date:20061009


On 09/10/06, Daniel Cyt [EMAIL PROTECTED] wrote:

Hi,

I can't get my INX line working for incoming (outgoing is working fine).
When  I dial this number from my home phone, asterisk sends the call
straight to extension 101, for some reason it doens't read what my
extensions.conf is saying.



SIP.CONF

register = number:[EMAIL PROTECTED]/101  ;number is a replacement for
my line number


Didn't you think that the '101' there might be a clue? Your register
statement tells the provider to deliver the call to '101'. Replace
that with a different number and something different will happen -
perhaps the rest of your dialplan is expecting the call to come in
with a destination which matches your DID - in which case, put the DID
number there instead of the 101.

Peter

--
Peter Bowyer
Email: [EMAIL PROTECTED]
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Asterisk 1.2.12 - Can NOT make call out / Asterisk terminate

2006-10-09 Thread Joseph

I just upgraded to Asterisk 1.2.12 from 1.0.1 and it seems to me
Asterisk 1.2 is not ready for PRIME TIME.

I have three Sipura units registered with Asterisk 1.2.12; registration
goes through just fine but I can not make calls out:

Got SIP response 503 Service Unavailable back from 10.0.0.102
-- SIP/pstn-1270-0819db50 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
-- Executing Hangup(SIP/11-08198610, ) in new stack

In addition Asterisk quits every time I try to make a call?
It just terminates. 

-- 
#Joseph
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk 1.2.12 - Can NOT make call out / Asterisk terminate


On 09/10/06, Joseph [EMAIL PROTECTED] wrote:

I just upgraded to Asterisk 1.2.12 from 1.0.1 and it seems to me
Asterisk 1.2 is not ready for PRIME TIME.


And that new-fangled electricity will never catch on - lets stick with
gas-lamps...

--
Peter Bowyer
Email: [EMAIL PROTECTED]
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] sip provider not working

2006-10-09 Thread [EMAIL PROTECTED]

Hi, 

Just thought i have a similiar situation. 

Im using TRIXBOX too, the latest version which has the asterisk 1.2.12.1. Im not able to get registered to some providers. It just saysRequest Sent.

The SIP logs has revealed the following.I could seetransport=UDinstead of UDP. 

Is this a bug in asterisk or something to do with TRIXBOX? 

Is this why the asterisk is not able to register to the provider?

Please let me know if anyone has a solution for this.

No clues how to solve it.

= REGISTER sip: callcentric.com SIP/2.0 Via: SIP/2.0/
UD 85.93.11.XXX:5060;branch=z9hG4bK79570937;rport From: 
sip:[EMAIL PROTECTED];tag=as656d7f9a To: sip:[EMAIL PROTECTED] 
Call-ID: [EMAIL PROTECTED] CSeq: 1006 REGISTER User-Agent: Asterisk PBX 
Max-Forwards: 70 Expires: 120 Contact: sip:[EMAIL PROTECTED];transport=
UD Event: registration Content-Length: 0 =Thanks


Dan




On 03/10/06, Jim Lynch [EMAIL PROTECTED]
 wrote: 
I am getting a couple of messages in the log I don't understand.Thefirst is:Unsupported transport 'UDP' 
The second isOct3 06:55:54 DEBUG[20372] channel.c: Nobody there, continuing...Oct3 06:55:54 DEBUG[20372] channel.c: Nobody there, continuing...Oct3 06:55:54 DEBUG[20372] channel.c: Nobody there, continuing... 
The second is repeated a number of times.I am unable to get any audioto or from my sip provider.I checked the firewall and the necessaryports are open.It used to work before I installed tribox.I guess I 
have to go back to [EMAIL PROTECTED]Thanks,Jim.___--Bandwidth and Colocation provided by 
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 list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk 1.2.12 - Can NOT make call out / Asterisk terminate

2006-10-09 Thread Dave Cotton

On Mon, 2006-10-09 at 08:59 +0100, Peter Bowyer wrote:
 On 09/10/06, Joseph [EMAIL PROTECTED] wrote:
  I just upgraded to Asterisk 1.2.12 from 1.0.1 and it seems to me
  Asterisk 1.2 is not ready for PRIME TIME.
 
 And that new-fangled electricity will never catch on - lets stick with
 gas-lamps...
 

Oh, that's why I'm not getting any phone calls.

-- 
Dave Cotton [EMAIL PROTECTED]

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Redefinition of transfer

2006-10-09 Thread john

Hi,

I redifined the transfer key in Asterisk 1.2.11 svn from the default # key 
to ** and when I do a show features in CLI I get:


Builtin Feature   Default Current
---   --- ---
Pickup*8  *8
Blind Transfer#   **
Attended Transfer *2
One Touch Monitor *1
Disconnect Call   *   *

Also, I have included:
include = featuremap

in my extensions.conf


But when I try to use the transfer feature, I only works on the # key.  And 
in other contexts the # key should be used to signify the end of a 
recording, but pressing that key activates the transfer.

By the way, the attended transfer does not work at all

Any ideas are more than welcomed.

Thanks for the help

John
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk 1.2.12 - Can NOT make call out / Asterisk terminate

2006-10-09 Thread Joseph

On Mon, 2006-10-09 at 08:59 +0100, Peter Bowyer wrote:
 On 09/10/06, Joseph [EMAIL PROTECTED] wrote:
  I just upgraded to Asterisk 1.2.12 from 1.0.1 and it seems to me
  Asterisk 1.2 is not ready for PRIME TIME.
 
 And that new-fangled electricity will never catch on - lets stick with
 gas-lamps...

Very funny!

Though, it seems to me that my crashes after today's upgrade to 1.2.12.1
are related to this bug:
http://bugs.digium.com/view.php?id=7972

-- 
#Joseph
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Beronet card strange log messages

2006-10-09 Thread Giorgio Incantalupo


Hi,
I have an Asterisk (1.2.9.1) box with a Beronet card and 
chan-misdn-queue version 0.3.1-rc23.


This morning I found  these messages inside my asterisk log (never got 
them before!!):


Oct  9 10:08:49 localhost -- MARK --
Oct  9 10:23:39 localhost kernel: mISDN: prim 280 addr 100 not implemented
Oct  9 10:23:39 localhost kernel: mISDNd: addr(f) prim(f1980) failed 
err(-22)
Oct  9 10:23:44 localhost kernel:  (skb-len=128 next_skb-len=128)
Oct  9 10:23:44 localhost kernel: channel_senddata: next_skb exist ERROR 
(skb-len=128 next_skb-len=128)
Oct  9 10:23:44 localhost last message repeated 53 times
Oct  9 10:23:44 localhost kernel: channel_ (skb-len=128 next_skb-len=128)
Oct  9 10:23:44 localhost kernel: channel_senddata: next_skb exist ERROR 
(skb-len=128 next_skb-len=128)
Oct  9 10:23:44 localhost last message repeated 217 times
Oct  9 10:23:56 localhost kernel: mISDN: prim 20080 addr 100 not implemented
Oct  9 10:24:22 localhost kernel: mISDN: prim 280 addr 100 not implemented
Oct  9 10:24:22 localhost kernel: mISDNd: addr(f) prim(f1980) failed 
err(-22)
Oct  9 10:24:27 localhost kernel:  (skb-len=128 next_skb-len=128)
Oct  9 10:24:27 localhost kernel: channel_senddata: next_skb exist ERROR 
(skb-len=128 next_skb-len=128)
Oct  9 10:24:27 localhost last message repeated 217 times
Oct  9 10:24:42 localhost kernel: mISDN: prim 280 addr 100 not implemented
Oct  9 10:24:42 localhost kernel: mISDN: prim 280 addr 100 not implemented
Oct  9 10:25:02 localhost kernel: mISDN: prim 20080 addr 100 not implemented
Oct  9 10:28:31 localhost kernel: MISDN free_device: entitylist not empty
Oct  9 10:46:16 localhost kernel: mISDN: prim 280 addr 100 not implemented

Anybody knows what do they mean?

TIA

Giorgio Incantalupo


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk 1.2.12 - Can NOT make call out / Asterisk terminate


On 09/10/06, Joseph [EMAIL PROTECTED] wrote:

On Mon, 2006-10-09 at 08:59 +0100, Peter Bowyer wrote:
 On 09/10/06, Joseph [EMAIL PROTECTED] wrote:
  I just upgraded to Asterisk 1.2.12 from 1.0.1 and it seems to me
  Asterisk 1.2 is not ready for PRIME TIME.

 And that new-fangled electricity will never catch on - lets stick with
 gas-lamps...

Very funny!

Though, it seems to me that my crashes after today's upgrade to 1.2.12.1
are related to this bug:
http://bugs.digium.com/view.php?id=7972


Fair enough - that's a bit different to 'Asterisk 1.2 is not ready for
PRIME TIME' though, isn't it? There are plenty of stable 1.2 releases,
all of which have many fewer bugs than your 1.0.x version.

Peter

--
Peter Bowyer
Email: [EMAIL PROTECTED]
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Incoming sip line with INX(internationalnumber.com)

2006-10-09 Thread Daniel Cyt


Hi Peter,

Thank you for your answer.
I did:

register = DID:[EMAIL PROTECTED]/DID

exten = DID,1,...

Now when I call the DID number It doesnt reach the Asterisk.
sip show registry shows me the line is registered but when I dial out from 
my softphone (eyeBeam) I get the 500 error - disconnected and the message 
the person you are calling is unavailable.


Please, what do you suggest me to do?


From: Peter Bowyer [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial 
Discussionasterisk-users@lists.digium.com
To: Asterisk Users Mailing List - Non-Commercial 
Discussionasterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Incoming sip line with 
INX(internationalnumber.com)

Date: Mon, 9 Oct 2006 08:48:58 +0100

On 09/10/06, Daniel Cyt [EMAIL PROTECTED] wrote:

Hi,

I can't get my INX line working for incoming (outgoing is working fine).
When  I dial this number from my home phone, asterisk sends the call
straight to extension 101, for some reason it doens't read what my
extensions.conf is saying.



SIP.CONF

register = number:[EMAIL PROTECTED]/101  ;number is a replacement 
for

my line number


Didn't you think that the '101' there might be a clue? Your register
statement tells the provider to deliver the call to '101'. Replace
that with a different number and something different will happen -
perhaps the rest of your dialplan is expecting the call to come in
with a destination which matches your DID - in which case, put the DID
number there instead of the 101.

Peter

--
Peter Bowyer
Email: [EMAIL PROTECTED]
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users


_
MSN Messenger: instale grátis e converse com seus amigos. 
http://messenger.msn.com.br


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Incoming sip line with INX(internationalnumber.com)


On 09/10/06, Daniel Cyt [EMAIL PROTECTED] wrote:

Hi Peter,

Thank you for your answer.
I did:

register = DID:[EMAIL PROTECTED]/DID

exten = DID,1,...

Now when I call the DID number It doesnt reach the Asterisk.
sip show registry shows me the line is registered but when I dial out from
my softphone (eyeBeam) I get the 500 error - disconnected and the message
the person you are calling is unavailable.

Please, what do you suggest me to do?


Have you matched up the 'context= ' entry for your SIP provider in
sip.conf with the right context in extensions.conf where the 'exten =
DID' is?

Do a sip debug and see what it's telling you about the call, post it
here if it doesn't help.

Peter

--
Peter Bowyer
Email: [EMAIL PROTECTED]
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Incoming sip line withINX(internationalnumber.com)

2006-10-09 Thread Daniel Cyt


Ok, I've got asterisk stop and start over again and Its working!!!
THANK YOU VERY MUCH



From: Daniel Cyt [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial 
Discussionasterisk-users@lists.digium.com

To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Incoming sip line 
withINX(internationalnumber.com)

Date: Mon, 09 Oct 2006 07:48:29 -0200

Hi Peter,

Thank you for your answer.
I did:

register = DID:[EMAIL PROTECTED]/DID

exten = DID,1,...

Now when I call the DID number It doesnt reach the Asterisk.
sip show registry shows me the line is registered but when I dial out from 
my softphone (eyeBeam) I get the 500 error - disconnected and the message 
the person you are calling is unavailable.


Please, what do you suggest me to do?


From: Peter Bowyer [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial 
Discussionasterisk-users@lists.digium.com
To: Asterisk Users Mailing List - Non-Commercial 
Discussionasterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Incoming sip line with 
INX(internationalnumber.com)

Date: Mon, 9 Oct 2006 08:48:58 +0100

On 09/10/06, Daniel Cyt [EMAIL PROTECTED] wrote:

Hi,

I can't get my INX line working for incoming (outgoing is working fine).
When  I dial this number from my home phone, asterisk sends the call
straight to extension 101, for some reason it doens't read what my
extensions.conf is saying.



SIP.CONF

register = number:[EMAIL PROTECTED]/101  ;number is a replacement 
for

my line number


Didn't you think that the '101' there might be a clue? Your register
statement tells the provider to deliver the call to '101'. Replace
that with a different number and something different will happen -
perhaps the rest of your dialplan is expecting the call to come in
with a destination which matches your DID - in which case, put the DID
number there instead of the 101.

Peter

--
Peter Bowyer
Email: [EMAIL PROTECTED]
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users


_
MSN Messenger: instale grátis e converse com seus amigos. 
http://messenger.msn.com.br


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users


_
MSN Messenger: instale grátis e converse com seus amigos. 
http://messenger.msn.com.br


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] password for vm users

2006-10-09 Thread Marco Mouta

just set your initial password to be equal to vm-account number, and Voicemail application will do that for you and will request users to setup a new password!On 10/9/06, 
stan ford [EMAIL PROTECTED] wrote:
how does one force mandatory password change on login? and a period of time to pass before mandating a password change?im using trixbox so if you have that info. that would be good, if you have it for asterisk, im sure i could figure it out as well
thx. 
		Yahoo! Messenger with Voice. 
Make PC-to-Phone Calls to the US (and 30+ countries) for 2¢/min or less.
___--Bandwidth and Colocation provided by Easynews.com
 --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:  
http://lists.digium.com/mailman/listinfo/asterisk-users-- Com os melhores cumprimentos,Marco Mouta
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] SIP vz IAX...

2006-10-09 Thread raviprakash sunkara

Hello Users.I'm in Dilemma with the performance on SIP and IAXCan any one help ... 1) Difference between the SIP and IAX... which one is Best... in VOIP service
I'm using only SIP protocol for my VOIP in OpenSER...And Also I using Asterisk in SIPwe can Communicate the SIP and IAX by below scenarioSIP (UA)  OPENSER - ASTERISK  IAX (UA)... this I can do...
IAX --- OPENSER - ASTERISK - SIP/IAX.But main problem is ...SupposeIAX -- ASTERISK--- openSER  SIP / IAX ... How ?
Help me this forgive me in English :P-- Thanks and RegardsRavi Prakash Sunkara		[EMAIL PROTECTED]
 	M:+91 9985077535O:+91 40 23114549F:+91 40 40208727 		[EMAIL PROTECTED]www.hyperion-tech.com

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Re: Where is the PlayDTMF command?

2006-10-09 Thread Jan du Toit

Benny Amorsen benny+usenet at amorsen.dk writes:

 
  JdT == Jan du Toit jan.du.toit at decisionworx.com writes:
 
 JdT PS: This reply will probably go under a new thread with the same
 JdT subject. I receive the digest mode of the mails on this list, and
 JdT replying to it breaks the thread. How can I avoid this in the
 JdT future? Thanks.
 
 Switch to a newsreader and use gmane.org...
 

Thanks for the tip about using a newsreader.

My problem of getting an error while executing the Manager PlayDTMF action still
persist.

Oct  6 09:31:06 WARNING[3449]: channel.c:1610 ast_waitfor_nandfds: 
Thread 294931 Blocking 'SIP/Jan-081ba140', already blocked by thread 
360468 in procedure ast_waitfor_nandfds

Can somebody please help me with this.

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Re: Where is the PlayDTMF command?

2006-10-09 Thread Jan du Toit

So I patch my asterisk (version 1.2.12.1) with the patch given by Moises.
http://galileo.ivsol.net/play_dtmf-1.2.12.1.patch
Thanks Moises.

When I type in show manager command PlayDTMF it is their. With the show manager
commands it is not within the list containing all the commands.
When I execute the manager PlayDTMF action, the manager response says DTMF
successfully queued. I don't hear anything on the phone, when I look at the CLI
I see the following warning message. Its produced everytime I execute the
PlayDTMF action.

Oct  6 09:31:06 WARNING[3449]: channel.c:1610 ast_waitfor_nandfds:
Thread 294931 Blocking 'SIP/Jan-081ba140', already blocked by thread
360468 in procedure ast_waitfor_nandfds

Am I doing something wrong? Is this a bug? Please help, I need this to
work as soon as possible...

Thanks for all the help.

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Odd echo issue with speaker phone

2006-10-09 Thread BerkHolz, Steven






I assume that this 
is from the echo canceller, but I am not sure.

A call is started 
via SIP speakerphone.
When the handset is 
picked up, there is a slight echo of your own voice after you speak.(duh, is 
there any other kind of echo)

If the call is made 
without the speaker phone, there is no echo.

I am stumped by this 
one.



Thank You,

Board member 
ofwww.glimasoutheast.org

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Re: PRI issues

2006-10-09 Thread Steven

We had that problem but changing busydetect from on to off fixed it.

It appears that you already have that covered.

-- 
-- 
Steven

http://www.glimasoutheast.org



Doug Lytle [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED]
 Hey everybody,

 I've, within the last 3 weeks, moved over to a PRI from SBC/ATT.  I've 
 received several complaints about dropped calls. 
 Reviewing the archives on PRI and dropped calls shows that I should set the 
 resetinterval=never in the zapata.conf and restart. 
 This hasn't helped.
 The dropped calls have to date only been on outbound calls.  Usually within 2 
 to 3 minutes of a call.  The full log shows 
 something about not getting a frame and stopping the bridge.

 On Saturday I put into place 1.2 Branch and have pri debug setup to log to a 
 file.  Is there anything else that I can do to get an 
 idea as to what is going on here?

 My zapata and zaptel below:

 [zaptel]

 # Zaptel Configuration File

 span=1,1,0,esf,b8zs
 defaultzone=us
 loadzone=us
 bchan=1-23
 dchan=24

 span=2,0,0,esf,b8zs
 fxsks=25-32
 fxoks=33-48
 defaultzone=us
 loadzone=us

 [zapata]

 [channels]
 ;
 context=default
 resetinterval = never
 musiconhold=tape

 switchtype=national
 context=pri
 signalling=pri_cpe
 group=1
 echocancel=yes
 echotraining=yes
 echocancelwhenbridged=yes
 rxgain=-1.0
 txgain=-2.0
 busydetect=no
 pridialplan=unknown
 usercallerid=yes
 callerid=asreceived
 channel = 1-23

 I see the following the full log:

 Oct  4 09:09:30 VERBOSE[29894] logger.c: -- Executing 
 Dial(SIP/4228-082131e8, ZAP/G1/1xx5800) in new stack
 Oct  4 09:09:30 DEBUG[29894] dsp.c: dsp busy pattern set to 0,0
 Oct  4 09:09:30 VERBOSE[29894] logger.c: -- Requested transfer 
 capability: 0x00 - SPEECH
 Oct  4 09:09:30 VERBOSE[29894] logger.c: -- Called G1/xx5800
 Oct  4 09:09:30 VERBOSE[29894] logger.c: -- Zap/23-1 is proceeding 
 passing it to SIP/4228-082131e8
 Oct  4 09:09:32 VERBOSE[29894] logger.c: -- Zap/23-1 is ringing
 Oct  4 09:09:37 VERBOSE[29894] logger.c: -- Zap/23-1 answered 
 SIP/4228-082131e8
 Oct  4 09:11:26 DEBUG[29894] channel.c: Didn't get a frame from channel: 
 SIP/4228-082131e8
 Oct  4 09:11:26 DEBUG[29894] channel.c: Bridge stops bridging channels 
 SIP/4228-082131e8 and Zap/23-1
 Oct  4 09:11:26 DEBUG[29894] chan_zap.c: Set option AUDIO MODE, value: ON(1) 
 on Zap/23-1
 Oct  4 09:11:26 DEBUG[29894] chan_zap.c: Hangup: channel: 23 index = 0, 
 normal = 40, callwait = -1, thirdcall = -1
 Oct  4 09:11:26 DEBUG[29894] chan_zap.c: Not yet hungup...  Calling hangup 
 once with icause, and clearing call
 Oct  4 09:11:26 DEBUG[29894] chan_zap.c: disabled echo cancellation on 
 channel 23
 Oct  4 09:11:26 DEBUG[29894] chan_zap.c: Set option TDD MODE, value: OFF(0) 
 on Zap/23-1
 Oct  4 09:11:26 DEBUG[29894] chan_zap.c: Updated conferencing on 23, with 0 
 conference users
 Oct  4 09:11:26 DEBUG[29894] chan_zap.c: Set option AUDIO MODE, value: OFF(0) 
 on Zap/23-1
 Oct  4 09:11:26 DEBUG[29894] chan_zap.c: disabled echo cancellation on 
 channel 23
 Oct  4 09:11:26 VERBOSE[29894] logger.c: -- Hungup 'Zap/23-1'
 Oct  4 09:11:26 DEBUG[29894] app_dial.c: Exiting with DIALSTATUS=ANSWER.
 Oct  4 09:11:26 VERBOSE[29894] logger.c:   == Spawn extension (sip, 
 x5800, 5) exited non-zero on 'SIP/4228-082131e8'
 Oct  4 09:11:26 VERBOSE[29894] logger.c: -- Executing 
 NoOp(SIP/4228-082131e8, Hungup) in new stack
 Oct  4 09:11:26 VERBOSE[29894] logger.c: -- Executing 
 Hangup(SIP/4228-082131e8, ) in new stack


 -- Ben Franklin quote: Those who would give up Essential Liberty to purchase 
 a little Temporary Safety, deserve neither Liberty 
 nor Safety.

 ___
 --Bandwidth and Colocation provided by 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 list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Odd echo issue with speaker phone


BerkHolz, Steven wrote:
 
 
If the call is made without the speaker phone, there is no echo.


This is caused by the port implementation of echo cancellation of the 
phone itself.  Grandstream 102's speaker phone sucks badly.  To the 
point where it's useless.


Doug

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Odd echo issue with speaker phone


Doug Lytle wrote:


This is caused by the port implementation of echo cancellation of the 
phone itself.  Grandstream 102's speaker phone sucks badly.  To the 
point where it's useless.


That should have read poor implementation.

Doug

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Re: Outbound FXO call, getting You must firstdial...

2006-10-09 Thread Steven

Are you are saying that you are adding a 'w' and leaving the 9 on the front?

If that is the case, remove the 9 and add more 'w's.

Any changes at the telco will break you system if you are relying on it 
dropping that first digit.

I believe it would be better for you to use the 'w's for consistency purposes.


-- 
-- 
Steven

http://www.glimasoutheast.org



Nick Ellson [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED]

 I did have a bit of trouble with searching (what to search on), though 
 looking for the w in the dial command did return quite a 
 few hits as you described. Thank you so much for taking the time to reanswer 
 a covered subject.

 I played with the settings and 1 w and removing the :1 after EXTEN (not 
 stripping the leading digit?) makes it reliable. Not 
 stripping the first digit worked about 2 in 5 attempts. I stumbled onto that 
 idea when I missdialed a number 9215037 digit 
 number and it worked! The 2 was a fat finger mistake. So I tried 
 90xx and that worked.

 As I have some success now, I can tune this so it works as the HowTo's list. 
 :)

 Thank you again!

 -- 
 Nick Ellson
 CCDA, CCNP, CCSP, CCAI,
 MCSE 2000, Security+, Network+
 Network Hobbyist, VFR Private Pilot.


 On Sat, 7 Oct 2006, Rich Adamson wrote:

 Nick Ellson wrote:

  I am not sure what I might be set up wrong, but dialing out with my Zap/1
  port seems to alwyas get the You must first dial a 1 when calling this
  number message from what sounds like the actual PSTN. My zapatel.conf and
  extensions.conf bits below. Any advice? (I do receive inbound calls, and
  it does sound like I am getting the PSTN error. I do notice that when I
  get an inbound call, I have 5 secs of sevear static before it suddenly
  becomes clear.. could that be happening on the outboud as well munging the
  first few digits?)

 signalling=fxs_ks
 language=us
 context=inbound_qwest
 sendcalleridafter=2
 callerid=yes
 threewaycalling=yes
 transfer=yes
 cancallforward=yes
 callreturn=yes
 echocancel=yes
 echocancelwhenbridged=yes
 rxgain=0.0
 txgain=0.0
 group=1
 channel=1

  exten = _9.,1,Dial(Zap/1/${EXTEN:1},60)

 You should probably do a little research before posting questions like this 
 as its been answered many many time.

 The problem is that some pstn central offices are not ready to receive 
 dtmf digits as quickly as what asterisk sends them. So, 
 an option w has been added to the Dial command to instruct asterisk to 
 wait about 200 milliseconds before sending dtmf. Try 
 something like this:
 exten = _9.,1,Dial(Zap/1/w${EXTEN:1},60)
 and notice that lower-case w in the string. If that doesn't fix the 
 problem, try two ww's in a row.


 ___
 --Bandwidth and Colocation provided by 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 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 list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Polycom 601 Expansion Module: Light the LEDs???

2006-10-09 Thread Jerry Jones


Enable buddy watch in your poly config files

also set each speed dial to have this enabled also


On Oct 9, 2006, at 12:04 AM, Doug wrote:


Hey Folks,

Been wrestling with the 601 and the expansion module.  Finally
figured out how to populate both with speed dial entries.  Also
hints are showing in Asterisk with the show hints command.

But how do I get the LEDs to light when one of these other
extensions is either off-hook, or ringing.

Reading the 'Net and Polycom's documentation doesn't give
a clear solution.  Is there a genius out there who has this
working??

Please help!!!

___
--Bandwidth and Colocation provided by 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 list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Re: Odd echo issue with speaker phone

2006-10-09 Thread Steven

Sorry for the double post.

Sometimes my posts do not show up, so I wait a day or two to make sure.

-- 
-- 
Steven

http://www.glimasoutheast.org



BerkHolz, Steven [EMAIL PROTECTED] wrote in message 
news:[EMAIL PROTECTED]

I assume that this is from the echo canceller, but I am not sure.

A call is started via SIP speakerphone.
When the handset is picked up, there is a slight echo of your own voice after 
you speak.(duh, is there any other kind of echo)

If the call is made without the speaker phone, there is no echo.

I am stumped by this one.


Thank You,

Board member of
www.glimasoutheast.org




___
--Bandwidth and Colocation provided by 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 list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Re: Asterisk 1.2.12 - Can NOT make call out / Asterisk terminate

2006-10-09 Thread Benny Amorsen

 PB == Peter Bowyer [EMAIL PROTECTED] writes:

PB Fair enough - that's a bit different to 'Asterisk 1.2 is not ready
PB for PRIME TIME' though, isn't it? There are plenty of stable 1.2
PB releases, all of which have many fewer bugs than your 1.0.x
PB version.

Unfortunately they also have security issues. It would be nice if
someone made a 1.2.7.2 with the security issues fixed. Either way it
is rather unfortunate that the latest version of 1.2 is unstable.


/Benny


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [asterisk-users] Options for moving to * friendly Business VSP

2006-10-09 Thread Gleim, Jason

Al,

If you want the most flexibility you can get and you want to (or can)
use a purely IP solution, then I would recommend looking at the
pay-as-you-go plans a lot of VoIP service providers offers. (Lots of
recommendations on this list) Most of them will allow you to pay a small
monthly fee (~ $5) for a public number and then $0.02/min for usage. The
nice thing is that there is no limit to the number of channels. So, if
you have a single 'public' number and someone calls it, the service
provider sends it to your box. While that call is in progress, if
someone else calls, the service provider just opens another channel to
your box and it rings. You don't have to maintain a block of numbers and
a hunt group which costs money and limits your max simultaneous calls.
Same thing with the other company that is merging... just port their
main number(s) over and go. You can have several numbers (including 800
TF) which all run over the same connection. The service provider will
set the call information during the call setup period. Asterisk can read
this and determine which 'line' has been called so you can route
appropriately. (Basically DID)

To smooth the transition, you would get a temp number from your VoIP
service provider and do all the testing. (Since you already have
Asterisk setup, this would be as easy as adding a new SIP or IAX trunk.)
Then, when you are ready, set your current lines to forward to that temp
number and order the number ports. When the ports go through, the
numbers will move which will drop the forwards and you should be left
with uninterrupted service.

You might also find that doing it this way saves you money. A pure IP
solution doesn't make you pay for hard-lines that are there strictly for
capacity purposes. How often do those last few lines get used in that
hunt group versus how much they cost? The real cost per call is much
higher on those lines but businesses keep them anyway because they have
to be ready for that one time a month when all the lines are busy. I
think you'll find with this solution that it scales automatically and as
long as you keep the account refilled, you can make and take as many
calls as you want. (I believe a number of providers support an account
threshold below which they will automatically refill your account with a
specific amount.)

In regards to your number portability problem... I would make your first
call to the public utility commission to find out if CableVision is even
allowed to hold that number. I believe a lot of the rules that opened
the markets to the CLECs required that a number be portable from the
ILEC to any CLEC and vice-versa. Your area may have regulations that
require your CLEC to make the number portable between service providers
and the person at CableVision you spoke with may either be unaware of it
or deliberately misleading you. In general, I find the phone companies
suddenly become very cooperative when you call them back with someone
from the PUC backing you up.

HTH!
Jason



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Al Stery
Sent: Saturday, October 07, 2006 12:51 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Options for moving to * friendly Business VSP

previuos post mangled.

Hi all, 

I have a client whose business is currently running on
[EMAIL PROTECTED] 2.6 with Cablevision' s (CV) Optimum Voice (OV)
and 3 lines. There are going to be 4 additional trunks
needed and I'd like to move/migrate them off of OV, to
a better more flexible/open/supportive VSP. OV does
not share SIP credentials and operates a closed system
which required the use of digium tdm-400b card in
order to get the trunks into * and limits what we can
achieve. There are two parts to this plan. Here are
some of the requirements for the first part.

The current 3 lines are setup as a hunt group so
there's only one published number. My client needs to
(at least for the time being) retain that phone number
(business continuity) and CV does NOT allow number's
in exchange blocks they own to be ported out. Due to
this fact, I was pondering keeping one of the OV
trunks open (the main number from the hunt group), and
set it to forward all calls to the new hunt group
number on the new VSP. This would be done until such
time as the majority of customers are updated with the
new phone number.

I'm not sure how something like this would function
but my concern would be how the hand-off on the
forward would behave. For example, can this scenario
handle multiple incoming calls simultaneously or would
one call be dumped off into OV's voicemail system?
Also, once a call is forwarded to the new number, is
the original OV trunk freed up to accept/forward more
incoming calls? or is it tied to that call?

Part two.

Another business is merging in, bringing with it 4
lines of their own, one of which is an 800 TF number,
all currently configured via Verizon POTS serivce.
Ideally, I'd like to get those 4 trunks

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

2006-10-09 Thread William Piper

No idea, I've never used Trixbox. 
I believe they have a support forum though... 

bp
On 10/9/06, Crazy Boy [EMAIL PROTECTED] wrote:
Hi William,Thank you for response. Sorry. I forgot to say that am configuring using Trixbox. Can you tell me the solution? Thank you.
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. 

___--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


How low will we go? Check out Yahoo! Messenger's low 
PC-to-Phone call rates. 
___--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 list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [asterisk-users] Blacklist to check http://whocalled.us

2006-10-09 Thread Michelle Dupuis

There's a program cid_rewrite (at www.generationd.com) which includes a
blacklist database field.  It also autopopulates based on 411.com reverse
lookup.


MD

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Noah Swint
Sent: Sunday, October 08, 2006 9:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Blacklist to check http://whocalled.us

There's a product below on the market that checks whocalled.us to
determine if a telmarketer should get the Zapteller.  Do you know if
that's something that could possibly be included into the blacklist or
in a macro.

http://venotec.com/product/tms/


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] H323 - SIP

2006-10-09 Thread tlott

Hi

The communcation between an alcatel telephone switchbox and a sip phone (using 
asterisk h.323 implementation) isnt working fully bidirectional.

The user at the alcatel telephone switchbox can hear the user who is speaking 
on the sip phone but not the other way around.

Could that be a miss-configuration or a incompatibility between asterisk h.323 
and pwlib/openh323?

The only allowed codec is alaw and the alcatel telephone switchbox is 
configured as gatekeeper.

Im using asterisk 1.2.12.1, pwlib 1.11.0 and openh323 1.19.0.1

Greetings
Tobi
-- 
Der GMX SmartSurfer hilft bis zu 70% Ihrer Onlinekosten zu sparen! 
Ideal für Modem und ISDN: http://www.gmx.net/de/go/smartsurfer
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] AstriCon Dallas in Two Weeks

2006-10-09 Thread Steven Sokol


Just a quick reminder that AstriCon is now only two weeks away.  If
you're interested in going, please see the site:
http://www.astricon.net

Thanks,

Steve

--
Steven Sokol
AstriCon 2006
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] How to make this easier

2006-10-09 Thread Tom Vile

Thanks James, I was close.On 10/8/06, James Jones [EMAIL PROTECTED] wrote:
exten = _*1XX,1,Set(CALLERID(all) = Nursery ${EXTEN:1})exten = _*1XX,2,Dial(SIP/400)Tom Vile wrote: I have a need for a dialplan that call for the ability for people to dial *1XX and it send a call
 to extension 400 with the calleridname of Nursery and the calleridnum of the *1XX number that was put in minus the *.Now I know how to do it individually but I now there must be an easier
 way to simply the code. Any help would be appreciated. Tom Vile___
--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
-- Tom VileBaldwin Technology Solutions, IncConsulting - Web Design - VoIP Telephonywww.baldwintechsolutions.comPhone: 518-631-2855 x205
Fax: 518-631-2856
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [asterisk-users] AstriCon Dallas in Two Weeks

2006-10-09 Thread Dean Collins

Hi Steven,
Any update on the number of people attending? 
I'm already booked arriving Tuesday through to Sunday but curious as to
size of the sessions.

 
Cheers,
 
Dean
 

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Steven Sokol
 Sent: Monday, 9 October 2006 9:50 AM
 To: Asterisk Users
 Subject: [asterisk-users] AstriCon Dallas in Two Weeks
 
 Just a quick reminder that AstriCon is now only two weeks away.  If
 you're interested in going, please see the site:
 http://www.astricon.net
 
 Thanks,
 
 Steve
 
 --
 Steven Sokol
 AstriCon 2006
 ___
 --Bandwidth and Colocation provided by 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 list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Lots and lots of log files

2006-10-09 Thread Ejay Hire

Hello all, and good morning

In my /var/log/asterisk directory I have 492,018 log files, most of which
are empty.
event_log.XXX queue_log.XXX messages.XXX where XXX is an integer.

I removed them all and restarted asterisk a few days ago, but they came
back.

This morning I turned off event and queue logging, but I would prefer to
have the messages log.  I didn't see an entry for theis in logrotate.

The root issue behind this is that I get a message about a signal that the
log files are too big, and asterisk stops working.  None of my log files are
 1mb though.  Restarting asterisk fixes it.  

Thanks,
Ejay


... System information 

[EMAIL PROTECTED] asterisk]# du -h /var/log/asterisk
1.3G.
840K./cdr-csv
956K./cdr-custom

Asterisk 1.2.11 
Linux version 2.6.9-42.0.2.ELsmp (gcc version 3.4.6 20060404 (Red Hat
3.4.6-3)) #1 SMP
Asterisk 1.2.11 built by buildcentos @ localhost.localdomain on a i686
running Linux 
localhost*CLI logger show channels 
Channel Type StatusConfiguration
---  ---
/var/log/asterisk/messages  File Enabled- Warning Notice
Error 
Console  Enabled- Warning Notice
Error 

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] how to play background music

2006-10-09 Thread ram

Hi all

when iam quering the Database, and till i get some results 
from the database, i want to play background music

and once i get the results, i should play the results


any examples or recomendation to achive this



Ram
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] G729 Licence Consumption Problem

2006-10-09 Thread Alvaro Parres

Yes i'm recording...On 10/8/06, Thomas Kenyon [EMAIL PROTECTED] wrote:
Alvaro Parres wrote: Hi List: I have the next diagram: GSM G729 G729IdeFisk -- Asterisk A - [INTERNET] Asterisk B - PSTN ( Via Unicall / Zap )
The user at IdeFisk Login as Agents on Asterisk B at this moment we have the next Licence Use: A) 1/1 B) 1/0When a Call from the QUEUE on Asterisk B is Bridge to the Agent I
 have the next Use:A) 1/1B) 1/3 Any one can explain me this ?, why the incress of licence consumptions. Thanks.Are you recording the call? There may be a separate process decoding the
call (each way) to make a recording.___--Bandwidth and Colocation provided by 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 list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Get user context from dialplan.

2006-10-09 Thread José Luis Gómez

Hello.
I want to know how to get the user context (set in sip.conf) from a
macro in the dialplan file. I need to use it in followme app. If user
don't have permission to dial to a cellphone he will not able to use
followme to redirect an incoming call to his cellphone.
I use outbound for generals outbound calls. So, I need to replace that
context instead the user context.

My config:
* sip.conf:
[1001]
type=friend
secret=
context=out

* extensions.conf:
[macro-stdexten]
exten = s,1,DBget(temp=CF/${ARG1})
exten = s,2,Goto(outbound,${temp},1)
..

[outbound]
exten = _9.,1,Dial(Zap/1/${EXTEN:1}) ;out calls
exten = _1XX,1,Dial(SIP/${EXTEN});calls to another users

Regards,
 José Luis

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Lots and lots of log files

2006-10-09 Thread J. Oquendo


Ejay Hire wrote:

Hello all, and good morning

In my /var/log/asterisk directory I have 492,018 log files, most of which
are empty.
event_log.XXX queue_log.XXX messages.XXX where XXX is an integer.

I removed them all and restarted asterisk a few days ago, but they came
back.

This morning I turned off event and queue logging, but I would prefer to
have the messages log.  I didn't see an entry for theis in logrotate.

The root issue behind this is that I get a message about a signal that the
log files are too big, and asterisk stops working.  None of my log files are
  
1mb though.  Restarting asterisk fixes it.  



Thanks,
Ejay


... System information 


[EMAIL PROTECTED] asterisk]# du -h /var/log/asterisk
1.3G.
840K./cdr-csv
956K./cdr-custom

Asterisk 1.2.11 
Linux version 2.6.9-42.0.2.ELsmp (gcc version 3.4.6 20060404 (Red Hat

3.4.6-3)) #1 SMP
Asterisk 1.2.11 built by buildcentos @ localhost.localdomain on a i686
running Linux 
localhost*CLI logger show channels 
Channel Type StatusConfiguration

---  ---
/var/log/asterisk/messages  File Enabled- Warning Notice
Error 
Console  Enabled- Warning Notice
Error 


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

  
Edit /etc/asterisk/logger.conf then asterisk -rx logger reload or do 
it in cron the dirty way:


0 * * * *  ls -ltha /var/log/asterisk/|awk '{print $5,$9}'|grep 0 
|grep -v [1-9]|xargs rm -rf


--

J. Oquendo
http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x1383A743
sil . infiltrated @ net http://www.infiltrated.net 


The happiness of society is the end of government.
John Adams



smime.p7s
Description: S/MIME Cryptographic Signature
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Lots and lots of log files

2006-10-09 Thread J. Oquendo


Ejay Hire wrote:

Hello all, and good morning

In my /var/log/asterisk directory I have 492,018 log files, most of which
are empty.
event_log.XXX queue_log.XXX messages.XXX where XXX is an integer.

I removed them all and restarted asterisk a few days ago, but they came
back.

This morning I turned off event and queue logging, but I would prefer to
have the messages log.  I didn't see an entry for theis in logrotate.

The root issue behind this is that I get a message about a signal that the
log files are too big, and asterisk stops working.  None of my log files are
  
1mb though.  Restarting asterisk fixes it.  



Thanks,
Ejay


... System information 


[EMAIL PROTECTED] asterisk]# du -h /var/log/asterisk
1.3G.
840K./cdr-csv
956K./cdr-custom

Asterisk 1.2.11 
Linux version 2.6.9-42.0.2.ELsmp (gcc version 3.4.6 20060404 (Red Hat

3.4.6-3)) #1 SMP
Asterisk 1.2.11 built by buildcentos @ localhost.localdomain on a i686
running Linux 
localhost*CLI logger show channels 
Channel Type StatusConfiguration

---  ---
/var/log/asterisk/messages  File Enabled- Warning Notice
Error 
Console  Enabled- Warning Notice
Error 


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

  
Edit /etc/asterisk/logger.conf then asterisk -rx logger reload or in 
cron the dirty way:


0 * * * *  ls -ltha /var/log/asterisk/|awk '{print $5,$9}'|grep 0 
|grep -v [1-9]|xargs rm -rf


--

J. Oquendo
http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x1383A743
sil . infiltrated @ net http://www.infiltrated.net 


The happiness of society is the end of government.
John Adams



smime.p7s
Description: S/MIME Cryptographic Signature
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Re: PRI issues

2006-10-09 Thread mavince

What ISDN cause code do you see when the call terminates abruptly?

Not sure if the Sangoma cards include a CSU... line errors can make strange things happenat random times... if you have a CSU, telephone company will test line to make sure that it is error free if you call in a trouble.

Do you know the type of CO switch serving you?

Mark --  Message: 8 Date: Mon, 9 Oct 2006 07:32:48 -0400 From: "Steven" <[EMAIL PROTECTED]> Subject: [asterisk-users] Re: PRI issues To: asterisk-users@lists.digium.com Message-ID: <[EMAIL PROTECTED]>  We had that problem but changing busydetect from on to off fixed it.  It appears that you already have that covered.  --  --  Steven  http://www.glimasoutheast.org"Doug Lytle" <[EMAIL PROTECTED]>wrote in message  news:[EMAIL PROTECTED] Hey everybody,   I've, within the last 3 weeks, moved over to a PRI from  SBC/ATT. I've received several complaints about dropped calls.   Reviewing the archives on PRI and dropped calls shows that I  should set the resetinterval=never in the zapata.conf and  restart.   This hasn't helped.  The dropped calls have to date only been on outbound calls.  Usually within 2 to 3 minutes of a call. The full log shows   something about not getting a frame and stopping the bridge.   On Saturday I put into place 1.2 Branch and have pri debug  setup to log to a file. Is there anything else that I can do to  get an   idea as to what is going on here?   My zapata and zaptel below:   [zaptel]   # Zaptel Configuration File   span=1,1,0,esf,b8zs  defaultzone=us  loadzone=us  bchan=1-23  dchan=24   span=2,0,0,esf,b8zs  fxsks=25-32  fxoks=33-48  defaultzone=us  loadzone=us   [zapata]   [channels]  ;  context=default  resetinterval = never  musiconhold=tape   switchtype=national  context=pri  signalling=pri_cpe  group=1  echocancel=yes  echotraining=yes  echocancelwhenbridged=yes  rxgain=-1.0  txgain=-2.0  busydetect=no  pridialplan=unknown  usercallerid=yes  callerid=asreceived  channel = 1-23   I see the following the full log:   Oct 4 09:09:30 VERBOSE[29894] logger.c: -- Executing  Dial("SIP/4228-082131e8", "ZAP/G1/1xx5800") in new stack  Oct 4 09:09:30 DEBUG[29894] dsp.c: dsp busy pattern set to 0,0  Oct 4 09:09:30 VERBOSE[29894] logger.c: -- Requested  transfer capability: 0x00 - SPEECH  Oct 4 09:09:30 VERBOSE[29894] logger.c: -- Called G1/xx5800  Oct 4 09:09:30 VERBOSE[29894] logger.c: -- Zap/23-1 is  proceeding passing it to SIP/4228-082131e8  Oct 4 09:09:32 VERBOSE[29894] logger.c: -- Zap/23-1 is ringing  Oct 4 09:09:37 VERBOSE[29894] logger.c: -- Zap/23-1  answered SIP/4228-082131e8  Oct 4 09:11:26 DEBUG[29894] channel.c: Didn't get a frame  from channel: SIP/4228-082131e8  Oct 4 09:11:26 DEBUG[29894] channel.c: Bridge stops bridging  channels SIP/4228-082131e8 and Zap/23-1  Oct 4 09:11:26 DEBUG[29894] chan_zap.c: Set option AUDIO  MODE, value: ON(1) on Zap/23-1  Oct 4 09:11:26 DEBUG[29894] chan_zap.c: Hangup: channel: 23  index = 0, normal = 40, callwait = -1, thirdcall = -1  Oct 4 09:11:26 DEBUG[29894] chan_zap.c: Not yet hungup...  Calling hangup once with icause, and clearing call  Oct 4 09:11:26 DEBUG[29894] chan_zap.c: disabled echo  cancellation on channel 23  Oct 4 09:11:26 DEBUG[29894] chan_zap.c: Set option TDD MODE,  value: OFF(0) on Zap/23-1  Oct 4 09:11:26 DEBUG[29894] chan_zap.c: Updated conferencing  on 23, with 0 conference users  Oct 4 09:11:26 DEBUG[29894] chan_zap.c: Set option AUDIO  MODE, value: OFF(0) on Zap/23-1  Oct 4 09:11:26 DEBUG[29894] chan_zap.c: disabled echo  cancellation on channel 23  Oct 4 09:11:26 VERBOSE[29894] logger.c: -- Hungup 'Zap/23-1'  Oct 4 09:11:26 DEBUG[29894] app_dial.c: Exiting with  DIALSTATUS=ANSWER. Oct 4 09:11:26 VERBOSE[29894] logger.c:  == Spawn extension (sip, x5800, 5) exited non-zero on  'SIP/4228-082131e8'  Oct 4 09:11:26 VERBOSE[29894] logger.c: -- Executing  NoOp("SIP/4228-082131e8", "Hungup") in new stack  Oct 4 09:11:26 VERBOSE[29894] logger.c: -- Executing  Hangup("SIP/4228-082131e8", "") in new stack-- Ben Franklin quote: "Those who would give up Essential  Liberty to purchase a little Temporary Safety, deserve neither  Liberty   nor Safety." 
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] isdn cross-over ...

2006-10-09 Thread Cesc


Hi,

I am sure this is a bit off topic, but maybe the people here have the knowledge.
Quick question: I have two ISDN S/T phones. What is the quickest way
to test them (call one from the other)? can i make an isdn cross-over
cable, taking the correct pinning, of course? What i need is to avoid
the need for an NT connection (via a PBX).

If the above is not possible ... where can I buy a cheap, small,
simple ISDN PBX with at least two NT ports, so that i can connect my
two phones and call each other?

Tks!

Cesc
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] VOIP with PSTN backup

2006-10-09 Thread Brian Candler

I'm looking for a way to set up a VOIP network in branch offices where one
or more phones have lifeline capability, i.e. can place calls if the IP
network or VOIP service dies, or even if power goes down. (I'm thinking of
business continuity here, not just emergency services)

This seems to limit my choice of products somewhat, and I was wondering if
anyone had recommendations for use in this scenario.

The approaches I'm thinking of are:

(1) Use an ATA with PSTN passthrough or FXO port, and connect an old
analogue telephone to the FXS port.

In this case, the analogue phone has lifeline. If there's a true FXO port
then PSTN calls can in principle be routed to/from other VOIP phones in the
office (but see below)

There seem to be a few to choose from, although far fewer with a true FXO
port.

(2) Find a VOIP phone with integrated PSTN or FXO port

In this case, the only one I have found so far by searching the web is
Clipcomm CP101.

I have also read that many FXO devices tend to be badly implemented; in
particular, on seeing ringing voltage, they actually pick up and answer the
call, instead of sending off a SIP INVITE and waiting for the OK before
connecting. I'd certainly like the device to behave properly in this regard.

As a second part of this question, it would be extremely desirable if the
backup PSTN service were available to all the phones in the office. That
means:

(a) incoming PSTN calls could ring *all* the VOIP phones in the office, not
just the one phone or ATA connected to the PSTN line; and

(b) any VOIP phone could route a call out over the LAN to the local FXO PSTN
port, e.g. by dialling a prefix to access it.

This isn't so essential but it's definitely desirable. Any recommendations
for how to do this too?

A large number of offices is going to be involved, and I want to keep as
much switching intelligence centralised as possible, both for ease of
management and to keep the cost down. That is, I don't want to install a
PC + TMD400P + Asterisk in each location, but just a small media gateway or
VOIP phone.

However I can see that the incoming ringing issue will require call forking,
so I am happy to install an OpenWrt box running Asterisk or siproxd or
whatever in each site. Being diskless and low power should mean little
maintenance is required. But such a box isn't going to be able to take an
FXS/FXO card, so I'll still rely on an ATA or VOIP phone to present a PSTN
interface. So that's the key part I'm looking for.

Finally, the devices must be robust (i.e. not need power cycling every 24
hours) and centrally manageable.

I think that's about it - many thanks for your ideas and experience!

Cheers,

Brian.
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] PRI TON/pridialplan digit prefixing

2006-10-09 Thread Jesse Peterson


Hello all,

I have a situation where I receive international calls from a legacy  
PBX into Asterisk.  These calls have a PRI TON of International and  
do not have the (US) international prefix 011 in the number.  I need  
to be able to either act on (in the dial plan) the PRI TON or be able  
to have the 011 re-added (prefixed) onto the called number so that I  
may route it properly.  Is there a way to do this?


This post:
http://lists.digium.com/pipermail/asterisk-users/2005-May/102837.html

Appears to suggest that this might be possible, but I was not able to  
get it working as suggested.


Thanks for your time,
- Jesse


--
Jesse Peterson [EMAIL PROTECTED]


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] VOIP with PSTN backup

2006-10-09 Thread Peter Lindquist


Brian,

Take a look at www.intertex.se I believe they have what you are looking for.

Peter

Brian Candler wrote:

I'm looking for a way to set up a VOIP network in branch offices where one
or more phones have lifeline capability, i.e. can place calls if the IP
network or VOIP service dies, or even if power goes down. (I'm thinking of
business continuity here, not just emergency services)

This seems to limit my choice of products somewhat, and I was wondering if
anyone had recommendations for use in this scenario.

The approaches I'm thinking of are:

(1) Use an ATA with PSTN passthrough or FXO port, and connect an old
analogue telephone to the FXS port.

In this case, the analogue phone has lifeline. If there's a true FXO port
then PSTN calls can in principle be routed to/from other VOIP phones in the
office (but see below)

There seem to be a few to choose from, although far fewer with a true FXO
port.

(2) Find a VOIP phone with integrated PSTN or FXO port

In this case, the only one I have found so far by searching the web is
Clipcomm CP101.

I have also read that many FXO devices tend to be badly implemented; in
particular, on seeing ringing voltage, they actually pick up and answer the
call, instead of sending off a SIP INVITE and waiting for the OK before
connecting. I'd certainly like the device to behave properly in this regard.

As a second part of this question, it would be extremely desirable if the
backup PSTN service were available to all the phones in the office. That
means:

(a) incoming PSTN calls could ring *all* the VOIP phones in the office, not
just the one phone or ATA connected to the PSTN line; and

(b) any VOIP phone could route a call out over the LAN to the local FXO PSTN
port, e.g. by dialling a prefix to access it.

This isn't so essential but it's definitely desirable. Any recommendations
for how to do this too?

A large number of offices is going to be involved, and I want to keep as
much switching intelligence centralised as possible, both for ease of
management and to keep the cost down. That is, I don't want to install a
PC + TMD400P + Asterisk in each location, but just a small media gateway or
VOIP phone.

However I can see that the incoming ringing issue will require call forking,
so I am happy to install an OpenWrt box running Asterisk or siproxd or
whatever in each site. Being diskless and low power should mean little
maintenance is required. But such a box isn't going to be able to take an
FXS/FXO card, so I'll still rely on an ATA or VOIP phone to present a PSTN
interface. So that's the key part I'm looking for.

Finally, the devices must be robust (i.e. not need power cycling every 24
hours) and centrally manageable.

I think that's about it - many thanks for your ideas and experience!

Cheers,

Brian.
___
--Bandwidth and Colocation provided by 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 list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Re: PRI issues


[EMAIL PROTECTED] wrote:

What ISDN cause code do you see when the call terminates abruptly?

None

The log gives indication that the SIP phone may have been at fault, 
Didn't get frame from channel SIP/4288).


 
 Not sure if the Sangoma cards include a CSU... line errors can make 
strange things happen at random times... if you have a CSU, telephone 
company will test line to make sure that it is error free if you call in 
a trouble.
 

Yes, integrated CSU/DSU in software.  The tools provided show now errors 
on the line.




 Do you know the type of CO switch serving you?
 



That I don't.  I've upgraded to 1.2 branch Saturday and nobody has 
complained as of yet (Like that means anything).


Doug

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [asterisk-users] disabling hardware echo can on tdm2400p

2006-10-09 Thread Ejay Hire

Hi.  

I don't believe this course of action will fix your issue.  It sounds like
crosstalk, where stray signals from one pair are transmitted onto another.

My standard voice quality troubleshooting fix is as follows.

1.  If the zaptel module is ancient, install the new release.
2.  Stop asterisk and run 'fxotune -I 5'.  This takes a while, 3-4 minutes
per line.
3.  Add 'fxotune -s' to /etc/init.d/asterisk
4.  Ask the Telco for the number to a milliwatt test line and use the
correct procedure to set the incoming gain in zaptel.conf.
5.  Loop two lines and use the correct procedure to set the outgoing gain
in zaptel.conf.
6.  Reset all the echo cancelling and training settings to default.

Fxotune does some magic that matches the internal settings of the zaptel
card to the lines.
Setting the gain properly in the zaptel.conf puts the incoming signal into
the optimal range of the echo filters.
Resetting the echocanceler and training settings back to default pulls
everything back to a known value, which usually works well.

This has fixed it for me every time except once.  In the case of the once,
a cheap phone cable (flat, not twisted) was run 35 feet around the edges of
the room, under space heaters, from the NID to the closet, butt-spliced and
hooked to the 66block, and then to the PBX.  Replacing this cable with a
category 5 cable run under the house (I didn't have any Cat3) fixed it.

Ejay Hire

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sean Kennedy
Sent: Saturday, October 07, 2006 5:31 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] disabling hardware echo can on tdm2400p

Hey list,

Short version:
I have a need to disable the hardware can on the tdm24xxp I have.  I figure
it's something in zconfig.h in the zaptel directory, but I'll be damned if I
can figure it out.

Long version:
I have a tdm2403e card which is experiencing an odd problem;  When several
lines are in use, there is a bleeding of lines.  My users call it the
'ghost'.  Regardless, they can hear other people's conversation on different
lines.  I've been told this has to do with the hardware echo can I have on
there, and that I should disable it if I continue having problems.  So
that's where I stand.

Answers and opinions welcome. 

Sean

___
--Bandwidth and Colocation provided by 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 list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [asterisk-users] isdn cross-over ...

2006-10-09 Thread Ejay Hire

Hi.  A cross-over cable won't work, the isdn network provides signalling
and adressing functions.

When I was studying for my CCIE, an ISDN simulator cost an arm-and-a-leg,
around $1k used from ebay.

-ejay 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Cesc
Sent: Monday, October 09, 2006 10:25 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] isdn cross-over ...

Hi,

I am sure this is a bit off topic, but maybe the people here have the
knowledge.
Quick question: I have two ISDN S/T phones. What is the quickest way to
test them (call one from the other)? can i make an isdn cross-over cable,
taking the correct pinning, of course? What i need is to avoid the need for
an NT connection (via a PBX).

If the above is not possible ... where can I buy a cheap, small, simple ISDN
PBX with at least two NT ports, so that i can connect my two phones and call
each other?

Tks!

Cesc
___
--Bandwidth and Colocation provided by 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 list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] polycom reboot script

2006-10-09 Thread Noah Miller


Hi Dean -


I actually just use the SIP notify command on the Asterisk console to
remotely reboot my Polycom phones. It requires a pre-configured
sip_notify.conf file and the Polycom option to reboot on config
check. You can then call it from a script using:


To concur with Avi, I used the script for quite a while, but found
situations where it just wouldn't reboot certain phones.  I never
really bothered to troubleshoot why.  Instead, I switched to the sip
notify method instead and it has always worked so far.

Note, though, that there's a setting in sip.cfg where you can have it
only reboot on a notify if the config files have changed (this is the
default setting).  If you prefer to always have the phone reboot when
you issue a sip notify, you can change the following line:

specialEvent ... voIpProt.SIP.specialEvent.checkSync.alwaysReboot=1

- Noah


On 10/8/06, Avi Miller [EMAIL PROTECTED] wrote:


On 09/10/2006, at 12:12 PM, Dean Collins wrote:

 can anyone give me an idea on how this reboot script works?

I actually just use the SIP notify command on the Asterisk console to
remotely reboot my Polycom phones. It requires a pre-configured
sip_notify.conf file and the Polycom option to reboot on config
check. You can then call it from a script using:

# asterisk -rx sip notify polycom-reboot 400

(Where 400 is the SIP ID of the phone).

I'm interstate at the moment, but if you send me an email, I can
lookup the settings when I'm back on Wednesday.

Ta,
Avi

--
National Manager - Special Projects

 Sydney / Melbourne / Canberra / Hobart / London /
   2/340 Gore StreetT: +61 (0) 3 9235 5400
   Fitzroy, VIC F: +61 (0) 3 9235 5444
   3065 W: http://www.squiz.net

.   Open Source - Own It - Squiz.net .. /




___
--Bandwidth and Colocation provided by 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 list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] connecting multiple servers with iax - authentication fails

2006-10-09 Thread Benko

Hello!

I'm having a problem which actually looks banal. I'm trying to
connect 3 servers via iax with each other. However, i've not been
successfull so far. Asterisk always tries to authenticate the calling
user with the credentials of the last entry in iax.conf, not the ones
that would actually belong to the calling user.

e.g. Server1 has peer/user entries for Server2 and Server3(in this
order), Server2 now tries to call Server1, but is asked for the
credentials of Server3(Because Server3 is the last entry in iax.conf),
which doesn't work of course.

The IAX debug for this example is attached(iax_server2.txt).

Please also take a look at the attached iax.conf-files for each server,
maybe i've missed some setting...

Currently i workaround this issue by using the same secret for all
servers, this is not very practicable however...

The asterisk versions in use are 1.2.9.1 on server3 and server2 and
1.4.0-beta2 on server1.

This guy seems to have had the same problem, unfortunately he received
no answer:
http://lists.digium.com/pipermail/asterisk-users/2003-August/011960.html


thx 
christian
[general]
register = server3:[EMAIL PROTECTED]
bindport=4569   ; bindport and bindaddr may be specified

bindaddr=10.1.99.157
bandwidth=high
allow=all
disallow=lpc10

jitterbuffer=no
forcejitterbuffer=no
autokill=yes


[server3]
type=peer
auth=md5
user=server3
secret=thirdsecret321
qualify=yes
host=XXX.XXX.XXX.XXX
context=iax_server3

[server3]
type=user
auth=md5
user=server3
secret=thirdsecret321
qualify=yes
host=XXX.XXX.XXX.XXX
context=iax_server3



[server2]
type=peer
auth=md5
user=server2
secret=othersecret123
qualify=yes
host=XXX.XXX.XXX.XXX
context=iax_server3

[server2]
type=user
auth=md5
user=server2
secret=othersecret123
qualify=yes
host=XXX.XXX.XXX.XXX
context=iax_server3
[general]
bindport=4569
bindaddr=213.208.4.99
bandwidth=high
allow=all
disallow=lpc10
jitterbuffer=no
forcejitterbuffer=no
tos=lowdelay
autokill=yes

[server1]
type=user
auth=md5
user=server1
secret=othersecret123
qualify=yes
host=dynamic
context=iax_server3

[server1]
type=peer
auth=md5
user=server1
secret=othersecret123
qualify=yes
host=dynamic
context=iax_server3


[server3]
type=user
auth=md5
user=mgw1
secret=12345678
qualify=yes
host=XXX.XXX.XXX.XXX
context=iax_server3

[server3]
type=peer
auth=md5
user=server3
secret=12345678
qualify=yes
host=XXX.XXX.XXX.XXX
context=iax_server3
[general]
bindport=4569
bindaddr=0.0.0.0
delayreject=yes
bandwidth=high
allow=all; same as bandwidth=high
allow=alaw
disallow=ulaw
disallow=lpc10  ; Icky sound quality...  Mr. Roboto.
jitterbuffer=no
forcejitterbuffer=no




[server1]
type=peer
auth=md5
secret=thirdsecret321  ; redundant when already embedded in 
Dial string
qualify=yes
host=81.XXX.XXX.XXX
user=server1   ; redundant when already embedded in Dial string
context=iax_server1

[server1]
type=user
auth=md5
secret=thirdsecret321  ; redundant when already embedded in 
Dial string
qualify=yes
host=81.XXX.XXX.XXX
user=server1   ; redundant when already embedded in Dial string
context=iax_server1




[server2]
type=peer
auth=md5
secret=12345678  ; redundant when already embedded in Dial 
string
qualify=yes
host=XXX.XXX.XXX.XXX
user=server2   ; redundant when already embedded in Dial string
context=iax_server3 ;yes, this context is the same as in iax.conf.server2.txt


[server2]
type=user
auth=md5
secret=12345678  ; redundant when already embedded in Dial 
string
qualify=yes
host=XXX.XXX.XXX.XXX
user=server2   ; redundant when already embedded in Dial string
context=iax_server3 ;yes, this context is the same as in iax.conf.server2.txt
Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW
   Timestamp: 00015ms  SCall: 6  DCall: 0 [81.XXX.XXX.XXX:4569]
   VERSION : 2
   CALLED NUMBER   : 004989153213126
   CODEC_PREFS : ()
   CALLING NUMBER  : 49896272423
   CALLING PRESNTN : 34
   CALLING TYPEOFN : 0
   CALLING TRANSIT : 0
   CALLING NAME: server1
   LANGUAGE: en
   FORMAT  : 4
   CAPABILITY  : 4194175
   ADSICPE : 2
   DATE TIME   : 2006-10-09  17:18:34

Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: AUTHREQ
   Timestamp: 2ms  SCall: 00030  DCall: 6 [81.XXX.XXX.XXX:4569]
   AUTHMETHODS : 2
   CHALLENGE   : 757581300
   USERNAME: server3

Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: AUTHREP
   Timestamp: 00040ms  SCall: 6  DCall: 00030 [81.XXX.XXX.XXX:4569]
   MD5 RESULT  : 94975d6e1044df7ddcafee71463fbfd9
server2*CLI 
Oct  9 17:15:53 NOTICE[11866]: chan_iax2.c:7229 socket_read: Host 
81.XXX.XXX.XXX failed to authenticate as server1
Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: REJECT 
   Timestamp: 00027ms  SCall: 00030  DCall: 6 [81.XXX.XXX.XXX:4569]
   CAUSE   : No

Re: [asterisk-users] ftp server

2006-10-09 Thread Noah Miller


Hi Avi -


 username and password is PlcmSpIp.  vsftpd cannot handle capitalized
 usernames, so if you want to use vsftpd, you have to manually
 re-configure the username on each phone.

I use vsftpd and I'm using the default PlcmSpIp username just
fine. :) Essentially, I configured PlcmSpIp as a Linux user and I'm
serving it out by using personalised FTP home directories in vsftp
and then chrooting per user. Works like a charm and no phone
configuration is required.


Quite right.  I'm blaming the inadequacies of my OS on vsftpd.  vsftpd
just uses your OS user accounts.  On the Tao linux box that I had it
installed on, you couldn't do capitals in user account names.  My bad.

- Noah
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Re: PRI issues

On Mon, Oct 09, 2006 at 12:01:02PM -0400, Doug Lytle wrote:
 Yes, integrated CSU/DSU in software.  The tools provided show now errors 
 on the line.

DSU maybe.

But if someone's figured out how to strip 130VDC off a copper pair and
convert a T-span to a DSX-1 in software, tell me about it, so I can
invest in them?

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_
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [asterisk-users] Voicemail MWI

2006-10-09 Thread Race Vanderdecken

I did write some code once that worked but was not used in the end. The
customer did a very clever and elaborate solution that did not involve
asterisk.

In the code when a call came in the local asterisk server would talk to
a server on a socket on the main voicemail machine. 

The local/client code would redirect the MWI counter code to read from
the main server socket instead of the local file system.  The server
side socket used similar asterisk code to count the messages on the
remote server instead of on the local asterisk server.

Then the normal asterisk code would trip the MWI light.

The server address and whether to use the remote voicemail server
counter were in the vm.conf file.

It could handle about 500 queries per second without much cpu time.

It was built back under the 1.2 or 1.1 code base, so it might not work
under the 1.4 code base(which I have not looked at yet.)

Race Vanderdecken
Code Tyrant
Somewhere near Asheville, NC.
 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Douglas
Garstang
Sent: Friday, October 06, 2006 1:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Voicemail MWI

I'd like to know if anyone has a suggested fix for this...

You have a 'cluster' of Asterisk servers that use DUNDi etc for
registration redundancy, finding other phones etc. You have a separate
Asterisk box for voicemail. For voicemail deposit/retrieval you trunk
the call over to the voicemail server. This all works fine. No issues
there.

What about MWI though? Your phones register with the cluster, not with
the voicemail server, and therefore the voicemail server has no
knowledge of where the phones are and therefore cannot send out SIP
NOTIFY messages to phones.

This is a general architectural problem with Asterisk. Has anyone solved
it? Are the developers working on fixing problems like this for 1.6?

Doug.
___
--Bandwidth and Colocation provided by 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 list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] isdn cross-over ...

On Mon, Oct 09, 2006 at 10:55:51AM -0500, Ejay Hire wrote:
 Hi.  A cross-over cable won't work, the isdn network provides signalling
 and adressing functions.
 
 When I was studying for my CCIE, an ISDN simulator cost an arm-and-a-leg,
 around $1k used from ebay.

They appear to have come down:

http://search.ebay.com/isdn-bri-simulator_W0QQfrppZ50QQfsopZ1QQmaxrecordsreturnedZ300

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_
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] isdn cross-over ...

2006-10-09 Thread Zoa



Any card using this chip:
http://www.colognechip.com/isdn/controllers/frame-hfc-s-usb.htm (=usb 1 
port doing TE and NT mode)


This would be the cheapest way to test them, hook the 1 port up to the 
asterisk server and call an echo application, then try the same with the 
second one.

I dont know any 1 port cards that do NT mode.

Zoa

Ejay Hire wrote:

Hi.  A cross-over cable won't work, the isdn network provides signalling
and adressing functions.

When I was studying for my CCIE, an ISDN simulator cost an arm-and-a-leg,
around $1k used from ebay.

-ejay 


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Cesc
Sent: Monday, October 09, 2006 10:25 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] isdn cross-over ...

Hi,

I am sure this is a bit off topic, but maybe the people here have the
knowledge.
Quick question: I have two ISDN S/T phones. What is the quickest way to
test them (call one from the other)? can i make an isdn cross-over cable,
taking the correct pinning, of course? What i need is to avoid the need for
an NT connection (via a PBX).

If the above is not possible ... where can I buy a cheap, small, simple ISDN
PBX with at least two NT ports, so that i can connect my two phones and call
each other?

Tks!

Cesc
___
--Bandwidth and Colocation provided by 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 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 list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Range Operator

How can I check a number is within a specified range in the dialplan? What's 
the greater than operator? How would I use a combination of greater than and 
less than in conjection with GotoIf()?

The following seems to break the dialplan. I need to check callerid is _5XXX.

_X./_5XXX,1,Set(CALLERID(number)=5551212)
_X./_5XXX,n,NoOp(Dialplan dies before here)

Presumably it's because we just changed the callerid number and the dialplan 
now has nowhere to go.

Doug.
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] snom 360: how to make record button working ?

2006-10-09 Thread Mojo with Horan Company, LLC


Jay R. Ashworth wrote:
 So, you're suggesting that the FXO channel driver generates outbound
 DTMF under the command of (eventually) the phone set?  That would be
 nice.

Yes, that _would_ be nice.  Are you suggesting that that's not what's 
happening?  I'm not sure I gather your meaning, or I could be 
incorrectly discerning sarcasm.  I tried to disclaim my ignorance _and_ 
answer Remco's concern regarding DTMF reaching remote IVRs.


When I press the monitor sequence on my phone, the remote party doesn't 
hear anything, not even a crackle.  Same with the blind transfer sequence.


Moj


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Polycom 601 Expansion Module: Light the LEDs???

2006-10-09 Thread Faris Raouf


Doug wrote:

Hey Folks,

Been wrestling with the 601 and the expansion module.  Finally
figured out how to populate both with speed dial entries.  Also
hints are showing in Asterisk with the show hints command.

But how do I get the LEDs to light when one of these other
extensions is either off-hook, or ringing.

Reading the 'Net and Polycom's documentation doesn't give
a clear solution.  Is there a genius out there who has this
working??

Please help!!!



I can't help with the extension module, but on the phone itself...I 
can't quite remember exactly what you do but the trick is NOT to have 
lines programmed for all the line key buttons on the phone itself. 
Any free line key buttons will then get populated by the speed dials, 
and the respective LEDs will show the status of those speed dials 
(assuming the corresponding hints are correctly configured in asterisk)


You also have to enable to buddy feature on the phone itself using the 
XML config file. I think that part at least is documented somewhere.


Search the mailing list for polycom buddy or polycom hints or 
similar and you will find more detailed instructions and a cry for 
similar help from me six months to a year or so ago.


My problem was that I had defined all the line keys as lines, and 
freeing up those solved the issue.


Faris.


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Re: connecting multiple servers with iax - authentication fails

2006-10-09 Thread Benko


e.g. Server1 has peer/user entries for Server2 and Server3(in this
order), Server2 now tries to call Server1, but is asked for the
credentials of Server3(Because Server3 is the last entry in iax.conf),
which doesn't work of course.


i beg your pardon, actually the description of the attached debug goes
like this:

e.g. Server2 has peer/user entries for Server1 and Server3(in this
order), Server1 now tries to call Server2, but is asked for the
credentials of Server3(Because Server3 is the last entry in iax.conf
of Server2)...
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Number Range

My extensions.conf has:

exten = _X.,1,GotoIf([${CALLERID(number)}  5000  ${CALLERID(number)} 
 5999]?true:false)

and the dialplan on execution evaluates it as

-- vmtest1*CLI
-- Executing GotoIf(SIP/3254101-081e2820, [3254101  5000]  [3254101  
5999]?true:false) in new stack
-- Goto (btck_CallStart,5551212,2)
-- Executing NoOp(SIP/3254101-081e2820, True) in new stack:

Obviously I have something wrong. 3254101 is NOT between 5000 and 5999.

Doug.



 
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Re: PRI issues


Jay R. Ashworth wrote:

On Mon, Oct 09, 2006 at 12:01:02PM -0400, Doug Lytle wrote:
  
Yes, integrated CSU/DSU in software.  The tools provided show now errors 
on the line.



DSU maybe.

But if someone's figured out how to strip 130VDC off a copper pair and
convert a T-span to a DSX-1 in software, tell me about it, so I can
invest in them?

Cheers,
-- jra
  

From the A102 spec sheet:

   * Available as *Single *T1/E1 port (A101) or *Dual *T1/E1 port
 (A102) with daughterboard (as shown in photograph).
   * Intelligent hardware: Downloadable Field Programmable Gate Array
 programming with multiple operating modes:
   * T1/E1 and fractional T1/E1, single channel HDLC per line
   * Power: 520mA at +5v
   * PCI 32 bit (5v and 64 bit (3.3v) compatible.
   * Temperature range: 0 - 45C.
   * All set-up and configuration is in software or by machine BIOS.
   * DSU/CSU set up entirely in software.
   * Line decoding: HDB3, AMI, B8ZS.
   * Framing: CRC4, non-CRC4, ESF, D4.
   * Clocking mode: Normal , Master.
   * Software controlled DSU/CSU test modes.
   * Remote monitoring of card and CSU/DSU operation.
   * Dimensions: 2U Form factor: 120mm x 55 mm.



--

Ben Franklin quote:

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


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Lots and lots of log files

2006-10-09 Thread Tzafrir Cohen

On Mon, Oct 09, 2006 at 10:52:52AM -0400, J. Oquendo wrote:
 Ejay Hire wrote:
 Hello all, and good morning
 
 In my /var/log/asterisk directory I have 492,018 log files, most of which
 are empty.
 event_log.XXX queue_log.XXX messages.XXX where XXX is an integer.
 
 I removed them all and restarted asterisk a few days ago, but they came
 back.
 
 This morning I turned off event and queue logging, but I would prefer to
 have the messages log.  I didn't see an entry for theis in logrotate.
 
 The root issue behind this is that I get a message about a signal that the
 log files are too big, and asterisk stops working.  None of my log files 
 are
   
 1mb though.  Restarting asterisk fixes it.  
 
 
 Thanks,
 Ejay
 
 
 ... System information 
 
 [EMAIL PROTECTED] asterisk]# du -h /var/log/asterisk
 1.3G.
 840K./cdr-csv
 956K./cdr-custom
 
 Asterisk 1.2.11 
 Linux version 2.6.9-42.0.2.ELsmp (gcc version 3.4.6 20060404 (Red Hat
 3.4.6-3)) #1 SMP
 Asterisk 1.2.11 built by buildcentos @ localhost.localdomain on a i686
 running Linux 
 localhost*CLI logger show channels 
 Channel Type StatusConfiguration
 ---  ---
 /var/log/asterisk/messages  File Enabled- Warning Notice
 Error 
 Console  Enabled- Warning Notice
 Error 
 
 ___
 --Bandwidth and Colocation provided by Easynews.com --
 
 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
 
   
 Edit /etc/asterisk/logger.conf then asterisk -rx logger reload or in 
 cron the dirty way:
 
 0 * * * *  ls -ltha /var/log/asterisk/|awk '{print $5,$9}'|grep 0 
 |grep -v [1-9]|xargs rm -rf

Huh?

Is it supposed to pick files in the csv dirs?

-- 
Tzafrir Cohen sip:[EMAIL PROTECTED]
icq#16849755  iax:[EMAIL PROTECTED]
+972-50-7952406  jabber:[EMAIL PROTECTED]
[EMAIL PROTECTED] http://www.xorcom.com
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] [EMAIL PROTECTED] problems

2006-10-09 Thread Edward0219




Hi, Alex...thank you for your response

How do you do that, at the Portal or using a dos command?

Thanks again.

Ed
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Is there a way to collect dtmf digits during a call? (inband)

2006-10-09 Thread MF


Hi,

Does anyone knows if I can collect DTMF digits (inband) during a bridged 
call   (E1 to E1),?

I see the DTMF tones on the debug file but does not activate the dialplan.

My problem is,  I need to signal from the second E1 to bridge the call 
to another E1 (a third one),  if I use the transfer capability, (dialing 
#)  it works fine with one or two calls but does not work fine with more 
than 5 calls,  (don't know what's wrong yet and why,  probably delays or 
something).
So I'm thinking on implementing another way to transfer by detecting an 
inband tone and performing a transfer comand on the dial plan,


any input will be greatly appreciated

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Range Operator


Douglas Garstang wrote:

How can I check a number is within a specified range in the dialplan? What's 
the greater than operator? How would I use a combination of greater than and 
less than in conjection with GotoIf()?

  


I'm not really sure what _X./_5XXX does.  I thought _X. would match 
anything that would start with a digit. What does the /_5XXX do on that 
same matching string?


Anyway, this is what I have setup to limit the number of calls going to 
our fax using GotoIf()


Doug

; 
; Front office FAX
; 

exten = 734261,1,Set(GROUP()=Office_Fax_Max)
exten = 734261,n,NoOP(Active Calls: ${GROUP_COUNT(Office_Fax_Max)})
exten = 734261,n,GotoIf($[ ${GROUP_COUNT(Office_Fax_Max)}  1 
]?734261,100)

exten = 734261,n,Set(CALLERID(name)=${DB(dnis/${CALLERIDNUM})})
exten = 734261,n,Dial(ZAP/37)
exten = 734261,100,Set(PRI_CAUSE=17)
exten = 734261,101,Hangup()

Doug

--

Ben Franklin quote:

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


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] External domain

2006-10-09 Thread Josemar Lohn


Hello:

There is a way to make asterisk reach external domains by SIP?
I am on domain example.org and want to talk to peers on any other domains.
But I don't know how to configure it on extensions.conf.
Or do I have to do some modifications on SIP.conf to create a new context?

Thanks

Josemar Lohn
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] GotoIfTime - much slowdown with 90 conditions?

2006-10-09 Thread Mike Dent


Hi,

I wonder if anybody can share their experience of this. I am designing
a system with 90
GotoIfTime conditions to check through for a match.

Bascially each month day will be split in to 3 time ranges and a month
has 31 days in, so this gives a possible 90+ combinations, nearer the
end of the months the dialplan will have to traverse through nearly
all of these.

How much load/slowdown (if any) can I expect from this? Assuming a low
end Celeron 3.0Ghz box with 512mb memory.

I'm guessing myself it won't be a big deal but I just want to check
before I commit too much.

thanks
Mike
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Number Range

2006-10-09 Thread Ira


At 09:59 AM 10/9/2006, you wrote:
seems to be missing a dollar sign.

exten = _X.,1,GotoIf($[${CALLERID(number)}  5000  
${CALLERID(number)}  5999]?true:false)


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Number Range


Douglas Garstang wrote:

My extensions.conf has:

exten = _X.,1,GotoIf([${CALLERID(number)}  5000  ${CALLERID(number)} 
 5999]?true:false)

and the dialplan on execution evaluates it as

-- vmtest1*CLI
-- Executing GotoIf(SIP/3254101-081e2820, [3254101  5000]  [3254101  
5999]?true:false) in new stack
-- Goto (btck_CallStart,5551212,2)
-- Executing NoOp(SIP/3254101-081e2820, True) in new stack:
  

**According to the entry on the wiki, it should be:


exten = _X.,1,GotoIf([${CALLERID(number)}  5000  ${CALLERID(number)}  
5999]?true:false)


Doug

--

Ben Franklin quote:

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


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Lots and lots of log files

2006-10-09 Thread J. Oquendo


Tzafrir Cohen wrote:

On Mon, Oct 09, 2006 at 10:52:52AM -0400, J. Oquendo wrote:
  

Ejay Hire wrote:


Hello all, and good morning

In my /var/log/asterisk directory I have 492,018 log files, most of which
are empty.
event_log.XXX queue_log.XXX messages.XXX where XXX is an integer.

I removed them all and restarted asterisk a few days ago, but they came
back.

This morning I turned off event and queue logging, but I would prefer to
have the messages log.  I didn't see an entry for theis in logrotate.

The root issue behind this is that I get a message about a signal that the
log files are too big, and asterisk stops working.  None of my log files 
are
 
  
1mb though.  Restarting asterisk fixes it.  
   


Thanks,
Ejay


... System information 


[EMAIL PROTECTED] asterisk]# du -h /var/log/asterisk
1.3G.
840K./cdr-csv
956K./cdr-custom

Asterisk 1.2.11 
Linux version 2.6.9-42.0.2.ELsmp (gcc version 3.4.6 20060404 (Red Hat

3.4.6-3)) #1 SMP
Asterisk 1.2.11 built by buildcentos @ localhost.localdomain on a i686
running Linux 
localhost*CLI logger show channels 
Channel Type StatusConfiguration

---  ---
/var/log/asterisk/messages  File Enabled- Warning Notice
Error 
   Console  Enabled- Warning Notice
Error 


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

 
  
Edit /etc/asterisk/logger.conf then asterisk -rx logger reload or in 
cron the dirty way:


0 * * * *  ls -ltha /var/log/asterisk/|awk '{print $5,$9}'|grep 0 
|grep -v [1-9]|xargs rm -rf



Huh?

Is it supposed to pick files in the csv dirs?

  


No: His original post:


In my /var/log/asterisk directory I have 492,018 log files, most of which
are empty.
event_log.XXX queue_log.XXX messages.XXX where XXX is an integer.




--

J. Oquendo
http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x1383A743
sil . infiltrated @ net http://www.infiltrated.net 


The happiness of society is the end of government.
John Adams



smime.p7s
Description: S/MIME Cryptographic Signature
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] [EMAIL PROTECTED] problems

2006-10-09 Thread Alex Robar

Within FreePBX, under the tools menu, there is an Asterisk CLI module. Select that one and type sip show peers when one of the phones isn't working. Paste the output that you get back here.Alex
On 10/9/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:





Hi, Alex...thank you for your response

How do you do that, at the Portal or using a dos command?

Thanks again.

Ed

___--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing listTo UNSUBSCRIBE or update options visit:  http://lists.digium.com/mailman/listinfo/asterisk-users
-- Alex Robar[EMAIL PROTECTED]
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] ftp server

On Mon, Oct 09, 2006 at 12:40:03PM -0400, Noah Miller wrote:
 I use vsftpd and I'm using the default PlcmSpIp username just
 fine. :) Essentially, I configured PlcmSpIp as a Linux user and I'm
 serving it out by using personalised FTP home directories in vsftp
 and then chrooting per user. Works like a charm and no phone
 configuration is required.
 
 Quite right.  I'm blaming the inadequacies of my OS on vsftpd.  vsftpd
 just uses your OS user accounts.  On the Tao linux box that I had it
 installed on, you couldn't do capitals in user account names.  My bad.

And, I would speculate, it wasn't the OS, it was whatever layered
management tool you had on top; Linux has never cared whether login
names had caps in them.

Unix, in general, has always had *login programs* which would note an
*all caps* login name, and turn on case folding with an obscure
protocol for signifying real caps, but I strongly suspect current
versions don't do that any more: it was an ASR-33 era thing, for
terminals which couldn't do lowercase (gotta spell GOD'S name
right[1]), and how many of *those* do you see anymore?

Cheers
-- jra

[1]http://www.elsewhere.org/jargon/html/entry/Great-Runes.html
-- 
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_
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Cisco 7970 Unbootable After FW Upgrade

2006-10-09 Thread Jeremiah Millay

I tried upgrading a used Cisco 7970 from the image it shipped with to 
SIP 8.0.2 SR1 but didn't have any luck so I followed the procedures to 
do a factory reset on the phone. The phone is grabbing an IP and 
attempting to grab my term70.default.loads file but not moving any 
further. The phone screen no longer shows anything. Has anyone else had 
the same problem? All of my other 7970s upgraded with no problems. Since 
our 7970s are all used I couldn't tell what image they shipped with or 
what the default is. I've tried grabbing a much older SCCP image version 
and placing that image in my tftp server hoping it would like that but 
still no success.
Does anyone have any suggestions as to how I can at least get this phone 
to boot some default SCCP image? As of right now this phone is 
unuseable. I get the feeling that if I can figure out what the default 
image is for one of these I may be able to get it to boot to that.

Thanks!
Jeremiah
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] VOIP with PSTN backup

2006-10-09 Thread Brian Candler

On Mon, Oct 09, 2006 at 11:01:30PM +0700, Peter Lindquist wrote:
 Brian,
 
 Take a look at www.intertex.se I believe they have what you are looking for.

Thanks - that one is on my shopping list already :-)

The unit is limited to 5 users, and they are very coy about letting you know
how much the extra licences are. Their e-commerce website detects whether
you are coming in via one of their routers, and if not, it hides all the
upgrade prices :-(

They do publish the price of the softswitch licence though - $500 (ouch).
Just have to hope that I can do all I need without that...

Cheers,

Brian.
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] snom 360: how to make record button working ?

On Mon, Oct 09, 2006 at 08:50:51AM -0800, Mojo with Horan  Company, LLC wrote:
 Jay R. Ashworth wrote:
  So, you're suggesting that the FXO channel driver generates outbound
  DTMF under the command of (eventually) the phone set?  That would be
  nice.
 
 Yes, that _would_ be nice.  Are you suggesting that that's not what's 
 happening?  I'm not sure I gather your meaning, or I could be 
 incorrectly discerning sarcasm.  I tried to disclaim my ignorance _and_ 
 answer Remco's concern regarding DTMF reaching remote IVRs.

You were incorrectly discerning sarcasm.  Sorry.

I was hoping it worked as described.

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_
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] isdn cross-over ...

2006-10-09 Thread Brian Candler

On Mon, Oct 09, 2006 at 10:55:51AM -0500, Ejay Hire wrote:
 Hi.  A cross-over cable won't work, the isdn network provides signalling
 and adressing functions.
 
 When I was studying for my CCIE, an ISDN simulator cost an arm-and-a-leg,
 around $1k used from ebay.

Couldn't you put two ISDN phones on the same NT though? (one B-channel each)

One way to get an NT is to rent an ISDN BRI circuit from your PTT, although
it's not portable :-)

I haven't tested one yet, but you could have a look at the Fritz!Box Fon WLAN:
http://www.avm.de/en/

This has an ISDN S0 NT interface for plugging in an ISDN phone. If you can't
plug both phones onto it, then I guess you just buy two and point them at a
convenient local SIP server. It's cheap enough to take a punt on.

Regards,

Brian.
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Cisco 7970 Unbootable After FW Upgrade

2006-10-09 Thread Greg Oliver

When you do a factory reset on a 41/61/70/71, it actually deletes ALL of
the firmware except the bootloader from the phone.  You would have to
have all of the 70s firmware files that come with them in order to boot
them.  The term70.default.loads tells the phone what version of software
to tftp.  Does the phone actually try to receive the file from your tftp
server?  

What does your tftp log say?

-Greg

On Mon, 2006-10-09 at 13:23 -0500, Jeremiah Millay wrote:
 I tried upgrading a used Cisco 7970 from the image it shipped with to 
 SIP 8.0.2 SR1 but didn't have any luck so I followed the procedures to 
 do a factory reset on the phone. The phone is grabbing an IP and 
 attempting to grab my term70.default.loads file but not moving any 
 further. The phone screen no longer shows anything. Has anyone else had 
 the same problem? All of my other 7970s upgraded with no problems. Since 
 our 7970s are all used I couldn't tell what image they shipped with or 
 what the default is. I've tried grabbing a much older SCCP image version 
 and placing that image in my tftp server hoping it would like that but 
 still no success.
 Does anyone have any suggestions as to how I can at least get this phone 
 to boot some default SCCP image? As of right now this phone is 
 unuseable. I get the feeling that if I can figure out what the default 
 image is for one of these I may be able to get it to boot to that.
 Thanks!
 Jeremiah
 ___
 --Bandwidth and Colocation provided by 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 list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Range Operator

2006-10-09 Thread Eric \ManxPower\ Wieling




Douglas Garstang wrote:
How can I check a number is within a specified range in the dialplan? 
What's the greater than operator? How would I use a combination of 
greater than and less than in conjection with GotoIf()?


README.variables should give you what you need.  It's in 
/path/to/src/asterisk/docs/


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] problem in ooh323

2006-10-09 Thread Andre Luiz Martins Rodrigues

Hello everbody,

I have a problem, I installed ooh323 in mine * but when I try to dial appears like itself be contacting the gatekeeper but nothing happens. follows my ooh323.conf

[general]
port=1720
bindaddr=0.0.0.0
gateway=no
h323id=XXX
e164=1234567890
callerid=h323id
gatekeeper = ipgatekeeper
context=voip-h323
disallow=all ;Note order of disallow/allow is important.allow=g729allow=gsmallow=ulawdtmfmode=rfc2833

in debug mode i have 

--- ooh323_request - data 55XX format 0x4 (ulaw)--- find_peer+++ find_peer+++ ooh323_request--- ooh323_call- 55XX+++ ooh323_call -- Called 55XX


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk RT on Disk On Module Performance and Durability

2006-10-09 Thread Erick Perez

Jeremy, Cohen, Kris, thanks to all of you.

Indeed after reading the Sandisk paper it shed a lot of light on this matter. The whole idea is to have a large scale system with no moving parts (we call a large system something with250 users, at least down here ;-) )

the whole idea is for a customer that needs an IVR in 4 languages with autoattendant, extensive CDR and plotted usage patterns as well as voicemail. Voicemail will be used *a lot*, probably about one thousand voicemails per day and the customer does not want VM-to-Email (God knows why!).

Oh, and the whole idea of the database is because the developers are working in an AJAX based interface that does the asterisk config/plotting/vm/day-to-day stuff with ARA, so a db is needed.
I started learning asterisk with flat files...it works for me...but hey...times are changing.

Who knows, maybe the whole thing can be fitted in ram (except for the vm part)...we'll see. I had to ask anyway, but i don't like Dbs eitherit adds and extra breakup layer (maybe Im kind of outdated).

Smaller iPBXs will definitely be CF and RAM based and I, at least, will force VMtoEmail and do all the processing in RAM.

Again,

Thanks to all of you.
P.D. I will later follow this thread with the full working configs that will take place at user premises. And for the sake of the test. I will try to kill a sandisk USB with the full config.

On 10/8/06, Kristian Kielhofner [EMAIL PROTECTED] wrote:
Jeremy McNamara wrote: Tzafrir Cohen wrote: H, I'm not sure that this is exactly the data you're after.
You're looking for the ammounts of writes for the disk block that gets the most writes. E.g: for a standard ext3 filesystem, the journal area would probably
have very frequent writes, whereas most of the system would remain mostly unchanged. Again, if the embedded system is setup properly, there is NO writing to the flash during normal operations, thus the device won't be killed by
its alleged 2 million write limitation. Kris and I had a quick discussion on this topic, off-list, and his original flash-based device is still in constant operation after 2 years and I have flash modules that I purposely tried to kill with writes. It
took significant effort to start causing error situations, which were very easily detected before the system would become unusable. Erick, you should focus on having a quick action restoration plan and
extra DOMs always readily available.Then when a failure situation is detected, you can react very quickly. Jeremy McNamaraJeremy, Erick - I have always pointed to this SanDisk whitepaper:
http://www.sandisk.com/Assets/File/OEM/WhitePapersAndBrochures/RS-MMC/WPaperWearLevelv1.0.pdf While it specifically discusses their industrial line of CF cards, it
is pretty obvious that flash can, and often does, last much longer thanother components in a system when properly implemented.You will noticethat the SHORTEST expected life of a CF card in their test scenarios was
over 70 years!How long is your power supply going to last?Even ifthe consumer level cards had 1/10 the life expectancy, that is stillseven years.I expect to get at least that from my original AstLinux
system.It's been two so far, I'll let you know how it is doing inanother five years :). JFFS (and similar FSs) are not appropriate for CF cards or DOMs.Theyare meant to be used directly on flash memory and do their own wear
leveling and in some cases, compression.All kinds of commercialdevices use JFFS2.If you are using a CF or DOM with Linux, ext2 is thebest FS to use.CF cards and DOMs use their own wear leveling, so none
is required in the operating system or file system.CF cards and DOMshide wear leveling from you and expose themselves as an ordinary IDE device. I echo Jeremy's conclusions.With a properly designed operating
system, decent flash memory, and a reasonable usage pattern, I can tellyou (with a great amount of certainty) that in most situations, CF cardswill outlast just about any hard drive (even SCSI) when used 24/7.
These days, it really is pretty tough to trash flash. However, if you are running a MySQL cluster or something with several,multi-gigabyte databases, no type of flash memory will last very long! :)
To get back to answering your question, I HIGHLY recommend that youavoid MySQL and realtime on your box with a DOM.Nothing against either(MySQL or Realtime), but they will probably make your device more
complicated than it needs to be while substantially shortening the lifeof your DOM.If you absolutely have to use MySQL, you might have betterluck using a MySQL storage engine that uses fewer writes than InnoDB,
but I am no expert on that.--Kristian Kielhofner___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing list
To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users
-- Erick PerezPanama

Re: [asterisk-users] [EMAIL PROTECTED] problems

2006-10-09 Thread Edward0219




Alex...I do not have FreePBX. What I have is this:

http://70.89.124.237/


Ed
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [asterisk-users] Asterisk RT on Disk On Module Performance andDurability

I'm
just going to jump in here, and ask a stoopid question.

How
could you possibly write a multi-user front end in AJAXwithout using a
database backend like MySQL?

Doug.

-Original Message-From: Erick Perez
[mailto:[EMAIL PROTECTED]Sent: Monday, October 09, 2006 1:58
PMTo: Asterisk Users Mailing List - Non-Commercial
DiscussionSubject: Re: [asterisk-users] Asterisk RT on Disk On
Module Performance andDurability
Jeremy, Cohen, Kris, thanks to all of you.

Indeed after reading the Sandisk paper it shed a lot of light on this
matter. The whole idea is to have a large scale system with no moving parts
(we call a large system something with250 users, at least down here
;-) )

the whole idea is for a customer that needs an IVR in 4 languages with
autoattendant, extensive CDR and plotted usage patterns as well as voicemail.
Voicemail will be used *a lot*, probably about one thousand voicemails per day
and the customer does not want VM-to-Email (God knows why!).

Oh, and the whole idea of the database is because the developers are
working in an AJAX based interface that does the asterisk
config/plotting/vm/day-to-day stuff with ARA, so a db is needed.
I started learning asterisk with flat files...it works for me...but
hey...times are changing.

Who knows, maybe the whole thing can be fitted in ram (except for the vm
part)...we'll see. I had to ask anyway, but i don't like Dbs eitherit adds
and extra breakup layer (maybe Im kind of outdated).

Smaller iPBXs will definitely be CF and RAM based and I, at least, will
force VMtoEmail and do all the processing in RAM.

Again,

Thanks to all of you.
P.D. I will later follow this thread with the full working configs
that will take place at user premises. And for the sake of the test. I will
try to kill a sandisk USB with the full config.

On 10/8/06, Kristian
Kielhofner [EMAIL PROTECTED]
wrote:
Jeremy
McNamara wrote: Tzafrir Cohen wrote: H,
I'm not sure that this is exactly the data you're after.
You're looking for the ammounts of writes for the
disk block that gets the most writes.
E.g: for a standard ext3 filesystem, the journal area would
probably have very frequent writes, whereas most of the system
would remain mostly unchanged. Again, if
the embedded system is setup properly, there is NO writing to the
flash during normal operations, thus the device won't be killed by
its alleged 2 million write limitation. Kris and I had a
quick discussion on this topic, off-list, and his original
flash-based device is still in constant operation after 2 years and
I have flash modules that I purposely tried to kill with writes. It
took significant effort to start causing error situations, which
were very easily detected before the system would become
unusable. Erick, you should focus on having a quick action
restoration plan and extra DOMs always readily
available.Then when a failure situation is detected, you
can react very quickly. Jeremy
McNamaraJeremy, Erick -
I have always pointed to this SanDisk whitepaper: http://www.sandisk.com/Assets/File/OEM/WhitePapersAndBrochures/RS-MMC/WPaperWearLevelv1.0.pdf
While it specifically discusses their industrial line of CF cards, it is
pretty obvious that flash can, and often does, last much longer
thanother components in a system when properly
implemented.You will noticethat the SHORTEST expected life
of a CF card in their test scenarios was over 70 years!How
long is your power supply going to last?Even ifthe consumer
level cards had 1/10 the life expectancy, that is stillseven
years.I expect to get at least that from my original
AstLinuxsystem.It's been two so far, I'll let you know how
it is doing inanother five years
:). JFFS (and similar FSs) are
not appropriate for CF cards or DOMs.Theyare meant to be
used directly on flash memory and do their own wear leveling and in some
cases, compression.All kinds of commercialdevices use
JFFS2.If you are using a CF or DOM with Linux, ext2 is
thebest FS to use.CF cards and DOMs use their own wear
leveling, so noneis required in the operating system or file
system.CF cards and DOMshide wear leveling from you and
expose themselves as an ordinary IDE
device. I echo Jeremy's
conclusions.With a properly designed operating system,
decent flash memory, and a reasonable usage pattern, I can tellyou (with
a great amount of certainty) that in most situations, CF cardswill
outlast just about any hard drive (even SCSI) when used 24/7. These
days, it really is pretty tough to trash
flash. However, if you are
running a MySQL cluster or something with several,multi-gigabyte
databases, no type of flash memory will last very long!
:) To

[asterisk-users] how to play pre-recorded file in meetme conference

2006-10-09 Thread Barry D. Hassler

Hey folks, Is it possible to play a pre-recorded file in a meetme
conference? That is, I'd like to get everyone into a conference, then
somehow play a previously recorded file (in this case, a podcast). This
isn't for individuals to call into to listen to the cast, but for it to
be played simultaneously for all in the conference. 

This would be handy for me!

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [asterisk-users] Asterisk RT on Disk On Module Performance andDurability

2006-10-09 Thread Aaron Daniel

Very very carefully ;) I'm thinking pizza, and maybe some red-bull...
very little time for sleep

Aaron

On Mon, 2006-10-09 at 14:19 -0600, Douglas Garstang wrote:
I'm just going to jump in here, and ask a stoopid question.

How could you possibly write a multi-user front end in AJAX without
using a database backend like MySQL?

Doug.
-Original Message-
From: Erick Perez [mailto:[EMAIL PROTECTED]
Sent: Monday, October 09, 2006 1:58 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk RT on Disk On Module
Performance andDurability

Jeremy, Cohen, Kris, thanks to all of you.

Indeed after reading the Sandisk paper it shed a lot of light
on this matter. The whole idea is to have a large scale system
with no moving parts (we call a large system something
with 250 users, at least down here ;-) )

the whole idea is for a customer that needs an IVR in 4
languages with autoattendant, extensive CDR and plotted usage
patterns as well as voicemail. Voicemail will be used *a lot*,
probably about one thousand voicemails per day and the
customer does not want VM-to-Email (God knows why!).

Oh, and the whole idea of the database is because the
developers are working in an AJAX based interface that does
the asterisk config/plotting/vm/day-to-day stuff with ARA, so
a db is needed.
I started learning asterisk with flat files...it works for
me...but hey...times are changing.

Who knows, maybe the whole thing can be fitted in ram (except
for the vm part)...we'll see. I had to ask anyway, but i don't
like Dbs eitherit adds and extra breakup layer (maybe Im
kind of outdated).

Smaller iPBXs will definitely be CF and RAM based and I, at
least, will force VMtoEmail and do all the processing in RAM.

Again,

Thanks to all of you.

P.D. I will later follow this thread with the full working
configs that will take place at user premises. And for the
sake of the test. I will try to kill a sandisk USB with the
full config.

On 10/8/06, Kristian Kielhofner [EMAIL PROTECTED] wrote:
Jeremy McNamara wrote:
Tzafrir Cohen wrote:
H, I'm not sure that this is exactly the data
you're after.

You're looking for the ammounts of writes for the
disk block that gets
the most writes.

E.g: for a standard ext3 filesystem, the journal
area would probably
have very frequent writes, whereas most of the
system would remain
mostly unchanged.

Again, if the embedded system is setup properly,
there is NO writing to
the flash during normal operations, thus the device
won't be killed by
its alleged 2 million write limitation.

Kris and I had a quick discussion on this topic,
off-list, and his
original flash-based device is still in constant
operation after 2 years
and I have flash modules that I purposely tried to
kill with writes. It
took significant effort to start causing error
situations, which were
very easily detected before the system would become
unusable.

Erick, you should focus on having a quick action
restoration plan and
extra DOMs always readily available. Then when a
failure situation is
detected, you can react very quickly.

Jeremy McNamara

Jeremy, Erick -

I have always pointed to this SanDisk
whitepaper:

http://www.sandisk.com/Assets/File/OEM/WhitePapersAndBrochures/RS-MMC/WPaperWearLevelv1.0.pdf

While it specifically discusses their
industrial line of CF cards, it
is pretty obvious that flash can, and often does, last
much longer than
other components in a system when properly
implemented. You will notice
that the

RE: [asterisk-users] Asterisk RT on Disk On Module PerformanceandDurability

I'd be curious to know what you come up with, because we're using MySQL, and 
I'd rather not!

 -Original Message-
 From: Aaron Daniel [mailto:[EMAIL PROTECTED]
 Sent: Monday, October 09, 2006 2:48 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: RE: [asterisk-users] Asterisk RT on Disk On Module
 PerformanceandDurability

 Very very carefully ;)  I'm thinking pizza, and maybe some red-bull...
 very little time for sleep

 Aaron

 On Mon, 2006-10-09 at 14:19 -0600, Douglas Garstang wrote:
  I'm just going to jump in here, and ask a stoopid question.

  How could you possibly write a multi-user front end in AJAX without
  using a database backend like MySQL?

  Doug.
  -Original Message-
  From: Erick Perez [mailto:[EMAIL PROTECTED]
  Sent: Monday, October 09, 2006 1:58 PM
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  Subject: Re: [asterisk-users] Asterisk RT on Disk On Module
  Performance andDurability

  Jeremy, Cohen, Kris, thanks to all of you.

  Indeed after reading the Sandisk paper it shed a 
 lot of light
  on this matter. The whole idea is to have a large 
 scale system
  with no moving parts (we call a large system something
  with 250 users, at least down here ;-)  ) 

  the whole idea is for a customer that needs an IVR in 4
  languages with autoattendant, extensive CDR and 
 plotted usage
  patterns as well as voicemail. Voicemail will be 
 used *a lot*,
  probably about one thousand voicemails per day and the
  customer does not want VM-to-Email (God knows why!). 

  Oh, and the whole idea of the database is because the
  developers are working in an AJAX based interface that does
  the asterisk config/plotting/vm/day-to-day stuff 
 with ARA, so
  a db is needed.
  I started learning asterisk with flat files...it works for
  me...but hey...times are changing.

  Who knows, maybe the whole thing can be fitted in 
 ram (except
  for the vm part)...we'll see. I had to ask anyway, 
 but i don't
  like Dbs eitherit adds and extra breakup layer (maybe Im
  kind of outdated).

  Smaller iPBXs will definitely be CF and RAM based and I, at
  least, will force VMtoEmail and do all the 
 processing in RAM.

  Again,

  Thanks to all of you.

  P.D. I will later follow this thread with the full working
  configs that will take place at user premises. And for the
  sake of the test. I will try to kill a sandisk USB with the
  full config.

  On 10/8/06, Kristian Kielhofner [EMAIL PROTECTED] wrote: 
  Jeremy McNamara wrote:
   Tzafrir Cohen wrote:
 H, I'm not sure that this is 
 exactly the data
  you're after. 

   You're looking for the ammounts of writes for the
  disk block that gets
   the most writes.

   E.g: for a standard ext3 filesystem, the journal
  area would probably
   have very frequent writes, whereas most of the
  system would remain
   mostly unchanged.

   Again, if the embedded system is setup properly,
  there is NO writing to
   the flash during normal operations, thus 
 the device
  won't be killed by 
   its alleged 2 million write limitation.

   Kris and I had a quick discussion on this topic,
  off-list, and his
   original flash-based device is still in constant
  operation after 2 years
   and I have flash modules that I purposely tried to
  kill with writes. It 
   took significant effort to start causing error
  situations, which were
   very easily detected before the system 
 would become
  unusable.

   Erick, you should focus on having a quick action
  restoration plan and 
   extra DOMs always readily available.  Then when a
  failure situation is
   detected, you can react very quickly.

   Jeremy McNamara

  Jeremy, Erick -

 I have always pointed to this SanDisk
  whitepaper:

Re: [asterisk-users] password for vm users

2006-10-09 Thread stan ford

easy enough. thanks!Marco Mouta [EMAIL PROTECTED] wrote:  just set your initial password to be equal to vm-account number, and Voicemail application will do that for you and will request users to setup a new password!  On 10/9/06, stan ford [EMAIL PROTECTED] wrote:how does one force mandatory password change on login? and a period of time to pass before mandating a password change?im using trixbox so if you have that info. that would be good, if you have it for asterisk, im sure i could figure it out as well
 thx.  Yahoo! Messenger with Voice. Make PC-to-Phone Calls to the US (and 30+ countries) for 2¢/min or less.   ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Com os melhores cumprimentos,Marco Mouta
 ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users 
		Get your email and more, right on the  new Yahoo.com 
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Function ENUMLOOKUP




Just 
playing around with Enum. What's wrong with this in 
Asterisk?

exten = 
555,1,Set(foo=${ENUMLOOKUP(+16049586111)})

 -- Executing 
Set("SIP/3254101-081e8c58", "foo=") in new stack -- Executing 
NoOp("SIP/3254101-081e8c58", "") in new stack -- Executing Hangup("SIP/3254101-081e8c58", 
"") in new stack

... 
because dig resolves it That's the number e164.org has as a callerid 
readback on their website.

dig +short 
1.1.1.6.8.5.9.4.0.6.1.e164.org any100 10 "u" "E2U+ADDRESS" 
"!^.*$!ADDRESS:CN=Matthew Asham\;STREET=Eduard-Bodem-Gasse 
9\;L=Burnaby\;ST=BC\;C=Canada!" .100 10 "u" "E2U+SIP" 
"!^\\+16049586111$!sip:[EMAIL PROTECTED]" .

Doug.


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] password for vm users

2006-10-09 Thread stan ford

how about password strength? or remembering and not allowing password? or password duration?Marco Mouta [EMAIL PROTECTED] wrote:  just set your initial password to be equal to vm-account number, and Voicemail application will do that for you and will request users to setup a new password!  On 10/9/06, stan ford [EMAIL PROTECTED] wrote:how does one force mandatory password change on login? and a period of time to pass before mandating a password change?im using trixbox so if you have that info. that would be good, if you have it for
 asterisk, im sure i could figure it out as well thx.  Yahoo! Messenger with Voice. Make PC-to-Phone Calls to the US (and 30+ countries) for 2¢/min or less.   ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Com os melhores cumprimentos,Marco Mouta ___--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.___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] isdn cross-over ...

2006-10-09 Thread Cesc


Thanks for the link.The VConsole equipment seems to be a close match
to what i am looking for, just quite expensive ... but i hope that the
company i work for will pay for it :)

I see many options here. It came to my knowledge that we have a PSTN +
1 ISDN pbx ... a small quattrovox box. How can I connect the 2 phones
in that one ISDN? and, can i call one from the other in this fashion?

Cesc

On 10/9/06, Jay R. Ashworth [EMAIL PROTECTED] wrote:

On Mon, Oct 09, 2006 at 10:55:51AM -0500, Ejay Hire wrote:
 Hi.  A cross-over cable won't work, the isdn network provides signalling
 and adressing functions.

 When I was studying for my CCIE, an ISDN simulator cost an arm-and-a-leg,
 around $1k used from ebay.

They appear to have come down:

http://search.ebay.com/isdn-bri-simulator_W0QQfrppZ50QQfsopZ1QQmaxrecordsreturnedZ300

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_
___
--Bandwidth and Colocation provided by 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 list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] ftp server

2006-10-09 Thread Avi Miller



On 10/10/2006, at 2:10 AM, Noah Miller wrote:


Quite right.  I'm blaming the inadequacies of my OS on vsftpd.  vsftpd
just uses your OS user accounts.  On the Tao linux box that I had it
installed on, you couldn't do capitals in user account names.  My bad.


Which is weird, because I thought Tao was like CentOS: A basic  
rebrand of RHEL. And I use CentOS. :)


cYa,
Avi

--
National Manager - Special Projects

 Sydney / Melbourne / Canberra / Hobart / London /
  2/340 Gore StreetT: +61 (0) 3 9235 5400
  Fitzroy, VIC F: +61 (0) 3 9235 5444
  3065 W: http://www.squiz.net

.   Open Source - Own It - Squiz.net .. /




___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [asterisk-users] Function ENUMLOOKUP

2006-10-09 Thread Watkins, Bradley




Does that entry exist also in e164.arpa (the 
default)? Have you tried explicitly pointing it at e164.org 
instead?

FWIW, I see nothing in particular wrong about your usage, 
but make sure we're talking about the right trees here.

Regards,
- Brad

  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Douglas 
  GarstangSent: Monday, October 09, 2006 5:13 PMTo: 
  Asterisk Users Mailing List - Non-Commercial DiscussionSubject: 
  [asterisk-users] Function ENUMLOOKUP
  
  Just 
  playing around with Enum. What's wrong with this in 
  Asterisk?
  
  exten = 
  555,1,Set(foo=${ENUMLOOKUP(+16049586111)})
  
   -- Executing 
  Set("SIP/3254101-081e8c58", "foo=") in new stack -- Executing 
  NoOp("SIP/3254101-081e8c58", "") in new stack -- Executing Hangup("SIP/3254101-081e8c58", 
  "") in new stack
  
  ... 
  because dig resolves it That's the number e164.org has as a callerid 
  readback on their website.
  
  dig +short 
  1.1.1.6.8.5.9.4.0.6.1.e164.org any100 10 "u" "E2U+ADDRESS" 
  "!^.*$!ADDRESS:CN=Matthew Asham\;STREET=Eduard-Bodem-Gasse 
  9\;L=Burnaby\;ST=BC\;C=Canada!" .100 10 "u" "E2U+SIP" 
  "!^\\+16049586111$!sip:[EMAIL PROTECTED]" .
  
  Doug.
  
  The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. 
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk RT on Disk On Module Performance andDurability

2006-10-09 Thread Erick Perez

Douglas, Im just the asterisk guy. If they decide to write a cross-browser multi-tier interface in AJAX, assembly language or Pascal, that's up to them (the programmers). I will let them know what can/can't be done.

Thinking of that...15 years ago...the last time i used pascal.

On 10/9/06, Douglas Garstang [EMAIL PROTECTED] wrote:

I'm just going to jump in here, and ask a stoopid question.

How could you possibly write a multi-user front end in AJAXwithout using a database backend like MySQL?

Doug.

-Original Message-From: Erick Perez [mailto:
[EMAIL PROTECTED]]Sent: Monday, October 09, 2006 1:58 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [asterisk-users] Asterisk RT on Disk On Module Performance andDurability

Jeremy, Cohen, Kris, thanks to all of you.

Smaller iPBXs will definitely be CF and RAM based and I, at least, will force VMtoEmail and do all the processing in RAM.

Again,

On 10/8/06, Kristian Kielhofner [EMAIL PROTECTED]
wrote:
Jeremy McNamara wrote: Tzafrir Cohen wrote: H, I'm not sure that this is exactly the data you're after.
You're looking for the ammounts of writes for the disk block that gets the most writes. E.g: for a standard ext3 filesystem, the journal area would probably
have very frequent writes, whereas most of the system would remain mostly unchanged. Again, if the embedded system is setup properly, there is NO writing to the flash during normal operations, thus the device won't be killed by
its alleged 2 million write limitation. Kris and I had a quick discussion on this topic, off-list, and his original flash-based device is still in constant operation after 2 years and I have flash modules that I purposely tried to kill with writes. It
took significant effort to start causing error situations, which were very easily detected before the system would become unusable. Erick, you should focus on having a quick action restoration plan and
extra DOMs always readily available.Then when a failure situation is detected, you can react very quickly. Jeremy McNamaraJeremy, Erick - I have always pointed to this SanDisk whitepaper:
http://www.sandisk.com/Assets/File/OEM/WhitePapersAndBrochures/RS-MMC/WPaperWearLevelv1.0.pdf
While it specifically discusses their industrial line of CF cards, it is pretty obvious that flash can, and often does, last much longer thanother components in a system when properly implemented.You will notice
that the SHORTEST expected life of a CF card in their test scenarios was over 70 years!How long is your power supply going to last?Even ifthe consumer level cards had 1/10 the life expectancy, that is still
seven years.I expect to get at least that from my original AstLinuxsystem.It's been two so far, I'll let you know how it is doing inanother five years :). JFFS (and similar FSs) are not appropriate for CF cards or DOMs.They
are meant to be used directly on flash memory and do their own wear leveling and in some cases, compression.All kinds of commercialdevices use JFFS2.If you are using a CF or DOM with Linux, ext2 is the
best FS to use.CF cards and DOMs use their own wear leveling, so noneis required in the operating system or file system.CF cards and DOMshide wear leveling from you and expose themselves as an ordinary IDE device.
I echo Jeremy's conclusions.With a properly designed operating system, decent flash memory, and a reasonable usage pattern, I can tellyou (with a great amount of certainty) that in most situations, CF cards
will outlast just about any hard drive (even SCSI) when used 24/7. These days, it really is pretty tough to trash flash. However, if you are running a MySQL cluster or something with several,multi-gigabyte databases, no type of flash memory will last very long! :)
To

Re: [asterisk-users] password for vm users

2006-10-09 Thread Aaron Daniel

You're digging a little further into it than any standard voicemail
system would ever go.  If you need more strenuous password
functionality, I would suggest dropping users through a voicemail macro
that does all the password functionality for you and then passed them to
the actual voicemail app.  That way it's more flexible, and you can
allow the voicemail app to do what it's good at, voicemail ;)

Aaron

On Mon, 2006-10-09 at 14:07 -0700, stan ford wrote:
 how about password strength? or remembering and not allowing password?
 or password duration?
 
 Marco Mouta [EMAIL PROTECTED] wrote: 
 just set your initial password to be equal to vm-account
 number, and Voicemail application will do that for you and
 will request users to setup a new password!
 
 
 
 On 10/9/06, stan ford [EMAIL PROTECTED] wrote: 
 how does one force mandatory password change on login?
 and a period of time to pass before mandating a
 password change?
  
  im using trixbox so if you have that info. that would
 be good, if you have it for asterisk, im sure i could
 figure it out as well 
  
 thx.
 
 
 __
 Yahoo! Messenger with Voice. Make PC-to-Phone Calls to
 the US (and 30+ countries) for 2¢/min or less. 
 
 
 ___
 --Bandwidth and Colocation provided by Easynews.com --
 
 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
 
 http://lists.digium.com/mailman/listinfo/asterisk-users
 
 
 
 
 
 -- 
 Com os melhores cumprimentos,
 
 Marco Mouta ___
 --Bandwidth and Colocation provided by Easynews.com --
 
 asterisk-users mailing list
 To 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.
 ___
 --Bandwidth and Colocation provided by Easynews.com --
 
 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
-- 
Aaron Daniel
Computer Systems Technician
Sam Houston State University
[EMAIL PROTECTED]
(936) 294-4198

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Range Operator

2006-10-09 Thread Mojo with Horan Company, LLC


Doug Lytle wrote:
I'm not really sure what _X./_5XXX does.  I thought _X. would match 
anything that would start with a digit. What does the /_5XXX do on that 
same matching string?


Doug, it's:
exten = exten#/cid_2_match,priority,application
for example, ex-girlfriend logic: (g/f number is 5551234)

exten = s/5551234,1,Zapateller
exten = s,1,Dial(SIP/housephone...
exten = s,2,Hangup

Notice how I have two priority 1s for the 's' extension. The first one 
ends up being most specific for an incoming call from that callerid and 
is used.  Then it jumps to priority 2.


OP-Doug is matching incoming calls from callers with a callerID of 5xxx.

Moj


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] MINNESOTA: TwinCities Asterisk Users Group - Saturday October 14th 2006 - 10:30am

2006-10-09 Thread asterisk_help

This is a reminder that the Twin Cities Asterisk Users Group will be
meeting this Saturday, October 14th at 10:30am. - Please note the time
change; we are meeting one hour earlier than our normal time.

This month is our bi-annual new user meeting. We'll show you how to get
started with Asterisk and answer your questions about what Asterisk can do
and if possible, we'll show you how, on the spot. The new Asterisk 1.4 is
currently in the beta test stage and if there is interest, we'll update
one of our systems from 1.2 to 1.4 and discuss what's new and what changes
you might need to make. If you're not a developer, this is now your
chance to contribute to the Asterisk development process. Beta testers are
needed now. Please try the new version on any non mission critical
systems.

Meetings are held monthly on the second Saturday of each month, excluding
July and December. The Agenda is posted online
http://www.voip-info.org/wiki/index.php?page=Twin+Cities+Asterisk+User+Group+Agenda

Meetings are held at Sound Choice Communications LLC...
-= 7839 12th Ave So, Bloomington Minnesota USA 55425 =-
http://maps.google.com/maps?oi=mapq=7839%2012th%20Ave%20S%2055425

Come to a meeting to meet other asterisk users, see asterisk solutions,
win a door prize, eat food, or for the good company, to look for work, if
your looking for employees, to go out for a drive, to get out of your
house, whatever, JUST COME TO THE MEETING!

New visitors can help themselves to FREE FXO Interface cards (So you can
connect your phone line, and have a timing source for meetme and IAX
protocols). Some members have been known to swap hardware at the meetings.
Have extra VoIP gear, looking for VoIP gear? There's plenty of hardware to
see. Have you been to a meeting recently?

Please come and share your own ideas and learn from others. As always,
free food.

We are always looking for help with meeting topics. If you feel like
taking the lead, please do and simply let me know if you need anything.

Meeting starts at 10:30am and parking is available in the rear of the
building. Runs about 2 hours or less, and we'll order Pizza to the meeting
for lunch. This month we will need to wrap up by 12:30pm or 12:45pm.

Look forward to seeing you there.

http://www.voip-info.org/tiki-index.php?page=Asterisk%20User%20Group%20TwinCities%20Minnesota%20USA

If you have a product or service you'd like to introduce to our members,
send a private message to ejo1(at)soundchoicecomm.com and we'll see if we
can't get you listed as next month's sponsor.

Sound Choice Communications is a reseller of Digium and Polycom products
and we have inventory on hand. Give us a call and your items will be
waiting for you on Saturday. Thank you! +1.651-999-0888

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Problem compiling libmfcr2.0.0.2 on Fedora Core 5

2006-10-09 Thread Carlos Chavez

I am getting an error in Fedora Core 5 when trying to compile
libmfcr2.0.0.2:

[EMAIL PROTECTED] libmfcr2-0.0.3]# make
make  all-am
make[1]: Entering directory `/usr/src/mfcr2/libmfcr2-0.0.3'
if /bin/sh ./libtool --tag=CC --mode=compile gcc -DHAVE_CONFIG_H -I. -I.
-I. -I/usr/include/libxml2-g -O2 -MT mfcr2.lo -MD -MP -MF
.deps/mfcr2.Tpo -c -o mfcr2.lo mfcr2.c; \
then mv -f .deps/mfcr2.Tpo .deps/mfcr2.Plo; else rm -f
.deps/mfcr2.Tpo; exit 1; fi
mkdir .libs
 gcc -DHAVE_CONFIG_H -I. -I. -I. -I/usr/include/libxml2 -g -O2 -MT
mfcr2.lo -MD -MP -MF .deps/mfcr2.Tpo -c mfcr2.c  -fPIC -DPIC
-o .libs/mfcr2.o
mfcr2.c: In function 'check_event':
mfcr2.c:2793: warning: passing argument 1 of 'r2_mf_tx' from
incompatible pointer type
mfcr2.c:2801: warning: passing argument 1 of 'r2_mf_tx' from
incompatible pointer type
mfcr2.c: In function 'load_r2_parameter_set':
mfcr2.c:2905: error: too few arguments to function 'r2_mf_tx_init'
make[1]: *** [mfcr2.lo] Error 1
make[1]: Leaving directory `/usr/src/mfcr2/libmfcr2-0.0.3'
make: *** [all] Error 2

Any idea what could be missing?

-- 
Carlos Chavez Prats
Director de Tecnología
Telecomunicaciones Abiertas de México S.A. de C.V.
Tel: +52-55-91169161 Ext 2001


signature.asc
Description: This is a digitally signed message part
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [asterisk-users] Asterisk RT on Disk On Module PerformanceandDurability

I was
thinking about it purely from the perspective of multiple access to flat
files :)

-Original Message-From: Erick Perez
[mailto:[EMAIL PROTECTED]Sent: Monday, October 09, 2006 3:44
PMTo: Asterisk Users Mailing List - Non-Commercial
DiscussionSubject: Re: [asterisk-users] Asterisk RT on Disk On
Module PerformanceandDurability
Douglas, Im just the asterisk guy. If they decide to write a
cross-browser multi-tier interface in AJAX, assembly language or Pascal,
that's up to them (the programmers). I will let them know what can/can't be
done.

Thinking of that...15 years ago...the last time i used pascal.

On 10/9/06, Douglas
Garstang [EMAIL PROTECTED]
wrote:

I'm just going to jump in
here, and ask a stoopid question.

How could you possibly
write a multi-user front end in AJAXwithout using a database backend
like MySQL?

Doug.

-Original
Message-From: Erick Perez [mailto:
[EMAIL PROTECTED]]Sent: Monday, October 09, 2006 1:58
PMTo: Asterisk Users Mailing List - Non-Commercial
DiscussionSubject: Re: [asterisk-users] Asterisk RT on Disk On
Module Performance andDurability
Jeremy, Cohen, Kris, thanks to all of you.

Indeed after reading the Sandisk paper it shed a lot of light on this
matter. The whole idea is to have a large scale system with no moving
parts (we call a large system something with250 users, at least down
here ;-) )

the whole idea is for a customer that needs an IVR in 4 languages
with autoattendant, extensive CDR and plotted usage patterns as well as
voicemail. Voicemail will be used *a lot*, probably about one thousand
voicemails per day and the customer does not want VM-to-Email (God knows
why!).

Oh, and the whole idea of the database is because the developers are
working in an AJAX based interface that does the asterisk
config/plotting/vm/day-to-day stuff with ARA, so a db is needed.
I started learning asterisk with flat files...it works for me...but
hey...times are changing.

Who knows, maybe the whole thing can be fitted in ram (except for the
vm part)...we'll see. I had to ask anyway, but i don't like Dbs
eitherit adds and extra breakup layer (maybe Im kind of
outdated).

Smaller iPBXs will definitely be CF and RAM based and I, at least,
will force VMtoEmail and do all the processing in RAM.

Again,

Thanks to all of you.
P.D. I will later follow this thread with the full working
configs that will take place at user premises. And for the sake of the
test. I will try to kill a sandisk USB with the full config.

Re: [asterisk-users] GotoIfTime - much slowdown with 90 conditions?


Mike Dent wrote:


I'm guessing myself it won't be a big deal but I just want to check
before I commit too much.


Don't know... Why don't you bench it and let us know.

Doug

-- Ben Franklin quote: Those who would give up Essential Liberty to 
purchase a little Temporary Safety, deserve neither Liberty nor Safety.


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Re: PRI issues