[asterisk-users] asterisk 1.8 app_monitor problem

[cfh]

Hi

I have just update my old asterisk 1.4 to 1.8 version and all works good 
but I have a problem with the monitor features.


With the old version this part of dialplan worked without problem :

exten => _90.,1,NoOp(call out interoute REC)
exten => _90.,n,Monitor(wav,${CALLERID(num)}-${EXTEN:2}})
exten => _90.,n,Dial(SIP/provider/${EXTEN},90)


now with 1.8 the same dialplan with the same hw sometimes causes lost 
audio call problem



server: debian 6.0.6

asterisk 1:1.8.11.1-1digium1~squeeze



what can I do ?


best

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Re: [asterisk-users] dahdi and digium debian package

[cfh]

On 02/22/2012 06:09 PM, ml asterisk wrote:

Hi

I'm trying to install dahdi. I just need the dahdi timer for conference.
I currently using digium debian package for asterisk 1.8.8.1.
When i install asterisk-dahdi , i've got several dependencies which came
for official debian repository (including the dahdi package) and are
outdated.
Is it normal than dahdi is not include into digium packages ? Do i have
to compil it before install asterisk-dahdi ?




You can install dahdi kernel modules with the deb package source and the 
debian tool module-assistant



use


apt-get install dahdi-source
m-a a-i dahdi
dpkg -i dahdi-modules-2.6.*.deb


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[asterisk-users] cisco 7945 sip lines on 2 different asterisk servers

[cfh]

hi all


If I try to register a cisco 7945 phone (firmware sip  v 9.2) to a 
asterisk server 1.8.5


I set in the xml file  SEPmac.cnf.xml

   
  IP_ADDRESS_ASTERISK
...

and in the line 1 settings

USECALLMANAGER


it works good .




Now I need to use the phone with 2 different sip lines on 2 different 
servers both with asterisk v.1.8.5


I tried to change with this config :

 
 
9
user1
user1
user1
user1
user1-secret
user1
IP_ADDRESS_ASTERISK1
5060

 

 
9
user2
user2
user2
user2
user2-secret
user2
IP_ADDRESS_ASTERISK2
5060

 




but the phone didn't send sip segnaling

Please help on this..




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[asterisk-users] presence with polycom DND

[cfh]

hi,

I have configured asterisk 1.4.21 to control the presence BLF (hint + 
watch buddy parameter)  of Polycom phones (650,550,330) and it works good.

But when I set the phones on Do Not Disturb (DND) on the server there 
arent sip notifications and the presence doesnt change.

On the Polycom configuration I have try to use the server based DND 
option but i dont know how to use this with asterik.

What can i do ? Are there some workaround to use the DND button and the 
BLF on asterisk?

thanks

cfh

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Re: [asterisk-users] Distinctive Ring for SIP?

[cfh]

> 
> Does anyone have experience with setting distinctive ring in SIP in such 
> a way?
> 

I have done this for internal call with

;;;grandstream;;;
exten => _12X,1,Set(_ALERT_INFO=\;info=internal)
exten => _12X,2,Dial(SIP/${EXTEN},30,tTr)

http://www.grandstream.com/asteriskfaqs.html


;;;linksys;;;
exten => _1XX,1,SetVar(_ALERT_INFO=Simple-1)
exten => _1XX,2,Dial(SIP/${EXTEN},30,Tr)



;;;polycom;;;
exten => s,n,Set(_ALERT_INFO=Custom 1)

or when it doesnt works best solution is

exten => s,n,,SIPAddHeader(Alert-Info: Custom 1)

http://threebit.net/mail-archive/asterisk-users/msg02875.html
http://www.voip-info.org/wiki/view/Polycom+auto-answer+config



LL









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Re: [asterisk-users] call pick up

[cfh]

Tharanga wrote:
> Hi List,
> 
> Iam using asterisk 1.2.14. when someone call to my ip-pbx, receptionist will
> transfer that to specific extension. If the person is not there.(ignore the
> voicemail part) i need to get the call back to receptionist again. so she
> can tell the calling party he is not there or line is busy.
> how should i do this in asterisk ? will call parking work for this ?

if the receptionist use blind transfer you can do this  :

To get call back from blind transfer

use the ${BLINDTRANSFER} variable to callback when no one answer the 
transfered call.

exten => _XXX,1,NoOp(${UNIQUEID} - ${DATETIME} - ${CALLERID} => 
${EXTEN}) ; Add some call tracing
exten => _XXX,2,Dial(SIP/${EXTEN},5,mTt)
exten => _XXX,3,Gotoif($["x${BLINDTRANSFER}"="x"]?4:5)
exten => _XXX,4,Hangup
exten => _XXX,5,Set(tx=${BLINDTRANSFER:0:7})
exten => _XXX,6,Set(CALLERID(all)=RET_${EXTEN} <${CALLERID(num)}>)
exten => _XXX,7,Dial(${tx},,mTt)
exten => _XXX,8,Hangup
;
exten => _XXX,103,Set(tx=${BLINDTRANSFER:0:7})
exten => _XXX,104,Wait(5)
exten => _XXX,105,Set(CALLERID(all)=ERR_${EXTEN} <${CALLERID(num)}>)
exten => _XXX,106,Dial(${tx},,mTt)
exten => _XXX,107,Hangup

example from
http://www.voip-info.org/wiki/view/Asterisk+config+features.conf


my macro for interal call (with custom ring different from external call)

[macro-int]
exten => s,1,Set(_ALERT_INFO=Custom 1)
exten => s,n,Dial(${ARG1},60)
exten => s,n,Gotoif($["x${BLINDTRANSFER}"="x"]?:jump)
exten => s,n,Hangup
exten => s,n(jump),Set(tx=${BLINDTRANSFER:0:7})
exten => s,n,Set(CALLERID(all)=RET_${EXTEN} <${CALLERID(num)}>)
exten => s,n,Dial(${tx},20)
exten => s,n,Hangup



bye

ll

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[asterisk-users] asterisk manager and perl

[cfh]

Hi,

I m trying to use perl script to generate call with a server asterik .
If I use telnet session to generate like this :

$telnet localhost 5038

Action: Login
Username: useroperator
Secret: password

Action: Originate
Context: context
Channel: Local/[EMAIL PROTECTED]
Exten: 221
Priority: 1

it works good :)


instead with a script perl like this :

use Net::Telnet ();

  $tn->print("Action: Login\nUsername:$USERNAME\n Secret:$SECRET\n\n");
  $tn->waitfor('/Authentication accept*/')

  $tn->print("Action: Originate\n");
  $tn->print("Context: context\n");
  $tn->print("Channel: Local/[EMAIL PROTECTED] \n");
  $tn->print("Exten: 221\n");
  $tn->print("Priority: 1 \n\n");


i get this error from asterisk cli :
chan_local.c:498 local_alloc: No such extension/context
[EMAIL PROTECTED]  creating local channel
channel.c:2491 __ast_request_and_dial: Unable to request channel
Local/[EMAIL PROTECTED]
and the call doesnt start.

What can i do to resolve this problem?

Thanks


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Re: [asterisk-users] SER/OpenSER as registrar to Asterisk (1500 SIP users)

[cfh]

> 
> Can anyone please point me in the right direction, provide me with
> OpenSER configuration, or any pointers on the subject. I tried to read
> all the material on how to write configuration files for OpenSER, but it
> is incomprehensible to me, and it is much harder that when I learning
> Asterisk 3 years ago.
> 
> Your help is much appreciated.
> 


http://www.voice-sistem.ro/downloads/2007.08.29-Admin-Course/von-italy-2007_admin-course.zip



LL

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[Asterisk-Users] snom programmable buttons

[cfh]

Hi,

I want to pick up a call with the snom's programmable buttons(snom190 
-SIP 3.60x, snom360-SIP 4.1)  with asterisk server (v 1.2.0), I tried 
with the option 'Destination' and  when the incoming call arrive to 
another snom phone the button blinking.

In this way I can only  pick down it pressing the blinking button.

The solution is call the *8 or parcking the call but my pbroblem is when 
the incoming call are 2 or 3 and I would press a programmable button to 
pick up the calls.


Is possible have configured asterisk and the snom phone with the 
function shared line?


Are there solutions ?


Thanks Luca L. [cfh]
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