date:20060103

In article [EMAIL PROTECTED], [EMAIL PROTECTED] 
says...
 /blockquote
 FYI, this is the relevant extensions_custom.conf entry on an AAH system:=

I'm not using [EMAIL PROTECTED] Thank you!


-- 

Tomislav Parcina
[EMAIL PROTECTED]

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[Asterisk-Users] Re: Re: Asterisk Christmas Help request

In article [EMAIL PROTECTED], [EMAIL PROTECTED] 
says...
 /blockquote
 FYI, this is the relevant extensions_custom.conf entry on an AAH system:=
 br

It works great on Asterisk 1.2.1

exten = 270,1,Answer
exten = 270,2,Playback(at-tone-time-exactly)
exten = 270,3,SayUnixTime(,/Europ/Zagreb,AdBY \'digits/at\' kM)
exten = 270,4,Playback(beep)
exten = 270,5,Hangup

Thank you!


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[EMAIL PROTECTED]

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RE: [Asterisk-Users] How to set features.conf to change thehangup key.

Hello,

The idea is the following:

For the 1.2.1 installation just set the parameter 
disconnect = *97
In your features.conf

For the 1.0.7 installation you either upgrade or patch the code. The patch
the code would require you a lot of knowledge of c programming. It would
consist of extracting from the 1.2.1 code the disconnect functionality and
add it to the 1.0.7 code base. But that is not straight forward...

If you need it badly we can do it for you as consulting. But I strongly
advise you to upgrade.

Upgrade,now, is not an easy task either, but it might be easier that the
code patch. Mainly because you would have to migrate the configuration or
test it... Do you have a test bed?

BR

Bogdan Moldovan
MODULO Consulting
The Future Is Not What It Used To Be
http://www.modulo.ro 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Obelix
Sent: Tuesday, January 03, 2006 2:01 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] How to set features.conf to change thehangup
key.

Quoting Bogdan Moldovan [EMAIL PROTECTED]:

I don't have this in my main installation, which is 1.0.7.
In the case of 1.0.7 where else can I effect that change?

I also have a 1.2.1 setup, what would I have to change in the code below?

What is the general idea?

 Indeed, this is 1.2.1

 But do the following:

 Go to the source tree, do a
 vi res/res_features.c

 Search for a :
 struct ast_call_feature builtin_features[]

 And you should see the builtin features:

 In 1.2.1 I have:

 #define FEATURES_COUNT (sizeof(builtin_features) /
 sizeof(builtin_features[0]))
 struct ast_call_feature builtin_features[] =  {
 { AST_FEATURE_REDIRECT, Blind Transfer, blindxfer, #, 
 #, builtin_blindtransfer, AST_FEATURE_FLAG_NEEDSDTMF },
 { AST_FEATURE_REDIRECT, Attended Transfer, atxfer, , , 
 builtin_atxfer, AST_FEATURE_FLAG_NEEDSDTMF },
 { AST_FEATURE_AUTOMON, One Touch Monitor, automon, , , 
 builtin_automonitor, AST_FEATURE_FLAG_NEEDSDTMF },
 { AST_FEATURE_DISCONNECT, Disconnect Call, disconnect, 
 *, *, builtin_disconnect, AST_FEATURE_FLAG_NEEDSDTMF }, };

 In case you do not have this, good changes are that, in case you need 
 badly this feature, you will upgrade or tweak the sources...

 Bogdan



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Re: [Asterisk-Users] call-limit kills hints

2006-01-03 Thread Eric \ManxPower\ Wieling


Joseph Rothstein wrote:

I am setting up 10 SNOM 320s for a customer, and there seems to be a problem
with call-limit and hints.

Here is my sip config for one phone:

[944]
type=friend
context=x
language=de
accountcode=x
notifyringing=yes
host=dynamic
dtmfmode=rfc2833
[EMAIL PROTECTED]   
callerid=x  944

canreinvite=no
disallow=all
allow=g729
nat=yes

If I add to this, call-limit=1, hint does not work at all. I get no status
change from the hinted devices/extensions.


I believe the incominglimit outgoinglimit and limit options will be 
removed in the next version of Asterisk.  They were replaced by the 
*Group applications in 1.0 and by the *GROUP functions in 1.2.

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Re: [Asterisk-Users] cisco 7960 registration fails

2006-01-03 Thread miguel saravia


What is your firmware version? I have a few problems with the release 7.5

Miguel

Ben Fitzgerald wrote:


Hi,

Apologies for hitting the list with such a long mail on my first post!
Having seen the archives this seems like a list that likes debugging
output. If I have left any information out please let me know.

I have recently begun using asterisk on debian.

[EMAIL PROTECTED] /usr/sbin/asterisk -V
Asterisk 1.0.7-BRIstuffed-0.2.0-RC7k

[EMAIL PROTECTED] dpkg -l asterisk
Desired=Unknown/Install/Remove/Purge/Hold
|
Status=Not/Installed/Config-files/Unpacked/Failed-config/Half-installed
|/ Err?=(none)/Hold/Reinst-required/X=both-problems (Status,Err:
uppercase=bad)
||/ NameVersion Description
+++-===-===-==
ii  asterisk1.0.7.dfsg.1-2  Private Branch Exchange (PBX)

I have an odd problem that may be known as I saw one similar posting.

I have the following config on my cisco 7940:

line1_name : localuser
line1_authname : localuser
line1_password : localpass
line1_shortname : asterisk
line1_displayname : myphone

Then in sip.conf:

[localuser]
type=friend
username=localuser
secret=localpass
auth=md5
host=dynamic
dtmfmode=rfc2833
nat=no
allow=all
canreinvite=no

Phone IP: 192.168.1.50.

I startup asterisk and connect to the console, and set:

sip debug ip 192.168.1.50
set verbose 255
set debug 255

The console output is as follows:

## Start asterisk debug output ###
Sip read:
REGISTER sip:192.168.1.4 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK4b1f5669
From: sip:[EMAIL PROTECTED]
To: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
Date: Mon, 02 Jan 2006 21:11:05 GMT
CSeq: 646 REGISTER
User-Agent: CSCO/7
Contact: sip:[EMAIL PROTECTED]:5060
Content-Length: 0
Expires: 3600


11 headers, 0 lines
Jan  2 21:11:05 DEBUG[6128]: chan_sip.c:2355 sip_alloc: Allocating new SIP call 
for [EMAIL PROTECTED]
Using latest request as basis request
Sending to 192.168.1.50 : 5060 (non-NAT)
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK4b1f5669
From: sip:[EMAIL PROTECTED]
To: sip:[EMAIL PROTECTED];tag=as01aba5cf
Call-ID: [EMAIL PROTECTED]
CSeq: 646 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:[EMAIL PROTECTED]
Content-Length: 0


to 192.168.1.50:5060
Transmitting (no NAT):
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK4b1f5669
From: sip:[EMAIL PROTECTED]
To: sip:[EMAIL PROTECTED];tag=as01aba5cf
Call-ID: [EMAIL PROTECTED]
CSeq: 646 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:[EMAIL PROTECTED]
WWW-Authenticate: Digest realm=mail.bfitzgerald.co.uk, nonce=773ad211
Content-Length: 0


to 192.168.1.50:5060
Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms
Urgent handler


Sip read:
REGISTER sip:192.168.1.4 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK1f807b05
From: sip:[EMAIL PROTECTED]
To: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
Date: Mon, 02 Jan 2006 21:11:06 GMT
CSeq: 647 REGISTER
User-Agent: CSCO/7
Contact: sip:[EMAIL PROTECTED]:5060
Authorization: Digest 
username=localuser,realm=mail.bfitzgerald.co.uk,uri=sip:192.168.1.4,response=56cf80cc6dc37af4e3f6e036cb45a7bd,nonce=773ad211,algorithm=md5
Content-Length: 0
Expires: 3600


12 headers, 0 lines
Using latest request as basis request
Sending to 192.168.1.50 : 5060 (non-NAT)
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK1f807b05
From: sip:[EMAIL PROTECTED]
To: sip:[EMAIL PROTECTED];tag=as01aba5cf
Call-ID: [EMAIL PROTECTED]
CSeq: 647 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:[EMAIL PROTECTED]
Content-Length: 0


to 192.168.1.50:5060
Urgent handler

*
## End asterisk debug output ###

The tethereal capture on my asterisk server is as below:

39.943108  192.168.1.4 - 192.168.1.50 SIP Status: 100 Trying(1 bindings)
39.943335  192.168.1.4 - 192.168.1.50 SIP Status: 401 Unauthorized(1 
bindings)
40.184716 192.168.1.50 - 192.168.1.4  SIP Request: REGISTER sip:192.168.1.4
40.185768  192.168.1.4 - 192.168.1.50 SIP Status: 100 Trying(1 bindings)
59.999521  192.168.1.1 - BroadcastARP Who has 192.168.1.50?  Tell 
192.168.1.1

The main problem is I cannot get my 7940 to register. But in attempting
to debug this I have seen another problem.

Asterisk stops outputting to the console after the above output. Even
when subsequent REGISTER requests are seen by tethereal I do not get any
more asterisk console messages. This makes me wonder if the debian
distro package is correct. Surely this is a problem with the package?

The phone starts to register but doesn't quite manage it:

SIP Phone sh reg

LINE REGISTRATION TABLE
Proxy Registration: ENABLED, state: IDLE
line  APR  state  timer   expires proxy:port
  ---

Re: [Asterisk-Users] call-limit kills hints

2006-01-03 Thread Paradise Dove

i have the same problem and also have submitted it as bug
http://bugs.digium.com/view.php?id=5281.
the  Patch-5281-v2.txt in the mentioned bug will solve your problem.

Paradise Dove

On 1/3/06, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
 Joseph Rothstein wrote:
  I am setting up 10 SNOM 320s for a customer, and there seems to be a problem
  with call-limit and hints.
 
  Here is my sip config for one phone:
 
  [944]
  type=friend
  context=x
  language=de
  accountcode=x
  notifyringing=yes
  host=dynamic
  dtmfmode=rfc2833
  [EMAIL PROTECTED]
  callerid=x  944
  canreinvite=no
  disallow=all
  allow=g729
  nat=yes
 
  If I add to this, call-limit=1, hint does not work at all. I get no status
  change from the hinted devices/extensions.

 I believe the incominglimit outgoinglimit and limit options will be
 removed in the next version of Asterisk.  They were replaced by the
 *Group applications in 1.0 and by the *GROUP functions in 1.2.
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[Asterisk-Users] outbound sip calls on asterisk

2006-01-03 Thread James Burke



hi,

i would like all my calls originating from asterisk users bound for 
external to route to one destination, a session border controller. 
protocol used is sip.


i have edited extensions_custom.conf with:

exten = _.,1,dial(sip/[EMAIL PROTECTED])

would this be correct to send any calls from internal to the x.x.x.x ip?

i get this from the cli:

== Spawn extension (from-external. then it just times out and dumps 
the calls?


the phone used is a cisco7960 using pos3-07-5-00

any help appreciated... :)

james
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Re: [Asterisk-Users] cisco 7960 registration fails

2006-01-03 Thread Ben Fitzgerald

On Tue, Jan 03, 2006 at 01:34:28PM +0200, Tzafrir Cohen wrote:
 On Tue, Jan 03, 2006 at 10:47:59AM +, Ben Fitzgerald wrote:
  Hi,
  
  Apologies for hitting the list with such a long mail on my first post!
  Having seen the archives this seems like a list that likes debugging
  output. If I have left any information out please let me know.
 
 What do you see on 'sip show peers' ?

That does show the device, but when I set qualify=yes in sip.conf I
get:

deb-tv*CLI sip show peers
Name/usernameHost Dyn Nat ACL MaskPort Status
localuser/local  192.168.1.50 D  255.255.255.255  5060 UNKNOWN

As I understand it the Status should not be UNKNOWN.

Many thanks for the pointer to Rapid. I will add this to my sources.list
and try re-installing asterisk, as I'm sure that the loss of console
output does not bode well, however poor my configuration may be!

I will let you know how I get on.

Cheers,

Ben.

-- 
Registered Linux user number 339435
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Re: [Asterisk-Users] cisco 7960 registration fails

2006-01-03 Thread Ben Fitzgerald

On Tue, Jan 03, 2006 at 09:46:32AM -0300, miguel saravia wrote:
 What is your firmware version? I have a few problems with the release 7.5

It's 7.4. I have read a few comments about 7.5 so only went to 7.4:

Loadid:  SW: P0S3-07-4-00  ARM: PAS3ARM1  Boot: PC030301  DSP: PS03AT45

Thanks,

Ben.

-- 
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Re: [Asterisk-Users] USB phone

2006-01-03 Thread Leo Ann Boon


Dushyanth Harinath wrote:


Hi,

Asterisk doesnt support USB phones directly. You need a softphone and
then a compatible USB phone.
 

Asterisk does support the Digium S100U USB analog FXS adapter. It's 
based on the TigerJet chipset found in many cheap USB phones. The S100U 
looks like the stock TJ reference design, so I believe it's possible to 
use other TJ based USB analog adapter. There was a bounty to get wcusb 
to work with another TJ based FXS adapter.


FYI.

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[Asterisk-Users] Re: Re: Congestion problem

In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
 It's a common complaint.
 
 Have you searched the archives?  Look for disconnect supervision.

I have now. And things are a litle bit more clear to me. Thank you for 
hint.


-- 

Tomislav Parcina
[EMAIL PROTECTED]

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[Asterisk-Users] Three Way Calling with HFC PCI Card

2006-01-03 Thread Henry Margies

Hello,

I'm using a TDM 400 together with two HFC PCI Cards in my Box. Three Way
Calling with a SIP or analog Phone is working perfectly.

But if I try to do Three Way Calling with my ISDN Phone I get an error
message: Facility Name requested on channel 0/2 not in use on span 1

I use bristuff with my HFC card and don't know why I get this message?

I'm using still asterisk 1.0 and can not update to the newest version at
the moment. Is there a simple trick to make it work or is this problem
already solved in asterisk 1.2?

Thanks in advance,

Henry
-- 
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~/.signature to help me spread!

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[Asterisk-Users] Where is the Prefix() application in Asterisk 1.2.1 ?

2006-01-03 Thread Bukoka Budoka



Hi,

i am trying to use the Prefix() application in my dialplan but ...it is not 
there:


pbx.c:1690 pbx_extension_helper: No application 'Prefix' for extension 
(test, 1233, 1)


My entry in extensions.conf is the following:

[outgoing-calls]
exten = _12xx,1,Prefix(0)
exten = _012xx,2,Dial(${TRUNK}/${EXTEN}|30,r)

As i also see in my Asterisk 1.2.1 there is no prefix() application.

localhost*CLI show applications like Prefix
   -= Matching Asterisk Applications =-
   -= 0 Applications Matching =-
localhost*CLI show applications like prefix
   -= Matching Asterisk Applications =-
   -= 0 Applications Matching =-


I read somewhere that the prefix command is only in CVS-HEAD...   Is this 
true?


Thank you,

Budoka.

_
Express yourself instantly with MSN Messenger! Download today it's FREE! 
http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/


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[Asterisk-Users] Re: voicemail storage over odbc and postgres

In article [EMAIL PROTECTED], 
[EMAIL PROTECTED] says...
 There is already a very good database for binary files,
 called a filesystem

Is there any how-to for filesystem and Asterisk voicemail storage?


-- 

Tomislav Parcina
[EMAIL PROTECTED]

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Re: [Asterisk-Users] Cell phone dock/switch as Asterisk FXO source

2006-01-03 Thread Noah Swint


Do you have a url for the device?



From: [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] Cell phone dock/switch as Asterisk FXO source
Date: Tue, 3 Jan 2006 11:02:30 +0200
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On Tuesday 03 January 2006 05:48, Paul Dugas wrote:
 On Mon, 2006-01-02 at 16:06 +, Jonathan Attwood wrote:
  I use a Dock-n-Talk in conjuction with a Sipura SPA3000  Asterisk.

 Does this unit require any funky dialing when placing outbound calls
 from * through the phone?  Do the docs indicate operation is any
 different between CDMA, TDMS, AMPS, or GSM phones?  I'd guess not or, if
 so, it was simple to handle it in the dialplan but I'm curious anyway.
 I've been considering this as a way to have work calls that come to my
 cell appear different to the server.  At the moment, I have my GSM phone
 forward calls to the house when it's off so I can't really tell between
 them.
I have good experience with a GSM-box I've bought from cybertelecom and
SPA3000. GSM-box acts as a Dock-n-Talk because is it allows in and out
dialing. The advantage is that one doesn't need even a mobile phone, but 
only

a SIM card. The whole thing is like porting a number.

There are 2 FXS ports. One could go to an ordinary phone, the other to
SPA3000.

The disadvantage is that you have one more number for your friends to
remember. Otherwise is stable, and as-easy- as-PnP instalation, if you 
don't

forget to disable the pin lock as I did :-)
benchev


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Re: [Asterisk-Users] cisco 7960 registration fails

2006-01-03 Thread Rich Adamson

 Apologies for hitting the list with such a long mail on my first post!
 Having seen the archives this seems like a list that likes debugging
 output. If I have left any information out please let me know.
 
 I have recently begun using asterisk on debian.
 
 [EMAIL PROTECTED] /usr/sbin/asterisk -V
 Asterisk 1.0.7-BRIstuffed-0.2.0-RC7k
 
 [EMAIL PROTECTED] dpkg -l asterisk
 Desired=Unknown/Install/Remove/Purge/Hold
 |
 Status=Not/Installed/Config-files/Unpacked/Failed-config/Half-installed
 |/ Err?=(none)/Hold/Reinst-required/X=both-problems (Status,Err:
 uppercase=bad)
 ||/ NameVersion Description
 +++-===-===-==
 ii  asterisk1.0.7.dfsg.1-2  Private Branch Exchange (PBX)
 
 I have an odd problem that may be known as I saw one similar posting.
 
 I have the following config on my cisco 7940:
 
 line1_name : localuser
 line1_authname : localuser
 line1_password : localpass
 line1_shortname : asterisk
 line1_displayname : myphone

If the above is a copy/paste, then remove the quotes. Format should
be like this: line1_authname: 1234
Quotes can be used in displayname and shortname.

 Then in sip.conf:
 
 [localuser]
 type=friend
 username=localuser
 secret=localpass
 auth=md5

Try removing auth=md5; not sure the 7940 supports it (never tried it).

 host=dynamic
 dtmfmode=rfc2833
 nat=no
 allow=all
 canreinvite=no

If you want the above definitions to fit a specific context, then add:
 context=from-sip
or whatever extensions.conf context you'd like. The above definitions
assume a default context which might not be all that obvious later
when you're playing with other functions. The rest of the definitions
in the above are fine.
 
 The phone starts to register but doesn't quite manage it:
 
 SIP Phone sh reg
 
 LINE REGISTRATION TABLE
 Proxy Registration: ENABLED, state: IDLE
 line  APR  state  timer   expires proxy:port
   ---  -  --  -- -
 1 11x  REGISTERING3600204 192.168.1.4:5060

Try 'sip show peers' from the CLI. You should see something like:

phoenix*CLI sip show peers
Name/username  HostDyn Nat ACL Port Status
3000/3000  206.222.193.90   D  5060 Unmonitored

when the phone successfully registers.

If you see something like this:
Name/username  HostDyn Nat ACL Port Status
3050/3050  (Unspecified)D  0Unmonitored

the registration process is a problem.

Rich


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Re: [Asterisk-Users] Where is the Prefix() application in Asterisk 1.2.1 ?

2006-01-03 Thread Pete Barnwell

On Tue, 2006-01-03 at 14:06 +0100, Bukoka Budoka wrote:
 Hi,
 
 i am trying to use the Prefix() application in my dialplan but ...it is not 
 there:
 
 pbx.c:1690 pbx_extension_helper: No application 'Prefix' for extension 
 (test, 1233, 1)
 
 My entry in extensions.conf is the following:
 
 [outgoing-calls]
 exten = _12xx,1,Prefix(0)
 exten = _012xx,2,Dial(${TRUNK}/${EXTEN}|30,r)
 
 As i also see in my Asterisk 1.2.1 there is no prefix() application.
 
 localhost*CLI show applications like Prefix
 -= Matching Asterisk Applications =-
 -= 0 Applications Matching =-
 localhost*CLI show applications like prefix
 -= Matching Asterisk Applications =-
 -= 0 Applications Matching =-
 
 
 I read somewhere that the prefix command is only in CVS-HEAD...   Is this 
 true?
 

Looks a plausible explanation:-

asterisk*CLI show applications like Prefix
-= Matching Asterisk Applications =-
Prefix: Prepend leading digits
-= 1 Applications Matching =-
asterisk*CLI show version
Asterisk CVS HEAD built by [EMAIL PROTECTED] on a i686 running Linux on
2005-10-13 15:42:41 UTC

Rgds

Pete

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[Asterisk-Users] Howto compile chan_h323 on macosx 10.3?

2006-01-03 Thread Antonio Marquez


I can not compile the h323 support for macosx 10.3?
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Re: [Asterisk-Users] Realtime Multiple Asterisk boxes, iaxusers


Simone Cittadini wrote:

What about IAX ? If I connect two asterisk servers to a common mysql 
backend (only iaxusers, no sip or extensions) will it :


There is no support for sharing dynamic peer registrations between 
Asterisk servers via Realtime for SIP or IAX2. Sharing the Realtime 
database for users and non-dynamic peers works fine, since there is no 
updating of the database required.

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RE: [Asterisk-Users] Problem with date time on Aastra 480i sincerelease 1.3

Does anyone know whether there is some sort of time zone option?  I've
emailed Aastra who didn't come back to me.  I would like to set the time
zone - e.g. Britain-London, in the cfg files so I don't have to set it
on 40 phones...

Regards

Lee 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Robert La
Ferla
Sent: 26 December 2005 16:51
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Problem with date  time on Aastra 480i
sincerelease 1.3


Jacques Leisy wrote:
 Thanks Robert. I tried of course with time server disabled: 0 too.
 Is it working for you? Which time server are you using, an external
one?

Works for me and I'm using an internal one which is then synced to an
external one.

Try ONLY these entries.  Remove the time format and date format and
backup ntp servers:

time server disabled: 0
time server1: 192.168.0.10

If this doesn't work, you should check your firewall rules (if any) and
the versions of ntpd (4.2?) that you are running.

Robert

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[Asterisk-Users] Re: connect more the one phone to ONE sip Acoount

2006-01-03 Thread Mikael Magnusson


Olle E Johansson wrote / skrev:


Andreas Koch wrote:


Hello,
how is it possible to connect (register)  more the one Phone to One 
Sip-Acoount.


With, for example sipgate.de this is not a special feature, it is common.
We have users, what like to have more then one Phone, - Homeoffice 
and Desk


Rigth now if a other phone registers whith the data, the first ist 
removed.


You have to consider that Asterisk is a multiprotocol PBX and that the 
PBX need to be in control of each device connected to the PBX. With 
multiple registrations for one account we would break the Asterisk 
architecture unless we did some very clever stuff. This has been 
discussed quite a lot of times, so please search the mailing list for 
more information.


I understand that allowing multiple registrations would break chan_sip, 
but how can it break the Asterisk architecture if the forking is done by 
the Dial application? Would it really matter if the dial string contains 
multiple SIP AOR:s/users, which is possible today, or multiple bindings 
for one SIP AOR?


Example of Dial with multiple SIP AOR:s/users, working today:
Dial(SIP/user1SIP/user2)

Example of Dial with multiple bindings for one SIP AOR, expanded by FOO:
Dial(${FOO(user1)}) = Dial(SIP/user1/10.1.1.1SIP/user1/10.2.2.2)

/Mikael

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RE: [Asterisk-Users] Problem with date time on Aastra 480i sincerelease 1.3

2006-01-03 Thread Dave Cotton

On Tue, 2006-01-03 at 14:13 +, Lee Archer wrote:
 Does anyone know whether there is some sort of time zone option?  I've
 emailed Aastra who didn't come back to me.  I would like to set the time
 zone - e.g. Britain-London, in the cfg files so I don't have to set it
 on 40 phones...
 
in aastra.cfg

time server disabled: 0
time server1: 192.168.1.253
time format: 1
date format: 0
time zone name: FR-Paris
time zone code: CET
time zone minutes: 60

works for me.
-- 
Dave Cotton [EMAIL PROTECTED]

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Re: [Asterisk-Users] Realtime Multiple Asterisk boxes, iaxusers

2006-01-03 Thread Rich Adamson


  What about IAX ? If I connect two asterisk servers to a common mysql 
  backend (only iaxusers, no sip or extensions) will it :
 
 There is no support for sharing dynamic peer registrations between 
 Asterisk servers via Realtime for SIP or IAX2. Sharing the Realtime 
 database for users and non-dynamic peers works fine, since there is no 
 updating of the database required.

If you take the word dynamic out of that, then can he effectively 
have primary/secondary/backup systems that allows the user to
re-register and/or redial his call on a different * server?


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RE: [Asterisk-Users] Problem with date time on Aastra 480i sincerelease 1.3

2006-01-03 Thread Pete Barnwell

On Tue, 2006-01-03 at 14:13 +, Lee Archer wrote:
 Does anyone know whether there is some sort of time zone option?  I've
 emailed Aastra who didn't come back to me.  I would like to set the time
 zone - e.g. Britain-London, in the cfg files so I don't have to set it
 on 40 phones...


time zone name: GB-London
time zone code: GMT
time zone minutes: 60

Rgds

Pete

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RE: [Asterisk-Users] Re: connect more the one phone to ONE sip Acoount

2006-01-03 Thread Juan Janczuk

Just a contribution coming from an Asterisk-Newbie ignorant

Couldn't this behaviuor (The fake 2 phones, with the same ext #), be
achieved via a gruop configuration?
At least, in my [EMAIL PROTECTED], you can configure a group pointing to 2 
different
extensions.

Regards.
Juan.

 -Mensaje original-
 De: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] nombre de
 Mikael Magnusson
 Enviado el: Martes, 03 de Enero de 2006 11:21 a.m.
 Para: asterisk-users@lists.digium.com
 Asunto: [Asterisk-Users] Re: connect more the one phone to ONE sip Acoount


 Olle E Johansson wrote / skrev:

  Andreas Koch wrote:
 
  Hello,
  how is it possible to connect (register)  more the one Phone to One
  Sip-Acoount.
 
  With, for example sipgate.de this is not a special feature, it
 is common.
  We have users, what like to have more then one Phone, - Homeoffice
  and Desk
 
  Rigth now if a other phone registers whith the data, the first ist
  removed.
 
  You have to consider that Asterisk is a multiprotocol PBX and that the
  PBX need to be in control of each device connected to the PBX. With
  multiple registrations for one account we would break the Asterisk
  architecture unless we did some very clever stuff. This has been
  discussed quite a lot of times, so please search the mailing list for
  more information.

 I understand that allowing multiple registrations would break chan_sip,
 but how can it break the Asterisk architecture if the forking is done by
 the Dial application? Would it really matter if the dial string contains
 multiple SIP AOR:s/users, which is possible today, or multiple bindings
 for one SIP AOR?

 Example of Dial with multiple SIP AOR:s/users, working today:
 Dial(SIP/user1SIP/user2)

 Example of Dial with multiple bindings for one SIP AOR, expanded by FOO:
 Dial(${FOO(user1)}) = Dial(SIP/user1/10.1.1.1SIP/user1/10.2.2.2)

 /Mikael

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Re: [Asterisk-Users] ACD with polycom ip phones

2006-01-03 Thread BJ Weschke

On 12/19/05, Kevin P. Fleming [EMAIL PROTECTED] wrote:
 Matthew wrote:

  For the uninitiated among us (myself included) what is ACD login/logout
  support?

 The Polycom phones can send XML NOTIFY messages to signal to the server
 the agent is logged in/out/paused. I know of no documentation on the
 messages (although they don't look that hard to parse), but nobody has
 come up with any sort of architecture that would allow chan_sip to do
 something useful with the messages.


 For all those interested, there's now a working implementation of
Polycom Agent login/logout integration with the Asterisk agent
infrastructure on mantis. We'll be adding avail/unavail in the next
few days.

 Testing assistance is greatly appreciated!

 http://bugs.digium.com/view.php?id=6119

--
Bird's The Word Technologies, Inc.
http://www.btwtech.com/
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RE: [Asterisk-Users] Problem with date time on Aastra 480isincerelease 1.3

Thanks, so would I be correct in assuming

time zone name: UK-London
time zone code: GMT
time zone minutes: 0

And will this have any affect on the daylight savings in march?

Regards

Lee

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dave
Cotton
Sent: 03 January 2006 14:24
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Problem with date  time on Aastra
480isincerelease 1.3

On Tue, 2006-01-03 at 14:13 +, Lee Archer wrote:
 Does anyone know whether there is some sort of time zone option?  I've

 emailed Aastra who didn't come back to me.  I would like to set the 
 time zone - e.g. Britain-London, in the cfg files so I don't have to 
 set it on 40 phones...
 
in aastra.cfg

time server disabled: 0
time server1: 192.168.1.253
time format: 1
date format: 0
time zone name: FR-Paris
time zone code: CET
time zone minutes: 60

works for me.
--
Dave Cotton [EMAIL PROTECTED]

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[Asterisk-Users] E1 with CAS but no call signalling?

2006-01-03 Thread Tony Mountifield

I'm investigating an application which the client says uses an E1 trunk
with 30 voice channels and a D channel on 16 as normal, but without any
call signalling on the D channel.

In other words, as soon as I originate an outgoing call to a Zap channel
on the E1, the call immediately succeeds (is considered answered) and
passes audio in and out on the specified channel. Obviously there will be
no such thing as an incoming call or a remotely initiated hangup.

Is this possible using one of the existing signalling types? I don't
understand the meaning of many of the types listed in zapata.conf.

If not, does anyone have any pointers on what would be required to
add a new signalling type with the behaviour described?

Thanks in advance
Tony
-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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Re: [Asterisk-Users] Realtime Multiple Asterisk boxes, iaxusers


Rich Adamson wrote:

If you take the word dynamic out of that, then can he effectively 
have primary/secondary/backup systems that allows the user to

re-register and/or redial his call on a different * server?


I don't understand the question.

'dynamic' is used for registrations; if the peer is not dynamic, then 
registration is not needed (nor allowed). Thus, there is no 
're-register' possible.


For IAX2, and SIP in 'type=user' mode, placing outbound calls via 
multiple servers would work fine, since the information required to 
support that is static (not changed by Asterisk itself).

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Re: [Asterisk-Users] Cell phone dock/switch as Asterisk FXO source


Matt Riddell wrote:

Morel Mosolff wrote:


Dear friends and business associates,

I will be out of office until January the 12th, 2006.
With kind regards,

Morel Mosolff



H1 more of these and I will start a loop on a spare high bandwidth
server :)


This person has been unsubscribed.
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[Asterisk-Users] machine load (was best dell a long time ago)

2006-01-03 Thread Simone Cittadini



(with no agi and transcoding) 80 alaw concurrent calls , cdr_mysql, 
terminating on one TE410

Mem:   3105772k total,   733928k used,  2371844k free,8k buffers
Cpu(s):   5.0% user,   5.5% system,   0.0% nice,  89.5% idle
load average: 0.37, 0.39, 0.41




So that is ~80 calls per GB of ram which is 20% of 400 users so that 
should be 5 or 6GB to handle 100% usage.


The load avg is the most important here.  You want to keep it under 
1.00 or you have processes waiting which increases jitter.  Your 
system will be at 80% usage with 160 calls, assuming linear scaling.


What are the specs for processor, memory and chipset that you pulled 
this stat from?

___


xeon 3 Ghz, kernel 2.4
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RE: [Asterisk-Users] Problem with date time on Aastra 480isincerelease 1.3

Still no joy, if I set my phone to a different time zone then reboot it
isn't being updated to use London. 

Lee

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Pete
Barnwell
Sent: 03 January 2006 14:30
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Problem with date  time on Aastra
480isincerelease 1.3

On Tue, 2006-01-03 at 14:13 +, Lee Archer wrote:
 Does anyone know whether there is some sort of time zone option?  I've

 emailed Aastra who didn't come back to me.  I would like to set the 
 time zone - e.g. Britain-London, in the cfg files so I don't have to 
 set it on 40 phones...


time zone name: GB-London
time zone code: GMT
time zone minutes: 60

Rgds

Pete

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Re: [Asterisk-Users] E1 with CAS but no call signalling?


Tony Mountifield wrote:


In other words, as soon as I originate an outgoing call to a Zap channel
on the E1, the call immediately succeeds (is considered answered) and
passes audio in and out on the specified channel. Obviously there will be
no such thing as an incoming call or a remotely initiated hangup.

Is this possible using one of the existing signalling types? I don't
understand the meaning of many of the types listed in zapata.conf.


I don't think so. All of them assume some sort of coordinated signaling 
between the two ends.


If the other end is going to ignore the signaling bits anyway, then you 
might be able to use one of the simpler modes like FXSLS... it'd be 
worth a try anyway.

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[Asterisk-Users] Question on SPA-2002

2006-01-03 Thread Matt

Hi,
I have about 53 SPA-2002 units out in the field.   I've seen two or
three of them, now, exhibit an odd happening.

Users plug their phones into LINE1 (unless they have two lines).   The
two users I've had issues with are both employees here who are fairly
knowledgeable in computers.  They both were using portable phones and
that was the only phone they used (so no trying to back feed the house
or anything).

Both of them started using the service with the device around October
of this past year (2005).

Just recently both of them have come to me and said they could place
outbound calls just fine, however inbound calls go to voicemail.  I
did some looking and the ATA is sending a Busy message back to
asterisk. (Even when the phone is on hook).. (Even when the phone is
unplugged!!).

I moved the config and the phone to LINE_2 and all is well.   Any
thoughts on this?
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Re: [Asterisk-Users] Cell phone dock/switch as Asterisk FXO source

2006-01-03 Thread bbench

On Tuesday 03 January 2006 15:37, Noah Swint wrote:

 Do you have a url for the device?
http://cyber-telecom.net/store/index.php?cPath=1


 From: [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] Cell phone dock/switch as Asterisk FXO
  source Date: Tue, 3 Jan 2006 11:02:30 +0200
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 On Tuesday 03 January 2006 05:48, Paul Dugas wrote:
   On Mon, 2006-01-02 at 16:06 +, Jonathan Attwood wrote:
I use a Dock-n-Talk in conjuction with a Sipura SPA3000  Asterisk.
  
   Does this unit require any funky dialing when placing outbound calls
   from * through the phone?  Do the docs indicate operation is any
   different between CDMA, TDMS, AMPS, or GSM phones?  I'd guess not or,
   if so, it was simple to handle it in the dialplan but I'm curious
   anyway. I've been considering this as a way to have work calls that
   come to my cell appear different to the server.  At the moment, I have
   my GSM phone forward calls to the house when it's off so I can't really
   tell between them.
 
 I have good experience with a GSM-box I've bought from cybertelecom and
 SPA3000. GSM-box acts as a Dock-n-Talk because is it allows in and out
 dialing. The advantage is that one doesn't need even a mobile phone, but
 only
 a SIM card. The whole thing is like porting a number.
 
 There are 2 FXS ports. One could go to an ordinary phone, the other to
 SPA3000.
 
 The disadvantage is that you have one more number for your friends to
 remember. Otherwise is stable, and as-easy- as-PnP instalation, if you
 don't
 forget to disable the pin lock as I did :-)
 benchev
 
 
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Re: [Asterisk-Users] Problem with date time on Aastra 480isincerelease 1.3

2006-01-03 Thread Tzafrir Cohen

On Tue, Jan 03, 2006 at 02:38:15PM -, Lee Archer wrote:
 Thanks, so would I be correct in assuming
 
 time zone name: UK-London
 time zone code: GMT
 time zone minutes: 0
 
 And will this have any affect on the daylight savings in march?

Those are part of the definition of the timezone.

  zdump -v /etc/localtime | grep 2006

-- 
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il |   | a Mutt's  
[EMAIL PROTECTED] |   |  best
ICQ# 16849755 |   | friend

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RE: [Asterisk-Users] GXP-2000 fw 1.0.1.13 and NTP

I had a problem which I spoke to Grandstream about.  It seemed that
around 7 seconds in it goes for time sync and if it fails it doesn't
retry.  This problem was highlighted by the .12 firmware and a Windows
DHCP server we were using.  Upon moving to a Linux DHCP server the
process was much quicker and NTP worked.  However there isn't an auto
DST mode  This upset a lot of people here where I work as all the
clocks were wrong.  Shame is these are reasonably cheap and fairly
descent phones but we are now moving towards the Aastra range.

I've tried out .13 and NTP worked fine. 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Peter
Bowyer
Sent: 31 December 2005 10:35
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] GXP-2000 fw 1.0.1.13 and NTP

Hi all

Slightly OT but I know a lot of GS experts hang out here - I just
upgraded a GXP-2000 to firmware 1.0.1.13 to try out the BLF
functionality with Asterisk (which so far works as expected), but as a
side-effect the phone won't sync with an NTP server - I've tried
different server names (time.nist.gov and
pool.ntp.org)  and IPs in the config, but it refuses to update the time
on the display.

Anyone heard of this? Any workarounds (other than go back to 1.0.1.12) ?

(Hmmm.. just regressed to 1.0.1.12 and it's still not working -
curiouser and curiouser said Alice...)

Thanks

Peter 

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RE: [Asterisk-Users] Problem with date time on Aastra480isincerelease 1.3

Actually it worked, but only after I defaulted all the settings on the
phone and let it pick the config up fresh.

Anyone know if there is any headset config options to default to
headset/speaker?

Thanks

Lee 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Lee Archer
Sent: 03 January 2006 14:49
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: RE: [Asterisk-Users] Problem with date  time on
Aastra480isincerelease 1.3

Still no joy, if I set my phone to a different time zone then reboot it
isn't being updated to use London. 

Lee

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Pete
Barnwell
Sent: 03 January 2006 14:30
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Problem with date  time on Aastra
480isincerelease 1.3

On Tue, 2006-01-03 at 14:13 +, Lee Archer wrote:
 Does anyone know whether there is some sort of time zone option?  I've

 emailed Aastra who didn't come back to me.  I would like to set the 
 time zone - e.g. Britain-London, in the cfg files so I don't have to 
 set it on 40 phones...


time zone name: GB-London
time zone code: GMT
time zone minutes: 60

Rgds

Pete

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[Asterisk-Users] Asterisk realtime mysql connection

2006-01-03 Thread Sig Lange

vmail*CLI realtime mysql status 
Jan 3 10:14:20 ERROR[13666]: res_config_mysql.c:623
mysql_reconnect: MySQL RealTime: Failed to reconnect. Check debug for
more info.
vmail*CLI realtime mysql status
Connected to [EMAIL PROTECTED], port 3306 with username voicemail for 2 days, 17 hours, 15 minutes, 37 seconds.
vmail*CLI realtime mysql status
Connected to [EMAIL PROTECTED], port 3306 with username voicemail for 8 seconds.
vmail*CLI 
vmail*CLI realtime mysql status
Connected to [EMAIL PROTECTED], port 3306 with username voicemail for 11 seconds.
All of which happened in a few seconds. The part that
really gets me is one second it says failed to reconnect, then says
it's connected for 2 days, then for only 8 seconds. After the 8 second
line, it seems to be working correctly.

Surely there is code to reconnect on a failed attempt but perhaps those counters aren't reset.

Anyone experiencing this problem?
-- Sig Lange

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Re: [Asterisk-Users] Question on SPA-2002

2006-01-03 Thread C F

Could be they pressed the DND code for the SPA (don't remember by
heart what it is, something like *xx). The easiest way to check is to
log into the http server of the SPA and check the status on the first
page.

On 1/3/06, Matt [EMAIL PROTECTED] wrote:
 Hi,
 I have about 53 SPA-2002 units out in the field.   I've seen two or
 three of them, now, exhibit an odd happening.

 Users plug their phones into LINE1 (unless they have two lines).   The
 two users I've had issues with are both employees here who are fairly
 knowledgeable in computers.  They both were using portable phones and
 that was the only phone they used (so no trying to back feed the house
 or anything).

 Both of them started using the service with the device around October
 of this past year (2005).

 Just recently both of them have come to me and said they could place
 outbound calls just fine, however inbound calls go to voicemail.  I
 did some looking and the ATA is sending a Busy message back to
 asterisk. (Even when the phone is on hook).. (Even when the phone is
 unplugged!!).

 I moved the config and the phone to LINE_2 and all is well.   Any
 thoughts on this?
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Re: [Asterisk-Users] Where is the Prefix() application in Asterisk 1.2.1 ?

2006-01-03 Thread C F

9 more days to go.

On 1/3/06, Morel Mosolff [EMAIL PROTECTED] wrote:

 Dear friends and business associates,

 I will be out of office until January the 12th, 2006.
 With kind regards,

 Morel Mosolff
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[Asterisk-Users] Where is the Prefix() application in Asterisk 1.2.1 ?

2006-01-03 Thread Bukoka Budoka


Hi ,

thank you for your answer,

If Prefix() command is only in CVS-HEAD, then how can you prepend leading 
digits in a stable version?


It does not make any sense not to have this feature in a version downloaded 
from thw Digium FTP site...


Budoka.

_
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Re: [Asterisk-Users] Three Way Calling with HFC PCI Card

2006-01-03 Thread Giovanni Miano

Use meetme appCheers, Giovanni Miano2006/1/3, Henry Margies [EMAIL PROTECTED]:
Hello,I'm using a TDM 400 together with two HFC PCI Cards in my Box. Three WayCalling with a SIP or analog Phone is working perfectly.But if I try to do Three Way Calling with my ISDN Phone I get an error
message: Facility Name requested on channel 0/2 not in use on span 1I use bristuff with my HFC card and don't know why I get this message?I'm using still asterisk 1.0 and can not update to the newest version at
the moment. Is there a simple trick to make it work or is this problemalready solved in asterisk 1.2?Thanks in advance,Henry--Hi! I'm a .signature virus! Copy me into your~/.signature to help me spread!
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[Asterisk-Users] IAXTEL??

2006-01-03 Thread Kerry Garrison

Is IAXTEL still around? I needed to call Digium and figured I would set it
up to save some miinutes when talking to them but I can't get it to
register.

-Kerry


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[Asterisk-Users] Anyone heard of this company? http://www.affinityvoiptelecom.com/

2006-01-03 Thread Matt

Does anyone know anything about this company?
http://www.affinityvoiptelecom.com/

They claim to offer 911 routing and PS/ALI updates, etc.
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Re: [Asterisk-Users] Question on SPA-2002

2006-01-03 Thread Matt

That's possible, and I didn't think about that.(I wil check). however
I did totally wipe the configuration on the device *** RESET and then
reprogrammed it and the same problem happened, so I kind of doubt that
was the issue.

On 1/3/06, C F [EMAIL PROTECTED] wrote:
 Could be they pressed the DND code for the SPA (don't remember by
 heart what it is, something like *xx). The easiest way to check is to
 log into the http server of the SPA and check the status on the first
 page.

 On 1/3/06, Matt [EMAIL PROTECTED] wrote:
  Hi,
  I have about 53 SPA-2002 units out in the field.   I've seen two or
  three of them, now, exhibit an odd happening.
 
  Users plug their phones into LINE1 (unless they have two lines).   The
  two users I've had issues with are both employees here who are fairly
  knowledgeable in computers.  They both were using portable phones and
  that was the only phone they used (so no trying to back feed the house
  or anything).
 
  Both of them started using the service with the device around October
  of this past year (2005).
 
  Just recently both of them have come to me and said they could place
  outbound calls just fine, however inbound calls go to voicemail.  I
  did some looking and the ATA is sending a Busy message back to
  asterisk. (Even when the phone is on hook).. (Even when the phone is
  unplugged!!).
 
  I moved the config and the phone to LINE_2 and all is well.   Any
  thoughts on this?
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RE: [Asterisk-Users] IAXTEL??

From:
http://www.iaxtel.com/

The IAXTel Server is currently under maintenance. Some technical
difficulties, such as connection timeouts, registration timeouts, and the
inability to make phone calls may be experienced. Thank you for your
patience.




:(

b

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kerry Garrison
Sent: Tuesday, January 03, 2006 5:55 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] IAXTEL??

Is IAXTEL still around? I needed to call Digium and figured I would set it
up to save some miinutes when talking to them but I can't get it to
register.

-Kerry


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Re: [Asterisk-Users] Howto compile chan_h323 on macosx 10.3?

2006-01-03 Thread Moises Silva

Hi Antonio. h323 support is composed from several versions and
packages, including compatibility between asterisk versions and
asterisk-oh323 is important. I guess more people will be able to help
you if you privide more info.

Kind Regards

On 1/3/06, Antonio Marquez [EMAIL PROTECTED] wrote:
 I can not compile the h323 support for macosx 10.3?
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Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org;
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Re: [Asterisk-Users] snom Firmware 5.0.

2006-01-03 Thread Joe Pukepail

I agree, I liked the old ringtone 2 also (just abeep), I use it at my desk, If I'm there I can pick it up and it wasn't obnoxious enough to disturb others. Please email it to me if you get it in the format needed.

On 1/2/06, Remco Barende [EMAIL PROTECTED] wrote:
Hi Usman,Thanks for the explanation.Could you make the old Ringer 2 available in some form, preferable
already in the format the phone understands?That would solve the problem too :)Thanks!!RemcoOn Mon, 2 Jan 2006, Usman Tahir wrote: Hi Remco, Old Ringer 2 is not there on the phone anymore, perhaps you can use another ring melody or a suitable custom melody:
The wav file itself should be a PCM encoded 8 KHz file at 16bit mono. The time for loading the file should not be longer then 3 seconds ! And the size should be below 250KB. To create this format from mp3:
/usr/bin/mpg123 -m -r 8000 -w - -n 190 -q test.mp3 test.wav To convert an existing WAV file: sox GENERIC.wav -c 1 -r 8000 -w SNOM.wav * The -c 1 flag makes the output mono.
* The -r 8000 flag makes the output a 8kHz sample. * The -w flag uses 16 bits (word) per sample. Regards, Usman. -
Usman Tahir snom technology AG Gradestraße 46 D-12347 Berlin. Tel: +49 30 398330 Fax: +49 30 39833111 [EMAIL PROTECTED]
www.snom.com This e-mail may contain confidential and/or privileged information. If you are not the intended recipient (or have received this e-mail in error) please notify the sender immediately and destroy this e-mail. Any unauthorized copying, disclosure or distribution of the material in this e-mail is strictly forbidden.
Diese E-Mail könnte vertrauliche und/oder rechtlich geschützte Informationen enthalten. Wenn Sie nicht der richtige Adressat sind oder diese E-Mail irrtümlich erhalten haben, informieren Sie bitte sofort den Absender und vernichten Sie diese Mail. Das unerlaubte Kopieren sowie die unbefugte Weitergabe dieser Mail sind nicht gestattet.
- -Original Message- From: Remco Barende [mailto:[EMAIL PROTECTED]
] Sent: Monday, January 02, 2006 2:29 PM To: Usman Tahir Cc: Asterisk Users List Subject: Re: [Asterisk-Users] snom Firmware 5.0. Thanks for the new firmware, finally some of the features are becoming available that make the phone more usable with Asterisk.
One question though, ringer tone #2 on the Snom 360 firmware has been replaced? How can I get the old ringtone back? I was using the ringtone on phones in locations like meeting rooms. The ringtone wasn't intrusive at all, yet well audible. Now when a phone rings everybody is disturbed with a loud noise.
Thanks! Remco On Thu, 22 Dec 2005, Usman Tahir wrote: Hi, Snom phones firmware 5.0 is now out. Try it if you like:
http://www.snom.com/wiki/index.php/Main_Page. Regards, - Usman Tahir snom technology AG
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[Asterisk-Users] Re: Where is the Prefix() application in Asterisk1.2.1 ?

2006-01-03 Thread Steven

Just do:
exten = _12xx,2,Dial(${TRUNK}/0${EXTEN}|30,r) ; adding zero
exten = _012xx,2,Dial(${TRUNK}/${EXTEN}|30,r) ; not adding zero
The zero is added before ${EXTEN}.

I have only ever used the stable versions and have always done it this way.

-- 
-- 
Steven

May you have the peace and freedom that come from abandoning all hope of having 
a better past.
----  ---  - - -   -- -   -   --  - - - --- - --   - - 
--- - - -- -  -- --   -   --
Bukoka Budoka [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED]
 Hi ,

 thank you for your answer,

 If Prefix() command is only in CVS-HEAD, then how can you prepend leading 
 digits in a stable version?

 It does not make any sense not to have this feature in a version downloaded 
 from thw Digium FTP site...

 Budoka.

 _
 Don't just search. Find. Check out the new MSN Search! 
 http://search.msn.click-url.com/go/onm00200636ave/direct/01/

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Re: [Asterisk-Users] IAXTEL??

2006-01-03 Thread Tom Vile

That message has been there for months.

On 1/3/06, Bogdan Moldovan [EMAIL PROTECTED] wrote:
 From:
 http://www.iaxtel.com/

 The IAXTel Server is currently under maintenance. Some technical
 difficulties, such as connection timeouts, registration timeouts, and the
 inability to make phone calls may be experienced. Thank you for your
 patience.

 


 :(

 b

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Kerry Garrison
 Sent: Tuesday, January 03, 2006 5:55 PM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: [Asterisk-Users] IAXTEL??

 Is IAXTEL still around? I needed to call Digium and figured I would set it
 up to save some miinutes when talking to them but I can't get it to
 register.

 -Kerry


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--
Tom Vile
Baldwin Technology Solutions, Inc
Consulting - Web Design - VoIP Telephony
www.baldwintechsolutions.com
Phone: 518-631-2855 x205
Fax: 518-631-2856
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[Asterisk-Users] AEL - Using a Macro in the Dial Command in AEL

2006-01-03 Thread John Melody


I cannot get the following to work in an AEL script on 1.2.1 

Dial(mynumber,timeout,M(mymacro))

Does anyone know if the Macro construction used above is supported in AEL? 

or should I use 

Dial(mynumber,timeout,mymacro) 



John Melody 
SyberNet Ltd. 
Galway Business Park, 
Dangan, 
Galway. 
Tel. No. +353 91 514400 
Fax. NO. +353 91 514409 
Mobile - 087-2345847 
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RE: [Asterisk-Users] IAXTEL??

2006-01-03 Thread Kerry Garrison

Yeah, saw that, and it had said that for like six months if I recall. You
would figure that since Digium features IAXTEL phone numbers so prominently,
that it would be a service that was actually capable of connecting to them.
-Kerry
 

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Bogdan Moldovan
 Sent: Tuesday, January 03, 2006 8:01 AM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: RE: [Asterisk-Users] IAXTEL??
 
 From:
 http://www.iaxtel.com/
 
 The IAXTel Server is currently under maintenance. Some 
 technical difficulties, such as connection timeouts, 
 registration timeouts, and the inability to make phone calls 
 may be experienced. Thank you for your patience.
 
 
 
 
 :(
 
 b
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Kerry Garrison
 Sent: Tuesday, January 03, 2006 5:55 PM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: [Asterisk-Users] IAXTEL??
 
 Is IAXTEL still around? I needed to call Digium and figured I 
 would set it up to save some miinutes when talking to them 
 but I can't get it to register.
 
 -Kerry
 
 
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RE: [Asterisk-Users] Question on SPA-2002

2006-01-03 Thread Damon Estep

If the ata is sending busy here sip response back to asterisk it IS
most likely a DND or other call redirect setting that was user
programmed at the ATA.

I have seen the Linksys/sipura ATA retain USER settings when ADMIN
settings are reset to default with certain firmware versions.

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Matt
 Sent: Tuesday, January 03, 2006 8:58 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] Question on SPA-2002
 
 That's possible, and I didn't think about that.(I wil check). however
 I did totally wipe the configuration on the device *** RESET and then
 reprogrammed it and the same problem happened, so I kind of doubt that
 was the issue.
 
 On 1/3/06, C F [EMAIL PROTECTED] wrote:
  Could be they pressed the DND code for the SPA (don't remember by
  heart what it is, something like *xx). The easiest way to check is
to
  log into the http server of the SPA and check the status on the
first
  page.
 
  On 1/3/06, Matt [EMAIL PROTECTED] wrote:
   Hi,
   I have about 53 SPA-2002 units out in the field.   I've seen two
or
   three of them, now, exhibit an odd happening.
  
   Users plug their phones into LINE1 (unless they have two lines).
The
   two users I've had issues with are both employees here who are
fairly
   knowledgeable in computers.  They both were using portable phones
and
   that was the only phone they used (so no trying to back feed the
house
   or anything).
  
   Both of them started using the service with the device around
October
   of this past year (2005).
  
   Just recently both of them have come to me and said they could
place
   outbound calls just fine, however inbound calls go to voicemail.
I
   did some looking and the ATA is sending a Busy message back to
   asterisk. (Even when the phone is on hook).. (Even when the phone
is
   unplugged!!).
  
   I moved the config and the phone to LINE_2 and all is well.   Any
   thoughts on this?
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RE: [Asterisk-Users] IAXTEL??

I know, this is the sad part :(
b 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tom Vile
Sent: Tuesday, January 03, 2006 6:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] IAXTEL??

That message has been there for months.

On 1/3/06, Bogdan Moldovan [EMAIL PROTECTED] wrote:
 From:
 http://www.iaxtel.com/

 The IAXTel Server is currently under maintenance. Some technical 
 difficulties, such as connection timeouts, registration timeouts, and 
 the inability to make phone calls may be experienced. Thank you for 
 your patience.

 


 :(

 b

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Kerry 
 Garrison
 Sent: Tuesday, January 03, 2006 5:55 PM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: [Asterisk-Users] IAXTEL??

 Is IAXTEL still around? I needed to call Digium and figured I would 
 set it up to save some miinutes when talking to them but I can't get 
 it to register.

 -Kerry


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--
Tom Vile
Baldwin Technology Solutions, Inc
Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com
Phone: 518-631-2855 x205
Fax: 518-631-2856
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[Asterisk-Users] FC3 or FC1 (or something else?)

2006-01-03 Thread Brett, Gary

Hi

I wish to install asterisk 1.2 (the latest tar.gz from the site not the
CVS version) on an HP box with a TE110P (single port E1/T1)

My question is which OS would be preferred in this configuration Fedora Core
1 or Fedora Core 3, and are there any install guides out there that are
recent enough for asterisk 1.2

I am also open to suggestions for other Operating Systems if any of you feel
that FC1/3 are not the best for the job, my only definates are that I use
the latest tar.gz from the asterisk.org website not the CVS and also that I
will be using the TE110p

Any help would be greatly appreciated
Gary
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Re: [Asterisk-Users] IAXTEL??

2006-01-03 Thread Ariel Batista


Iaxtel has been down for some time now.

But to get in contact with digium via your asterisk box all you need is to 
set this dialing rule up.


exten = 500,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]) ;Call Digium
exten = 500,2,Congestion

Kerry Garrison wrote:

Is IAXTEL still around? I needed to call Digium and figured I would
set it up to save some miinutes when talking to them but I can't get
it to register.

-Kerry


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[Asterisk-Users] Sipura SPA-1001 question

2006-01-03 Thread burke

Asterisk-Users,

Is anyone out there using the SPA-1001 for integrating existing analog
phones into a VoIP setup? My question has to do with the MWI. From the
datasheet it says that it provides MWI Tones, and then that it provides
Visual MWL via FSK. What does via FSK mean? My exsting phone has an
answering machine built in and I am debating using Asterisk as the
Voicemail, or just the exsting answering machine. Any comments or insight
into the SPA-1001 would be appreciated.

Thanks,
Ryan
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Re: [Asterisk-Users] IAXTEL??

2006-01-03 Thread Rich Adamson

 Is IAXTEL still around? I needed to call Digium and figured I would set it
 up to save some miinutes when talking to them but I can't get it to
 register.

That hasn't worked for many many months.

Much easier to reach digium by using the Demo that is/was installed in
all asterisk installs. When the voice prompt indicates its connecting
to a demonstation server at digium, it is a real * server that can
connect you to tech support, etc, etc. Try it.


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RE: [Asterisk-Users] FC3 or FC1 (or something else?)

IMHO use FC4.

Also after the install of the OS and all the required packages do a 'yum
update'.

Bogdan Moldovan
MODULO Consulting
The Future Is Not What It Used To Be
http://www.modulo.ro 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brett, Gary
Sent: Tuesday, January 03, 2006 6:34 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] FC3 or FC1 (or something else?)

Hi

I wish to install asterisk 1.2 (the latest tar.gz from the site not the
CVS version) on an HP box with a TE110P (single port E1/T1)

My question is which OS would be preferred in this configuration Fedora Core
1 or Fedora Core 3, and are there any install guides out there that are
recent enough for asterisk 1.2

I am also open to suggestions for other Operating Systems if any of you feel
that FC1/3 are not the best for the job, my only definates are that I use
the latest tar.gz from the asterisk.org website not the CVS and also that I
will be using the TE110p

Any help would be greatly appreciated
Gary
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[Asterisk-Users] Re IAXTEL

2006-01-03 Thread Dave Cotton

For those bemoaning the lack of IAXTEL and wanting to contact Digium
what's wrong with:-


exten = ${DIGIUM},1,Dial(IAX2/[EMAIL PROTECTED])

worked 2 minutes ago.
-- 
Dave Cotton [EMAIL PROTECTED]

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Using *RT for HA purposes was: [Asterisk-Users] Realtime Multiple Asterisk boxes, iaxusers

2006-01-03 Thread Mike Fedyk


Kevin P. Fleming wrote:


Rich Adamson wrote:

If you take the word dynamic out of that, then can he effectively 
have primary/secondary/backup systems that allows the user to

re-register and/or redial his call on a different * server?



I don't understand the question.


I don't know if it was Rich's intention, but I'm interested in using RT 
for HA (High Availability) purposes.


Think of this scenario: You have two * RT servers running heartbeat and 
one goes down.  If the SIP registration information was kept in the DB 
tables, the backup server could take over the ethernet and IP addresses 
and continue without forcing the phones to re-register.

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Re: [Asterisk-Users] Asterisk PRI problems.

2006-01-03 Thread William Lloyd


It sounds like it might be dialplan instead of PRI related.

-bill

On 2-Jan-06, at 10:43 AM, Kristian Larsson wrote:


On Mon, Jan 02, 2006 at 03:36:57PM +0200, [EMAIL PROTECTED] wrote:



On Mon, 2 Jan 2006, Kristian Larsson wrote:


I have an Avaya IP Office PBX connected to an
Asterisk system via a Sangoma ISDN PRI card.
Dialing from the as
terisk system into the avaya works just fine but
when trying to call from a phone connected to the
avaya syste
m something goes wrong. After punching the first
four digits the Avaya calls out, shouldn't it wait
for all di
gits and then dial out?
If I try to dial a three digit number it waits for
a while then dials.

Is this some feature to let the CO know of which
area code the calls is going ahead of time?
Is there some way to circumvent this using hacks
on the asterisk side?



Looks like you need to enable overlapdial=yes on the Asterisk  
side.  It
will then wait for additional digits sent from the Avaya after the  
initial

ones sent with the SETUP.

I did try enabling overlapdial=yes but I saw no
real change. Is there any other variable to go
with it that I might need to tune?

I am quite new to the whole PRI thing. What does
it do when setting up a call?

First a SETUP and after that it dials?

Regards,
Kristian
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[Asterisk-Users] GSM Gateway / Terminal for sale

2006-01-03 Thread Sam Tam



Single port GSM Gateway support 900 / 1800 GSM mode with external antenna.

Brand new unit and all of them will be tested before dispatch.

Extremely easy to setup and can be used out of the box without any
configuration. So should be good alternatively of phonecell or nokia pbx
etc..

Units are located in UK and £60 GBP per unit excluding shipping.

I have limited stock therefore please act quick to avoid disappointment 

Working mode: GSM 900 MHz or GSM 1800MHz double frequency 
Peak power: 2 W
Power consume: static state 25mA, launch 600mA
Sencitivity:-104dB
Inner pressure :DC 12V/1.5A
Condition temperature:0C~+40C
Working humidity:45%-90%
Atmosphere pressure:86~106Pka
Circumstance noise:60 dB
Wireless decibel :3.5dB or 12dB
AC power:220V ac+-10%,frequency 47-54Hz;110Vac/60Hz(optional)
Power port: China, USA, UK, (by customer s optional)
Connection means:RJ-11 telephone line plug
Antenna connection: SMA antenna tie-in, N type port(optional).TNC
port(optional)

For more info please email gsm AT cyber-telecom.net for more info or visit
www.cyber-telecom.net to purchase right away.

Sam


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Re: [Asterisk-Users] asterisk AVM C2 again

2006-01-03 Thread stéphane plichon

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

happy new year, evrething work now :-D
the error came from France Telecom


thanks everybody

- --
Stephane Plichon | HASGARD
tel: +33 (0)472529881
fax: +33 (0)472177764
web: http://www.hasgard.net
email: [EMAIL PROTECTED]
jabber: [EMAIL PROTECTED]
~
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.1 (GNU/Linux)

iD8DBQFDuq/pMI/jEEfAy/4RApbhAJ9uG0EYuwaG0uFRc5uP9h2HosPJYQCgyyCw
HwffC7Kc8/iMWC5QzCQx0dw=
=Uhml
-END PGP SIGNATURE-
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RE: [Asterisk-Users] Regular Crashes

2006-01-03 Thread Andrew Gough

By hints do you mean comments??

Seems a very odd solution, but I'm willing to give anything a go. 

Regards
 
Andrew Gough
Senior Partner
 
GCD Technologies
Unit 414
Lisburn Enterprise Park
Ballinderry Road
Lisburn
Co Antrim
BT28 2BP
 
E:  [EMAIL PROTECTED] 
W: www.gcdtech.com 
T:  028 9264 1144
 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Paradise Dove
 Sent: 02 January 2006 14:13
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] Regular Crashes
 
 i have the same problem. but when i remove all hints from my dialplan
 in extensions.conf.
 on more crash will occur.
 
 Paradise Dove
 
 On 1/2/06, Andrew Gough [EMAIL PROTECTED] wrote:
  I don't think this is the same problem I am experiencing. As you can
see
 below the two BT's are almost identical and I have others the same
too. so
 the fault is fairly consistent, unfortunately I have been unable to
 determine the exact reason for it yet. It is not the whole box
crashing it
 is merely Asterisk core dumps. sometimes in the middle of a call and
 sometimes when there is no-one even in the office. Unless I get
solution
 soon I'll be forced to give up on asterisk, which would be a real
shame.
 
  Regards
 
  Andrew
 
  
 
  From: [EMAIL PROTECTED] on behalf of Zafer
Khodr
  Sent: Fri 30/12/2005 15:32
  To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
  Subject: RE: [Asterisk-Users] Regular Crashes
 
 
 
  I have been experiencing a similar problem.
 
  I have not yet been able to figure out what the exact problem is but
I
 know that the errors are inconsitant.
 
  Sometimes nothing for 2 days and sometimes 5 times a day.
 
 
 
  I thought about it a lot and I have found only one thing in common.
 
 
 
  The area where my server is stored gets pretty stuffy, especially on
a
 hot day.
 
 
 
  I occasionally turn on the aircon as I need to go in and do some
work.
 
  From my best recollection the server has never crashed when the
aircon
 has been on.
 
  This is my third day of testing my theory, and with the aircon
 controlling the room tempreture to make sure it is always nice and
cool in
 there I have not seen any errors for 3 days (Keeping in mind that the
day
 I decided to try this theory by constantly keeping the room cool my
server
 encountered around 4 errors in just a few hours).
 
 
 
  So to put in short I think but cant be sure that somehow when the
room
 gets too hot the server goes awol and somehow causes this error.
 
  Don't ask me how or why... all I know is that now with controlled
room
 temp I have not had a problem.
 
 
 
  Good Luck
 
 
 
 
 
  
 
  From: [EMAIL PROTECTED]
[mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Andrew Gough
  Sent: Saturday, 31 December 2005 1:43 AM
  To: asterisk-users@lists.digium.com
  Subject: [Asterisk-Users] Regular Crashes
 
 
 
  I have just setup asterisk on a debian sarge box. I am running
Asterisk
  1.21 with AMP and chan_capi_cm 0.6.1  using a BT Speedway (AVM
Fritz)
  ISDN card, connected to a BT ISDN2e line. Currently we have 6
extensions
  (SIP) configured all using CounterPath(Xten) eyebeam softphone.
 
  After many hours of Googling I have finally got it all setup and
  working. We can transfer calls internally and make and receive
external
  calls. Its all great except for stability issues!!
 
  Essentially  every now and again, asterisk simply dies (2-3 times a
  day). No warning, no error, just my console session outputs a
  disconnected from console message.
 
  Sometimes the crashes happen when you are on a call, other times
when
  there is no-one in the office.
 
  The server is a brand new AMD 3400+ with 512Mb RAM. The other issue
  experienced is occasional break up on inbound sound quality.
 
  Below are traces of the last two crashes
 
  Any Help much appreciated
 
  Regards
 
  Andrew Gough
 
  FIRST TRACE
 
  #0  0x400268b7 in pthread_mutex_trylock () from
/lib/tls/libpthread.so.0
  No symbol table info available.
  #1  0x0806c146 in ast_mutex_trylock (pmutex=0x672e33fc) at
lock.h:597
  No locals.
  #2  0x0806175a in ast_queue_hangup (chan=0x672e3330) at
channel.c:671
  f = {frametype = 4, subclass = 1, datalen = 0, samples = 0,
mallocd = 0, offset = 0, src = 0x0, data = 0x0, delivery = {tv_sec
=
  0,
  tv_usec = 0}, prev = 0x0, next = 0x0}
  #3  0x408fc2d9 in __sip_autodestruct (data=0x81be208) at
chan_sip.c:1315
  p = (struct sip_pvt *) 0x81be208
  #4  0x08056c3e in ast_sched_runq (con=0x8172f28) at sched.c:373
  current = (struct sched *) 0x8174868
  tv = {tv_sec = 1135275568, tv_usec = 989877}
  x = 0
  res = 1083432672
  #5  0x40927e28 in do_monitor (data=0x0) at chan_sip.c:11253
  res = 0
  sip = (struct sip_pvt *) 0x0
  peer = (struct sip_peer *) 0x0
  t = 1135275568
  fastrestart = 0

Re: [Asterisk-Users] FC3 or FC1 (or something else?)

2006-01-03 Thread Robert La Ferla


Brett, Gary wrote:

My question is which OS would be preferred in this configuration Fedora Core
1 or Fedora Core 3, and are there any install guides out there that are
recent enough for asterisk 1.2
  

Try Fedora Core 4 (FC4).  Works great.
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[Asterisk-Users] Asterisk on Dell blade servers

2006-01-03 Thread Alistair Cunningham

We've been asked to quote for a large cluster running Asterisk and our 
ITSP in a box product. The system will be SIP throughout, with mixed 
codecs.


We're considering using Dell blade servers, 1855 or similar, on the 
grounds that we normally use Dell machines and they work well, but we 
need higher rack density.


Has anyone used these? Any feedback on whether they're 
good/bad/indifferent? What scalability do you get on simple SIP-SIP 
forwarding either with or without RTP passing through Asterisk?


--
Alistair Cunningham,
Integrics Ltd,
+44 (0)7870 699 479
http://integrics.com/
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Re: [Asterisk-Users] Regular Crashes

2006-01-03 Thread Robert La Ferla

Did you try running * under gdb?  When it crashes, do a bt to get a 
back trace and post it to the mailing list.


e.g.

% gdb /usr/sbin/asterisk
GNU gdb Red Hat Linux (6.3.0.0-1.84rh)
Copyright 2004 Free Software Foundation, Inc.
GDB is free software, covered by the GNU General Public License, and you are
welcome to change it and/or distribute copies of it under certain 
conditions.

Type show copying to see the conditions.
There is absolutely no warranty for GDB.  Type show warranty for details.
This GDB was configured as i386-redhat-linux-gnu...Using host 
libthread_db library /lib/libthread_db.so.1.

(gdb) run

 wait for crash 

(gdb) bt

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Re: [Asterisk-Users] AEL - Using a Macro in the Dial Command in AEL


John Melody wrote:
I cannot get the following to work in an AEL script on 1.2.1 


Dial(mynumber,timeout,M(mymacro))


AEL does not affect the syntax of arguments passed to applications, so 
if this does not work then it is a bug in the AEL parser.

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Re: Using *RT for HA purposes was: [Asterisk-Users] Realtime Multiple Asterisk boxes, iaxusers