_pbx_run (c=0x82782b0) at pbx.c:2227

#24 0x0808f6cc in pbx_thread (data="" at pbx.c:2514

#25 0x4004a297 in start_thread () from /lib/tls/libpthread.so.0

#26 0x401c737e in clone () from /lib/tls/libc.so.6

#27 0x41485bb0 in ?? ()

#0 0xe410 in __kernel_vsyscall ()


2nd extra process


(gdb) info thread

 1 Thread 1096059824 (LWP 14214) 0xe410 in ?? ()

(gdb) thread apply all bt


Thread 1 (Thread 1096059824 (LWP 14214)):

#0 0xe410 in ?? ()

#1 0x41533594 in ?? ()

#2 0x0002 in ?? ()

#3 0x in ?? ()

#4 0x4004f13e in __lll_mutex_lock_wait () from /lib/tls/libpthread.so.0

#5 0x4004be41 in _L_mutex_lock_191 () from /lib/tls/libpthread.so.0

#6 0x0001 in ?? ()

#7 0x40e16778 in ?? () from /usr/lib/asterisk/modules/cdr_odbc.so

#8 0x40e16818 in __dso_handle () from /usr/lib/asterisk/modules/cdr_odbc.so

#9 0x0002 in ?? ()

#10 0x in ?? ()

#11 0x080a20b7 in ast_cdr_unregister (name=0x40e1455c ODBC) at lock.h:592

#12 0x40e13299 in odbc_unload_module () at cdr_odbc.c:240

#13 0x40e13978 in reload () at cdr_odbc.c:465

#14 0x0805be32 in ast_module_reload (name=0x0) at loader.c:257

#15 0x080b4623 in hup_handler (num=-4) at asterisk.c:754

#16 signal handler called

#17 0xe410 in ?? ()

#0 0xe410 in ?? ()


Regards


Lee


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Re: [Asterisk-Users] Global variables - collision?

2006-05-31 Thread Koen Van Impe

Sounds like a reasonable explanation.
But this means that I should limit the incrementing stuff to one line in the dialplan.

This would be bad:
exten = s,1,Set(Chan_Var=${GlobalVar})
exten = s,2,Set(Chan_Var=$[${Chan_Var} + 1])
exten = s,3,Set(GlobalVar=Chan_Var,g)

Better:
exten = s,1,Set(GlobalVar=$[${GlobalVar} + 1])exten = s,2,Set(Chan_Var=${GlobalVar})
Please confirm...

K


On 5/31/06, Filip Drągowski [EMAIL PROTECTED] wrote:


Each variable is specyfied by name and callid Call number 1. Executing Set(SIP/X-2749, DL=0) in new stackCall number 2. Executing Set(SIP/X-9100, DL=0) in new stack
X - sip provider login and there is -number (i think that this number is in HEX)so every local variable have diffrent identityAs You can see Asterisk uses stack so there should be:1. Executing Set(global_VAR) in new stack 
2. Executing Set(global_VAR) in new stackNo.1 resolves then next ... ther is no simultaneous operationit's my opinion. Try it and see what is shown in * console.-FD


If I edit the value of a global variable in my dialplan, could there be a risk of collision between calls?

More in details: could a global var be used to build a counter that will be incremented by every call that passes.
I think when 2 calls come in almost sumiltaneously, they could both be incrementing and saving the same value... which is bad!

Anybody knows how asterisk handles this?

K

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[Asterisk-Users] Incoming IAX going to wrong context

2006-05-31 Thread Thomas Kenyon

I have (more than 1) provider that I receive calls from using IAX, and I
have 2 IAX deskphones, all work fine except for some reason with 1
provider, when the call comes in, it doesn't match up with the
incomingcall context. (A bit worrying, since I don't want people to be
able to relay calls off me.)

in iax.conf I have:

[ipcomms]
type=user
nat=yes
dtmfmode=rfc2833
host=71.16.179.149
context=incomingcall
disallow=all
allow=gsm
allow=alaw
allow=ulaw

in extensions.conf I have at the bottom of [incomingcall] :

exten = 9546782688,1,AGI,calleridlookup.agi 
(script I wrote to rewrite callerid based on phone number, works on all
other incoming extensions).
exten = 9546782688,2,Answer
exten =
9546782688,3,Dial(SIP/AdamSIP/GarySIP/TomsoneSIP/GarageSIP/BedroomIAX2/OfficeIAX2/Conservatory,90,t)
  
   (household extensions)


From a debug output I get:

Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW
   Timestamp: 00015ms  SCall: 1  DCall: 0 [71.16.179.149:4569]
   VERSION : 2
   CALLED NUMBER   : 9546782688
   CALLING NUMBER  : 441489858029
   LANGUAGE: en
   FORMAT  : 4
   CAPABILITY  : 63502
   ADSICPE : 2
   DATE TIME   : 2006-05-31  08:24:58

-- Accepting UNAUTHENTICATED call from 71.16.179.149:
requested format = ulaw,
requested prefs = (),
actual format = alaw,
host prefs = (alaw|ulaw),
priority = mine

Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass:
ACCEPT
   Timestamp: 00018ms  SCall: 6  DCall: 1 [71.16.179.149:4569]
   FORMAT  : 8

As soon as it passes this, the call gets passed to one of my outgoing
contexts.
I know I must be doing something very silly, but I'm damned if I can see it.

TIA for any help with this.
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RE: [Asterisk-Users] Polycom replacement handset

2006-05-31 Thread Cory Andrews









Ryan  Shoot me an email off list, I
can help you out with a replacement handset.



Thanks





Cory Andrews

Executive Vice President

++

VoIPSupply.com

PBXSelect.com

++

454 Sonwil Drive

Buffalo, NY 14225

voice - 800.398.VoIP X3402

fax - 716.630.1548

e - [EMAIL PROTECTED]

m - 716.907.4059

aim - B2Cory











From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ryan Stark
Sent: Tuesday, May 30, 2006 8:29
PM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Polycom
replacement handset





Does anyone know where I can get replacement handsets for the Polycom
SoundPoint IP phones? Or does anyone have any they want to sell?
>From the looks of it you have to buy a whole new phone to get a new
handset. My vendor, TriaTechCOA, told me I had to buy a whole new phone
to get a handset, which is pretty ridiculous. Maybe there is a more sane
vendor I should be buying from? 

Thanks,
-Ryan






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Re: [Asterisk-Users] Zap Channels , for round-robin search and call

2006-05-31 Thread Koen Van Impe

depending on your zapata.conf file, you should use 

exten = _9X.,1,Dial(Zap/r1/${EXTEN:1})

The little 'r' means round robin, starting at the next highest channel than last time.
Have a look in extensions.conf from the samples for more options.
Make sure you have your 4 channels in one group (group=1).
K

On 5/31/06, John Joseph [EMAIL PROTECTED] wrote:
HiI am using a 4FXO , TDM400P cardI am able to call outside , after modifiyingextensions.conf
withexten = _9X.,1,Dial(ZAP/1/${EXTEN:1})using this , I can only dial through one of theport , Actually I want todial outside using round -robinsearchAfter reading the manuals , I have plans to
modified the above line asexten =_9X.,1,Dial(ZAP/1/${EXTEN:1}|ZAP/2/${EXTEN:1}|ZAP/3/${EXTEN:1}|ZAP/4/${EXTEN:1})Please let me know wheter the above line ,iscorrect to useI think , it will dial any one of the four
channel which is available Pleasegive your comments on theputtingthe lineexten =_9X.,1,Dial(ZAP/1/${EXTEN:1}|ZAP/2/${EXTEN:1}|ZAP/3/${EXTEN:1}|ZAP/4/${EXTEN:1}) Thanks
Joseph John___Yahoo! Messenger - with free PC-PC calling and photo sharing. http://uk.messenger.yahoo.com
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Re: [Asterisk-Users] Multiple processes

2006-05-31 Thread Moises Silva


check the output of getconf GNU_LIBPTHREAD_VERSION

If you see as output linuxthreads-version

then is highly probably that those extra processes that you say, are
nothing more than some of the threads Asterisk needs for other
services.

If you see as output nptl-version

then I think you should see only one Asterisk process.

Regards

On 5/31/06, Lee Archer [EMAIL PROTECTED] wrote:




Can someone shed any light on the following.  I have 2 identical systems, 1
of which seems to spawn multiple processes which have to be killed manually.
 It recently kicked up 2 so I ran gdb on them and this is the thread output.
 I current use FreePBX with these systems.

1st extra process

(gdb) info thread
  1 Thread 1095261104 (LWP 14213)  0xe410 in __kernel_vsyscall ()
(gdb) thread apply all bt

Thread 1 (Thread 1095261104 (LWP 14213)):
#0  0xe410 in __kernel_vsyscall ()
#1  0x4004f13e in __lll_mutex_lock_wait () from /lib/tls/libpthread.so.0
#2  0x4004be41 in _L_mutex_lock_191 () from /lib/tls/libpthread.so.0
#3  0x0001 in ?? ()
#4  0x40e16778 in ?? () from
/usr/lib/asterisk/modules/cdr_odbc.so
#5  0x40e16818 in __dso_handle () from
/usr/lib/asterisk/modules/cdr_odbc.so
#6  0x0002 in ?? ()
#7  0x in ?? ()
#8  0x080a20b7 in ast_cdr_unregister (name=0x40e1455c ODBC) at lock.h:592
#9  0x40e13299 in odbc_unload_module () at cdr_odbc.c:240
#10 0x40e13978 in reload () at cdr_odbc.c:465
#11 0x0805be32 in ast_module_reload (name=0x0) at loader.c:257
#12 0x080b4623 in hup_handler (num=-4) at asterisk.c:754
#13 signal handler called
#14 0xe410 in __kernel_vsyscall ()
#15 0x401afd99 in sched_setscheduler () from /lib/tls/libc.so.6
#16 0x080b4743 in ast_set_priority (pri=0) at asterisk.c:803
#17 0x40445ee8 in agi_exec_full (chan=0x82782b0, data=value optimized out,
enhanced=0, dead=0) at res_agi.c:300
#18 0x0808d521 in pbx_extension_helper (c=0x82782b0, con=value optimized
out, context=0x8278400 macro-record-enable,

exten=0x82784f4 s, priority=4, label=0x0, callerid=0x8159f38
0163861, action=1) at pbx.c:553
#19 0x40c44851 in macro_exec (chan=0x82782b0, data=0x4147c768) at
app_macro.c:210
#20 0x0808d521 in pbx_extension_helper (c=0x82782b0, con=value optimized
out, context=0x8278400 macro-record-enable,

exten=0x82784f4 s, priority=7, label=0x0, callerid=0x8159f38
0163861, action=1) at pbx.c:553
#21 0x40c44851 in macro_exec (chan=0x82782b0, data=0x41482fd8) at
app_macro.c:210
#22 0x0808d521 in pbx_extension_helper (c=0x82782b0, con=value optimized
out, context=0x8278400 macro-record-enable,

exten=0x82784f4 s, priority=1, label=0x0, callerid=0x8159f38
0163861, action=1) at pbx.c:553
#23 0x0808ead4 in __ast_pbx_run (c=0x82782b0) at pbx.c:2227
#24 0x0808f6cc in pbx_thread (data=0x0) at pbx.c:2514
#25 0x4004a297 in start_thread () from /lib/tls/libpthread.so.0
#26 0x401c737e in clone () from /lib/tls/libc.so.6
#27 0x41485bb0 in ?? ()
#0  0xe410 in __kernel_vsyscall ()

2nd extra process

(gdb) info thread
  1 Thread 1096059824 (LWP 14214)  0xe410 in ?? ()
(gdb) thread apply all bt

Thread 1 (Thread 1096059824 (LWP 14214)):
#0  0xe410 in ?? ()
#1  0x41533594 in ?? ()
#2  0x0002 in ?? ()
#3  0x in ?? ()
#4  0x4004f13e in __lll_mutex_lock_wait () from /lib/tls/libpthread.so.0
#5  0x4004be41 in _L_mutex_lock_191 () from /lib/tls/libpthread.so.0
#6  0x0001 in ?? ()
#7  0x40e16778 in ?? () from
/usr/lib/asterisk/modules/cdr_odbc.so
#8  0x40e16818 in __dso_handle () from
/usr/lib/asterisk/modules/cdr_odbc.so
#9  0x0002 in ?? ()
#10 0x in ?? ()
#11 0x080a20b7 in ast_cdr_unregister (name=0x40e1455c ODBC) at lock.h:592
#12 0x40e13299 in odbc_unload_module () at cdr_odbc.c:240
#13 0x40e13978 in reload () at cdr_odbc.c:465
#14 0x0805be32 in ast_module_reload (name=0x0) at loader.c:257
#15 0x080b4623 in hup_handler (num=-4) at asterisk.c:754
#16 signal handler called
#17 0xe410 in ?? ()
#0  0xe410 in ?? ()

Regards

Lee ###

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Re: [Asterisk-Users] Global variables - collision?

2006-05-31 Thread Filip Drągowski





This looks very well:
exten = s,1,Set(GlobalVar=$[${GlobalVar} + 1])
exten = s,2,Set(Chan_Var=${GlobalVar})

s,1 gives You incrementing in one line
setting ChanVar (local variable i presume) don't bother GlobalVar.

-FD

Użytkownik Koen Van Impe napisał:

  Sounds like a reasonable explanation.
  But this means that I should limit the incrementing stuff to one
line in the dialplan.
   
  This would be bad:
  exten = s,1,Set(Chan_Var=${GlobalVar})
  exten = s,2,Set(Chan_Var=$[${Chan_Var} + 1])
  exten = s,3,Set(GlobalVar=Chan_Var,g)
   
  Better:
  exten = s,1,Set(GlobalVar=$[${GlobalVar} + 1])
exten = s,2,Set(Chan_Var=${GlobalVar})
 
  Please confirm...
   
  K
   
   
  On 5/31/06, Filip Drągowski [EMAIL PROTECTED] wrote:
  

Each variable is specyfied by
name and callid 
Call number 1. Executing Set("SIP/X-2749", "DL=0") in new stack
Call number 2. Executing Set("SIP/X-9100", "DL=0") in new stack

X - sip provider login and there is -number (i think that this number
is in HEX)
so every "local" variable have diffrent identity

As You can see Asterisk uses stack so there should be:
1. Executing Set(global_VAR) in new stack 
2. Executing Set(global_VAR) in new stack
No.1 resolves then next ... ther is no simultaneous operation
it's my opinion. 
Try it and see what is shown in * console.

-FD


If I edit the value of a global variable in my dialplan, could
there be a risk of collision between calls?
 
More in details: could a global var be used to build a counter
that will be incremented by every call that passes.
I think when 2 calls come in almost sumiltaneously, they could
both be incrementing and saving the same value... which is bad!
 
Anybody knows how asterisk handles this?
 
K





  
  





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RE: [Asterisk-Users] Multiple processes

2006-05-31 Thread Lee Archer

I get NPTL 2.3.5.  It's only on 1 box and after a while there are so
many that it stops calls.  On the other box and the other test boxes I
have its only 1 asterisk process.

Regards

Lee 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Moises
Silva
Sent: 31 May 2006 14:56
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Multiple processes

check the output of getconf GNU_LIBPTHREAD_VERSION

If you see as output linuxthreads-version

then is highly probably that those extra processes that you say, are
nothing more than some of the threads Asterisk needs for other services.

If you see as output nptl-version

then I think you should see only one Asterisk process.

Regards

On 5/31/06, Lee Archer [EMAIL PROTECTED] wrote:



 Can someone shed any light on the following.  I have 2 identical 
 systems, 1 of which seems to spawn multiple processes which have to be
killed manually.
  It recently kicked up 2 so I ran gdb on them and this is the thread
output.
  I current use FreePBX with these systems.

 1st extra process

 (gdb) info thread
   1 Thread 1095261104 (LWP 14213)  0xe410 in __kernel_vsyscall ()
 (gdb) thread apply all bt

 Thread 1 (Thread 1095261104 (LWP 14213)):
 #0  0xe410 in __kernel_vsyscall ()
 #1  0x4004f13e in __lll_mutex_lock_wait () from 
 /lib/tls/libpthread.so.0
 #2  0x4004be41 in _L_mutex_lock_191 () from /lib/tls/libpthread.so.0
 #3  0x0001 in ?? ()
 #4  0x40e16778 in ?? () from
 /usr/lib/asterisk/modules/cdr_odbc.so
 #5  0x40e16818 in __dso_handle () from 
 /usr/lib/asterisk/modules/cdr_odbc.so
 #6  0x0002 in ?? ()
 #7  0x in ?? ()
 #8  0x080a20b7 in ast_cdr_unregister (name=0x40e1455c ODBC) at 
 lock.h:592
 #9  0x40e13299 in odbc_unload_module () at cdr_odbc.c:240 #10 
 0x40e13978 in reload () at cdr_odbc.c:465
 #11 0x0805be32 in ast_module_reload (name=0x0) at loader.c:257
 #12 0x080b4623 in hup_handler (num=-4) at asterisk.c:754
 #13 signal handler called
 #14 0xe410 in __kernel_vsyscall ()
 #15 0x401afd99 in sched_setscheduler () from /lib/tls/libc.so.6
 #16 0x080b4743 in ast_set_priority (pri=0) at asterisk.c:803
 #17 0x40445ee8 in agi_exec_full (chan=0x82782b0, data=value optimized

 out, enhanced=0, dead=0) at res_agi.c:300
 #18 0x0808d521 in pbx_extension_helper (c=0x82782b0, con=value 
 optimized
 out, context=0x8278400 macro-record-enable,

 exten=0x82784f4 s, priority=4, label=0x0, callerid=0x8159f38 
 0163861, action=1) at pbx.c:553
 #19 0x40c44851 in macro_exec (chan=0x82782b0, data=0x4147c768) at 
 app_macro.c:210 #20 0x0808d521 in pbx_extension_helper (c=0x82782b0, 
 con=value optimized
 out, context=0x8278400 macro-record-enable,

 exten=0x82784f4 s, priority=7, label=0x0, callerid=0x8159f38 
 0163861, action=1) at pbx.c:553
 #21 0x40c44851 in macro_exec (chan=0x82782b0, data=0x41482fd8) at 
 app_macro.c:210
 #22 0x0808d521 in pbx_extension_helper (c=0x82782b0, con=value 
 optimized
 out, context=0x8278400 macro-record-enable,

 exten=0x82784f4 s, priority=1, label=0x0, callerid=0x8159f38 
 0163861, action=1) at pbx.c:553
 #23 0x0808ead4 in __ast_pbx_run (c=0x82782b0) at pbx.c:2227
 #24 0x0808f6cc in pbx_thread (data=0x0) at pbx.c:2514
 #25 0x4004a297 in start_thread () from /lib/tls/libpthread.so.0
 #26 0x401c737e in clone () from /lib/tls/libc.so.6
 #27 0x41485bb0 in ?? ()
 #0  0xe410 in __kernel_vsyscall ()

 2nd extra process

 (gdb) info thread
   1 Thread 1096059824 (LWP 14214)  0xe410 in ?? ()
 (gdb) thread apply all bt

 Thread 1 (Thread 1096059824 (LWP 14214)):
 #0  0xe410 in ?? ()
 #1  0x41533594 in ?? ()
 #2  0x0002 in ?? ()
 #3  0x in ?? ()
 #4  0x4004f13e in __lll_mutex_lock_wait () from 
 /lib/tls/libpthread.so.0
 #5  0x4004be41 in _L_mutex_lock_191 () from /lib/tls/libpthread.so.0
 #6  0x0001 in ?? ()
 #7  0x40e16778 in ?? () from
 /usr/lib/asterisk/modules/cdr_odbc.so
 #8  0x40e16818 in __dso_handle () from 
 /usr/lib/asterisk/modules/cdr_odbc.so
 #9  0x0002 in ?? ()
 #10 0x in ?? ()
 #11 0x080a20b7 in ast_cdr_unregister (name=0x40e1455c ODBC) at 
 lock.h:592
 #12 0x40e13299 in odbc_unload_module () at cdr_odbc.c:240
 #13 0x40e13978 in reload () at cdr_odbc.c:465
 #14 0x0805be32 in ast_module_reload (name=0x0) at loader.c:257
 #15 0x080b4623 in hup_handler (num=-4) at asterisk.c:754
 #16 signal handler called
 #17 0xe410 in ?? ()
 #0  0xe410 in ?? ()

 Regards

 Lee ###

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Re: [Asterisk-Users] Compiling Asterisk-addons

2006-05-31 Thread Armin Schindler

On Wed, 31 May 2006, Kevin P. Fleming wrote:
 Armin Schindler wrote:
 
  Actually, the error is not in the addons. The Asterisk-trunk installation
  produces incomplete/misconfigured headers, which prevents building of 
  external modules.
  Or is it digiums intention to make life more difficult for external modules?
 
 Are you serious? How could that possibly be the case?
 
 I just don't understand why you persist in making statements like this,
 when it has been clearly stated that trunk is a _development_ area and
 we will fix these problems before any beta releases are made.

I know and it is alright that the trunk is in development of course.

I don't persist in anything. The first mail was just that I noted about the 
fact that the trunk doesn't allow external modules at this time.
But when I read your statement:

   The answer is: asterisk-addons will not be brought up to date with SVN
   trunk until SVN trunk enters the beta phase, which will occur in the
   next week.

I got the impression that trunk won't be changed and you will make the 
addons compile against this trunk status (which I would find ugly).
But if I misunderstood you and you plan to fix trunk, then I'm sorry
for causing any noise.

regards
Armin
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Re: [Asterisk-Users] Handset recommendations

2006-05-31 Thread Philippe Lindheimer

I am also a VERY happy with the Polycom 501. I will disagree with the comment "They Just Work." when it comes to NAT. Once they are setup properly (e.g. set registration timeout to 60 sec so it registers every 30 seconds and keeps NAT holes open) then they work fine. There is good info on the wiki on how to set them up.Question I have on the Aastra - isn't it 2.4Ghz, so potential interference with wireless?One thing to consider with Polycom or other phones, is you can get a wireless headset arrangement as a compromise. Another option is to get a second extension with an ATA and your favorite 5.8Ghz cordless phone.p From: Andrew Kohlsmith [EMAIL PROTECTED]To:
asterisk-users@lists.digium.comDate: Wed, 31 May 2006 06:34:31 -0400Subject: Re: [Asterisk-Users] Handset recommendationsOn Tuesday 30 May 2006 23:13, George A. Roberts IV wrote: Any other thoughts on good reasonably priced handsets? This is for just a couple of people who work from home offices and will be connecting to an Asterisk server hosted in a datacenter.I am a *huge* fan of the Polycom ip501. The 301 works just as well, but the display is significantly crappier. If you've got the cash, go 601.I have never used Cisco, but I've used the cheaper phones enough to know that this is one place were spending a little more is WELL worth it. And I know from personal experience that the Polycom phones have *zero* issues with being behind NAT and talking to a public-IP Asterisk box. No firewall configuration, no screwing around whatsoever. They Just Work.-A.
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[Asterisk-Users] Pickup problem

2006-05-31 Thread Bogdan Tocu

Hi,I have the folowing setup:[incoming]exten = s,1,Wait(3)exten = s,2,Answerexten = s,3,Background(welcome)exten = 1,1,Noop(call for operators)exten = 1,2,Dial(SIP/10SIP/11|60|tr)
exten = 1,3,Hungup;this is for pulse phones exten = t,1,NoOp(.call for .60)exten = t,2,Dial(SIP/10,60,mtr)exten = t,3,Background(busy-retrylater)exten = t,4,Hungup
[take_call]exten = _6ZX,1,Background(pickup)exten = _6ZX,2,Pickup(${EXTEN:1})[sip_users]include = take_call;this is the context for sip usersNow when an incoming caller press 1 ... it cals sip 10 and sip 11. If me sip 22 want to pickup sip/10 or sip/11 by dialing 611 or 610 noting happends.
On asterisk CLI says:-- Executing Pickup(SIP/22-e7f0, 11) in new stackAny ideea why it does'nt work? BTW on internal calls pickup works just fine.
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Re: [Asterisk-Users] Handset recommendations

2006-05-31 Thread Andrew Kohlsmith

On Wednesday 31 May 2006 10:12, Philippe Lindheimer wrote:
 I am also a VERY happy with the Polycom 501. I will disagree with the
 comment They Just Work. when it comes to NAT. Once they are setup
 properly (e.g. set registration timeout to 60 sec so it registers every 30
 seconds and keeps NAT holes open) then they work fine. There is good info
 on the wiki on how to set them up.

I made *no* registration changes to the default values.  Perhaps your router 
had its NAT timeout window set really short?  I am using a 
totally-factory-standard WRT54G.

-A.
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Re: [Asterisk-Users] Pickup problem

2006-05-31 Thread Joshua Colp


Bogdan Tocu wrote:

Hi,

I have the folowing setup:

[incoming]
exten = s,1,Wait(3)
exten = s,2,Answer
exten = s,3,Background(welcome)

exten = 1,1,Noop(call for operators)
exten = 1,2,Dial(SIP/10SIP/11|60|tr)
exten = 1,3,Hungup


;this is for pulse phones
exten = t,1,NoOp(.call for .60)
exten = t,2,Dial(SIP/10,60,mtr)
exten = t,3,Background(busy-retrylater)
exten = t,4,Hungup



[take_call]
exten = _6ZX,1,Background(pickup)
exten = _6ZX,2,Pickup(${EXTEN:1})


[sip_users]
include = take_call
;this is the context for sip users

Now when an incoming caller press 1 ... it cals sip 10 and sip 11. If me 
sip 22 want to pickup sip/10 or sip/11 by dialing 611 or 610 noting 
happends.

On asterisk CLI says:
 -- Executing Pickup(SIP/22-e7f0, 11) in new stack

Any ideea why it does'nt work? BTW on internal calls pickup works just fine.




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That's because the extension dialed was 1. Using the Pickup application 
you can't do a Pickup on the device called (ie: SIP/10 or SIP/11) but 
the extension, which is 1.


--
Joshua Colp
Software Developer
Digium
P - 256-428-6066
C - 506-878-0147
[EMAIL PROTECTED]
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Re: [Asterisk-Users] instalacion

2006-05-31 Thread Hermann Wecke


samuel wrote:
I am of Argentina, and I do not speak very well English, I cannot 
install asterisk in red hat 9.


Don't send HTML messages to the list.
Install [EMAIL PROTECTED] Please remember that [EMAIL PROTECTED] will erase all data on 
your HD.

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RE: [Asterisk-Users] AEL #include

 -Original Message-
 From: Michael Collins [mailto:[EMAIL PROTECTED]
 Sent: Tuesday, May 30, 2006 10:28 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: RE: [Asterisk-Users] AEL #include

  How would goto work if all the priorities where n?
 ...
  Example from one of my dial plans:

  exten = talk,1,ForkCDR
  exten = talk,n,Set(NUMTRIES=1)
  exten = talk,n,GotoIf($[${NUMTRIES} = 1]?first)
  exten = talk,n(repeat),Background(Initial-greeting)
  exten = talk,n(first),Wait(.1)
  exten = talk,n,Festival(${fname})
  exten = talk,n,Festival(${lname})
  exten = talk,n,Background(If-person1)
  exten = talk,n,GotoIf($[${NUMTRIES}  2]?repeat)
  exten = talk,n,Set(NUMTRIES=$[${NUMTRIES}+1])
  exten = talk,n,Goto(t,1)

 Doug,

 Like Kevin said, the label takes the place of the hard-numbered
 priorities.  In the example of my talk extension, I have a pair of
 GotoIf() commands.  This particular extension is used for an
 experimental voice-broadcasting system that I'm playing with.  

 As you can see, the first priority is hard-coded as '1' but each
 subsequent priority is simply 'n'.  It starts at 1, does the ForkCDR,
 then moves on to the next priority.  Since there is no '2' 
 priority, it
 simply moves to the next 'n' priority the extension.  In this case, it
 just sets my NUMTRIES variable then moves to the next 'n' priority,
 which is the first of the two GotoIf() commands.  Notice the first
 after the ? in the GotoIf().  Instead of putting in a 
 numbered priority,
 I put in a label.  In this case, the GotoIf() is saying, If 
 this is the
 first attempt, i.e. NUMTRIES equals 1, then goto the priority labeled
 'first', otherwise just move on to the next priority.  If NUMTRIES is
 '1' then the GotoIf sends the processing to this priority:
 exten = talk,n(first),Wait(.1)

 From there the processing continues.

 I have a feeling that if you aren't using labels and you have many
 Goto()'s and GotoIf()'s then you'll LOVE labels.  Once you get your
 labels in place you will almost never have to renumber your 
 priorities. 

Thanks Michael. I was not aware that labels where available.
In converting though, I've already hit a limitation. There's a single name 
space for all labels I assume?
When you have multiple loops and things of a similar nature, you have to start 
making you label names unique, to the point where they are no longer simple, 
and don't make a lot of sense anymore.
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Re: [Asterisk-Users] Handset recommendations

2006-05-31 Thread Michael Graves



I have a 480i CT as my primary deskphone now for about a year. Prior to that my favorite was the Polycom IP600. I also have a handful of IP500s.



The 480i CT is a GREAT phone for the money. The cordless works well, much better than the early sip wifi handset I tried (Hitachi WIP-5000.) 



The Polycoms are great too...but no cordless and harder to provision.



Cisco hardware seemed like a lot of hassle to license as well as very costly. 



Early on I used SPA-3000 and SPA-2000 units connected to a Panasonic key system. Once you've tried real SIP hard phones ATA's just kinda lose their appeal.



Michael







--Original Message Text---

From: George A. Roberts IV

Date: Tue, 30 May 2006 22:13:25 -0500



Seeking recommendations on handsets for use with Asterisk. 

 

I've been looking at the Aastra 480i CT because of its cordless handset and also the new Linksys SPA-942.  Anyone using either one of these with comments on them? 

 

Any other thoughts on good reasonably priced handsets?  This is for just a couple of people who work from home offices and will be connecting to an Asterisk server hosted in a datacenter. 

 

Thanks! 

 

Regards, 

 

George A. Roberts IV 

President and CEO, Interjuncture Corp. 

http://www.interjuncture.com/ 








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Re: [Asterisk-Users] XML to monitor queues on Cisco display ?

2006-05-31 Thread Jean-Louis curty

I'm still looking for this missing feature of a good asterisk ACD,
I have a qview.pl which display the queue state but it's on the pc not on the phone display,

shall we start a bounty , is there any developper here who knows how to do it , where to start ?
thanks
jl
2004/12/7, Brian Roy [EMAIL PROTECTED]:
On Sat, 4 Dec 2004 13:08:19 -0600, Joe Dennick [EMAIL PROTECTED] wrote:
 I, too would be very interested in this application.We are also building an application to handle this. The desktop app isbuilt in Java and will have a java proxy component (running inwebsphere) that talks to the manager. We are probably 3 weeks away
from putting anything usable out there, but I would be glad to giveback once it does.BTW, ours also has screen pop functionality so that it calls ourvertical software package. I've also created a perl app that reads the
queue.log and pipes all that info in to SQL server.-Chuji___Asterisk-Users mailing listAsterisk-Users@lists.digium.com
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Re: [Asterisk-Users] Handset recommendations

2006-05-31 Thread Michael Graves

--Original Message Text---

From: Philippe Lindheimer

Date: Wed, 31 May 2006 07:12:01 -0700 (PDT)

Question I have on the Aastra - isn't it 2.4Ghz, so potential interference with wireless?

I have no trouble with this is my office. It's only 500 sq ft. I have a G type AP right in the same room and a B type nearby as well. The range on the 480 cordless handset is litterally 5x that of a wifi handset.

Michael

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[Asterisk-Users] DTMF Again

2006-05-31 Thread Akpome Akpoguma

After reading couple of threads on this list on DTMF, I dont expect to send 
this mail ..anyway I have a problem that has to do with dtmf.


I have both a sip phone and an fxs card connected to my asterisk. The fxs is 
connected to an Alcatel PBX. On the sip phone, dtmf is working after I 
included the following line in sip.conf


dtmfmode = inband

But when I tried to pickup dtmf from an Alcatel digital phone connected to 
the Alcatel PBX on my asterisk I noticed some problems. I cound interrupt 
Background() sound with any key on the phone but that is abount all I can 
do.


I set the line --relaxdtmf = no- and toggled to yes in 
zapata.conf and the problem remains.


Question is, is there any other config file that can be edited to solve this 
problem? or anything else to be done to solve this??


Your response would be appriciated

_
FREE pop-up blocking with the new MSN Toolbar - get it now! 
http://toolbar.msn.click-url.com/go/onm00200415ave/direct/01/


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Re: [Asterisk-Users] Pickup problem

2006-05-31 Thread Bogdan Tocu

Thank you, you are perfectly right..:) It's logic, but, didn't cross my mind.. On 5/31/06, Joshua Colp [EMAIL PROTECTED]
 wrote:Bogdan Tocu wrote: Hi, I have the folowing setup:
 [incoming] exten = s,1,Wait(3) exten = s,2,Answer exten = s,3,Background(welcome) exten = 1,1,Noop(call for operators) exten = 1,2,Dial(SIP/10SIP/11|60|tr)
 exten = 1,3,Hungup ;this is for pulse phones exten = t,1,NoOp(.call for .60) exten = t,2,Dial(SIP/10,60,mtr) exten = t,3,Background(busy-retrylater)
 exten = t,4,Hungup [take_call] exten = _6ZX,1,Background(pickup) exten = _6ZX,2,Pickup(${EXTEN:1}) [sip_users] include = take_call
 ;this is the context for sip users Now when an incoming caller press 1 ... it cals sip 10 and sip 11. If me sip 22 want to pickup sip/10 or sip/11 by dialing 611 or 610 noting happends.
 On asterisk CLI says:-- Executing Pickup(SIP/22-e7f0, 11) in new stack Any ideea why it does'nt work? BTW on internal calls pickup works just fine.
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That's because the extension dialed was 1. Using the Pickup applicationyou can't do a Pickup on the device called (ie: SIP/10 or SIP/11) butthe extension, which is 1.--Joshua ColpSoftware Developer
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[Asterisk-Users] Labels and Goto()

I just discovered labels in the dialplan.

Maybe someone (hint: the author) could like, do something crazy, and say, 
update the unofficial docs on voip-user? There's nothing there about labels in 
the pages for extensions.conf OR the Goto() command.

I'm not going to do it. I've realised that when people post documentation about 
something they think they understand, on incorrect or incomplete assumptions, 
that they invariably post something wrong, which just fuels more incorrect 
information, and hurts others. That's why it's the developers that should write 
the docs.

Anyway... How can I use Goto() to jump to a label in a different extension or 
context?

When you have a lot of loops and such in a single extension, you end up wanting 
to use multiple labels called 'start', 'next' etc. I assume(hope!) that the 
namespace of labels is in a single context? ie can you use the same label name 
in another context? You then have to break a single extension up into multiple 
ones JUST to be able to use labels effectively.

Doug.




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Re: [Asterisk-Users] XML to monitor queues on Cisco display ?

2006-05-31 Thread Andrew Latham


This wouldn't be hard, you would neet to raise the refresh via the XML
document but all would work well.  Some one wrote some stuff on this
for the VoIP hacks book from Oreilly and I think that would be a great
place to start.  :)

Keep in mind that XML services are rather universal and should work on
many devices so if you bounty it or write it do so in a way to allow
device configuration later.



On 9/12/05, Jean-Louis curty [EMAIL PROTECTED] wrote:

anybody succeeded with this issue ? I mean writing xml code to display
queue info on the Cisco screen

jl


2004/12/7, Brian Roy [EMAIL PROTECTED]:
 On Sat, 4 Dec 2004 13:08:19 -0600, Joe Dennick [EMAIL PROTECTED] wrote:
  I, too would be very interested in this application.

 We are also building an application to handle this. The desktop app is
 built in Java and will have a java proxy component (running in
 websphere) that talks to the manager. We are probably 3 weeks away
 from putting anything usable out there, but I would be glad to give
 back once it does.

 BTW, ours also has screen pop functionality so that it calls our
 vertical software package. I've also created a perl app that reads the
 queue.log and pipes all that info in to SQL server.

 -Chuji
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--
---
Andrew Latham - AKA: LATHAMA (lay-th-ham-eh)
[EMAIL PROTECTED] - [EMAIL PROTECTED]
If any of the above are down we have bigger problems than my email!
Hind sight is most always 20/20 or better.
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Re: [Asterisk-Users] Multiple processes

2006-05-31 Thread Moises Silva


On 5/31/06, Lee Archer [EMAIL PROTECTED] wrote:

I get NPTL 2.3.5.  It's only on 1 box and after a while there are so
many that it stops calls.  On the other box and the other test boxes I
have its only 1 asterisk process.

Regards

Lee

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Moises
Silva
Sent: 31 May 2006 14:56
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Multiple processes

check the output of getconf GNU_LIBPTHREAD_VERSION

If you see as output linuxthreads-version

then is highly probably that those extra processes that you say, are
nothing more than some of the threads Asterisk needs for other services.

If you see as output nptl-version

then I think you should see only one Asterisk process.

Regards

On 5/31/06, Lee Archer [EMAIL PROTECTED] wrote:

I think you should consider to fill a bug in http://bugs.digium.com/,
make sure you read the bug guidelines before proceeding.
http://www.digium.com/bugguidelines.html

Regards




 Can someone shed any light on the following.  I have 2 identical
 systems, 1 of which seems to spawn multiple processes which have to be
killed manually.
  It recently kicked up 2 so I ran gdb on them and this is the thread
output.
  I current use FreePBX with these systems.

 1st extra process

 (gdb) info thread
   1 Thread 1095261104 (LWP 14213)  0xe410 in __kernel_vsyscall ()
 (gdb) thread apply all bt

 Thread 1 (Thread 1095261104 (LWP 14213)):
 #0  0xe410 in __kernel_vsyscall ()
 #1  0x4004f13e in __lll_mutex_lock_wait () from
 /lib/tls/libpthread.so.0
 #2  0x4004be41 in _L_mutex_lock_191 () from /lib/tls/libpthread.so.0
 #3  0x0001 in ?? ()
 #4  0x40e16778 in ?? () from
 /usr/lib/asterisk/modules/cdr_odbc.so
 #5  0x40e16818 in __dso_handle () from
 /usr/lib/asterisk/modules/cdr_odbc.so
 #6  0x0002 in ?? ()
 #7  0x in ?? ()
 #8  0x080a20b7 in ast_cdr_unregister (name=0x40e1455c ODBC) at
 lock.h:592
 #9  0x40e13299 in odbc_unload_module () at cdr_odbc.c:240 #10
 0x40e13978 in reload () at cdr_odbc.c:465
 #11 0x0805be32 in ast_module_reload (name=0x0) at loader.c:257
 #12 0x080b4623 in hup_handler (num=-4) at asterisk.c:754
 #13 signal handler called
 #14 0xe410 in __kernel_vsyscall ()
 #15 0x401afd99 in sched_setscheduler () from /lib/tls/libc.so.6
 #16 0x080b4743 in ast_set_priority (pri=0) at asterisk.c:803
 #17 0x40445ee8 in agi_exec_full (chan=0x82782b0, data=value optimized

 out, enhanced=0, dead=0) at res_agi.c:300
 #18 0x0808d521 in pbx_extension_helper (c=0x82782b0, con=value
 optimized
 out, context=0x8278400 macro-record-enable,

 exten=0x82784f4 s, priority=4, label=0x0, callerid=0x8159f38
 0163861, action=1) at pbx.c:553
 #19 0x40c44851 in macro_exec (chan=0x82782b0, data=0x4147c768) at
 app_macro.c:210 #20 0x0808d521 in pbx_extension_helper (c=0x82782b0,
 con=value optimized
 out, context=0x8278400 macro-record-enable,

 exten=0x82784f4 s, priority=7, label=0x0, callerid=0x8159f38
 0163861, action=1) at pbx.c:553
 #21 0x40c44851 in macro_exec (chan=0x82782b0, data=0x41482fd8) at
 app_macro.c:210
 #22 0x0808d521 in pbx_extension_helper (c=0x82782b0, con=value
 optimized
 out, context=0x8278400 macro-record-enable,

 exten=0x82784f4 s, priority=1, label=0x0, callerid=0x8159f38
 0163861, action=1) at pbx.c:553
 #23 0x0808ead4 in __ast_pbx_run (c=0x82782b0) at pbx.c:2227
 #24 0x0808f6cc in pbx_thread (data=0x0) at pbx.c:2514
 #25 0x4004a297 in start_thread () from /lib/tls/libpthread.so.0
 #26 0x401c737e in clone () from /lib/tls/libc.so.6
 #27 0x41485bb0 in ?? ()
 #0  0xe410 in __kernel_vsyscall ()

 2nd extra process

 (gdb) info thread
   1 Thread 1096059824 (LWP 14214)  0xe410 in ?? ()
 (gdb) thread apply all bt

 Thread 1 (Thread 1096059824 (LWP 14214)):
 #0  0xe410 in ?? ()
 #1  0x41533594 in ?? ()
 #2  0x0002 in ?? ()
 #3  0x in ?? ()
 #4  0x4004f13e in __lll_mutex_lock_wait () from
 /lib/tls/libpthread.so.0
 #5  0x4004be41 in _L_mutex_lock_191 () from /lib/tls/libpthread.so.0
 #6  0x0001 in ?? ()
 #7  0x40e16778 in ?? () from
 /usr/lib/asterisk/modules/cdr_odbc.so
 #8  0x40e16818 in __dso_handle () from
 /usr/lib/asterisk/modules/cdr_odbc.so
 #9  0x0002 in ?? ()
 #10 0x in ?? ()
 #11 0x080a20b7 in ast_cdr_unregister (name=0x40e1455c ODBC) at
 lock.h:592
 #12 0x40e13299 in odbc_unload_module () at cdr_odbc.c:240
 #13 0x40e13978 in reload () at cdr_odbc.c:465
 #14 0x0805be32 in ast_module_reload (name=0x0) at loader.c:257
 #15 0x080b4623 in hup_handler (num=-4) at asterisk.c:754
 #16 signal handler called
 #17 0xe410 in ?? ()
 #0  0xe410 in ?? ()

 Regards

 Lee ###

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[Asterisk-Users] SIP Presence

2006-05-31 Thread Forrest Beck









Does anyone have a working implementation of SIP Presence?
I have a new Grandstream GX-2000 phone with the supported hardware and I am not
sure how to setup presence with asterisk. 






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Re: [Asterisk-Users] Labels and Goto()

2006-05-31 Thread Julian Lyndon-Smith


Um, followed the link on the extensions.conf page on voip-info

Since Asterisk 1.2 there is a new way to work around this. Number the 
first priority and name the following priorities n. See Asterisk 
Priorities for further details!


to

http://www.voip-info.org/wiki/index.php?page=Asterisk%20priorities

Explains labels there ...

Douglas Garstang wrote:

I just discovered labels in the dialplan.

Maybe someone (hint: the author) could like, do something crazy, and say, 
update the unofficial docs on voip-user? There's nothing there about labels in 
the pages for extensions.conf OR the Goto() command.

I'm not going to do it. I've realised that when people post documentation about 
something they think they understand, on incorrect or incomplete assumptions, 
that they invariably post something wrong, which just fuels more incorrect 
information, and hurts others. That's why it's the developers that should write 
the docs.

Anyway... How can I use Goto() to jump to a label in a different extension or 
context?

When you have a lot of loops and such in a single extension, you end up wanting 
to use multiple labels called 'start', 'next' etc. I assume(hope!) that the 
namespace of labels is in a single context? ie can you use the same label name 
in another context? You then have to break a single extension up into multiple 
ones JUST to be able to use labels effectively.

Doug.




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RE: [Asterisk-Users] AEL #include

 Thanks Michael. I was not aware that labels where available.
 In converting though, I've already hit a limitation. There's a single
name
 space for all labels I assume?

Doug,

According to TFOT's Goto() application reference entry (page 254) the
namespace is actually the current extension:
Named priorities only work within the current extension.

So you can have 50 different labels called start as long as you use
start only once per extension.  If you're in extension 555 and you
Goto(start) it will look for exten = 555,n(start),...  If it doesn't
find a label start in exten 555 then the Goto() will fail, even if you
have start in another extension:
exten = 556,n(start),Noop(this start good only from exten 556)

HtH.

I can see a potential issue if you need to jump from one exten to
another exten using Goto().  You still need to use
Goto(context,extension,priority) to jump around like that.  Do you have
any scenarios like that?  If so, it might be possible to create numbered
jump-to points that will never change, and therefore won't require
renumbering each time you make an addition to the dialplan.

Example:

[test_context]
exten = 555,1,Noop(Starting exten 555)
exten = 555,n,dialplan stuff
exten = 555,n,Goto(test_context,556,999) 
; previous line will end up at 556,n(start)

exten = 556,1,Noop(Starting exten 556)
exten = 556,n,dialplan stuff
exten = 556,n(start),Noop(This is where I want to be)
exten = 556,n,more dialplan stuff
exten = 556,999,Goto(start)
; previous line used to allow other exten's to jump to 556,n(start)



FYI, your other post just came in.  I think I just answered a few of
your questions.  Let us know if this helps!

-MC
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RE: [Asterisk-Users] Labels and Goto()

 
 Anyway... How can I use Goto() to jump to a label in a different
extension
 or context?
 
 When you have a lot of loops and such in a single extension, you end
up
 wanting to use multiple labels called 'start', 'next' etc. I
assume(hope!)
 that the namespace of labels is in a single context? ie can you use
the
 same label name in another context? You then have to break a single
 extension up into multiple ones JUST to be able to use labels
effectively.
 

Doug, please see my previous post in the 'AEL # include thread' - I was
able to answer these questions.

-MC
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Re: [Asterisk-Users] Labels and Goto()

2006-05-31 Thread Aaron Daniel

I've never messed with labels myself... good old priorities tend to keep 
me on my toes :)


But here's a page specifically for priorities, see if it helps:
http://www.voip-info.org/wiki/index.php?page=Asterisk%20priorities

On Wed, 31 May 2006, Douglas Garstang wrote:


I just discovered labels in the dialplan.

Maybe someone (hint: the author) could like, do something crazy, and say, 
update the unofficial docs on voip-user? There's nothing there about labels in 
the pages for extensions.conf OR the Goto() command.

I'm not going to do it. I've realised that when people post documentation about 
something they think they understand, on incorrect or incomplete assumptions, 
that they invariably post something wrong, which just fuels more incorrect 
information, and hurts others. That's why it's the developers that should write 
the docs.

Anyway... How can I use Goto() to jump to a label in a different extension or 
context?

When you have a lot of loops and such in a single extension, you end up wanting 
to use multiple labels called 'start', 'next' etc. I assume(hope!) that the 
namespace of labels is in a single context? ie can you use the same label name 
in another context? You then have to break a single extension up into multiple 
ones JUST to be able to use labels effectively.

Doug.




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--
Aaron Daniel
Computer Systems Technician
Sam Houston State University
[EMAIL PROTECTED]
(936) 294-4198
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Re: [Asterisk-Users] Multiple processes

2006-05-31 Thread Anthony Cennami

Were these upgrades or fresh installs? Earlier versions of asterisk ran with multiple threads. If you upgraded asterisk versions, but did not upgrade the associated startup scripts, multiple processes would still be spawned even if not appropriate.
AnthonyOn 5/31/06, Lee Archer [EMAIL PROTECTED] wrote:










Can someone shed any light on the following. I have 2 identical systems, 1 of which seems to spawn multiple processes which have to be killed manually. It recently kicked up 2 so I ran gdb on them and this is the thread output. I current use FreePBX with these systems.


1st extra process


(gdb) info thread

 1 Thread 1095261104 (LWP 14213) 0xe410 in __kernel_vsyscall ()

(gdb) thread apply all bt


Thread 1 (Thread 1095261104 (LWP 14213)):

#0 0xe410 in __kernel_vsyscall ()

#1 0x4004f13e in __lll_mutex_lock_wait () from /lib/tls/libpthread.so.0

#2 0x4004be41 in _L_mutex_lock_191 () from /lib/tls/libpthread.so.0

#3 0x0001 in ?? ()

#4 0x40e16778 in ?? () from /usr/lib/asterisk/modules/cdr_odbc.so

#5 0x40e16818 in __dso_handle () from /usr/lib/asterisk/modules/cdr_odbc.so

#6 0x0002 in ?? ()

#7 0x in ?? ()

#8 0x080a20b7 in ast_cdr_unregister (name=0x40e1455c ODBC) at lock.h:592

#9 0x40e13299 in odbc_unload_module () at cdr_odbc.c:240

#10 0x40e13978 in reload () at cdr_odbc.c:465

#11 0x0805be32 in ast_module_reload (name=0x0) at loader.c:257

#12 0x080b4623 in hup_handler (num=-4) at asterisk.c:754

#13 signal handler called

#14 0xe410 in __kernel_vsyscall ()

#15 0x401afd99 in sched_setscheduler () from /lib/tls/libc.so.6

#16 0x080b4743 in ast_set_priority (pri=0) at asterisk.c:803

#17 0x40445ee8 in agi_exec_full (chan=0x82782b0, data="" optimized out, enhanced=0, dead=0) at res_agi.c:300

#18 0x0808d521 in pbx_extension_helper (c=0x82782b0, con=value optimized out, context=0x8278400 macro-record-enable,

 exten=0x82784f4 s, priority=4, label=0x0, callerid=0x8159f38 0163861, action="" at pbx.c:553

#19 0x40c44851 in macro_exec (chan=0x82782b0, data="" at app_macro.c:210

#20 0x0808d521 in pbx_extension_helper (c=0x82782b0, con=value optimized out, context=0x8278400 macro-record-enable,

 exten=0x82784f4 s, priority=7, label=0x0, callerid=0x8159f38 0163861, action="" at pbx.c:553

#21 0x40c44851 in macro_exec (chan=0x82782b0, data="" at app_macro.c:210

#22 0x0808d521 in pbx_extension_helper (c=0x82782b0, con=value optimized out, context=0x8278400 macro-record-enable,

 exten=0x82784f4 s, priority=1, label=0x0, callerid=0x8159f38 0163861, action="" at pbx.c:553

#23 0x0808ead4 in __ast_pbx_run (c=0x82782b0) at pbx.c:2227

#24 0x0808f6cc in pbx_thread (data="" at pbx.c:2514

#25 0x4004a297 in start_thread () from /lib/tls/libpthread.so.0

#26 0x401c737e in clone () from /lib/tls/libc.so.6

#27 0x41485bb0 in ?? ()

#0 0xe410 in __kernel_vsyscall ()


2nd extra process


(gdb) info thread

 1 Thread 1096059824 (LWP 14214) 0xe410 in ?? ()

(gdb) thread apply all bt


Thread 1 (Thread 1096059824 (LWP 14214)):

#0 0xe410 in ?? ()

#1 0x41533594 in ?? ()

#2 0x0002 in ?? ()

#3 0x in ?? ()

#4 0x4004f13e in __lll_mutex_lock_wait () from /lib/tls/libpthread.so.0

#5 0x4004be41 in _L_mutex_lock_191 () from /lib/tls/libpthread.so.0

#6 0x0001 in ?? ()

#7 0x40e16778 in ?? () from /usr/lib/asterisk/modules/cdr_odbc.so

#8 0x40e16818 in __dso_handle () from /usr/lib/asterisk/modules/cdr_odbc.so

#9 0x0002 in ?? ()

#10 0x in ?? ()

#11 0x080a20b7 in ast_cdr_unregister (name=0x40e1455c ODBC) at lock.h:592

#12 0x40e13299 in odbc_unload_module () at cdr_odbc.c:240

#13 0x40e13978 in reload () at cdr_odbc.c:465

#14 0x0805be32 in ast_module_reload (name=0x0) at loader.c:257

#15 0x080b4623 in hup_handler (num=-4) at asterisk.c:754

#16 signal handler called

#17 0xe410 in ?? ()

#0 0xe410 in ?? ()


Regards


Lee


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Re: [Asterisk-Users] Polycom replacement handset

2006-05-31 Thread olivier.taylor





heya Cory,

did u receive and test the siemens handsets?

Olivier

Cory Andrews a crit:

  
  

  
  
  
  
  
  Ryan  Shoot
me an email off list, I
can help you out with a replacement handset.
  
  Thanks
  
  
  Cory Andrews
  Executive
Vice President
  ++
  VoIPSupply.com
  PBXSelect.com
  ++
  454 Sonwil
Drive
  Buffalo, NY 14225
  voice -
800.398.VoIP X3402
  fax -
716.630.1548
  e - [EMAIL PROTECTED]
  m -
716.907.4059
  aim - B2Cory
  
  
  
  
  From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]] On Behalf Of Ryan Stark
  Sent: Tuesday, May 30,
2006 8:29
PM
  To:
asterisk-users@lists.digium.com
  Subject:
[Asterisk-Users] Polycom
replacement handset
  
  
  Does anyone know where I can get replacement
handsets for the Polycom
SoundPoint IP phones? Or does anyone have any they want to sell?
From the looks of it you have to buy a whole new phone to get a new
handset. My vendor, TriaTechCOA, told me I had to buy a whole new
phone
to get a handset, which is pretty ridiculous. Maybe there is a more
sane
vendor I should be buying from? 
  
Thanks,
-Ryan
  
  

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RE: [Asterisk-Users] SIP Presence




It's 
doable if you are only going to be usinga single, non redundant, Asterisk 
box. If you intend to use more Asterisk boxes in a 'cluster', your about to 
enter a whole world of hurt if you try and get SIP presence to work with 
it.

Doug

  -Original Message-From: Forrest Beck 
  [mailto:[EMAIL PROTECTED]Sent: Wednesday, May 31, 2006 
  9:15 AMTo: 'Asterisk Users Mailing List - Non-Commercial 
  Discussion'Subject: [Asterisk-Users] SIP 
  Presence
  
  Does anyone have a working 
  implementation of SIP Presence? I have a new Grandstream GX-2000 phone 
  with the supported hardware and I am not sure how to setup presence with 
  asterisk. 
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Re: [Asterisk-Users] Handset recommendations

2006-05-31 Thread Philippe Lindheimer

Andrew,it depends on the routers/firewalls involved. There are plenty of people who have problems if they don't set this up and it is something that Polycom has acknowledged (not providing any type of keepalive or similar) and may fix in future firmware. However - there is also very good documentation available and furthermore, many others fall in your situation where it does just work.pFrom: Andrew Kohlsmith [EMAIL PROTECTED]To: asterisk-users@lists.digium.comDate: Wed, 31 May 2006 10:34:00 -0400Subject: Re: [Asterisk-Users] Handset recommendationsOn Wednesday 31 May 2006 10:12, Philippe Lindheimer wrote: I am also a VERY happy with the Polycom 501. I will disagree with the comment "They Just Work." when it comes
 to NAT. Once they are setup properly (e.g. set registration timeout to 60 sec so it registers every 30 seconds and keeps NAT holes open) then they work fine. There is good info on the wiki on how to set them up.I made *no* registration changes to the default values. Perhaps your router had its NAT timeout window set really short? I am using a totally-factory-standard WRT54G.-A.
		Ring'em or ping'em. Make  PC-to-phone calls as low as 1¢/min with Yahoo! Messenger with Voice.___
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Re: [Asterisk-Users] SIP Presence

2006-05-31 Thread William Piper

Check out http://www.voip-info.org/wiki-Asterisk+standard+extensions
and look for hint, this will giveyou the presence that you are looking for.

bp
On 5/31/06, Forrest Beck [EMAIL PROTECTED] wrote:




Does anyone have a working implementation of SIP Presence? I have a new Grandstream GX-2000 phone with the supported hardware and I am not sure how to setup presence with asterisk. 
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RE: [Asterisk-Users] AEL #include

Oh Crud. So, if I want to jump to another extension or context, I have to 
specify the full context, extension and priority? I can't specify a label? It's 
a bit tricky trying to jump to a specific priority in an extension when they're 
all called 'n' !

Why is something so simple such a mess...


 -Original Message-
 From: Michael Collins [mailto:[EMAIL PROTECTED]
 Sent: Wednesday, May 31, 2006 9:22 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: RE: [Asterisk-Users] AEL #include
 
 
  Thanks Michael. I was not aware that labels where available.
  In converting though, I've already hit a limitation. 
 There's a single
 name
  space for all labels I assume?
 
 Doug,
 
 According to TFOT's Goto() application reference entry (page 254) the
 namespace is actually the current extension:
 Named priorities only work within the current extension.
 
 So you can have 50 different labels called start as long as you use
 start only once per extension.  If you're in extension 555 and you
 Goto(start) it will look for exten = 555,n(start),...  If 
 it doesn't
 find a label start in exten 555 then the Goto() will fail, 
 even if you
 have start in another extension:
 exten = 556,n(start),Noop(this start good only from exten 556)
 
 HtH.
 
 I can see a potential issue if you need to jump from one exten to
 another exten using Goto().  You still need to use
 Goto(context,extension,priority) to jump around like that.  
 Do you have
 any scenarios like that?  If so, it might be possible to 
 create numbered
 jump-to points that will never change, and therefore won't require
 renumbering each time you make an addition to the dialplan.
 
 Example:
 
 [test_context]
 exten = 555,1,Noop(Starting exten 555)
 exten = 555,n,dialplan stuff
 exten = 555,n,Goto(test_context,556,999) 
 ; previous line will end up at 556,n(start)
 
 exten = 556,1,Noop(Starting exten 556)
 exten = 556,n,dialplan stuff
 exten = 556,n(start),Noop(This is where I want to be)
 exten = 556,n,more dialplan stuff
 exten = 556,999,Goto(start)
 ; previous line used to allow other exten's to jump to 556,n(start)
 
 
 
 FYI, your other post just came in.  I think I just answered a few of
 your questions.  Let us know if this helps!
 
 -MC
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RE: [Asterisk-Users] Multiple processes

2006-05-31 Thread Lee Archer




It's only been a problem since I updated to Asterisk 1.2 a 
few months ago. It was a fresh install of OS, Asterisk, FreePBX and other 
scripts. I've recently just updating FreePBX but the problem hasn't 
gone.

Regards

Lee


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Anthony 
CennamiSent: 31 May 2006 16:34To: Asterisk Users Mailing 
List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] 
Multiple processes
Were these upgrades or fresh installs? Earlier versions of 
asterisk ran with multiple threads. If you upgraded asterisk versions, but 
did not upgrade the associated startup scripts, multiple processes would still 
be spawned even if not appropriate. Anthony
On 5/31/06, Lee 
Archer [EMAIL PROTECTED] 
wrote:

  
  
  Can someone shed any light on the following. 
  I have 2 identical systems, 1 of which seems to spawn multiple processes which 
  have to be killed manually. It recently kicked up 2 so I ran gdb on them 
  and this is the thread output. I current use FreePBX with these systems. 
  
  1st extra process 
  (gdb) info thread  1 Thread 1095261104 (LWP 14213) 0xe410 in 
  __kernel_vsyscall () (gdb) thread apply all 
  bt 
  Thread 1 (Thread 1095261104 (LWP 14213)): 
  #0 0xe410 in __kernel_vsyscall () 
  #1 0x4004f13e in __lll_mutex_lock_wait () 
  from /lib/tls/libpthread.so.0 #2 
  0x4004be41 in _L_mutex_lock_191 () from /lib/tls/libpthread.so.0 
  #3 0x0001 in ?? () #4 0x40e16778 in ?? () from 
  /usr/lib/asterisk/modules/cdr_odbc.so #5 0x40e16818 in __dso_handle () from 
  /usr/lib/asterisk/modules/cdr_odbc.so #6 0x0002 in ?? () #7 0x in ?? () #8 0x080a20b7 in ast_cdr_unregister (name=0x40e1455c "ODBC") at 
  lock.h:592 #9 0x40e13299 in 
  odbc_unload_module () at cdr_odbc.c:240 #10 
  0x40e13978 in reload () at cdr_odbc.c:465 #11 0x0805be32 in ast_module_reload (name=0x0) at loader.c:257 
  #12 0x080b4623 in hup_handler (num=-4) at 
  asterisk.c:754 #13 signal handler 
  called #14 0xe410 in 
  __kernel_vsyscall () #15 0x401afd99 in 
  sched_setscheduler () from /lib/tls/libc.so.6 #16 0x080b4743 in ast_set_priority (pri=0) at asterisk.c:803 
  #17 0x40445ee8 in agi_exec_full (chan=0x82782b0, 
  data="" optimized out, enhanced=0, dead=0) at res_agi.c:300 
  #18 0x0808d521 in pbx_extension_helper 
  (c=0x82782b0, con=value optimized out, context=0x8278400 
  "macro-record-enable",
   exten=0x82784f4 "s", priority=4, 
  label=0x0, callerid=0x8159f38 "0163861", action="" at pbx.c:553 
  #19 0x40c44851 in macro_exec (chan=0x82782b0, 
  data="" at app_macro.c:210 #20 
  0x0808d521 in pbx_extension_helper (c=0x82782b0, con=value optimized 
  out, context=0x8278400 "macro-record-enable",
   exten=0x82784f4 "s", priority=7, 
  label=0x0, callerid=0x8159f38 "0163861", action="" at pbx.c:553 
  #21 0x40c44851 in macro_exec (chan=0x82782b0, 
  data="" at app_macro.c:210 #22 
  0x0808d521 in pbx_extension_helper (c=0x82782b0, con=value optimized 
  out, context=0x8278400 "macro-record-enable",
   exten=0x82784f4 "s", priority=1, 
  label=0x0, callerid=0x8159f38 "0163861", action="" at pbx.c:553 
  #23 0x0808ead4 in __ast_pbx_run (c=0x82782b0) at 
  pbx.c:2227 #24 0x0808f6cc in pbx_thread 
  (data="" at pbx.c:2514 #25 0x4004a297 in 
  start_thread () from /lib/tls/libpthread.so.0 #26 0x401c737e in clone () from /lib/tls/libc.so.6 #27 0x41485bb0 in ?? () #0 0xe410 in __kernel_vsyscall () 
  2nd extra process 
  (gdb) info thread  1 Thread 1096059824 (LWP 14214) 0xe410 in ?? () 
  (gdb) thread apply all bt 
  Thread 1 (Thread 1096059824 (LWP 14214)): 
  #0 0xe410 in ?? () #1 0x41533594 in ?? () #2 0x0002 in ?? () #3 0x in ?? () #4 0x4004f13e in __lll_mutex_lock_wait () from 
  /lib/tls/libpthread.so.0 #5 
  0x4004be41 in _L_mutex_lock_191 () from /lib/tls/libpthread.so.0 
  #6 0x0001 in ?? () #7 0x40e16778 in ?? () from 
  /usr/lib/asterisk/modules/cdr_odbc.so #8 0x40e16818 in __dso_handle () from 
  /usr/lib/asterisk/modules/cdr_odbc.so #9 0x0002 in ?? () #10 
  0x in ?? () #11 0x080a20b7 in 
  ast_cdr_unregister (name=0x40e1455c "ODBC") at lock.h:592 #12 0x40e13299 in odbc_unload_module () at 
  cdr_odbc.c:240 #13 0x40e13978 in reload () 
  at cdr_odbc.c:465 #14 0x0805be32 in 
  ast_module_reload (name=0x0) at loader.c:257 #15 0x080b4623 in hup_handler (num=-4) at asterisk.c:754 
  #16 signal handler called #17 0xe410 in ?? () #0 0xe410 in ?? () 
  Regards 
  Lee 
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Re: [Asterisk-Users] I guess my server capacity is ok

2006-05-31 Thread Henry J. Cobb

 All I know is that it is very processor intensive and either not using
 it or just passing it through is your best bet.  I will be working alot
 with G729 in the near future and will post my findings but until then I
 am just relying on the dimensioning page on the wiki.

 Thanks,
 Steve Totaro

Which DSP based boards does Asterisk support for G729 and are any of these
more cost effective than piling on Pentiums?

BTW: Can AMD CPUs handle a higher G729 load in 64 bit mode?

-- 
Henry J. Cobb
http://www.io.com/~hcobb/

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[Asterisk-Users] Explicit Dialplan Exit




So, 
I've kind of converted my dialplan from:

exten 
= 
custcare,1,GotoIfTime(8:00-17:00|mon-fri|*|*?acd_one_queue,custcare-open,1)exten 
= custcare,2,Goto(custcare-closed,1)

exten 
= custcare-open,1

exten 
= custcare-open,99


exten 
= custcare-closed,1

exten 
= custcare-closed,99

to:


exten 
= custcare,1,GotoIfTime(8:00-17:00|mon-fri|*|*?custcare_open)exten = 
custcare,n,Goto(custcare-closed,1)

exten 
= custcare,n(open_start),...
exten = custcare,n...



exten 
= custcare,n(closed_start)...

exten 
= custcare,n...

I 
don't like having those final statements in each block. Previously, execution 
would implicitly end because there was no priorities left in each extension. Now 
however, everything is in one extension and I can't be sure that execution will 
not continue at the end of a section (open,closed etc). Is there some sort of 
explicit dialplan command that stops execution and immediately ends the 
dialplan? Something like MacroExit() in a macro Can't see it in the 
docs.

Doug.


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[Asterisk-Users] Asterisk crashes at startup

2006-05-31 Thread Jean-Michel Hiver


Hi List,

Yesterday night after a power off due to a faulty UPS my asterisk 
doesn't want to start anymore. Here is what I get on the CLI:


Asterisk Ready.
*CLI
Disconnected from Asterisk server: Bad file descriptor.
Executing last minute cleanups
 == Destroying musiconhold processes
Asterisk uncleanly ending (0).

I use 1.2.7 I think on a debian sarge and cdr_pgsql too.

Any ideas?

Cheers,
Jean-Michel.

--
Jean-Michel Hiver - http://ykoz.net/
Découvrez la Réunion des Technologies IP  Telecom
TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE

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RE: [Asterisk-Users] SIP Presence

2006-05-31 Thread Aaron Daniel

For once I agree with Doug.  Better make sure any phones in the presence 
group are on the same server.


On Wed, 31 May 2006, Douglas Garstang wrote:


It's doable if you are only going to be using a single, non redundant, Asterisk 
box. If you intend to use more Asterisk boxes in a 'cluster', your about to 
enter a whole world of hurt if you try and get SIP presence to work with it.

Doug

-Original Message-
From: Forrest Beck [mailto:[EMAIL PROTECTED]
Sent: Wednesday, May 31, 2006 9:15 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] SIP Presence



Does anyone have a working implementation of SIP Presence?  I have a new 
Grandstream GX-2000 phone with the supported hardware and I am not sure how to 
setup presence with asterisk.




--
Aaron Daniel
Computer Systems Technician
Sam Houston State University
[EMAIL PROTECTED]
(936) 294-4198
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[Asterisk-Users] Hold Status

2006-05-31 Thread Sean Cook

Is there a way through AMI or AGI to determine whether a channel is on
hold?  Or if a channel has a call on hold?

Sean
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Re: [Asterisk-Users] Compilation issues with s390

2006-05-31 Thread Frank Pani


Kevin P. Fleming wrote:


This is not an appropriate place for this discussion; the best thing to
do at this point is to either open a bug in the tracker at
bugs.digium.com or find a bug marshal on the #asterisk channel on IRC
(in fact, getting a bug marshal remote access to your system is the
fastest way to get this fixed, as they can run the build process
repeatedly until these problems are solved).

I was not under the impression that this wasn't an appropriate place for 
this discussion, as we have already exchanged several emails on this 
without any prior indication.   I will move the topic as suggested to a 
bug marshal or tracker at bugs.digium.com.

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Re: [Asterisk-Users] AEL #include

Douglas Garstang wrote:
 Oh Crud. So, if I want to jump to another extension or context, I have to 
 specify the full context, extension and priority? I can't specify a label? 
 It's a bit tricky trying to jump to a specific priority in an extension when 
 they're all called 'n' !

No. Labels are interpreted in their target context.

 Why is something so simple such a mess...

It's not. If instead of posting all these message you spent two minutes
actually trying it, you would have seen that it already works exactly
the way you want it to. assuming there is only a single namespace for
labels does not mean it is that way, and when it is so easy to determine
that your assumption is incorrect it seems rather pointless as well.
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[Asterisk-Users] RE: Explicit Dialplan Exit




Eh, 
I'm thinking I don't like labels very much. They aren't all they are cracked up 
to be.

Previously, using extensions of the format extension-function, like 
2944000-open or 2944000-closed for example, I could break up an extension into 
logical units based on function, and it made sense. By exclusively using labels, 
everthing is in the one extension and it isn't as easy to read at a glance. 
There's also the chance that statements from one section could over-run into 
another.

or... 
am I missing something?

Doug

  -Original Message-From: Douglas Garstang 
  Sent: Wednesday, May 31, 2006 10:06 AMTo: 'Asterisk 
  Users Mailing List - Non-Commercial Discussion'Subject: Explicit 
  Dialplan Exit
  So, 
  I've kind of converted my dialplan from:
  
  exten = 
  custcare,1,GotoIfTime(8:00-17:00|mon-fri|*|*?acd_one_queue,custcare-open,1)exten 
  = custcare,2,Goto(custcare-closed,1)
  
  exten = custcare-open,1
  
  exten = custcare-open,99
  
  
  exten = custcare-closed,1
  
  exten = custcare-closed,99
  
  to:
  
  
  exten = 
  custcare,1,GotoIfTime(8:00-17:00|mon-fri|*|*?custcare_open)exten = 
  custcare,n,Goto(custcare-closed,1)
  
  exten = custcare,n(open_start),...
  exten = 
  custcare,n...
  
  
  
  exten = custcare,n(closed_start)...
  
  exten = custcare,n...
  
  I 
  don't like having those final statements in each block. Previously, execution 
  would implicitly end because there was no priorities left in each extension. 
  Now however, everything is in one extension and I can't be sure that execution 
  will not continue at the end of a section (open,closed etc). Is there some 
  sort of explicit dialplan command that stops execution and immediately ends 
  the dialplan? Something like MacroExit() in a macro Can't see it in the 
  docs.
  
  Doug.
  
  
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Re: [Asterisk-Users] Centos cause Asterisk crash

2006-05-31 Thread Sean Kennedy

chan,
Run each script seperately to determine which one causes the crash. 
From there, check your logs to see any error messages.  There should be
something. 

My hunch is that prelink will cause the crash.

chan (Alpha Trilogies Networks) wrote:
 Hi,
 Can some one who experience that does those file necessary for the CentOS
 and Asterisk installation
 /etc/cron.daily/00-makewhatis.cron
 /etc/cron.daily/slocate.cron
 /etc/cron.daily/prelink
 /etc/cron.daily/rpm
 /etc/cron.weekly/00-makewhatis.cron

 I experience that those file cause my Asterisk Server crash.
 Can I just disable them and run the Asterisk stable? 


 Any reply will be appreciated.

 Thank you in advance.
begin:vcard
fn:Sean Kennedy
n:Kennedy;Sean
org:Rickey  Wong DDS Inc
adr;dom:A115;;2937 Veneman Ave;Modesto;CA;95356
email;internet:[EMAIL PROTECTED]
title:Chief Information Officer
tel;work:209-577-0777 x44
tel;fax:209-529-3209
tel;cell:209-485-2834
x-mozilla-html:TRUE
url:http://www.qualitydentists.com
version:2.1
end:vcard

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Re: [Asterisk-Users] AEL #include

2006-05-31 Thread Mojo with Horan Company, LLC

That's why you *name* certain n priorities, so you can Goto them later 
easily


Douglas Garstang wrote:

Oh Crud. So, if I want to jump to another extension or context, I have to 
specify the full context, extension and priority? I can't specify a label? It's 
a bit tricky trying to jump to a specific priority in an extension when they're 
all called 'n' !

Why is something so simple such a mess...



-Original Message-
From: Michael Collins [mailto:[EMAIL PROTECTED]
Sent: Wednesday, May 31, 2006 9:22 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] AEL #include



Thanks Michael. I was not aware that labels where available.
In converting though, I've already hit a limitation. 

There's a single
name

space for all labels I assume?

Doug,

According to TFOT's Goto() application reference entry (page 254) the
namespace is actually the current extension:
Named priorities only work within the current extension.

So you can have 50 different labels called start as long as you use
start only once per extension.  If you're in extension 555 and you
Goto(start) it will look for exten = 555,n(start),...  If 
it doesn't
find a label start in exten 555 then the Goto() will fail, 
even if you

have start in another extension:
exten = 556,n(start),Noop(this start good only from exten 556)

HtH.

I can see a potential issue if you need to jump from one exten to
another exten using Goto().  You still need to use
Goto(context,extension,priority) to jump around like that.  
Do you have
any scenarios like that?  If so, it might be possible to 
create numbered

jump-to points that will never change, and therefore won't require
renumbering each time you make an addition to the dialplan.

Example:

[test_context]
exten = 555,1,Noop(Starting exten 555)
exten = 555,n,dialplan stuff
exten = 555,n,Goto(test_context,556,999) 
; previous line will end up at 556,n(start)


exten = 556,1,Noop(Starting exten 556)
exten = 556,n,dialplan stuff
exten = 556,n(start),Noop(This is where I want to be)
exten = 556,n,more dialplan stuff
exten = 556,999,Goto(start)
; previous line used to allow other exten's to jump to 556,n(start)



FYI, your other post just came in.  I think I just answered a few of
your questions.  Let us know if this helps!

-MC
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--
Mojo [EMAIL PROTECTED]
Office Manger, Horan  Company, LLC
(907) 747- x112
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Re: [Asterisk-Users] AEL #include

2006-05-31 Thread Mojo with Horan Company, LLC


I apologize for my silly prior response, I didn't read the thread enough :(

Douglas Garstang wrote:

Oh Crud. So, if I want to jump to another extension or context, I have to 
specify the full context, extension and priority? I can't specify a label? It's 
a bit tricky trying to jump to a specific priority in an extension when they're 
all called 'n' !

Why is something so simple such a mess...



-Original Message-
From: Michael Collins [mailto:[EMAIL PROTECTED]
Sent: Wednesday, May 31, 2006 9:22 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] AEL #include



Thanks Michael. I was not aware that labels where available.
In converting though, I've already hit a limitation. 

There's a single
name

space for all labels I assume?

Doug,

According to TFOT's Goto() application reference entry (page 254) the
namespace is actually the current extension:
Named priorities only work within the current extension.

So you can have 50 different labels called start as long as you use
start only once per extension.  If you're in extension 555 and you
Goto(start) it will look for exten = 555,n(start),...  If 
it doesn't
find a label start in exten 555 then the Goto() will fail, 
even if you

have start in another extension:
exten = 556,n(start),Noop(this start good only from exten 556)

HtH.

I can see a potential issue if you need to jump from one exten to
another exten using Goto().  You still need to use
Goto(context,extension,priority) to jump around like that.  
Do you have
any scenarios like that?  If so, it might be possible to 
create numbered

jump-to points that will never change, and therefore won't require
renumbering each time you make an addition to the dialplan.

Example:

[test_context]
exten = 555,1,Noop(Starting exten 555)
exten = 555,n,dialplan stuff
exten = 555,n,Goto(test_context,556,999) 
; previous line will end up at 556,n(start)


exten = 556,1,Noop(Starting exten 556)
exten = 556,n,dialplan stuff
exten = 556,n(start),Noop(This is where I want to be)
exten = 556,n,more dialplan stuff
exten = 556,999,Goto(start)
; previous line used to allow other exten's to jump to 556,n(start)



FYI, your other post just came in.  I think I just answered a few of
your questions.  Let us know if this helps!

-MC
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--
Mojo [EMAIL PROTECTED]
Office Manger, Horan  Company, LLC
(907) 747- x112
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Re: [Asterisk-Users] I guess my server capacity is ok

Henry J. Cobb wrote:

 Which DSP based boards does Asterisk support for G729 and are any of these
 more cost effective than piling on Pentiums?

There are none at this time.

 BTW: Can AMD CPUs handle a higher G729 load in 64 bit mode?

Yes. The G.729 codec we distribute is marginally (6-7%) faster on AMD64
in 64-bit mode than in 32-bit mode.
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Re: [Asterisk-Users] Explicit Dialplan Exit

Douglas Garstang wrote:

 exten = custcare,1,GotoIfTime(8:00-17:00|mon-fri|*|*?custcare_open)
 exten = custcare,n,Goto(custcare-closed,1)
  
 exten = custcare,n(open_start),...

Use 'n+1' or 'n+10' or something here, to force a break in the sequence.
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Re: [Asterisk-Users] Explicit Dialplan Exit

2006-05-31 Thread Aaron Daniel


A) I think hangup will do just nicely for that.
B) Your first n, Goto(custcare-closed,1) is going to cause you problems, 
unless you have a custcare-closed somewhere else.


On Wed, 31 May 2006, Douglas Garstang wrote:


So, I've kind of converted my dialplan from:

exten = 
custcare,1,GotoIfTime(8:00-17:00|mon-fri|*|*?acd_one_queue,custcare-open,1)
exten = custcare,2,Goto(custcare-closed,1)

exten = custcare-open,1
exten = custcare-open,99

exten = custcare-closed,1
exten = custcare-closed,99

to:

exten = custcare,1,GotoIfTime(8:00-17:00|mon-fri|*|*?custcare_open)
exten = custcare,n,Goto(custcare-closed,1)

exten = custcare,n(open_start),...
exten = custcare,n...

exten = custcare,n(closed_start)...
exten = custcare,n...

I don't like having those final statements in each block. Previously, execution 
would implicitly end because there was no priorities left in each extension. 
Now however, everything is in one extension and I can't be sure that execution 
will not continue at the end of a section (open,closed etc). Is there some sort 
of explicit dialplan command that stops execution and immediately ends the 
dialplan? Something like MacroExit() in a macro Can't see it in the docs.

Doug.





--
Aaron Daniel
Computer Systems Technician
Sam Houston State University
[EMAIL PROTECTED]
(936) 294-4198
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RE: [Asterisk-Users] AEL #include

 
 Oh Crud. So, if I want to jump to another extension or context, I have
to
 specify the full context, extension and priority? I can't specify a
label?
 It's a bit tricky trying to jump to a specific priority in an
extension
 when they're all called 'n' !
 
 Why is something so simple such a mess...

Doug,

I believe that it has to be one or the other - either labels are unique
across the entire dialplan or they are not.  However, you may have
uncovered a great feature request: allowing the Goto() commands to jump
outside the extension and priority while still using a label.

I'll post this on the wish list and see what happens.

-MC

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Re: [Asterisk-Users] Compiling Asterisk-addons

2006-05-31 Thread Matthew Fredrickson



On May 31, 2006, at 12:55 AM, Armin Schindler wrote:


On Tue, 30 May 2006, Kevin P. Fleming wrote:

Douglas Garstang wrote:

svn checkout http://svn.digium.com/svn/asterisk-addons/trunk 
asterisk-addons

svn checkout http://svn.digium.com/svn/asterisk/trunk asterisk
svn checkout http://svn.digium.com/svn/zaptel/trunk zaptel
svn checkout http://svn.digium.com/svn/libpri/trunk libpri


This has been covered at least 6 times on this list in the last couple
of months; asking the same questions others have already asked and
answered does not make people want to help you very much.

The answer is: asterisk-addons will not be brought up to date with SVN
trunk until SVN trunk enters the beta phase, which will occur in the
next week.


Actually, the error is not in the addons. The Asterisk-trunk 
installation

produces incomplete/misconfigured headers, which prevents building of
external modules.
Or is it digiums intention to make life more difficult for external 
modules?


You can't be serious.  I can't believe you just said that.  We're 
working really hard to get trunk ready for 1.4, there are other things 
that are more pressing to work on right now.  This is an open source 
project, so if you wish to provide any patches to fix -addons, that is 
certainly welcome.  Otherwise, sit back and enjoy the ride. :-)



Matthew Fredrickson

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[Asterisk-Users] Re: Still can't get asterisk to play voicemail files occasionally

2006-05-31 Thread Dan Elder

Thanks Bill, will check this out, but from the bug report the problem linked
to below seems to happen when people are checking their vm as it's being
recorded. In the situation I'm looking at, the file will 'skip' to the end
even after it's been sitting in the inbox for hours. I'm going to do more
testing, but I'm guessing I'm not the best one to be trying to diagnose the
issue.. Searched the bug tracker as well  didn't see any existing issues
that seem similar.. Am I the only one who's ever had this problem?

Thanks again!

 A few times a week I will get a call from a user who has a new voicemail,
 but they cannot play it. They go through the menus, hit 1 to play the
 message, and immediately the 'post message' menu prompts them to delete
the
 message. The actual voicemail file never gets played.
 I've downloaded these voicemail wav
 files from the server to my local desktop, and they play fine... This is *
 1.2.0 -- anyone have any clues what might be causing this or has anyone
had
 any sort of similar occurance? It's driving both me  my users nuts!!!

...maybe this is the issue: 

http://bugs.digium.com/view.php?id=6714 

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RE: [Asterisk-Users] SIP Presence

Yay! :)

 -Original Message-
 From: Aaron Daniel [mailto:[EMAIL PROTECTED]
 Sent: Wednesday, May 31, 2006 10:09 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: RE: [Asterisk-Users] SIP Presence

 For once I agree with Doug.  Better make sure any phones in 
 the presence 
 group are on the same server.

 On Wed, 31 May 2006, Douglas Garstang wrote:

  It's doable if you are only going to be using a single, non 
 redundant, Asterisk box. If you intend to use more Asterisk 
 boxes in a 'cluster', your about to enter a whole world of 
 hurt if you try and get SIP presence to work with it.

  Doug

  -Original Message-
  From: Forrest Beck [mailto:[EMAIL PROTECTED]
  Sent: Wednesday, May 31, 2006 9:15 AM
  To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
  Subject: [Asterisk-Users] SIP Presence

  Does anyone have a working implementation of SIP Presence?  
 I have a new Grandstream GX-2000 phone with the supported 
 hardware and I am not sure how to setup presence with asterisk.

 -- 
 Aaron Daniel
 Computer Systems Technician
 Sam Houston State University
 [EMAIL PROTECTED]
 (936) 294-4198
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Re: [Asterisk-Users] Compilation issues with s390

Frank Pani wrote:

 I was not under the impression that this wasn't an appropriate place for
 this discussion, as we have already exchanged several emails on this
 without any prior indication.   I will move the topic as suggested to a
 bug marshal or tracker at bugs.digium.com.

Yeah, sorry about that... when I saw the first message I thought 'hey,
this is a simple fix, I'll just do it'. Turns out I was wrong :-) Don't
be offended, it just got more involved and will require some more direct
interaction to solve it.
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Re: [Asterisk-Users] Asterisk crashes at startup

2006-05-31 Thread Robert Rawlinson


You may have file damage. Run the file repair.
Bob Rawlinson

Jean-Michel Hiver wrote:

Hi List,

Yesterday night after a power off due to a faulty UPS my asterisk 
doesn't want to start anymore. Here is what I get on the CLI:


Asterisk Ready.
*CLI
Disconnected from Asterisk server: Bad file descriptor.
Executing last minute cleanups
 == Destroying musiconhold processes
Asterisk uncleanly ending (0).

I use 1.2.7 I think on a debian sarge and cdr_pgsql too.

Any ideas?

Cheers,
Jean-Michel.



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RE: [Asterisk-Users] Asterisk crashes at startup

2006-05-31 Thread Brian C. Fertig

run asterisk with asterisk -c   and see if it gives anymore 
information.  You can also get it to produce a core dump and see if it gives 
you anymore information.

brian


-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jean-Michel Hiver
Sent: Wednesday, May 31, 2006 12:10 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Asterisk crashes at startup

Hi List,

Yesterday night after a power off due to a faulty UPS my asterisk 
doesn't want to start anymore. Here is what I get on the CLI:

Asterisk Ready.
*CLI
Disconnected from Asterisk server: Bad file descriptor.
Executing last minute cleanups
  == Destroying musiconhold processes
Asterisk uncleanly ending (0).

I use 1.2.7 I think on a debian sarge and cdr_pgsql too.

Any ideas?

Cheers,
Jean-Michel.

-- 
Jean-Michel Hiver - http://ykoz.net/
Découvrez la Réunion des Technologies IP  Telecom
TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE

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Re: [Asterisk-Users] RE: Explicit Dialplan Exit

Douglas Garstang wrote:

 Previously, using extensions of the format extension-function, like 
 2944000-open or 2944000-closed for example, I could break up an extension 
 into logical units based on function, and it made sense. By exclusively using 
 labels, everthing is in the one extension and it isn't as easy to read at a 
 glance. There's also the chance that statements from one section could 
 over-run into another.

Why does it have to be exclusive? Use whatever works for you...
appropriately named extensions with labels on their priorities is a good
combination and very easy to work with.
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Re: [Asterisk-Users] Centos cause Asterisk crash

2006-05-31 Thread olivier.taylor


use freebsd, not just a kernel as linux, just a real complete os :)

Sean Kennedy a écrit :

chan,
Run each script seperately to determine which one causes the crash. 
From there, check your logs to see any error messages.  There should be
something. 


My hunch is that prelink will cause the crash.

chan (Alpha Trilogies Networks) wrote:
  

Hi,
Can some one who experience that does those file necessary for the CentOS
and Asterisk installation
/etc/cron.daily/00-makewhatis.cron
/etc/cron.daily/slocate.cron
/etc/cron.daily/prelink
/etc/cron.daily/rpm
/etc/cron.weekly/00-makewhatis.cron

I experience that those file cause my Asterisk Server crash.
Can I just disable them and run the Asterisk stable? 



Any reply will be appreciated.

Thank you in advance.

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[Asterisk-Users] Queue falls through to personal voicemail

2006-05-31 Thread Paul Tinsley


This seems like a simple configuration but I must be missing something.

I have a queue setup with three members, if I do ring-all every phone 
gets one ring then one of the three continues to ring and the other two 
show missed calls.  After a few seconds the call goes to that persons 
voicemail box.


If I set it to round robin it will call the next available agent and put 
the call into their voicemail if they don't answer.


Am I missing something or doesn't it seem like it should try the next 
available agent instead of giving up and going to voicemail?


Thanks,
Paul

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RE: [Asterisk-Users] RE: Explicit Dialplan Exit









Do you have a before-and-after example? I
think wed like to see a sample of a context  extensions with
hard-coded priorities and the subsequent translation into unnumbered priorities
with labels. There are some creative people out there who might have the key
to getting your dialplan simplified without losing its power.



-MC













From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Douglas Garstang
Sent: Wednesday, May 31, 2006 9:27
AM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: [Asterisk-Users] RE:
Explicit Dialplan Exit







Eh, I'm thinking I don't like labels very
much. They aren't all they are cracked up to be.











Previously, using extensions of the format
extension-function, like 2944000-open or 2944000-closed for example, I could
break up an extension into logical units based on function, and it made sense.
By exclusively using labels, everthing is in the one extension and it isn't as
easy to read at a glance. There's also the chance that statements from one
section could over-run into another.











or... am I missing something?











Doug





-Original Message-
From: Douglas Garstang 
Sent: Wednesday, May 31, 2006
10:06 AM
To: 'Asterisk Users Mailing List -
Non-Commercial Discussion'
Subject: Explicit Dialplan Exit



So, I've kind of converted my dialplan
from:











exten =
custcare,1,GotoIfTime(8:00-17:00|mon-fri|*|*?acd_one_queue,custcare-open,1)
exten = custcare,2,Goto(custcare-closed,1)











exten = custcare-open,1







exten = custcare-open,99













exten = custcare-closed,1







exten = custcare-closed,99











to:

















exten =
custcare,1,GotoIfTime(8:00-17:00|mon-fri|*|*?custcare_open)
exten = custcare,n,Goto(custcare-closed,1)











exten = custcare,n(open_start),...





exten = custcare,n...















exten = custcare,n(closed_start)...







exten = custcare,n...





















I don't like having those final statements
in each block. Previously, execution would implicitly end because there was no
priorities left in each extension. Now however, everything is in one extension
and I can't be sure that execution will not continue at the end of a section
(open,closed etc). Is there some sort of explicit dialplan command that stops
execution and immediately ends the dialplan? Something like MacroExit() in a
macro Can't see it in the docs.











Doug.
























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RE: [Asterisk-Users] AEL #include ( Now Labels Goto() )