date:20061016


Hi,
I'm using mysql to store my cdr data. I compiled asterisk-addon module 
without problems and I see nothing unusual in my cdr_mysql.conf but when 
I do a reload I get this messages (never seen before):


Oct 16 09:43:16 WARNING[8576]: db.c:67 dbinit: Unable to open Asterisk 
database

Oct 16 09:43:16 WARNING[8576]: db.c:423 ast_db_gettree: Database unavailable

But If I try to connect from shell it works without any problem.

Does anybody know why?

Giorgio Incantalupo
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Re: [asterisk-users] Unable to open Asterisk database

2006-10-16 Thread Benjamin Jacob


Giorgio Incantalupo wrote:


Hi,
I'm using mysql to store my cdr data. I compiled asterisk-addon module 
without problems and I see nothing unusual in my cdr_mysql.conf but 
when I do a reload I get this messages (never seen before):


Oct 16 09:43:16 WARNING[8576]: db.c:67 dbinit: Unable to open Asterisk 
database
Oct 16 09:43:16 WARNING[8576]: db.c:423 ast_db_gettree: Database 
unavailable


But If I try to connect from shell it works without any problem.

Does anybody know why?

Giorgio Incantalupo
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cross check ur modules.conf and cdr_mysql.conf
modules.conf shud have 'load = cdr_addon_mysql.so'

neway..  its something to do with your sql configuration,
wrong passwords, usernames, host and permissions on sql.

seen these errors too often.  n the reasons r human error.

cheerz
Ben.

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[asterisk-users] Quescom 400

2006-10-16 Thread Giordano Grandis

Hi all,
I just configured a quescom 400 to route all gsm incoming calls to
asterisk, now i would route all outgoing asterisk calls to gsm port of
the quescom. 

Anyone has any idea how implement it?

I did a configuration but i always get this error

-- Got SIP response 503 Service Unavailable back from
ip_add_quescom400

Thanks in advance.

Giordano
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[asterisk-users] FOP run control for CentOS/RHEL

2006-10-16 Thread Eric Bishop

Anyone have a sane rc script for FOP on CentOS/RHEL systems?
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[asterisk-users] Re: 1.4 beta2 on intel mac

2006-10-16 Thread Martin Joseph


On 2006-10-15 23:50:34 -0700, Tim Panton [EMAIL PROTECTED] said:



On 16 Oct 2006, at 07:15, Martin Joseph wrote:


On 2006-10-12 02:40:35 -0700, Tim Panton [EMAIL PROTECTED] said:


On 11 Oct 2006, at 19:35, Dean Collins wrote:

Lol - use a real PC maybe :P

Nah, that would be dull.
In some ways the mac intel is nearer to a 'normal PC'
(whatever that is) than the systems I normally run asterisk on
- a NatSemi Nemiah and an arm5 :-)
Asterisk 1.2.X runs fine on the intel macs, so
I guess there must be a bug in 1.4beta2 that stops it running.


Did you need to update the version of Make?  My PowerPC mac seems  to 
be complaining about version 3.80.


I don't have any Intel mac's to test with (yet).


Yes. I had to install a new make from source (with configure -- prefix=/usr)

I've got some stuff to get ready for Astricon Dallas next week,
where we will be launching Corraleta SDK - our zero install web-based  
Java softphone.


Once that's done and I get back I'll look into what the problem is  
(unless someone solves it for
me while I'm there  - drop by our stand in the exhibition If you have  
got 1.4beta2 working on

an intel mac - or if you want to see Corraleta in action! )

Tim Panton

www.mexuar.com


I just built 1.4b2 on a powerpc mac system, and although it seems to 
build ok, and starts up,  the command line is completely non-reponsive 
(although exit works).


I have head people describe this kind of dead CLI in the past, but 
never saw it before.


1.4b2 doesn't accept registrations or do anything.  If this sounds like 
what you saw, then I guess it's an OSX issue and not an Intel OSX issue 
specifically.


Hopefully beta 3 fixes it?

Sorry I can't come check out the demos, I would love to, but I am Mr. mom.

Marty

PS This was a working 1.2.12 system before 1.4b2 was installed.


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Re: [asterisk-users] Asterisk 1.2.12.1 and snom 360 6.2.3 no audio

2006-10-16 Thread Olivier

I know it won't help much but we use now a bristuffed Asterisk along with Snom 320 phones.It works now most of the time but we had to patch Asterisk keep calls from being cut (5% of calls were hit by that - symptom is voice cut in the middle of call).
Now we still have calls being hanged while ringing (0,5% occurence rate) but when a call is established, everything works fine.The setup is :0.3.0-PRE-1s brituffed 1.2.10 asterisk along with 6.2.3 Snom 320.
For further details, contact me offlist.Regards
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Re: [asterisk-users] Unable to open Asterisk database


Hi Benjamin,
I checked in every place and it seems all right. The strangest thing is 
that _Asterisk is writing CDR records inside the right table_that's 
why I do not understand this message. I would expect Asterisk not to 
fill in the DB.


Can I ignore the warning?

TIA

Giorgio Incantalupo




Benjamin Jacob wrote:

Giorgio Incantalupo wrote:


Hi,
I'm using mysql to store my cdr data. I compiled asterisk-addon 
module without problems and I see nothing unusual in my 
cdr_mysql.conf but when I do a reload I get this messages (never seen 
before):


Oct 16 09:43:16 WARNING[8576]: db.c:67 dbinit: Unable to open 
Asterisk database
Oct 16 09:43:16 WARNING[8576]: db.c:423 ast_db_gettree: Database 
unavailable


But If I try to connect from shell it works without any problem.

Does anybody know why?

Giorgio Incantalupo
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cross check ur modules.conf and cdr_mysql.conf
modules.conf shud have 'load = cdr_addon_mysql.so'

neway..  its something to do with your sql configuration,
wrong passwords, usernames, host and permissions on sql.

seen these errors too often.  n the reasons r human error.

cheerz
Ben.

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Re: [asterisk-users] Re: 1.4 beta2 on intel mac

2006-10-16 Thread Tzafrir Cohen

On Sun, Oct 15, 2006 at 11:15:55PM -0700, Martin Joseph wrote:
 On 2006-10-12 02:40:35 -0700, Tim Panton [EMAIL PROTECTED] said:
 
 
 On 11 Oct 2006, at 19:35, Dean Collins wrote:
 
 Lol - use a real PC maybe :P
 
 Nah, that would be dull.
 
 In some ways the mac intel is nearer to a 'normal PC'
 (whatever that is) than the systems I normally run asterisk on
 - a NatSemi Nemiah and an arm5 :-)
 
 Asterisk 1.2.X runs fine on the intel macs, so
 I guess there must be a bug in 1.4beta2 that stops it running.
 
 Did you need to update the version of Make?  

Or alternatively, use recent 1.4 branch from svn, that should not need
the latest nad greatest make.

-- 
Tzafrir Cohen sip:[EMAIL PROTECTED]
icq#16849755  iax:[EMAIL PROTECTED]
+972-50-7952406  jabber:[EMAIL PROTECTED]
[EMAIL PROTECTED] http://www.xorcom.com
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Re: [asterisk-users] Quescom 400

2006-10-16 Thread anban

 Hi all,
 I just configured a quescom 400 to route all gsm incoming calls to
 asterisk, now i would route all outgoing asterisk calls to gsm port of
 the quescom.

 Anyone has any idea how implement it?

 I did a configuration but i always get this error

On the quescom, under the objects section, you have to add the Asterisk
box as a foreign gatekeeper. Then you have to add it as a GSM service in
the services section.


 -- Got SIP response 503 Service Unavailable back from
 ip_add_quescom400

This usually happens when it isn't registered as a service.

Cheers



-- 
This message has been scanned for viruses and
dangerous content by MailScanner, and is
believed to be clean.

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SV: [asterisk-users] How do you like TrixBox?

2006-10-16 Thread Amund Nygaard









I love TrixBox, with the custom config
files you can tweak pretty much with TrixBox too, I have at least done some. Plan
to do a plain Asterisk install later, but for now I learn a lot about the
config files just with TrixBox. Some things might be a bit harder with TrixBox
due to some of the premade dial plans, but can get it to work J











Fra:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] På vegne av Chris Ramsey
Sendt: 16. oktober 2006 05:11
Til: Asterisk Users Mailing List -
Non-Commercial Discussion
Emne: Re: [asterisk-users] How do
you like TrixBox?





I agree with Mohamed.
TrixBox is an excellent way to start, but in the long run, if you attempt to
use Asterisk in a business setting, you will probably want to be able to
hardcode the conf files yourself. I have only recently changed over to TrixBox
from a standard installation on a debian system. Honestly, I really don't use
FreePBX much at all. I use it to create new extentions for VMB's since I need
to create a number of them every day for users on my website, but overall I
code my own dialplan. I don't really understand FreePBX well enough, nor do I
really want to put in the effort of learning it since I can already hand-code.
TrixBox just has a couple of nifty features that I enjoy to make daily life a
tad easier. 



On 10/15/06, Mohamed
A. Gombolaty [EMAIL PROTECTED]
wrote:

Dear All, 

I am have
experimented asterisk long before any gui was available and also currently
working with trixbox, ofcourse working with asterisk directly makes you more
aware but when you start deploying the system you will face management issues
for asterisk, as anyone who deals with asterisk must be experienced enough with
it and that will make the people who support the users a few, while with
trixbox those few people can be left as escalation points and through GUI you
can make other less aware of asterisk administer the day to day tasks. 

Trixbox
in my belief is making more people everyday depend on asterisk ofcourse knowing
how to deal directly with asterisk will be a plus but yet this could come by
time with trix box and everyday experience being gained will make them someday
reach that level. 

Trixbox
is a great start point to implement asterisk but learning asterisk
configs must also be in schedule to maintain a persistent environment. 

Thx 
MAG 



Dovid B wrote: 

Yes but they will never understand the configs. They need to learn step
by 
step. 

-
Original Message - 
From: joe, at j4computers [EMAIL PROTECTED] 
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com

Sent: Friday, October 13, 2006 4:11 PM 
Subject: Re: [asterisk-users] How do you like TrixBox? 

Dovid
B[EMAIL PROTECTED]
Wrote on: 10/13/2006 9:51 AM: 
. . . A)If something goes wrong they wont know where to 
 start. They only know the GUI. B)They will never know the real
way of 
 working asterisk.. . . 
 

But,
can't it be one way of learning? Can't one setup and modify 
a Trixbox setup, then peruse the conf files, to get familiar with 
(almost) all things Asterisk? 

Spoke as
one who was not very pleased with their own foray into Trixbox 
and is still creeping up to speed on Asterisk. 

joe 

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--
Thx
MAG

 


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-- 
www.AsteriskBlog.com
Your home for easy to learn Asterisk stuff. 






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[asterisk-users] Re: unauthenticated calls

2006-10-16 Thread Tomislav Parčina

In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
 is it possible on asterisk to reject unauthenticated calls or not registered
 phones to call?

You can send them to [default] context that has only extensions like this:

exten = i,1,Hangup
exten = s,1,Hangup


--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)270248
Mob.: +385(91)1212148
SIP: [EMAIL PROTECTED]
e-mail: tparcina#lama.hr
http://www.lama.hr
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Re: [asterisk-users] Polycom IP 501 phone randomly resets itself (loses Received call log, Missed calls, placed calls)

2006-10-16 Thread James Andrewartha

Mike Garey wrote:
 I've been noticing that my group of Polycom IP 501 phones seems to
 randomly reset themselves nearly every night (I guess it usually
 happens at night, since I've never seen it happen while I've been at
 work during the day)..
 
 When I say reset, I mean, the hands free volume and ring volume are
 set to the default and the call logs (received calls, missed calls,
 placed calls) are all reset.  It does, however, keep certain settings
 such as the specific ring tone used for incoming calls.. But most
 other settings are being reset..  Has anyone else experienced this, or
 know why it might be happening?  Thanks,

What firmware version are you using? There is an option to make the phones
check their config and reboot every day:

provisioning prov.fileSystem.rfs0.minFreeSpace=5
prov.fileSystem.ffs0.4meg.minFreeSpace=420
prov.fileSystem.ffs0.2meg.minFreeSpace=48 prov.polling.enabled=0
prov.polling.mode=abs prov.polling.period=86400 prov.polling.time=03:00/

if prev.polling.enabled is set then it will check for new configs and reboot
if there are any.

Also, if you set
  volume voice.volume.persist.handset=1
voice.volume.persist.headset=1 voice.volume.persist.handsfree=1/

then the headset and speaker volumes will be saved.

-- 
James Andrewartha
Systems Administrator
Data Analysis Australia Pty Ltd
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[asterisk-users] member queue refresh

2006-10-16 Thread nik600


hi

i've got this problem:

queue A (ringall strategy)
- sip/200
- sip/201
- sip/202

suppose that sip/200 is busy and a call is received, 201 and 202 start
ringing. After some seconds 200 becomes free but 201 and 202 are still
ringing and 200 not!

where am i wrong?

i need that when 200 becomes free it will immediately receive the call
that is ringing on the other members of the queue!

wrapuptime is set to 0
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Re: [asterisk-users] Reception Console

2006-10-16 Thread Crazy Boy

Hi,I am interested in test and work with your Reception appliation. Looking forward to your response. Thank you.Regards,Chaandra.Peter Lindquist [EMAIL PROTECTED] wrote: Sure thing, count me inPaul Hales wrote: We are currently writing a reception console for Asterisk - if anyone is interested in beta testing it, feel free to ask. Paul Hales   ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users 
		Talk is cheap. Use Yahoo! Messenger to make PC-to-Phone calls.  Great rates starting at 1¢/min.___
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[asterisk-users] Page hangs up after 5 seconds

2006-10-16 Thread Torbjörn Abrahamsson


Hi asterisk-users,

We are using Asterisk 1.2.12.1, and are trying to use the Page 
application. It seems to work but after approx 4-5 seconds the call is 
hung up.


The dialplan code look like this:

exten = _*2XX,1,AGI(get-paging-devices.agi,${EXTEN:2})
exten = _*2XX,n,GotoIf($[ ${PAGING_DEVICES} = invalid ]?i,1)
exten = _*2XX,n,SIPAddHeader(Call-Info: sip:192.168.20.1\; answer-after=0)
exten = _*2XX,n,Page(${PAGING_DEVICES},dq)


The CLI outputs the following:

-- Executing AGI(SIP/snom1-b7d0c328, get-paging-devices.agi|01) 
in new stack

-- Launched AGI Script /var/lib/asterisk/agi-bin/get-paging-devices.agi
-- AGI Script get-paging-devices.agi completed, returning 0
-- Executing GotoIf(SIP/snom1-b7d0c328, 0?i|1) in new stack
-- Executing SIPAddHeader(SIP/snom1-b7d0c328, Call-Info: 
sip:192.168.20.1; answer-after=0) in new stack
-- Executing Page(SIP/snom1-b7d0c328, SIP/snom1SIP/snom3|dq) 
in new stack

-- Created MeetMe conference 1023 for conference '2028709590d'
-- Launching MeetMe(2028709590d|qxdw(5)) on SIP/snom3-08984140
-- Hungup 'Zap/pseudo-1436409106'
  == Spawn extension (wx3trunk2, *201, 4) exited non-zero on 
'SIP/snom1-b7d0c328'

-- Executing Hangup(SIP/snom1-b7d0c328, ) in new stack


The 'full' log has this contents:

Oct 16 11:01:12 DEBUG[6767] pbx.c: Launching 'Goto'
Oct 16 11:01:12 VERBOSE[6767] logger.c: -- Executing 
Goto(SIP/snom1-b7d0c328, wx3trunk2|*201|1) in new stack

Oct 16 11:01:12 VERBOSE[6767] logger.c: -- Goto (wx3trunk2,*201,1)
Oct 16 11:01:12 DEBUG[6767] pbx.c: Launching 'AGI'
Oct 16 11:01:12 VERBOSE[6767] logger.c: -- Executing 
AGI(SIP/snom1-b7d0c328, get-paging-devices.agi|01) in new stack
Oct 16 11:01:12 VERBOSE[6767] logger.c: -- Launched AGI Script 
/var/lib/asterisk/agi-bin/get-paging-devices.agi
Oct 16 11:01:12 VERBOSE[6767] logger.c: -- AGI Script 
get-paging-devices.agi completed, returning 0

Oct 16 11:01:12 DEBUG[6767] pbx.c: Expression result is '0'
Oct 16 11:01:12 DEBUG[6767] pbx.c: Launching 'GotoIf'
Oct 16 11:01:12 VERBOSE[6767] logger.c: -- Executing 
GotoIf(SIP/snom1-b7d0c328, 0?i|1) in new stack

Oct 16 11:01:12 DEBUG[6767] pbx.c: Not taking any branch
Oct 16 11:01:12 DEBUG[6767] pbx.c: Launching 'SIPAddHeader'
Oct 16 11:01:12 VERBOSE[6767] logger.c: -- Executing 
SIPAddHeader(SIP/snom1-b7d0c328, Call-Info: sip:192.168.20.1; 
answer-after=0) in new stack

Oct 16 11:01:12 DEBUG[6767] pbx.c: Launching 'Page'
Oct 16 11:01:12 VERBOSE[6767] logger.c: -- Executing 
Page(SIP/snom1-b7d0c328, SIP/snom1SIP/snom3|dq) in new stack

Oct 16 11:01:12 DEBUG[6767] chan_sip.c: sip_answer(SIP/snom1-b7d0c328)
Oct 16 11:01:12 DEBUG[6767] app_meetme.c: Building dynamic conference 
'2028709590d'

Oct 16 11:01:12 DEBUG[6767] chan_zap.c: Using channel -2
Oct 16 11:01:12 VERBOSE[6767] logger.c: -- Created MeetMe conference 
1023 for conference '2028709590d'
Oct 16 11:01:12 DEBUG[6767] channel.c: Set channel SIP/snom1-b7d0c328 to 
write format slin
Oct 16 11:01:12 DEBUG[6767] channel.c: Set channel SIP/snom1-b7d0c328 to 
read format slin
Oct 16 11:01:12 DEBUG[6767] app_meetme.c: Placed channel 
SIP/snom1-b7d0c328 in ZAP conf 1023
Oct 16 11:01:12 DEBUG[6772] app_queue.c: Device 'SIP/snom1' changed to 
state '2' (In use) but we don't care because they're not a member of any 
queue.
Oct 16 11:01:12 DEBUG[6773] app_queue.c: Device 'Zap/pseudo' changed to 
state '2' (In use) but we don't care because they're not a member of any 
queue.
Oct 16 11:01:12 DEBUG[6771] chan_sip.c: Allocating new SIP dialog for 
(No Call-ID) - INVITE (With RTP)
Oct 16 11:01:12 DEBUG[6771] res_config_mysql.c: MySQL RealTime: 
Everything is fine.
Oct 16 11:01:12 DEBUG[6771] res_config_mysql.c: MySQL RealTime: Retrieve 
SQL: SELECT * FROM sipusers WHERE name = 'snom3'
Oct 16 11:01:12 VERBOSE[6771] logger.c: -- SIP Seeding peer from 
astdb: 'snom3' at [EMAIL PROTECTED]:59283 for 60
Oct 16 11:01:12 DEBUG[6771] chan_sip.c: Allocating new SIP dialog for 
(No Call-ID) - OPTIONS (No RTP)

Oct 16 11:01:12 DEBUG[6771] chan_sip.c: Setting NAT on RTP to 524288
Oct 16 11:01:12 DEBUG[6771] chan_sip.c: Outgoing Call for snom3
Oct 16 11:01:12 DEBUG[6771] chan_sip.c: Updating call counter for 
outgoing call

Oct 16 11:01:12 DEBUG[6767] rtp.c: Ooh, format changed from unknown to ulaw
Oct 16 11:01:12 DEBUG[6767] rtp.c: RTCP NAT: Got RTCP from other end. 
Now sending to address 212.247.4.149:49435

Oct 16 11:01:12 DEBUG[6767] rtp.c: Got RTCP report of 52 bytes
Oct 16 11:01:12 DEBUG[6767] app_meetme.c: Got unrecognized frame on 
channel SIP/snom1-b7d0c328, f-frametype=5,f-subclass=0
Oct 16 11:01:12 DEBUG[6767] rtp.c: RTP NAT: Got audio from other end. 
Now sending to address 212.247.4.149:49434
Oct 16 11:01:12 DEBUG[6767] app_meetme.c: Got unrecognized frame on 
channel SIP/snom1-b7d0c328, f-frametype=5,f-subclass=0
Oct 16 11:01:12 DEBUG[6771] rtp.c: RTCP NAT: Got RTCP from other end. 
Now sending to address 212.247.4.149:58421

Oct 16 11:01:12

[asterisk-users] asterisk upgrade

2006-10-16 Thread nik600


hi

i've got a production system running asterisk-1.2.4/ with
zaptel-1.2.4/ using a beronet  Beronet BN8S0 and  TE205P .

at the moment (fortunately) i'm not experiencing any kind of
particular problem, do you suggest me to upgrade asterisk?

and zaptel?

and misdn?

thanks
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Re: [asterisk-users] Reception Console

2006-10-16 Thread Steve Totaro


Paul,

I would love to test it out in a busy environment.  I am sure I can 
provide quite alot of feedback from a real receptionist


Thanks,
Steve Totaro

Crazy Boy wrote:

Hi,

I am interested in test and work with your Reception appliation. 
Looking forward to your response. Thank you.


Regards,
Chaandra.

*/Peter Lindquist [EMAIL PROTECTED]/* wrote:

Sure thing, count me in

Paul Hales wrote:
 We are currently writing a reception console for Asterisk - if
anyone is
 interested in beta testing it, feel free to ask.

 Paul Hales





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Re: [asterisk-users] 3way calling / codec problem

2006-10-16 Thread Thomas Kenyon


Mr. Jones wrote:

I'm having problems with conference calls (3-way) when I have my codec
forced to g729 in sip.conf.

I'm using Grandstream 2000s.

If enable both g711 and g729 then 3 way calling and transfers work.

I'm not sure why this would matter?

Here's the error:

Oct 13 13:54:45 NOTICE[31184] chan_sip.c: No compatible codecs!

Any help is greatly appreciated!


Are you out of licences? From memory when in a console each channel 
needs to be able to be transcoded to SLIN. (where it is mixed and 
transcoded back again).

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R: [asterisk-users] Quescom 400

2006-10-16 Thread Giordano Grandis

How do u call the quescom? With Dial() command?

exten = s,1,Dial(SIP/172.30.1.199:1123/${ARG2},Tt)

Did u set any port, or just call the ip address witout 1123 port ?

Thanks in advance


-Messaggio originale-
Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di [EMAIL PROTECTED]
Inviato: lunedì 16 ottobre 2006 10.59
A: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Oggetto: Re: [asterisk-users] Quescom 400

 Hi all,
 I just configured a quescom 400 to route all gsm incoming calls to
 asterisk, now i would route all outgoing asterisk calls to gsm port of
 the quescom.

 Anyone has any idea how implement it?

 I did a configuration but i always get this error

On the quescom, under the objects section, you have to add the Asterisk
box as a foreign gatekeeper. Then you have to add it as a GSM service in
the services section.


 -- Got SIP response 503 Service Unavailable back from
 ip_add_quescom400

This usually happens when it isn't registered as a service.

Cheers



-- 
This message has been scanned for viruses and
dangerous content by MailScanner, and is
believed to be clean.

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Re: [asterisk-users] Re: 1.4 beta2 on intel mac

2006-10-16 Thread Tim Panton



On 16 Oct 2006, at 09:09, Martin Joseph wrote:


On 2006-10-15 23:50:34 -0700, Tim Panton [EMAIL PROTECTED] said:


On 16 Oct 2006, at 07:15, Martin Joseph wrote:

On 2006-10-12 02:40:35 -0700, Tim Panton [EMAIL PROTECTED] said:

On 11 Oct 2006, at 19:35, Dean Collins wrote:

Lol - use a real PC maybe :P

Nah, that would be dull.
In some ways the mac intel is nearer to a 'normal PC'
(whatever that is) than the systems I normally run asterisk on
- a NatSemi Nemiah and an arm5 :-)
Asterisk 1.2.X runs fine on the intel macs, so
I guess there must be a bug in 1.4beta2 that stops it running.
Did you need to update the version of Make?  My PowerPC mac  
seems  to be complaining about version 3.80.

I don't have any Intel mac's to test with (yet).
Yes. I had to install a new make from source (with configure --  
prefix=/usr)

I've got some stuff to get ready for Astricon Dallas next week,
where we will be launching Corraleta SDK - our zero install web- 
based  Java softphone.
Once that's done and I get back I'll look into what the problem  
is  (unless someone solves it for
me while I'm there  - drop by our stand in the exhibition If you  
have  got 1.4beta2 working on

an intel mac - or if you want to see Corraleta in action! )
Tim Panton
www.mexuar.com


I just built 1.4b2 on a powerpc mac system, and although it seems  
to build ok, and starts up,  the command line is completely non- 
reponsive (although exit works).


I have head people describe this kind of dead CLI in the past, but  
never saw it before.


1.4b2 doesn't accept registrations or do anything.  If this sounds  
like what you saw, then I guess it's an OSX issue and not an Intel  
OSX issue specifically.


Yep exactly that, I'll grab an SVN head when I get back from Astricon  
and try that (unless beta3 comes out first)


Thanks.

Tim Panton

www.mexuar.com



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[asterisk-users] Re: Cisco 7970 SIP won't update?

2006-10-16 Thread Tomislav Parčina

In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
   Does anyone know what triggers the 7970 to update its config? I
 was able to get it to update to SIP, but the config I used initially
 won't go away. I am making small changes to the SEPxxx.cnf.xml file and
 rebooting the phone, the phone is downloading the (TFTP) new config
 file, but I don't see any change on the phone itself. 
   I've looked at the VersionStamp and incremented that, but still
 no go.

You setup versionStamp in cnf.xml file, but how do you check it on the phone?

--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)270248
Mob.: +385(91)1212148
SIP: [EMAIL PROTECTED]
e-mail: tparcina#lama.hr
http://www.lama.hr
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[asterisk-users] quality control

2006-10-16 Thread Juraj Bednar


Hello,


  I would like to create some form of reporting of call quality. Is
there a way to collect quality of RTP data (for SIP calls) to gather
some statistics (packet loss, ...). I would like to know when calls
are of lower quality and if I should blame ISP, operator or look for
some problems on my setup.


 Thanks,

Juraj.
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Re: [asterisk-users] VoicePulse Connect 4 Channel Limit?

2006-10-16 Thread Dovid B




I use myphonecompany.com. They have DID's for $5.00 
a month and they 'let you' use 2 channels for per did (you can use more but they 
dont like it if you abuse it). I had a client that needed 4 concurent channels 
so they told him to just purchase 2 did's. So if you need 8 concurent incoming 
channels it will cost you a total of $20.00 for inbound services :) 


  - Original Message - 
  From: 
  Nate Kapi 
  
  To: asterisk-users@lists.digium.com 
  
  Sent: Monday, October 16, 2006 7:13 
  AM
  Subject: [asterisk-users] VoicePulse 
  Connect 4 Channel Limit?
  Does anyone know what happens if you try to have 5 concurrent 
  outgoing channels with VoicePulse Connect? Does it give you an error message 
  or a reorder or something? I'm worried about using them as my primary carrier 
  if this is the case. I noticed that they supposedly only allow 4 
  channels for free and then you have to pay $20 a month extra per channel. I'm 
  guessing this is for inbound and outbound channels. If you wanted to be able 
  to have 8 concurrent channels then this could get costly. Too costly in my 
  opinion. I meanthat seems like a LOT to me, when you can go with other 
  providers who don't limit you to 4 channels, like Voxee, NuFone or SixTel, for 
  around the same price. I can understand the channel restrictions for inbound 
  calls, but not for outbound calls. VoicePulse, I know you read these 
  lists! You should be able to provide us VoicePulse Connect users with more 
  than 4 concurrent channels for free!
  
  

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Re: [asterisk-users] asterisk upgrade

2006-10-16 Thread Simone Ruffilli




at the moment (fortunately) i'm not experiencing any kind of
particular problem, do you suggest me to upgrade asterisk?

#1 sysadmin rule:
If it's not broken, just don't fix it.


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Re: [asterisk-users] asterisk upgrade

2006-10-16 Thread Conrad Wood

On Mon, 2006-10-16 at 13:13 +0200, Simone Ruffilli wrote:
  at the moment (fortunately) i'm not experiencing any kind of
  particular problem, do you suggest me to upgrade asterisk?
 #1 sysadmin rule:
 If it's not broken, just don't fix it.

That will get you into trouble when it _does_ break. 
I rather *test* new versions, fix any configuration problems and then
keep the live versions uptodate.
It can be quite a nightmare to skip lots of versions, particularly under
timepressure with a broken system at hand.

Conrad

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Re: [asterisk-users] Call bridged, but no sound

2006-10-16 Thread Norbert Zawodsky





Hi Brian, hi list,

Brian Candler wrote:

  On Fri, Oct 13, 2006 at 01:35:04AM +0200, Norbert Zawodsky wrote:
  
  
I've set canreinvite=no on the channel to the SIP provider and it
immediately worked. O.k., I'm happy about that but I want to
*understand* what's going on here.
.
My setup is:

Asterisk is connected on one side via eth1 to the "outside world" (IP
adress 81.223.xxx.xxx) and on the other side via eth0 to the internal
LAN (eth0 has IP 192.168.1.200, SNOM phone has 192.168.1.201, ...).

  
  
A good question, for which it's hard to give a short answer :-) 

  

Thanks for your explainations. Now all that is far more clear to me!
And my *-Box starts working now...

But I have another wierd problem to solve:

I reduced my sip.conf and extension.conf to an absolute minimum.

sip.conf:

[general]
context=from-inode ; Default context for incoming
calls
realm=zawodsky.at ; Realm for digest
authentication
defaultexpirey=14400
bindport=5060 ; UDP Port to bind to (SIP
standard port is 5060)
bindaddr=0.0.0.0 ; IP address to bind to
(0.0.0.0 binds to all)
srvlookup=yes ; Enable DNS SRV lookups on
outbound calls
tos=lowdelay ;
lowdelay,throughput,reliability,mincost,none
disallow=all ; First disallow all codecs
allow=alaw ; Allow codecs in order of
preference
allow=ulaw ; Allow codecs in order of
preference
allow=gsm ; Allow codecs in order of
preference

register = user:passwor@voip.inode.at:5060

externip = 81.223.241.115 ; Address that we're going to
put in outbound SIP messages
localnet=192.168.1.0/255.255.255.0 ; All RFC 1918 addresses are
local networks
nat=yes ; Global NAT settings (Affects
all peers and users)

;
; 10 - Chef (Snom360)
;
[10]
type=friend
context=local-clients
host=dynamic
secret=WdCm1g
dtmfmode=rfc2833
callerid=Chef 10
; callgroup=1
; pickupgroup=1
subscribecontext=local-clients

extensions.conf:

[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
priorityjumping=no

[globals]

[from-inode]
exten = s,1,NoOp(from-inode, EXTEN=${EXTEN})
exten = s,n,Answer()
exten = s,n,Echo()
exten = s,n,Hangup()

[local-clients]

[default]

Now, the behavior I don' understand.

I would assume that all inbound calls should be routed into the 's'
extension. I called * from another phone. The number my SIP provider
gave to me is 8904676, areacode 01. But

 if I call my box dialing my number "8904676", the call is routed to
's'. (I can hear the Echo application talking back to me)

 if I append an extension, regardless of using '10' or any other (fox
example 89046760, 89046761, 890467610, 890467612345), asterisk simply
rejects the
call. (The calling phones display says "not possible")

I turned on sip debugging and noted folowing differences in the output
(1st='8904676', 2nd='890467610'):


1st: INVITE sip:[EMAIL PROTECTED] SIP/2.0
2nd: INVITE sip:[EMAIL PROTECTED]
SIP/2.0

1st: To: sip:[EMAIL PROTECTED]
2nd: To: sip:[EMAIL PROTECTED]

1st:
From: sip:[EMAIL PROTECTED];tag=f6554db4ac48a72


2nd: From: sip:[EMAIL PROTECTED];tag=b9295878fc2630d

1st: Looking for s in from-inode (domain 81.223.241.115)
2nd: Looking for 01890467610 in from-inode (domain 81.223.241.115)

From this point on, debug output is completely different because
1st answers and 2nd hangs up.
But why is this so?

Regards Norbert
(And thank you for your patience wiht my beginners questions!)







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Re: [asterisk-users] Call bridged, but no sound

2006-10-16 Thread Brian Candler

 I turned on sip debugging and noted folowing differences in the output
 (1st='8904676', 2nd='890467610'):
 
 1st: INVITE sip:s at 81.223.241.115 SIP/2.0
 2nd: INVITE sip:01890467610 at 81.223.241.115 SIP/2.0
 
 1st: To: sip:8904676 at p1.voip.inode.at
 2nd: To: sip:890467610 at p1.voip.inode.at
 
 1st: From: sip:0132079780 at p1.voip.inode.at;tag=f6554db4ac48a72
 2nd: From: sip:0132079780 at p1.voip.inode.at;tag=b9295878fc2630d
 
 1st: Looking for s in from-inode (domain 81.223.241.115)
 2nd: Looking for 01890467610 in from-inode (domain 81.223.241.115)
 
 From this point on, debug output is completely different because 1st
 answers and 2nd hangs up.
 But why is this so?

From this, it looks to me like your SIP provider is being very kind and
sending you the full DDI in the INVITE when you dial the longer version of
the number. And so Asterisk is looking for this number in extensions.conf,
and currently fails to match.

Just try matching it in extensions.conf:

[from-inode]
exten = 01890467610,1,Answer()
exten = 01890467610,n,Echo()
exten = 01890467610,n,Hangup()

If that works, then you can use a pattern match to match everything
beginning with 018904676. More likely, you'll want to route to your internal
extensions using this number, or by stripping off the first 9 digits, so
that everyone gets their own DDI for free!

Regards,

Brian.
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[asterisk-users] Weird problem with beep.wav!

2006-10-16 Thread James Dyer

This is really doing my head in!

For some reason, my asterisk box can't playback beep.wav.

I have this extension defined in my internal context:

'10001' =1. Answer()   [pbx_config]
2. Wait(2)  [pbx_config]
3. Record(/tmp/asterisk/10001:gsm)  [pbx_config]
4. Wait(2)  [pbx_config]
5. Playback(/tmp/asterisk/10001)[pbx_config]
6. Wait(2)  [pbx_config]
7. Hangup() [pbx_config]


When I call ext 10001 from a phone, I get the following:

-- Executing Answer(IAX2/308-4, ) in new stack
-- Executing Wait(IAX2/308-4, 2) in new stack
-- Executing Record(IAX2/308-4, /tmp/asterisk/10001:gsm) in new 
stack
Oct 16 12:49:41 WARNING[8581]: format_wav.c:153 check_header: Not a wav 
file 49
Oct 16 12:49:41 WARNING[8581]: file.c:436 ast_filehelper: Unable to open 
file on /var/lib/asterisk/sounds/beep.wav
Oct 16 12:49:41 WARNING[8581]: file.c:824 ast_streamfile: Unable to open 
beep (format ulaw): No such file or directory
Oct 16 12:49:41 WARNING[8581]: app_record.c:247 record_exec: 
ast_streamfile failed on IAX2/308-4


I'm getting the same issue with voicemail when that application tries to 
play the beep.

Don't think it'a actually a problem with the playing back of WAV files as 
such, since ext. 666:
  '666' =  1. Answer()   [pbx_config]
2. Background(carried-away-by-monkeys)[pbx_config]
3. Hangup()   [pbx_config]

works fine. 

Permissionsn on the file (-rw-r--r--) are fine.

I've tried copying beep.wav to my windows box, where it played fine, so it 
is presumably not corrupted or anything.

Anyone got any ideas??


james


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[asterisk-users] Multiple 'routes' to extension in different contextes. How to influence search oder?

2006-10-16 Thread Benoit Panizzon

Hi all

I share my Asterisk Server with a few friends. It is connected to PSTN, and 
various SIP Providers.

I offer Free Calls to my friends, but myself I would like to be able to make 
calls to non free destinations via my PSTN Line.

Now I do this in my dialplan:

---
[myself]
; National Destinations
exten = _0z.,1,Dial(SIP/someisp/${EXTEN});
exten = _0z.,n,Dial(Zap/g1/${EXTEN});

; International Destinations
exten = _00z.,1,Dial(SIP/someisp/${EXTEN});
exten = _00z.,n,Dial(Zap/g1/${EXTEN});

include = freedestinations;

[freedestinations]
; Local Free Destionations
exten = _0800.,1,Dial(Zap/g1/${EXTEN});

; International Free Destionations
exten = _0049.,1,Dial(SIP/FWD/*${EXTEN}:2);
--

Now I get into this situation. I would like to call a german Free Numer: 
0049800xxx

This is best matched in the context [freedestinations], and also the cheapest. 
My Telco charges a fee to call free destionations abroad.
But still: exten = _00z. is being matched.

Is there a way to solve this in a clever way? I have started just copying all 
[freedestination] extensions into [myself] but every time I have to change 
anything I have to change it everywhere.

Regards

Benoit Panizzon
-- 
I m p r o W a r e   A G-System Services
__

Zurlindenstrasse 29 Tel  +41 61 826 93 00
CH-4133 PrattelnFax  +41 61 826 93 01
Schweiz Web  http://www.imp.ch
__


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[asterisk-users] Cisco 7970 strange Xml , but upgrade success.

2006-10-16 Thread nigma nigmus

When I try to upgrade 7970 phone to sip 8.0.4SR1, Im getting this error 
all time:


 Read request for file .loads. Mode octet [16/10 15:14:12.187]
File .loads : error 2 in system call CreateFile The system cannot find 
the file specified. [16/10 15:14:12.187]   


But I found this  inside SEP(MAC).cnf.xml :

loadInformationSIP70.8-0-4SR1S./loadInformationcare for .
When I add .(dot) at the end of version information ; upgrade started 
and successfully finished.


I hope this help.

Best Regards.

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[asterisk-users] Monitor stops recording midstream?

2006-10-16 Thread Tim Connolly

   Asterisk SVN-trunk-r7230 built by root @ pbx01.timsnet.com on a i686 
running Linux on 2006-06-17


   When I used monitor, I seem to get most calls cut off if they run 
very long. Sometimes two minutes, sometimes 5 or 15.. Seems random. Any 
ideas what might kill the recording process? I'm beginning to wonder if 
soxmix is truncating the file when it blends the in/outbound streams 
together due to bad data or something.

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RE: [asterisk-users] Reception Console

2006-10-16 Thread Viktor Tatianin

Hello Paul

Yes, I very interesting

Viktor Tatianin
[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul Hales
Sent: Monday, October 16, 2006 7:33 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Reception Console


We are currently writing a reception console for Asterisk - if anyone is
interested in beta testing it, feel free to ask.

Paul Hales

-- 
Paul Hales
Technical Manager
AsteriskIT
www.asteriskit.com.au
bus: 03 8320 8106
mob: 0434 673 529

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Re: [asterisk-users] tdm2400p question

2006-10-16 Thread Lito Lampitoc

I see, thank you very much for all your answers. Btw, the interface looks different than the ordinary rj45, so how are you going to plug in the rj45 plug to it?On 10/16/06, 
George Pajari [EMAIL PROTECTED] wrote:
The TDM2400P supports up to six quad modules -- each quad modulesupports EITHER four FXS ports OR four FXO ports...THEREFOREwith 6 quad FXO modules one has 24 FXO ports,with 5 quad FXO modules and 1 quad FXS module one has 20 FXO ports and
4 FXS ports...the remainder of these examples is left as an exercise for the reader.The board does not have to be fully populated (i.e. you do not need tohave all six quad module positions filled).
g.--George Pajari, netVOICE communications604 484 VOIP (484 8647 x102)Open Source VoIP/Telephony Specialists1 877 NET VOIP (638 8647 x102)
www.netvoice.cawww.ip-centrex.cawww.digium.ca www.grandstream.ca 
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Re: [asterisk-users] Bridging of PRI calls

2006-10-16 Thread Johann Steinwendtner


Matthew Fredrickson schrieb:


On Oct 12, 2006, at 1:17 PM, Johann Steinwendtner wrote:


Hello !

I 've some questions how bridging of ISDN calls is done.
Assume an asterisk system with a TE405 card equipped.
(PRI1 - PRI4)
An incoming ISDN call on PRI1 is transfered back to
PRI3. Unless there is DTMF detection or other things
involved, the bridging is done without Asterisk. Does
this card have a some sort of cross connection ? Does
the PCM leave the card ? Or is there some DMA magic in-
volved ?
Assume an asterisk system with two TE405 cards equipped.
An incoming ISDN call is transfered back to the second
TE405 card. Does this card have a seperate bus like H.100 ?
How is the bridging done in this scenario ?



If the call is between two spans on the card, there is an internal 
H.100-like bus that cross connects the timeslots.


Matthew Fredrickson



How is the situation between two cards ?  Is there a kind of
DMA mechanism involved, or does asterisk cross connect the timeslots ?

Thanks !

Hans
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RE: [asterisk-users] Reception Console

2006-10-16 Thread Senad Jordanovic

Viktor Tatianin wrote:
 Hello Paul
 
 Yes, I very interesting


Hi

We have MS Windows based operator consol/ panel available :)

http://www.bicomsystems.com/products/C/P/319/154_2571/#


Senad

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Re: [asterisk-users] tdm2400p question


Hi Lito,
you need a particular cable to connect TDM2400 (which has 1 big port) to 
a patch panel. Try Google on internet for a retailer.



Giorgio Incantalupo



Lito Lampitoc wrote:
I see, thank you very much for all your answers. Btw, the interface 
looks different than the ordinary rj45, so how are you going to plug 
in the rj45 plug to it?


On 10/16/06, * George Pajari* [EMAIL PROTECTED] 
mailto:[EMAIL PROTECTED] wrote:


The TDM2400P supports up to six quad modules -- each quad module
supports EITHER four FXS ports OR four FXO ports...

THEREFORE
  with 6 quad FXO modules one has 24 FXO ports,
  with 5 quad FXO modules and 1 quad FXS module one has 20 FXO
ports and
4 FXS ports...

the remainder of these examples is left as an exercise for the reader.

The board does not have to be fully populated (i.e. you do not need to
have all six quad module positions filled).

g.

--
George Pajari, netVOICE communications604 484 VOIP (484 8647 x102)
Open Source VoIP/Telephony Specialists  1 877 NET VOIP (638 8647 x102)
   www.netvoice.ca
http://www.netvoice.ca  www.ip-centrex.ca http://www.ip-centrex.ca
  www.digium.ca http://www.digium.ca www.grandstream.ca
http://www.grandstream.ca www.sipura.ca http://www.sipura.ca
www.snom.ca http://www.snom.ca

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[asterisk-users] Tellabs and PRI

2006-10-16 Thread Doug Lytle

Can anybody that is currently using a Tellabs 2572 E.C. with a PRI/ISDN 
with success, please let me know how they have the card (Wiring and 
Settings) setup.  I still have random local echo on our PRI.


Thanks,

Doug

--

Ben Franklin quote:

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deserve neither Liberty nor Safety.


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Re: [asterisk-users] Psst... Top secret information: Codename Pineapple

2006-10-16 Thread Eric \ManxPower\ Wieling


Brian Candler wrote:

On Fri, Oct 13, 2006 at 07:00:54PM -0500, Eric ManxPower Wieling wrote:

* Phones = stations, regardless of where they are

Asterisk = SIP Server, Phone = SIP Client


* Trunks = trunks to other SIP servers, bilateral

Asterisk and the other server is peer to peer


* Services = services you register for, like BroadVoice, Voop or FWD.
  (where asterisk acts as a phone)

Asterisk = SIP Client, Other End = SIP Server


Hmm, but I don't see how these ideas map to formal SIP concepts (RFC 3261).

Phone = User Agent Client (places outgoing calls) and also User Agent Server
(accepts incoming calls)

But then Asterisk is both of these too.

The term SIP Client does not appear in RFC 3261 at all. The term SIP
Server does, in a loose generic way, when they mean SIP Proxy and/or SIP
Registrar.

Asterisk is never a SIP Proxy, it's a SIP endpoint (UAC/UAS). I think it
*is* a registrar though.

So what I'm asking is: what's fundamentally different between a phone, and
trunk, and a service? How does Asterisk treat them differently?

After all, placing a SIP call to a phone (via a dialplan) and routing a SIP
call down a trunk (via a dialplan) are the same operation, aren't they


These ideas don't map to formal SIP concepts.  Olle's ideas seemt to map 
to more formal Asterisk concepts.  My terms are more generic and try 
to map to layman's internet concepts.


Really, a SIP device is a SIP device.  All SIP devices are clients and 
all SIP devices are servers.  It's how you USE the device.

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[asterisk-users] Some Warning in Asterisk for Voicemail intgreting,

2006-10-16 Thread raviprakash sunkara

Hello Users,I doing on Voicemail in Asterisk For my RealTime, By using the ODBC connectivity For Voicemessages.in Made the Change in res_odbc.conf,odbc.ini, odbcinst.ini and voicemail.conf
When I start My Asterisk server it give me Some Warning,When I googled , a proper Docummentation is not found, it found in some there languages,the First Warning is. 
Warning [30188] res_odbc.c 565 odbc_obj_connect: res_object:Error SqlConnect =-1 Error=0 [UnixODBC][Driver Manager]Data Source Name Not Found and No default Driver Specified
And Second one is Warning [30202] app_application.c 2107 MessageCount : Failed to Obtained for ' Asterisk' ! help me this..
-- Thanks and RegardsRavi Prakash Sunkara		[EMAIL PROTECTED]
 	M:+91 9985077535O:+91 40 23114549F:+91 40 40208727 		[EMAIL PROTECTED]www.hyperion-tech.com

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RE: [asterisk-users] Reception Console

2006-10-16 Thread Damon Estep

Secure multi-tenant partitioning capabilities?
What is your distribution intentions, commercial or GPL?

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Paul Hales
 Sent: Sunday, October 15, 2006 10:33 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [asterisk-users] Reception Console
 
 
 We are currently writing a reception console for Asterisk - if anyone
is
 interested in beta testing it, feel free to ask.
 
 Paul Hales
 
 --
 Paul Hales
 Technical Manager
 AsteriskIT
 www.asteriskit.com.au
 bus: 03 8320 8106
 mob: 0434 673 529
 
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[asterisk-users] Asterisk - Live Communications Server Integration

2006-10-16 Thread William Mandra

Hi all,
We are getting ready to release our Call Control Gateway application
which allows for both remote phone control and PC to phone integration
between LCS and an Asterisk PBX. The gateway is scheduled to be released
in the beginning of Nov. Currently we are looking for Beta Testers that
are interested in this solution. More information on the product, along
with the Beta Application can be found on our website at
http://www.m-networks.net/uccg/


William Mandra
M-Networks
973.559.5200 x1001
862.371.8661 (mobile)
[EMAIL PROTECTED]


---

Advance your business with the power of technology.

http://www.m-networks.net


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Re: [asterisk-users] Increase VoiceMail Messages Recording Gain - AudioCalls are Ok

2006-10-16 Thread Marco Mouta

Thanks!But i've solved my problem only using g(#) gain argument from voicemail application! For me was enough.Voicemail([EMAIL PROTECTED],b,g(10)) ; where 10 is the gain in dBthks guys for all your replies
On 10/16/06, kjcsb [EMAIL PROTECTED] wrote:







The 
  problem is:Right now, and i'm referring only to calls directly handled by 
  VoiceMail application, the users get their audio files in email but the audio 
  is very very low. I've thought about changing RX gain on PRI interface 
  between legacy pbx and asterisk, but until now no complaining with audio 
  calls.


There's a patch for this: 
http://bugs.digium.com/file_download.php?file_id=10824type=bug

Cameron



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http://lists.digium.com/mailman/listinfo/asterisk-users-- Com os melhores cumprimentos,Marco Mouta
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Re: [asterisk-users] Increase VoiceMail Messages Recording Gain - AudioCalls are Ok

2006-10-16 Thread Marco Mouta

Error syntax:is Voicemail([EMAIL PROTECTED],bg(10)) ; for busy announce and 10dB record gainOn 10/16/06, Marco Mouta 
[EMAIL PROTECTED] wrote:Thanks!But i've solved my problem only using g(#) gain argument from voicemail application! For me was enough.
Voicemail([EMAIL PROTECTED],b,g(10)) ; where 10 is the gain in dBthks guys for all your replies
On 10/16/06, kjcsb 
[EMAIL PROTECTED] wrote:








The 
  problem is:Right now, and i'm referring only to calls directly handled by 
  VoiceMail application, the users get their audio files in email but the audio 
  is very very low. I've thought about changing RX gain on PRI interface 
  between legacy pbx and asterisk, but until now no complaining with audio 
  calls.


There's a patch for this: 

http://bugs.digium.com/file_download.php?file_id=10824type=bug

Cameron



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Marco Mouta

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Re: [asterisk-users] tdm2400p question

2006-10-16 Thread Dave Schardin

The cable is an Amphenol Cable. This may help some.On Oct 16, 2006, at 8:34 AM, Giorgio Incantalupo wrote:Hi Lito,you need a particular cable to connect TDM2400 (which has 1 big port) to a patch panel. Try Google on internet for a retailer.Giorgio IncantalupoLito Lampitoc wrote: I see, thank you very much for all your answers. Btw, the interface looks different than the ordinary rj45, so how are you going to plug in the rj45 plug to it?On 10/16/06, * George Pajari* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote:    The TDM2400P supports up to six quad modules -- each quad module    supports EITHER four FXS ports OR four FXO ports...    THEREFORE      with 6 quad FXO modules one has 24 FXO ports,      with 5 quad FXO modules and 1 quad FXS module one has 20 FXO    ports and    4 FXS ports...    the remainder of these examples is left as an exercise for the reader.    The board does not have to be fully populated (i.e. you do not need to    have all six quad module positions filled).    g.    --    George Pajari, netVOICE communications    604 484 VOIP (484 8647 x102)    Open Source VoIP/Telephony Specialists  1 877 NET VOIP (638 8647 x102)                       www.netvoice.ca    http://www.netvoice.ca  www.ip-centrex.ca http://www.ip-centrex.ca          www.digium.ca http://www.digium.ca www.grandstream.ca    http://www.grandstream.ca www.sipura.ca http://www.sipura.ca    www.snom.ca http://www.snom.ca    ___    --Bandwidth and Colocation provided by Easynews.com    http://Easynews.com --    asterisk-users mailing list    To UNSUBSCRIBE or update options visit:       http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users   ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:  http://lists.digium.com/mailman/listinfo/asterisk-users  Looking for voice over IP products?  Visit our VoIP store at http://voipstore.atacomm.com ___
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Re: [asterisk-users] Multiple 'routes' to extension in different contextes. How to influence search oder?

2006-10-16 Thread Brian Candler

On Mon, Oct 16, 2006 at 02:08:05PM +0200, Benoit Panizzon wrote:
 [myself]
 ; National Destinations
 exten = _0z.,1,Dial(SIP/someisp/${EXTEN});
 exten = _0z.,n,Dial(Zap/g1/${EXTEN});
 
 ; International Destinations
 exten = _00z.,1,Dial(SIP/someisp/${EXTEN});
 exten = _00z.,n,Dial(Zap/g1/${EXTEN});
 
 include = freedestinations;
 
 [freedestinations]
 ; Local Free Destionations
 exten = _0800.,1,Dial(Zap/g1/${EXTEN});
 
 ; International Free Destionations
 exten = _0049.,1,Dial(SIP/FWD/*${EXTEN}:2);
 --
 
 Now I get into this situation. I would like to call a german Free Numer: 
 0049800xxx
 
 This is best matched in the context [freedestinations], and also the 
 cheapest. 
 My Telco charges a fee to call free destionations abroad.
 But still: exten = _00z. is being matched.
 
 Is there a way to solve this in a clever way? I have started just copying all 
 [freedestination] extensions into [myself] but every time I have to change 
 anything I have to change it everywhere.

Try moving the [myself] destinations into another context, say [pstn], then
do

[myself]
include = freedestinations
include = pstn

whilst your friends' contexts only include = freedestinations

Check the wiki:
http://www.voip-info.org/wiki/index.php?page=Asterisk+config+extensions.conf+sorting

Regards,

Brian.
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[asterisk-users] tdm2400p question

2006-10-16 Thread Cavanna, Richard



Richard G. Cavanna
Information Technology Manager
SyChip Inc.
P - 972.202.8840
F - 972.633.0327
You can buy a pre made breakout box or go directly to a patch panel.  

I have used this one form VoIP supply with success

http://www.voipsupply.com/product_info.php?products_id=1164searchid=111
839

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[asterisk-users] Remote UNIX connection, Remote UNIX disconnected displayed every second


Hi,
every second I get on the console:
Remote UNIX connection
Remote UNIX disconnected
which gives no problem but makes console unusable.
Is there anybody who has encountered the same problem? How did you solve it?

TIA

Giorgio Incantalupo
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RE: [asterisk-users] Reception Console

2006-10-16 Thread Henry.L.Coleman

I have a bata site we can use to test your software.
Please contact me [EMAIL PROTECTED]

Henry L.Coleman CEO
*VoIP-PBX* 1-866-415-5355
Toronto Ontario
Canada


 Secure multi-tenant partitioning capabilities?
 What is your distribution intentions, commercial or GPL?

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Paul Hales
 Sent: Sunday, October 15, 2006 10:33 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [asterisk-users] Reception Console


 We are currently writing a reception console for Asterisk - if anyone
 is
 interested in beta testing it, feel free to ask.

 Paul Hales

 --
 Paul Hales
 Technical Manager
 AsteriskIT
 www.asteriskit.com.au
 bus: 03 8320 8106
 mob: 0434 673 529

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Re: [asterisk-users] tdm2400p question

2006-10-16 Thread C F


This:
http://en.wikipedia.org/wiki/RJ-21
and this:
http://en.wikipedia.org/wiki/66_block
will get you there.

On 10/16/06, Giorgio Incantalupo [EMAIL PROTECTED] wrote:

Hi Lito,
you need a particular cable to connect TDM2400 (which has 1 big port) to
a patch panel. Try Google on internet for a retailer.


Giorgio Incantalupo



Lito Lampitoc wrote:
 I see, thank you very much for all your answers. Btw, the interface
 looks different than the ordinary rj45, so how are you going to plug
 in the rj45 plug to it?

 On 10/16/06, * George Pajari* [EMAIL PROTECTED]
 mailto:[EMAIL PROTECTED] wrote:

 The TDM2400P supports up to six quad modules -- each quad module
 supports EITHER four FXS ports OR four FXO ports...

 THEREFORE
   with 6 quad FXO modules one has 24 FXO ports,
   with 5 quad FXO modules and 1 quad FXS module one has 20 FXO
 ports and
 4 FXS ports...

 the remainder of these examples is left as an exercise for the reader.

 The board does not have to be fully populated (i.e. you do not need to
 have all six quad module positions filled).

 g.

 --
 George Pajari, netVOICE communications604 484 VOIP (484 8647 x102)
 Open Source VoIP/Telephony Specialists  1 877 NET VOIP (638 8647 x102)
www.netvoice.ca
 http://www.netvoice.ca  www.ip-centrex.ca http://www.ip-centrex.ca
   www.digium.ca http://www.digium.ca www.grandstream.ca
 http://www.grandstream.ca www.sipura.ca http://www.sipura.ca
 www.snom.ca http://www.snom.ca

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[asterisk-users] Asterisk-ooh323c Video ?

2006-10-16 Thread Patrick

I know this question has been asked a great deal, but 
does any1 have a simple way Of getting video to work 
using this particular channel...

Or at least is it possible just using the conf files, or do I
Have to have a separate decoder to encode the video
Thanks again


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RE: [asterisk-users] asterisk upgrade

2006-10-16 Thread Tim Sharp

I am currently running 1.2.7.1 and it works just fine.  I personally like to 
stay 3 or 4 months behind the current release.  This time it is a bit longer 
because I don't feel comfortable with the stability of later releases. 
Tim

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Conrad Wood
Sent: Monday, October 16, 2006 7:25 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] asterisk upgrade

On Mon, 2006-10-16 at 13:13 +0200, Simone Ruffilli wrote:
  at the moment (fortunately) i'm not experiencing any kind of
  particular problem, do you suggest me to upgrade asterisk?
 #1 sysadmin rule:
 If it's not broken, just don't fix it.

That will get you into trouble when it _does_ break. 
I rather *test* new versions, fix any configuration problems and then
keep the live versions uptodate.
It can be quite a nightmare to skip lots of versions, particularly under
timepressure with a broken system at hand.

Conrad

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[asterisk-users] Asterisk-ooh323c Video ?

2006-10-16 Thread Patrick









I know this question has been asked a great deal, but
does any1 have a simple way Of getting video to work using this particular
channel...



Or at least is it possible just using the conf files, or
do I Have to have a separate decoder to encode the video 



Thanks again








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[asterisk-users] Re: Asterisk (meetme) and SMP/HT OK?

2006-10-16 Thread Steve Edwards


More info...

All calls come in from a Tekelec-7000/r4.0.

The box has 2 te410p's left over from when calls came in from PRI. They 
were left in for a timing source since I don't have physical access.


On Fri, 13 Oct 2006, Steve Edwards wrote:


In the past, there have been reports of problems with Asterisk with
multiple processors and/or HyperThreading.

I'm having a [EMAIL PROTECTED] of a problem with an HPDL380 with 2 3.4gHz Xeon processors, 
2 gb RAM -- if I got 24 hours I'd think I had died and gone to heaven :)


Am I missing something obvious like Asterisk is single CPU, single core? I 
can't access the ILO so I can't just try it.


I'm currently running Asterisk SVN-branch-1.2-r43977, but Asterisk has never 
been stable, regardless of the version, release or SVN.


I have submitted a bug report, but it's been over 2 months and nobody seems 
interested in fixing a problem that has crashed 75 times (yes, seventy-five 
times) in the last 10 days!


The vast majority of crashes are in meetme. The bt's look like this:

#0  0x005e67a2 in _dl_sysinfo_int80 () from /lib/ld-linux.so.2
#0  0x005e67a2 in _dl_sysinfo_int80 () from /lib/ld-linux.so.2
#1  0x006267a5 in raise () from /lib/tls/libc.so.6
#2  0x00628209 in abort () from /lib/tls/libc.so.6
#3  0x0065a71a in __libc_message () from /lib/tls/libc.so.6
#4  0x00660fbf in _int_free () from /lib/tls/libc.so.6
#5  0x0066133a in free () from /lib/tls/libc.so.6
#6  0x080615f3 in ast_channel_free (chan=0xb7904e00) at channel.c:959
#7  0x08062bd7 in ast_hangup (chan=0xb7904e00) at channel.c:1392
#8  0x001aa4fb in conf_free (conf=0xb7901d98) at app_meetme.c:789
#9  0x001acfa3 in conf_run (chan=0x96e94a0, conf=0xb7901d98, confflags=4224, 
optargs=0xb7ddcd4c) at app_meetme.c:1607
#10 0x001aeb26 in conf_exec (chan=0x96e94a0, data=0xb7de1070) at 
app_meetme.c:2031
#11 0x08083d43 in pbx_exec (c=0x96e94a0, app=0x9587840, data=0xb7de1070, 
newstack=1) at pbx.c:553


Any clues leading to the arrest and conviction of this bug will earn you a 
case of Sierra Nevada at the next west coast Astricon :)


Thanks in advance,

Steve Edwards  [EMAIL PROTECTED]  Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000



Thanks in advance,

Steve Edwards  [EMAIL PROTECTED]  Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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[asterisk-users] Do you encounter this REC alarm before?

2006-10-16 Thread Xue Liangliang


We deployed a PABX in China, orginally it used Netcom（网通）'s E1, the
zaptel.conf is as following:


span=1,0,0,ccs,hdb3,crc4
bchan=1-15
bchan=17-31
dchan=16
loadzone=cn
defaultzone=cn

However, recently customer changed to use China Telecom（中国电信）'s E1, it
always show REC, RED/REC, RED,  cycling alarm when I run zttool in
console. They sometimes still can make call, but the quality was quite
bad. China Telecom's engineers already checked the cable using some E1
test tools, it works perfect, and they even plug E1 into a Panasonic
PABX, it didn't have any quality problem. FYI, The card model is
TE412P.

--
Regards!
Liangliang
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Re: [asterisk-users] 3way calling / codec problem

2006-10-16 Thread Mr. Jones


Is there some way I can tell?

On 10/16/06, Thomas Kenyon [EMAIL PROTECTED] wrote:

Mr. Jones wrote:
 I'm having problems with conference calls (3-way) when I have my codec
 forced to g729 in sip.conf.

 I'm using Grandstream 2000s.

 If enable both g711 and g729 then 3 way calling and transfers work.

 I'm not sure why this would matter?

 Here's the error:

 Oct 13 13:54:45 NOTICE[31184] chan_sip.c: No compatible codecs!

 Any help is greatly appreciated!

Are you out of licences? From memory when in a console each channel
needs to be able to be transcoded to SLIN. (where it is mixed and
transcoded back again).
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[asterisk-users] ZapHFC quadBRI D-Channel going down randomly

2006-10-16 Thread Alberto Pastore


Hi.

I'm running some asterisk boxes on different sites,
some equipped with a couple of ZapHFC cards, others with
Junghanns quadBRI cards.

All boxes were compiled with Asterisk 1.2.10 (libpri 1.2.3 zaptel 1.2.6)
and bristuff 0.3.0 pre 1s, distribution is Fedora Core 4 with
kernel 2.6.17.3

The cards are connected to Telecom Italia's NT1/NT1+ S/T lines;
some of them are point-to-point, others are point-to-multipoint.

I keep getting always the same problem: after some hours of regular
working, some boxes report the usual message


   Primary D-Channel on span n down


(where n is different every time, depending on the number of
active bri spans)

I've read on previous postings that having layer 1 down on ptmp
spans is normal.

However after getting a down message (on ptp spans too!) I'm no
more able to place outgoing calls on that span, until
I restart asterisk  zaptel drivers.

Sometimes, they get back working by themselves (with the related
span up notification) after a random time period.

During the down period, incoming calls are regularly served.
However these calls do not change the status of the span, i.e.
as soon as the calls are hung up, the span gets down again.

I've tried to capture the dialog between the card and NT1 equipment,
and during the down state, I got this repeated over and over:


Sending Set Asynchronous Balanced Mode Extended
 [ 00 8b 7f ]
Unnumbered frame:
SAPI: 00  C/R: 0 EA: 0
 TEI: 069EA: 1
  M3: 3   P/F: 1 M2: 3 11: 3  [ SABME (set asynchronous balanced mode 
extended) ]

  == Primary D-Channel on span 1 down


In zapata.conf I'm pretty sure I've always set the correct signalling 
settings
(switchtype = euroisdn, signalling = bri_cpe_ptmp or bri_cpe depending 
on the case)


In /etc/zaptel.conf, I've tried many combinations with no difference; my 
current

settings are like this:

span=1,1,0,ccs,ami
bchan=1-2
dchan=3

span=2,1,0,ccs,ami
bchan=4-5
dchan=6

etc


Any clue?

Thanks,
Alberto

--
Alberto Pastore
B-Press Srl - Gruppo MSoft
P.IVA 01697420030
P.le Lombardia, 4 - 28100 Novara - Italy
Tel. 0321-499508 
Fax 0321-492974

http://www.msoft.it

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Re: [asterisk-users] tdm2400p question

2006-10-16 Thread Jay R. Ashworth

On Mon, Oct 16, 2006 at 11:27:51AM -0400, C F wrote:
 This:
 http://en.wikipedia.org/wiki/RJ-21
 and this:
 http://en.wikipedia.org/wiki/66_block
 will get you there.

If the TDM card's connector is actually an Amphenol 50 (which *just*
fits into a card bracket hole, IIRC) and it's actually wired in
accordance with RJ21X, then you can get pre-stubbed, pre-labeled 66 and
110 blocks with the connectorized cable attached, which -- if you're
doing an install of any size -- will save you endless grief.

Cheers,
-- jra
-- 
Jay R. Ashworth[EMAIL PROTECTED]
Designer  Baylink RFC 2100
Ashworth  AssociatesThe Things I Think'87 e24
St Petersburg FL USA  http://baylink.pitas.com +1 727 647 1274

That's women for you; you divorce them, and 10 years later,
  they stop having sex with you.  -- Jennifer Crusie; _Fast_Women_
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Re: [asterisk-users] FOP run control for CentOS/RHEL

2006-10-16 Thread Anthony Rodgers


Like the one that comes with it?

[EMAIL PROTECTED] ~]$ sudo more /etc/init.d/op_panel
#!/bin/bash
#
# chkconfig: 2345 99 15
# description: Flash Operator Panel
# processname: op_server.pl

# source function library
. /etc/rc.d/init.d/functions

DAEMON=/usr/local/op_panel/op_server.pl
OPTIONS=-d
RETVAL=0

case $1 in
  start)
echo -n Starting Flash Operator Panel: 
daemon $DAEMON $OPTIONS
RETVAL=$?
echo
[ $RETVAL -eq 0 ]  touch /var/lock/subsys/op_server.pl
;;
  stop)
echo -n Shutting dows Flash Operator Panel: 
killproc op_server.pl
RETVAL=$?

echo
[ $RETVAL -eq 0 ]  rm -f /var/lock/subsys/op_server.pl
;;
  restart)
$0 stop
$0 start
RETVAL=$?
;;
  reload)
echo -n Reloading Flash Operator Panel configuration: 
killproc op_server.pl -HUP
RETVAL=$?
echo
;;
  status)
status op_server.pl
RETVAL=$?
;;
  *)
echo Usage: op_panel {start|stop|status|restart|reload}
exit 1
esac

exit $RETVAL

CP

On 16-Oct-06, at 1:08 AM, Eric Bishop wrote:


Anyone have a sane rc script for FOP on CentOS/RHEL systems?


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RE: [asterisk-users] ZapHFC quadBRI D-Channel going down randomly

2006-10-16 Thread asterisk

On most traditional pabx's it's possible to set layer 1 to permanent or
call. It sounds like your system is configured for permanent and your lines
to call. How you would set this on asterisk I have no idea.

fadge

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alberto
Pastore
Sent: 16 October 2006 17:26
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] ZapHFC  quadBRI D-Channel going down randomly

Hi.

I'm running some asterisk boxes on different sites,
some equipped with a couple of ZapHFC cards, others with
Junghanns quadBRI cards.

All boxes were compiled with Asterisk 1.2.10 (libpri 1.2.3 zaptel 1.2.6)
and bristuff 0.3.0 pre 1s, distribution is Fedora Core 4 with
kernel 2.6.17.3

The cards are connected to Telecom Italia's NT1/NT1+ S/T lines;
some of them are point-to-point, others are point-to-multipoint.

I keep getting always the same problem: after some hours of regular
working, some boxes report the usual message


Primary D-Channel on span n down


(where n is different every time, depending on the number of
active bri spans)

I've read on previous postings that having layer 1 down on ptmp
spans is normal.

However after getting a down message (on ptp spans too!) I'm no
more able to place outgoing calls on that span, until
I restart asterisk  zaptel drivers.

Sometimes, they get back working by themselves (with the related
span up notification) after a random time period.

During the down period, incoming calls are regularly served.
However these calls do not change the status of the span, i.e.
as soon as the calls are hung up, the span gets down again.

I've tried to capture the dialog between the card and NT1 equipment,
and during the down state, I got this repeated over and over:


Sending Set Asynchronous Balanced Mode Extended
  [ 00 8b 7f ]
Unnumbered frame:
SAPI: 00  C/R: 0 EA: 0
  TEI: 069EA: 1
   M3: 3   P/F: 1 M2: 3 11: 3  [ SABME (set asynchronous balanced mode 
extended) ]
   == Primary D-Channel on span 1 down


In zapata.conf I'm pretty sure I've always set the correct signalling 
settings
(switchtype = euroisdn, signalling = bri_cpe_ptmp or bri_cpe depending 
on the case)

In /etc/zaptel.conf, I've tried many combinations with no difference; my 
current
settings are like this:

span=1,1,0,ccs,ami
bchan=1-2
dchan=3

span=2,1,0,ccs,ami
bchan=4-5
dchan=6

etc


Any clue?

Thanks,
Alberto

--
Alberto Pastore
B-Press Srl - Gruppo MSoft
P.IVA 01697420030
P.le Lombardia, 4 - 28100 Novara - Italy
Tel. 0321-499508 
Fax 0321-492974
http://www.msoft.it

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Re: [asterisk-users] DID is not working (call is not routing)

2006-10-16 Thread R.R Libera

Hello Chandra,

What about Teliax´s service? Is it recommended? How´s their call quality? Thanks in advance...


On 10/10/06, Crazy Boy [EMAIL PROTECTED] wrote:
Hi William,My DID is working and am receiving calls. The problem is with Teliax settings from their end. Thank you for spending your valuable time for me. 
Regards,Chandra.William Piper [EMAIL PROTECTED]
 wrote: 


Your server seems to be doing exactly what you are telling it to do:

-- Executing Playback(SIP/216.89.79.2-09e1d020, ss-noservice) in new stack-- Playing 'ss-noservice' (language 'en')
Read the extensions.conf directions on the wiki site:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.conf


bp
On 10/8/06, Crazy Boy [EMAIL PROTECTED]
 wrote: 
Hi,I have created SIP extenstions and created Teliax Trunk using IAX2. I am making outgoing calls to USA successfully. 
When I am making a call to my DID number from outside, its telling that The number you have dialed is not inservice. Here I am giving the output from Asterisk server console: 
*CLI -- IAX2/teliax-2 answered SIP/350-09e3b540 -- Executing GotoIf(SIP/216.89.79.2 
-09e1d020, 0?from-trunk||1) in new stack -- Executing Set(SIP/216.89.79.2-09e1d020, TIMEOUT(absolute)=15) in new stack  -- Channel will hangup at 2006-10-06 11:27:55 UTC. 
 -- Executing Answer(SIP/216.89.79.2-09e1d020, ) in new stack -- Executing Wait(SIP/216.89.79.2-09e1d020, 2) in new stack  -- Executing Playback(SIP/216.89.79.2-09e1d020, ss-noservice) in new stack 
 -- Playing 'ss-noservice' (language 'en') -- Executing Congestion(SIP/216.89.79.2-09e1d020, ) in new stack  == Spawn extension (from-sip-external, s, 6) exited non-zero on 'SIP/216.89.79.2-09e1d020' 
 -- Executing NoOp(SIP/216.89.79.2-09e1d020, Hangup) in new stack -- Executing Set(SIP/216.89.79.2-09e1d020, DID=s) in new stack  -- Executing Goto(SIP/216.89.79.2-09e1d020, s|1) in new stack 
 -- Goto (from-sip-external,s,1) -- Executing GotoIf(SIP/216.89.79.2-09e1d020, 0?from-trunk|s|1) in new stack  -- Executing Set(SIP/216.89.79.2-09e1d020, TIMEOUT(absolute)=15) in new stack 
 -- Channel will hangup at 2006-10-06 11:28:04 UTC. -- Executing Answer(SIP/216.89.79.2-09e1d020, ) in new stack  == Spawn extension (from-sip-external, s, 3) exited non-zero on 'SIP/216.89.79.2-09e1d020' 
When I am calling from outside phone, call is coming to my server and is not routing. I am making calls to USA and between SIP extensions successfully. Please tell me the solution. Looking forward to your response. Thank you. 
Regards,Chandra.


Talk is cheap. Use Yahoo! Messenger to make PC-to-Phone calls. 
Great rates starting at 1¢/min. 

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Check it out. 
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RE: [asterisk-users] Inhouse SIP to ZAP has echo sometimes.

2006-10-16 Thread Ejay Hire




Hello.

I had the same problem, and was able to fix it as 
follows.

1. Run fxotune
2. Call your XO rep and get a milliwatt test line 
number
3. set the gain in the zaptel.conf incoming with the 
milliwatt test line
4. loop a call through the pbx and set the outgoing 
gain.

Withthese setup properly, the echo problem goes 
away.

-Ejay


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of BerkHolz, 
StevenSent: Friday, October 13, 2006 12:51 PMTo: 
asterisk-users@lists.digium.comSubject: [asterisk-users] Inhouse SIP 
to ZAP has echo sometimes.

Sometimes we get 
echo heard on SIP phone when dialing out.

Zap channel is on 
aTE411 Card.
It is using a PRI to 
XO.

As far as I know 
echo is created on the far side.
Could the Zaptel 
card be the far side as far as the SIP phone is concerned?

Calls from our soon 
to retire legacy PBX do not have this problem.
Those calls are 
Legacy - PRI - asterisk -PRI - XO - Dest.


Any 
suggestions?






Zaptel.conf:

context=from-pstnswitchtype=nationalpridialplan=unknown 
prilocaldialplan=unknownpriindication=inbandsignalling=pri_cpeusecallerid=yeshidecallerid=no
usecallingpres=yesechocancel=yesechocancelwhenbridged=yesechotraining=yesgroup=0callgroup=1pickupgroup=1useincomingcalleridonzaptransfer=yescallerid=asreceivedaccountcode=Imusiconhold=defaultoverlapdial=nofacilityenable=yesnsf=nonechannel 
= 1-23

I also set "static int vpmdtmfsupport = 0;" in wct4xxp.c to remove sporotic DTMF 
tones.







Thank You,
Steven 
BerkHolz- MCSA 
- MCSE -Manager of Information SystemsTESCO Group 
CompaniesFax. 248-836-5101www.TESCOGroup.com
Board member 
ofwww.glimasoutheast.org

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Re: [asterisk-users] Remote UNIX connection, Remote UNIX disconnected displayed every second

2006-10-16 Thread Brian Candler

On Mon, Oct 16, 2006 at 04:47:31PM +0200, Giorgio Incantalupo wrote:
 Hi,
 every second I get on the console:
 Remote UNIX connection
 Remote UNIX disconnected
 which gives no problem but makes console unusable.
 Is there anybody who has encountered the same problem? How did you solve it?

Have you got more than one copy of safe_asterisk running?
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Re: [asterisk-users] Remote UNIX connection, Remote UNIX disconnected displayed every second


Hi,
every second I get on the console:
Remote UNIX connection
Remote UNIX disconnected
which gives no problem but makes console unusable.
Is there anybody who has encountered the same problem? How did you solve it?

You probably have some script that use the console to query something,
like the WebMeetme application.

hth
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[asterisk-users] Why is this happening?


In my IAX config file I have:
[general]
bindport = 4569   ; Port to bind to (IAX is 4569)
bindaddr = 0.0.0.0; Address to bind to (all addresses on machine)
delayreject=yes
disallow=all
allow=ulaw
allow=gsm
jitterbuffer=yes
forcejitterbuffer=yes
mailboxdetail=yes
dropcount=3
minexcessbuffer=80
jittershrinkrate=1
notransfer=yes


allanrobertson-  209.23.224.97   (D)  255.255.255.255  1207  OK (33 ms)

Why is it running on port 1207?
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Re: [asterisk-users] asterisk upgrade

2006-10-16 Thread Mojo with Horan Company, LLC

I concur with Conrad.  As I understand it, as long as you stick with 
1.2.x versions, there should be no new 'features' to worry about 
implementing, only bugfixes.  So I'd recommend keeping up with them, and 
the 'upgrade' should go smoothly because it's generally not too much of 
an upgrade.


I'd recommend you stay away from 1.4.x until you've had ample time to 
test it (and it's had ample time to be claimed mature enough for 
production).  I'm not claiming 1.4.x is not ready for production, I bet 
there is a large number of peoples on the list using it happily in 
production, but you rightly seem to desire as little trouble as possible 
in the upgrade...  and the implementation of a few features has changed 
considerably in 1.4.


Moj

Conrad Wood wrote:

On Mon, 2006-10-16 at 13:13 +0200, Simone Ruffilli wrote:

at the moment (fortunately) i'm not experiencing any kind of
particular problem, do you suggest me to upgrade asterisk?

#1 sysadmin rule:
If it's not broken, just don't fix it.


That will get you into trouble when it _does_ break. 
I rather *test* new versions, fix any configuration problems and then

keep the live versions uptodate.
It can be quite a nightmare to skip lots of versions, particularly under
timepressure with a broken system at hand.

Conrad

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!DSPAM:500,45336c6c291961385410434!



--
Mojo [EMAIL PROTECTED]
Office Manager, Horan  Company, LLC
(907) 747- x112
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Re: [asterisk-users] Why is this happening?


Why is it running on port 1207?

because Asterisk is listening on port 4569 and when a connection comes
in, it as handed to another port so it can continue listening on port
4569. Otherwise you would only be handling 1 connection at a time.

Pretty basic networking stuff I think :c)
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Re: [asterisk-users] Why is this happening?


On 10/16/06, Time Bandit [EMAIL PROTECTED] wrote:

 Why is it running on port 1207?
because Asterisk is listening on port 4569 and when a connection comes
in, it as handed to another port so it can continue listening on port
4569. Otherwise you would only be handling 1 connection at a time.

Pretty basic networking stuff I think :c)


Thanks for the answer, but I don't buy it.  There are currently 0
calls up on that bridge, while another connection which has calls up
on it is on Port 4569.. please try again.  IAX2 is suppose to run on
ONLY one port.. this is why it is so nice for use in firewall
situations.
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Re: [asterisk-users] Unable to open Asterisk database

2006-10-16 Thread Andrea Spadaccini

Ciao Giorgio,

 I'm using mysql to store my cdr data. I compiled asterisk-addon
 module without problems and I see nothing unusual in my
 cdr_mysql.conf but when I do a reload I get this messages (never seen
 before):
 
 Oct 16 09:43:16 WARNING[8576]: db.c:67 dbinit: Unable to open
 Asterisk database
 Oct 16 09:43:16 WARNING[8576]: db.c:423 ast_db_gettree: Database
 unavailable
 
 But If I try to connect from shell it works without any problem.
 
 Does anybody know why?

I think that the error message refers to the Asterisk internal
database (AstDB), and not to MySQL.

This doesn't clarify the error, but might explain why you get a working
CDR.

Try to issue the db get and db put CLI commands, to see if AstDB is
working.

HTH,

-- 
Andrea Spadaccini
Multimedia Technologies Institute s.r.l.
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Re: [asterisk-users] VoipSupply? [Semi-Urgent]

2006-10-16 Thread VaibhaV Sharma

I don't think this is a problem because of the snow storm.

I just got off the phone with them. The sales guy I used to deal with left a
few months back and since then, its been a pain to get anything done with
them. People I have dealt with had no clue.

I called them this morning for a problem to be told that a technical support
person will call me back within an hour. Then no one calls back for 5
hours. So when I call them back, I am told We don't do technical support on
the phone. I don't know who told you that.

The lady who I was speaking with had no clue of what I was asking for. She
kept putting me on hold to ask someone for an answer.

What was my question?

Q. We purchased 25 polycom IP 601/501 from you a while back and one of them
   has a faulty power supply. How do I get a new one?
A. Hold on Oh! You have to speak with RMA and not technical support. Go
   to our website / rma and submit an RMA.

Q. Well, power supplies don't have serial numbers!
A. Hold on. .. No you will have to obtain an RMA!

Q. Well, what do I send to you? Can I speak with a technical support person?
A. Hold on. .. Send us the power supply *and* the phone.

Q. It will cost me the money for a power supply to ship the phone to you.
   Can you tell me somewhere else I can get just the power supply?
A. If I had the answer I would have told you, sir.

Gah!

This is just one case. I am really disappointed with their service. I am
worried about our technical support options for the polycom phones after the
last few expereinces with Voipsupply.

--
VaibhaV


On 10/14/06 10:36 AM, Matt [EMAIL PROTECTED] wrote:

 Contact them again... they have always been very good... I'm chocking
 this up to the snow storm.
 
 On 10/13/06, Shaw Terwilliger [EMAIL PROTECTED] wrote:
 Matt wrote:
 Hi,
 Does anyone know what is going on with voipsupply?   My sales guy
 hasn't been online in several days, their 800 number is fasy busy, as
 are their direct lines.  And the canadian store website is down.  What
 the heck is going on?
 
 If you search the archives from a few months ago you'll find a few
 unhappy voipsupply customers (including me).  They never shipped what I
 ordered, didn't respond to any e-mail or calls.  The president saw the
 list traffic and sent me a long apology (stating his commitment to
 service) and offered to send me an extra component that I had cancelled
 the order for--free of charge--as a show of good will.
 
 It's been two or three months since that promise, and I never received
 the part.  He hasn't responded to my follow-up did you really mean it?
 e-mail either.
 
 --
 Shaw Terwilliger [EMAIL PROTECTED]
 SourceGear LLC
 
 
 
 
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[asterisk-users] Accessing MySQL DB to set variables in Asterisk

2006-10-16 Thread Jean-Marc Salsa

Hi,

I have set up a fax to email capability on our Asterisk Server (which is used for voicemail, and fax2email only),
and would like to improve it !

Everytime Asterisk receives a fax, I would like it to go and search in a DB if the Extension is defined.
If it is, then it will set the Email address where to send the fax.

And actually, it would be the same for voicemail,
Our users have different extensions. And we want them to have only one Mailbox.
Thus we can configure the primary extension as the main voicemail number,
but we need to have all the secondaries extension to be sent to the primary one.
Thus I would like to seach into a DB,
ifsuch a secondaryextension exists, it sends back the primary extension, 
so we can route the call to the appropriate mailbox ( primary )

I would like to use Asterisk + MySQL Realtime.
I have set up in res_mysql.conf the way to access MySQL
I have set up in extconfig.conf :
 fax2email = mysql,asteriskrealtime,fax2email
And hen in my DB :
 DB Name = asteriskrealtime
 Table Name = fax2email

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[asterisk-users] Accessing MySQL DB to set variables in Asterisk

2006-10-16 Thread Jean-Marc Salsa

Hi, ( Sorry for previous post, it was incomplete :o(

I have set up a fax to email capability on our Asterisk Server (which is used for voicemail, and fax2email only),
and would like to improve it !

Everytime Asterisk receives a fax, I would like it to go and search in a DB if the Extension is defined.
If it is, then it will set the Email address where to send the fax.
I do not want to use extension.conf ...

And actually, it would be the same for voicemail,
Our users have different extensions. And we want them to have only one Mailbox.
Thus we can configure the primary extension as the main voicemail number,
but we need to have all the secondaries extension to be sent to the primary one.
Thus I would like to seach into a DB,
ifsuch a secondaryextension exists, it sends back the primary extension, 
so we can route the call to the appropriate mailbox ( primary )

I would like to use Asterisk + MySQL Realtime.
I have set up in res_mysql.conf the way to access MySQL
I have set up in extconfig.conf :
 fax2email = mysql,asteriskrealtime,fax2email
And then in my DB :
 DB Name = asteriskrealtime
 Table Name = fax2email
Table fax2email:
 Field EXT which contains extension numbers
 Field email which contains the email where to send the fax

If I use DBGet, how to specify that I want to retrieve the email address from fax2email table, which matches the extension in Asterisk ?

Thanks for your help guys !

Yours,

Jean-Marc ( can cannot send more than 10 lines in one email :o)
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Re: [asterisk-users] Why is this happening?

2006-10-16 Thread burke

Do me a favor and try running netstat -aplntu | grep asterisk and see
what ports are actually being used. Are you connected to another ITSP? If
so then that may be the local port of that connection... just an idea, i
don't have Asterisk access right now to double check.

Ryan


 On 10/16/06, Time Bandit [EMAIL PROTECTED] wrote:
  Why is it running on port 1207?
 because Asterisk is listening on port 4569 and when a connection comes
 in, it as handed to another port so it can continue listening on port
 4569. Otherwise you would only be handling 1 connection at a time.

 Pretty basic networking stuff I think :c)

 Thanks for the answer, but I don't buy it.  There are currently 0
 calls up on that bridge, while another connection which has calls up
 on it is on Port 4569.. please try again.  IAX2 is suppose to run on
 ONLY one port.. this is why it is so nice for use in firewall
 situations.
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Re: [asterisk-users] Why is this happening?

2006-10-16 Thread Andrew Kohlsmith

On Monday 16 October 2006 16:15, Matt wrote:
 Thanks for the answer, but I don't buy it.  There are currently 0

Whether you buy it or not is irrelevant.  That is the port that this asterisk 
box is seeing the other one up on.  It is seeing it that way (most likely) 
due to NAT between the two boxes.  i.e. the far end box is on port 4569/udp 
but it's being natted to 1207/udp on the outside.

I see this all the time in both my SIP and IAX2 registrations, although the 
port numbers are generally NATted much higher.  I only use Linux NAT though, 
so others could be acting quite differently.

-A.
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Re: [asterisk-users] Why is this happening?

2006-10-16 Thread Tim Panton



On 16 Oct 2006, at 20:43, Matt wrote:


In my IAX config file I have:
[general]
bindport = 4569   ; Port to bind to (IAX is 4569)
bindaddr = 0.0.0.0; Address to bind to (all addresses on machine)
delayreject=yes
disallow=all
allow=ulaw
allow=gsm
jitterbuffer=yes
forcejitterbuffer=yes
mailboxdetail=yes
dropcount=3
minexcessbuffer=80
jittershrinkrate=1
notransfer=yes


allanrobertson-  209.23.224.97   (D)  255.255.255.255   
1207  OK (33 ms)


Why is it running on port 1207?


I'm guessing here, since you haven't told us where you ran the  
command to generate that line

or what the command was, but it was probably
iax2 show peers
on your local machine.
This then tells you about the status of the peers (and friends) in  
iax.conf


It tells you what your local asterisk sees. So it is telling you that  
the _far_ asterisk
that has registered as peer allanrobertson from ipaddress  
209.23.224.97 on port 1207.


The port number may not be 4569 because :   
	1) it isn't asterisk at the far end - iaxclients (like ours) may use  
any port to connect.
	2) the remote asterisk is on 4569 but there is a nat/port mapping  
router

in-between
3) the remote asterisk has been configured to use 1207 (unlikely)


Tim Panton

www.mexuar.com



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Re: [asterisk-users] Accessing MySQL DB to set variables in Asterisk

2006-10-16 Thread Andrea Spadaccini

Ciao Jean-Marc,

 Everytime Asterisk receives a fax, I would like it to go and search
 in a DB if the Extension is defined.
 If it is, then it will set the Email address where to send the fax.

You can use app_addon_mysql for your purposes.
See: http://www.voip-info.org/wiki/view/Asterisk+cmd+MYSQL .

HTH,

-- 
Andrea Spadaccini
Multimedia Technologies Institute s.r.l.
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Re: [asterisk-users] VoipSupply? [Semi-Urgent]

2006-10-16 Thread Jessee J Holmes

Dear VaibhaV,You can purchase this part from pretty much any certified Polycom reseller.For the IP 30x/50x you would want the Mfg Part Number 2200-07496-001For the IP 430/60x you would want the Mfg Part Number 2200-17492-101We among many other certified resellers sell this part.Being a reseller ourselves I can understand why VoIPSupply does this (as far as wanting the phone and the power supply shipped back whole), but I also understand your frustrations with this kind of setup.Additionally, being a Minnesota based company, we can understand how these kind of weather related conditions can affect quality of service and with such we offer our sincerest wishes that everyone at VoIPSupply stays warm and safe. Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED]Looking for voice over IP products? Visit our VoIP store at http://voipstore.atacomm.com/ On Oct 16, 2006, at 3:25 PM, VaibhaV Sharma wrote:I don't think this is a problem because of the snow storm.I just got off the phone with them. The sales guy I used to deal with left afew months back and since then, its been a pain to get anything done withthem. People I have dealt with had no clue.I called them this morning for a problem to be told that a technical supportperson will call me back "within an hour". Then no one calls back for 5hours. So when I call them back, I am told "We don't do technical support onthe phone. I don't know who told you that".The lady who I was speaking with had no clue of what I was asking for. Shekept putting me on hold to ask someone for an answer.What was my question?Q. We purchased 25 polycom IP 601/501 from you a while back and one of them has a faulty power supply. How do I get a new one?A. Hold on Oh! You have to speak with RMA and not technical support. Go to our website / rma and submit an RMA.Q. Well, power supplies don't have serial numbers!A. Hold on. .. No you will have to obtain an RMA!Q. Well, what do I send to you? Can I speak with a technical support person?A. Hold on. .. Send us the power supply *and* the phone.Q. It will cost me the money for a power supply to ship the phone to you. Can you tell me somewhere else I can get just the power supply?A. If I had the answer I would have told you, sir.Gah!This is just one case. I am really disappointed with their service. I amworried about our technical support options for the polycom phones after thelast few expereinces with Voipsupply.--VaibhaVOn 10/14/06 10:36 AM, "Matt" [EMAIL PROTECTED] wrote: Contact them again... they have always been very good... I'm chockingthis up to the snow storm.On 10/13/06, Shaw Terwilliger [EMAIL PROTECTED] wrote: Matt wrote: Hi,Does anyone know what is going on with voipsupply? My sales guyhasn't been online in several days, their 800 number is fasy busy, asare their direct lines. And the canadian store website is down. Whatthe heck is going on? If you search the archives from a few months ago you'll find a fewunhappy voipsupply customers (including me). They never shipped what Iordered, didn't respond to any e-mail or calls. The president saw thelist traffic and sent me a long apology (stating his commitment toservice) and offered to send me an extra component that I had cancelledthe order for--free of charge--as a show of good will.It's been two or three months since that promise, and I never receivedthe part. He hasn't responded to my follow-up "did you really mean it?"e-mail either.--Shaw Terwilliger [EMAIL PROTECTED]SourceGear LLC ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___
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Re: [asterisk-users] Why is this happening?


Thanks for the answer, but I don't buy it.  There are currently 0
calls up on that bridge, while another connection which has calls up
on it is on Port 4569.. please try again.  IAX2 is suppose to run on
ONLY one port.. this is why it is so nice for use in firewall
situations.


It doesn't change a thing !

Same thing happens with a webserver. It listen for connections on port
80 (default port) and when a connection comes in, it is handed to
another free port on the server so the main server can continue
listening on port 80. Same thing with FTP, etc. All TCP servers that
accept more than one connection

I think that what iax2 show peers display is the remote port from
which the client connected. iaxclient library defaults to using port
4569 as the originating port but there is a function to specify
another port.

Check on your machine while you're surfing the web, your browser
doesn't use port 80 as the originating port. Connect to an FTP server
and check your netstats, you'll see that you're not connected to port
21 on the remote server
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Re: [asterisk-users] Why is this happening?


Andrew,
I totally buy YOUR explination and that is what I think is happening..
the NAT box on the far end (not ours) is changing the port.

My question is... if both machiens are set to listen on 4569, will the
fact that that router is mangeling the port cause any issues?

-- Forwarded message --
From: Andrew Kohlsmith [EMAIL PROTECTED]
Date: Oct 16, 2006 4:54 PM
Subject: Re: [asterisk-users] Why is this happening?
To: asterisk-users@lists.digium.com


On Monday 16 October 2006 16:15, Matt wrote:

Thanks for the answer, but I don't buy it.  There are currently 0


Whether you buy it or not is irrelevant.  That is the port that this asterisk
box is seeing the other one up on.  It is seeing it that way (most likely)
due to NAT between the two boxes.  i.e. the far end box is on port 4569/udp
but it's being natted to 1207/udp on the outside.

I see this all the time in both my SIP and IAX2 registrations, although the
port numbers are generally NATted much higher.  I only use Linux NAT though,
so others could be acting quite differently.

-A.
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Re: [asterisk-users] asterisk upgrade

2006-10-16 Thread Peter Bowyer


On 16/10/06, Simone Ruffilli [EMAIL PROTECTED] wrote:


 at the moment (fortunately) i'm not experiencing any kind of
 particular problem, do you suggest me to upgrade asterisk?
#1 sysadmin rule:
If it's not broken, just don't fix it.


Slightly older and wiser sysadmins consider the importance of staying
with a supportable version of software, especially if it's open
source. If there's a security-related bug found in your version, will
it get patched, or will you have a forced upgrade several versions
ahead on your hands in a hurry?

Peter

--
Peter Bowyer
Email: [EMAIL PROTECTED]
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Re: [asterisk-users] Accessing MySQL DB to set variables in Asterisk

2006-10-16 Thread Jean-Marc Salsa

Thanks,
it seems to be not easy to use, but ... should do what's needed !

Thanks.
On 10/16/06, Andrea Spadaccini [EMAIL PROTECTED] wrote:
Ciao Jean-Marc, Everytime Asterisk receives a fax, I would like it to go and search in a DB if the Extension is defined.
 If it is, then it will set the Email address where to send the fax.You can use app_addon_mysql for your purposes.See: http://www.voip-info.org/wiki/view/Asterisk+cmd+MYSQL
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[asterisk-users] Is 1.2.12.1 production ready

2006-10-16 Thread shadowym


I am getting ready to image a production system.  Right now I am planning on
using Centos 4.4, Asterisk 1.2.12.1, Freepbx 2.1.3.  I will be using a
Sangoma A200D card.

I read of some people having problems with Asterisk 1.2.12.1 crashing.  Is
this across the board or is there anyone out there with no problems.  If you
have 24/7 uptime and no nightly reboot crons I would definitely appreciate
hearing  about it.

Cheers

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Re: [asterisk-users] VoipSupply? [Semi-Urgent]

2006-10-16 Thread Doug

Or try WalMart. Just make ABSOLUTELY CERTAIN that you
use the correct voltage and polarity. Also make
certain that the current rating is adequate.

At 16:16 10/16/2006, Jessee J Holmes wrote:

Dear VaibhaV,

You can purchase this part from pretty much any certified Polycom reseller.

For the IP 30x/50x you would want the Mfg Part Number 2200-07496-001
For the IP 430/60x you would want the Mfg Part Number 2200-17492-101

We among many other certified resellers sell this part.

Being a reseller ourselves I can understand why VoIPSupply does this
(as far as wanting the phone and the power supply shipped back
whole), but I also understand your frustrations with this kind of setup.

Additionally, being a Minnesota based company, we can understand how
these kind of weather related conditions can affect quality of
service and with such we offer our sincerest wishes that everyone at
VoIPSupply stays warm and safe.

Jessee Holmes

Atacomm / Ataractic Corporation

www.atacomm.com

V: 1-877-700-VOIP

mailto:[EMAIL PROTECTED][EMAIL PROTECTED]

Looking for voice over IP products? Visit our VoIP store at
http://voipstore.atacomm.comhttp://voipstore.atacomm.com/

On Oct 16, 2006, at 3:25 PM, VaibhaV Sharma wrote:

I don't think this is a problem because of the snow storm.

I just got off the phone with them. The sales guy I used to deal with left a
few months back and since then, its been a pain to get anything done with
them. People I have dealt with had no clue.

I called them this morning for a problem to be told that a technical support
person will call me back within an hour. Then no one calls back for 5
hours. So when I call them back, I am told We don't do technical support on
the phone. I don't know who told you that.

The lady who I was speaking with had no clue of what I was asking for. She
kept putting me on hold to ask someone for an answer.

What was my question?

Q. We purchased 25 polycom IP 601/501 from you a while back and one of them
has a faulty power supply. How do I get a new one?
A. Hold on Oh! You have to speak with RMA and not technical support. Go
to our website / rma and submit an RMA.

Q. Well, power supplies don't have serial numbers!
A. Hold on. .. No you will have to obtain an RMA!

Q. Well, what do I send to you? Can I speak with a technical support person?
A. Hold on. .. Send us the power supply *and* the phone.

Q. It will cost me the money for a power supply to ship the phone to you.
Can you tell me somewhere else I can get just the power supply?
A. If I had the answer I would have told you, sir.

Gah!

This is just one case. I am really disappointed with their service. I am
worried about our technical support options for the polycom phones after the
last few expereinces with Voipsupply.

--
VaibhaV

On 10/14/06 10:36 AM, Matt
mailto:[EMAIL PROTECTED][EMAIL PROTECTED] wrote:

Contact them again... they have always been very good... I'm chocking
this up to the snow storm.

On 10/13/06, Shaw Terwilliger
mailto:[EMAIL PROTECTED][EMAIL PROTECTED] wrote:

Matt wrote:

Hi,
Does anyone know what is going on with voipsupply? My sales guy
hasn't been online in several days, their 800 number is fasy busy, as
are their direct lines. And the canadian store website is down. What
the heck is going on?

If you search the archives from a few months ago you'll find a few
unhappy voipsupply customers (including me). They never shipped what I
ordered, didn't respond to any e-mail or calls. The president saw the
list traffic and sent me a long apology (stating his commitment to
service) and offered to send me an extra component that I had cancelled
the order for--free of charge--as a show of good will.

It's been two or three months since that promise, and I never received
the part. He hasn't responded to my follow-up did you really mean it?
e-mail either.

--
Shaw Terwilliger mailto:[EMAIL PROTECTED][EMAIL PROTECTED]
SourceGear LLC

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Re[2]: [asterisk-users] Why is this happening?

2006-10-16 Thread Melcon Moraes

OMG, please read more about network ports.

:c)
MM

 -Original Message-
From:   Time Bandit [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Cc: 
Sent:  Mon, 16 Oct 2006 17:25:22 -0400
Delivered:  
Subject:[asterisk-users] Why is this happening?

 Thanks for the answer, but I don't buy it.  There are currently 0
 calls up on that bridge, while another connection which has calls up
 on it is on Port 4569.. please try again.  IAX2 is suppose to run on
 ONLY one port.. this is why it is so nice for use in firewall
 situations.

It doesn't change a thing !

Same thing happens with a webserver. It listen for connections on port
80 (default port) and when a connection comes in, it is handed to
another free port on the server so the main server can continue
listening on port 80. Same thing with FTP, etc. All TCP servers that
accept more than one connection

I think that what iax2 show peers display is the remote port from
which the client connected. iaxclient library defaults to using port
4569 as the originating port but there is a function to specify
another port.

Check on your machine while you're surfing the web, your browser
doesn't use port 80 as the originating port. Connect to an FTP server
and check your netstats, you'll see that you're not connected to port
21 on the remote server
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E-mail classificado pelo Identificador de Spam Inteligente Terra.
Para alterar a categoria classificada, visite
http://mail.terra.com.br/protected_email/imail/imail.cgi?+_u=levelz_l=1,1161036006.564822.910.ambrose.hst.terra.com.br,5048,Des15,Des15

 --Original Message Ends--

-- 
Melcon Moraes [EMAIL PROTECTED]

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Re: Re[2]: [asterisk-users] Why is this happening?


On 10/16/06, Melcon Moraes [EMAIL PROTECTED] wrote:

OMG, please read more about network ports.

Could you tell me what is wrong with my explanation ?
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[asterisk-users] Stopping putgoing calls after working hours

2006-10-16 Thread Mohamed A. Gombolaty



Dear All,
I am trying to find a way to stop people who use phones after
business hours (a policy the company wants to implement), we have cisco
7940 and 7910 phones and sadly they don't have a phone lock password system
(on these ciscos it locks config menu changes but not the calls but the
cisco 7920 has this feauture).
So I was wondering is there a way to make this happen in asterisk??
--
Thx
MAG

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[asterisk-users] Asterisk/VOIP to PSTN (?)

2006-10-16 Thread joe, at j4computers

I'm researching an asterisk implementation for a client.  Originally, they 
wanted a T1 (as other vendors had quoted such).  Now tho, they are asking about 
just doing VOIP, cause fortune 500's seem to be so successful at it.

That questionable assertion aside, I see there are a lot of outfits 
(Asterisk2PSTN, for one) that seem to offer what I think it required, a means 
for asterisk to go to the PSTN world.

Is there an unbiased evaluation service around for this sort of thing?

Yes, I do have my fire extinguisher standing by.

joe

++
www.j4computers.com
[EMAIL PROTECTED]
   845-687-4563
Stone Ridge, NY 12484
++ 

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Re: [asterisk-users] Stopping putgoing calls after working hours

2006-10-16 Thread Mojo with Horan Company, LLC

Sure, in the context the phones live in, play around with the 
GotoIfTime() application:


Completely pseudocoded, will not work without research:

[internal]
priority 1 : gotoiftime(8:00-17:00|mon-fri?priority 3)
priority 2 : goto 10
priority 3 : dial(out_trunk, ${EXTEN})
priority 4 : hangup
priority 10: play a message outgoing call restricted
priority 11: hangup

The next move in your text adventure might be Show Application 
GotoIfTime from the CLI :)


Moj


Mohamed A. Gombolaty wrote:

Dear All,

I  am trying to find a way to stop people who use phones after business 
hours (a policy the company wants to implement), we have cisco 7940 and 
7910 phones and sadly they don't have a phone lock password system (on 
these ciscos it locks config menu changes but not the calls but the 
cisco 7920 has this feauture).


So I was wondering is there a way to make this happen in asterisk??

--
Thx
MAG

  !DSPAM:500,4534119649042068143078!




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!DSPAM:500,4534119649042068143078!


--
Mojo [EMAIL PROTECTED]
Office Manager, Horan  Company, LLC
(907) 747- x112
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Re: [asterisk-users] Why is this happening?

2006-10-16 Thread Mojo with Horan Company, LLC

You're a little backwards.  When you connect to a remote server via HTTP 
protocol, for example, you ARE connected to their remote port 80.  They 
do not send data to YOUR port 80 though.


Moj

Time Bandit wrote:

On 10/16/06, Melcon Moraes [EMAIL PROTECTED] wrote:

OMG, please read more about network ports.

Could you tell me what is wrong with my explanation ?
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Mojo [EMAIL PROTECTED]
Office Manager, Horan  Company, LLC
(907) 747- x112
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Re: [asterisk-users] Stopping putgoing calls after working hours

2006-10-16 Thread Mohamed A. Gombolaty



Dear Moj,
Thanks a lot fo the tip, it seems I can do that it is very flexible
and easy to use, I will try to add it to the trixbox files in a nice fashion
but that will be after I get some sleep ;-)
Thx
MAG

"Mojo with Horan  Company, LLC" wrote:
Sure, in the context the phones live in, play around
with the
GotoIfTime() application:
Completely pseudocoded, will not work without research:
[internal]
priority 1 : gotoiftime(8:00-17:00|mon-fri?priority 3)
priority 2 : goto 10
priority 3 : dial(out_trunk, ${EXTEN})
priority 4 : hangup
priority 10: play a message "outgoing call restricted"
priority 11: hangup
The next move in your text adventure might be "Show Application
GotoIfTime" from the CLI :)
Moj
Mohamed A. Gombolaty wrote:
> Dear All,
>
> I am trying to find a way to stop people who use phones after
business
> hours (a policy the company wants to implement), we have cisco 7940
and
> 7910 phones and sadly they don't have a phone lock password system
(on
> these ciscos it locks config menu changes but not the calls but the
> cisco 7920 has this feauture).
>
> So I was wondering is there a way to make this happen in asterisk??
>
> --
> Thx
> MAG
>
> !DSPAM:500,4534119649042068143078!
>
>
> 
>
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Thx
MAG

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Re: [asterisk-users] Stopping putgoing calls after working hours

2006-10-16 Thread Lacy Moore - Aspendora



So I was wondering is there a way to make this happen in asterisk?? 

Depending on where you are located, you might want to allow emergency calls to go through. The bloodsuckers, I mean attorneys, here in the US would have a field day if something were to happen to someone at a company that did not allow emergency numbers to be dialed.


Translated: If something were to happen to someone outside of business hours (in the US), and the phones did not allow emergency calls, it would cost your company millions of dollars.

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Re: [asterisk-users] Why is this happening?