AW: [asterisk-users] Cisco 7960

2007-02-27 Thread Roland Ndaka Fru
Hi Carlos,

 

Check out Asterisk LDAP authentication:
http://www.voip-info.org/wiki/index.php?page=Asterisk+LDAP

 

Greetz,

[EMAIL PROTECTED]

 

  _  

Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Im Auftrag von Mohamed A.
Gombolaty
Gesendet: 27 February 2007 13:03
An: Asterisk Users Mailing List - Non-Commercial Discussion
Betreff: Re: [asterisk-users] Cisco 7960

 

Dear Khaled, 

What is the softphone u r using? 

Thx 
MAG 
  

Khaled wrote: 

I am using firmware version pos3-07-500 

Kindly can you provide me with  the basic configuration for cisco ip phone
and asterisk config file 

*I have nat=never at my asterisk config file and nat enabled N0 at cisco
phone  

*I have an out bound proxy ip and port 5060 at cisco phone 

*Voip control port is 5061  

My problem is  my soft phone can call the cisco phone with normal RTP and
Bye message,but my cisco phone cant dial my soft phone. 

Asterisk sends bye message for my soft phone. 

Thanks 



  _  


From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Wireless


Sent: Tuesday, February 27, 2007 12:48 PM 
To: Asterisk Users Mailing List - Non-Commercial Discussion 
Subject: Re: [asterisk-users] Cisco 7960 

can you give a bit more info?  I know that you need nat=never for example

- Original Message - 

From:Khaled mailto:[EMAIL PROTECTED] 

To:'Asterisk Users Mailing List - mailto:asterisk-users@lists.digium.com
Non-Commercial Discussion'

Cc:[EMAIL PROTECTED]

Sent: Tuesday, February 27, 2007 10:03 AM

Subject: [asterisk-users] Cisco 7960

Hi 

I have cisco 7960 connected to asterisk ,using tftp xml config file,my
problem is it can receive any call but it cant call any extension.

Please can you send me ,how to solve this issue  

Regards 

Khaled Chehab 

System Integration Engineer 

Xplorium Offshore. 

Sakiet Al Janzir 

Postal Code: 1102-2080 

Tel: (961) 1- 868 686 

Fax :(961) 1-808 810 

GSM: (961) 3-979 343 



  _  


*


No employee or agent is authorized to conclude any binding agreement on
behalf of Xplorium with another party by e-mail without express written
confirmation by an officer of Xplorium. Any views expressed by an individual
in this electronic message do not necessarily reflect views of Xplorium or
its subsidiaries and associates. 

This electronic message and its attachments are solely addressed to the
addressee(s), and contain confidential information protected from disclosure
belonging to Xplorium. 

If you are not the intended addressee of this electronic message and its
attachments, kindly delete it immediately from your system and notify the
sender by electronic mail. You must not copy this message or attachment or
disclose its content to any other person. 

Xplorium does not guarantee the integrity of this electronic message and any
of its attachments, or that they are free from computer viruses or other
defects. 
* 

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


  _  


___


--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


  _  


* 
No employee or agent is authorized to conclude any binding agreement on
behalf of Xplorium with another party by e-mail without express written
confirmation by an officer of Xplorium. Any views expressed by an individual
in this electronic message do not necessarily reflect views of Xplorium or
its subsidiaries and associates. 

This electronic message and its attachments are solely addressed to the
addressee(s), and contain confidential information protected from disclosure
belonging to Xplorium. 

If you are not the intended addressee of this electronic message and its
attachments, kindly delete it immediately from your system and notify the
sender by electronic mail. You must not copy this message or attachment or
disclose its content to any other person. 

Xplorium does not guarantee the integrity of this electronic message and any
of its attachments, or that they are free from computer viruses or other
defects. 
* 

 





  _  



___
--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

-- 
Thx
MAG

  

___
--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


AW: AW: [asterisk-users] ReceiveText()?

2007-02-25 Thread Roland Ndaka Fru
...You can declare a variable whose values gets set/used anywhere in the
dialplan.

Regards,
Roland.

-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Im Auftrag von Yuan LIU
Gesendet: 25 February 2007 08:41
An: asterisk-users@lists.digium.com
Betreff: RE: AW: [asterisk-users] ReceiveText()?

From: Roland Ndaka Fru [EMAIL PROTECTED]
Date: Sun, 25 Feb 2007 07:45:57 +0100

Here is how you can send/receive text in the DialPlan using an AGI script:

print STDERR 1.  Testing 'sendtext'...;
print SEND TEXT \hello world\\n;
my $result = STDIN;
checkresult($result);


print STDERR 2.  Receiving Text 'receivetext'...;
print RECEIVE TEXT 3000\n;
my $result = STDIN;
checkresult($result);

Greetz,
Roland.

That's cool.  Thanks for the pointer, Roland.  Gotta go back to test-agi 
again.

Now, if only one can pass value back into dial plan...

Yuan Liu

-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Im Auftrag von Olle E
Johansson
Gesendet: 24 February 2007 10:52
An: Asterisk Users Mailing List - Non-Commercial Discussion
Betreff: Re: [asterisk-users] ReceiveText()?


24 feb 2007 kl. 03.15 skrev Yuan LIU:

  How do I receive text sent from SendText() application?  Asterisk
  lists text capability, so SendText() is successful.  But I don't
  see an application to actually use it.

EyeBeam and several SIP phones does receive those messages.

We need to make sure that the application and the parser supports
UTF8 messages, as both SIP and
IAX2 is standardized on UTF8 text messaging.

/O


___
--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


AW: [asterisk-users] ReceiveText()?

2007-02-24 Thread Roland Ndaka Fru
Here is how you can send/receive text in the DialPlan using an AGI script:

print STDERR 1.  Testing 'sendtext'...;
print SEND TEXT \hello world\\n;
my $result = STDIN;
checkresult($result);


print STDERR 2.  Receiving Text 'receivetext'...;
print RECEIVE TEXT 3000\n;
my $result = STDIN;
checkresult($result);

Greetz,
Roland.

-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Im Auftrag von Olle E
Johansson
Gesendet: 24 February 2007 10:52
An: Asterisk Users Mailing List - Non-Commercial Discussion
Betreff: Re: [asterisk-users] ReceiveText()?


24 feb 2007 kl. 03.15 skrev Yuan LIU:

 How do I receive text sent from SendText() application?  Asterisk  
 lists text capability, so SendText() is successful.  But I don't  
 see an application to actually use it.

EyeBeam and several SIP phones does receive those messages.

We need to make sure that the application and the parser supports  
UTF8 messages, as both SIP and
IAX2 is standardized on UTF8 text messaging.

/O
___
--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


AW: [asterisk-users] Small CDR Billing Program

2007-02-12 Thread Roland Ndaka Fru
Hi Mark,

 

Take a look at the YakaVOIP solution from  http://www.yakasoftware.com/
http://www.yakasoftware.com. Probably suits your requirements.

 

Greetz,

Roland.

 

  _  

Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Im Auftrag von MBIT
Technologies
Gesendet: 12 February 2007 22:23
An: asterisk-users@lists.digium.com
Betreff: [asterisk-users] Small CDR Billing Program

 

Hi Guys

 

I am just looking around for a small billing program but can't really find
what I am looking for. 

 

It needs to bill straight off the CDR. It should grab all the CDR records
from the asteriskcdrdb mysql database then have a rates table to that it
calculate a bill from. Is there any open source packages or commercial
packages that will account for billing say only 5 extensions?

 

 

Regards

 

 

Mark Brooker

T: 02 4959 8670

M: 0415 846 865

F: 02 4950 5609

E: [EMAIL PROTECTED]

W: http://www.mbit.com.au

 

___
--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


AW: [asterisk-users] IAX2 softphones can't (won't?) use PRI trunks....

2007-02-05 Thread Roland Ndaka Fru
Try latest IAX2 YakaPhone which you can get from www.yakasoftware.com.

 

  _  

Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Im Auftrag von ismail loo
Gesendet: 05 February 2007 17:16
An: Asterisk Users Mailing List - Non-Commercial Discussion
Betreff: Re: [asterisk-users] IAX2 softphones can't (won't?) use PRI
trunks

 

Try latest version of iaxLite softphone. The testing result here is that it
could work with PRI or IAX2 trunks.



 

2007/1/16, Patrick W. Foster [EMAIL PROTECTED]: 

I have call center PCs that switch between an IBEAM SIP softphone and a NEBU
IAX softphone (for reasons

that aren't germane here).   The SIP softphones work fine, but the IAX
softphones get a fast busy unless I give

them an IAX trunk to use, instead of the PRI trunks that all the other
phones are using.  I am using Asterisk 1.2.3.

svn rev 47264.

 

I've appended a sample call trace.   The call fails through all the
configured PRI trunks to the IAX trunk with a CHANUNAVAIL error, whilst

the SIP phones are actively calling out on those same PRI trunks.   The
numbers dialed are 10 digits with no prefix.  I am hopeful that

someone will recognize the issue and give me a pointer on where to look for
the problem.

 

- Registered IAX2 '4414' (AUTHENTICATED) at 192.168.1.102:4569
http://192.168.1.102:4569/ 
-- Accepting AUTHENTICATED call from 192.168.1.102
http://192.168.1.102/ :
requested format = alaw,
requested prefs = (),
actual format = ulaw, 
host prefs = (ulaw|alaw|gsm),
priority = mine
-- Executing Set(IAX2/4414-6, EMERGENCYROUTE=YES) in new stack
-- Executing Macro(IAX2/4414-6, dialout-trunk|4|xxxnnn||) in new
stack 
-- Executing GotoIf(IAX2/4414-6, 1?3:2) in new stack
-- Goto (macro-dialout-trunk,s,3)
-- Executing Macro(IAX2/4414-6, user-callerid) in new stack
-- Executing GotoIf(IAX2/4414-6, 0?report) in new stack 
-- Executing GotoIf(IAX2/4414-6, 0?start) in new stack
-- Executing Set(IAX2/4414-6, REALCALLERIDNUM=4414) in new stack
-- Executing NoOp(IAX2/4414-6, REALCALLERIDNUM is 4414) in new stack

-- Executing Set(IAX2/4414-6, AMPUSER=4414) in new stack
-- Executing Set(IAX2/4414-6, AMPUSERCIDNAME=User32-IAX) in new
stack
-- Executing GotoIf(IAX2/4414-6, 0?report) in new stack 
-- Executing Set(IAX2/4414-6, CALLERID(all)=User32-IAX 4414) in
new stack
-- Executing NoOp(IAX2/4414-6, Using CallerID User32-IAX 4414)
in new stack 
-- Executing Macro(IAX2/4414-6, record-enable|4414|OUT) in new stack
-- Executing GotoIf(IAX2/4414-6, 0  0?2:4) in new stack
-- Goto (macro-record-enable,s,4) 
-- Executing AGI(IAX2/4414-6,
recordingcheck|20070115-121440|1168881280.2233) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
  recordingcheck|20070115-121440|1168881280.2233: Outbound recording not
enabled 
-- AGI Script recordingcheck completed, returning 0
-- Executing NoOp(IAX2/4414-6, No recording needed) in new stack
-- Executing Macro(IAX2/4414-6, outbound-callerid|4) in new stack 
-- Executing GotoIf(IAX2/4414-6, 1?start) in new stack
-- Goto (macro-outbound-callerid,s,3)
-- Executing NoOp(IAX2/4414-6, REALCALLERIDNUM is 4414) in new stack

-- Executing Set(IAX2/4414-6, USEROUTCID=Business Name
xxx-nnn-) in new stack
-- Executing Set(IAX2/4414-6, EMERGENCYCID=) in new stack 
-- Executing Set(IAX2/4414-6, TRUNKOUTCID=Business Name
xxx-nnn-) in new stack
-- Executing GotoIf(IAX2/4414-6, 0?trunkcid) in new stack 
-- Executing GotoIf(IAX2/4414-6, 1?trunkcid) in new stack
-- Goto (macro-outbound-callerid,s,11)
-- Executing GotoIf(IAX2/4414-6, 0?usercid) in new stack 
-- Executing Set(IAX2/4414-6, CALLERID(all)=Business Name
xxx-nnn-) in new stack
-- Executing GotoIf(IAX2/4414-6, 0?report) in new stack 
-- Executing Set(IAX2/4414-6, CALLERID(all)=Business Name
xxx-nnn-) in new stack
-- Executing NoOp(IAX2/4414-6, CallerID set to Business Name
xxx-nnn-) in new stack 
-- Executing Set(IAX2/4414-6, GROUP()=OUT_4) in new stack
-- Executing GotoIf(IAX2/4414-6, 0?108) in new stack
-- Executing Set(IAX2/4414-6, DIAL_NUMBER=xxxnnn) in new stack 
-- Executing Set(IAX2/4414-6, DIAL_TRUNK=4) in new stack
-- Executing AGI(IAX2/4414-6, fixlocalprefix) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix 
-- AGI Script fixlocalprefix completed, returning 0
-- Executing Set(IAX2/4414-6, OUTNUM=xxxnnn) in new stack
-- Executing Set(IAX2/4414-6, custom=ZAP/g0) in new stack 
-- Executing GotoIf(IAX2/4414-6, 0?16) in new stack
-- Executing Dial(IAX2/4414-6, ZAP/g0/xxxnnn|120|r) in new stack
  == Everyone is busy/congested at this time (1:0/0/1) 
-- Executing Goto(IAX2/4414-6, s-CHANUNAVAIL|1) in new stack
-- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
-- Executing NoOp(IAX2/4414-6, Dial failed due to 

AW: [asterisk-users] conditional dialplan

2006-12-24 Thread Roland Ndaka Fru
Classical if-then functionality can be achieved in the dialpan with the
GotoIf(...) function.

Regards,
Roland.

-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Im Auftrag von nik600
Gesendet: 23 December 2006 15:16
An: Asterisk Users Mailing List - Non-Commercial Discussion
Betreff: [asterisk-users] conditional dialplan

Hi

can i set up some conditions in my dialplan?

For example:

exten = 99,1,Answer
exten = 99,2, ... if {RECORD}=yes
then:
monitor...
Dial
else:
Dial.

Or something similar... ?

Many thanks in advance

nik
___
--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] IAX Realtime MD5 authentication

2006-11-01 Thread Roland Ndaka Fru
Hi,

Is there any possibility to have md5 encoded passwords in the IAX users
database? I notice the secret AND/OR md5secret columns always have to
contain the password in plain text even when you set the auth column value
to md5?!?

Am I missing out something? Any ideas on how to correct this? Having plain
text passwords in the realtime database is not very suitable for me and
poses a security vulnerability.

Thanks,
Pedros

___
--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] IAX Realtime MD5 authentication

2006-11-01 Thread Roland Ndaka Fru
Hi,

Is there any possibility to have md5 encoded passwords in the IAX users
database? I notice the secret AND/OR md5secret columns always have to
contain the password in plain text even when you set the auth column value
to md5?!?

Am I missing out something? Any ideas on how to correct this? Having plain
text passwords in the realtime database is not very suitable for me and
poses a security vulnerability.

Thanks,
Pedros

___
--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