[OFFLIST] Re: [Asterisk-Users] run with incorrect E1/T1 jumper settings

2006-02-28 Thread Micke Andersson

Robert Andersson wrote:

Hi,

I have installed an TE110P but forgot to change the jumper
settings to E1. I don't have easy physical access to ther server
at the moment so I wonder if it will be possible to run it without changing
the jumper settings with a configuration like below or will it be
impossible
to use the card at all before I fix the jumper? I can't try it myself yet
since the operator isn't ready yet, but I would like to know in advance
if it is impossible.

bchan=1-15,17-24
dchan=16

instead of

bchan=1-15,17-31
dchan=16

best regards
Robert



Tjena.

Fick du det att fungera?

Jag är lite osäker om det går att tvinga det.  Såg något svar på listan 
där, men jag har faktiskt aldrig provat det själv.




/Mvh  Micke

-
Mikael Andersson
dCAp Certified

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

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


Re: [Asterisk-Users] Prepaid / postpaid solution

2006-02-27 Thread Micke Andersson

Alexander Burke wrote:

At 05:03 PM 02/26/2006, you wrote:

I want to match the user from the users callerid.  All users have DIDs.


You probably shouldn't do that for security reasons -- rather, match 
them according to the SIP username/password pair they provide when they 
register.




Hm, Maybe you're right.

The Idea is to get the same solution, The user is automaticlly 
identified in the billingsystem...



/Mike

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

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


[Asterisk-Users] Prepaid / postpaid solution

2006-02-26 Thread Micke Andersson

Hi All..

I've noticed that there are quite a few different billing solutions 
availible.


If I want to have both prepaid and postpaid accounts with only ATAs (or 
other SIP devices) which one should I use ?


Some users are prepaid, and some are postpaid accounts (invoice)


I do not want the user to enter any Callingcard info, just place the 
call..


I want to match the user from the users callerid.  All users have DIDs.

And I want to use mysql.

What is the the best, and easiest solution to use ?

Any suggestions ?


/Mike

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

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


[Asterisk-Users] Asterisk and Cisco AS5350

2006-02-08 Thread Micke Andersson


Hiyas,

Does anybody have som good examples on how to configure the Cisco AS5350 
as a pstn gw to asterisk?


I get some really strange sounds, eg. when testing with milliwatt.

I have two E1 that shoud be bundled  (60 channels)


/Regards Mike


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

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


RE: [Asterisk-Users] Nitsuko 124i interface, anyone?

2004-03-11 Thread Micke Andersson
Andrew Thompson  wrote on the Thursday, March 11, 2004 6:06 PM 

 Comments from anyone who has worked with this hardware and knows more
 about it than myself are appreciated, even if you've not actually
 tried to swap it out with *.  
 


I have a Nitsuka system here at home.. somewhere in a box.

I'm not sure wich model.  I dont have any voicemail system to that though,
so I'm interessted in your idea to use a *

/Mike

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


[Asterisk-Users] having users in sql

2004-03-02 Thread Micke Andersson

Hi.

If I want to have all my users (sip) in q mysql 

I've tried a few thingies.. but I didn't gett all the needed fields..
like nat, callerid, etc etc

Is there a good way to solve this ?

/Mike
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


[Asterisk-Users] Cisco 7960

2004-03-01 Thread Micke Andersson

Does anybody know or have good examples of using all functions in a 7960
(SIP)

/Mike

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


RE: [Asterisk-Users] SIP REGISTER

2004-02-18 Thread Micke Andersson
John Fraizer  wrote on the Tuesday, February 17, 2004 7:14 PM 

 Try sending us the registry line and context information from your
   sip.conf. It is much easier to figure out what you're doing wrong
 from there.  I register my Asterisk server with 6 different SIP
 providers with no problems at all.  
 
 John



this is the line:

register = pstn-number:passwd:[EMAIL PROTECTED]/pstn-number

/Mike




 Micke Andersson wrote:
 Hiyas..
 
 I have a little problem ..
 
 I try to register my Asterisk at a sip provider.. but it just wont
 work. 
 
 It works fine with eg xlite or Grandstream.. .but not with Asterisk.
 
 
 I think it is in the Register process:
 
 This is the difference I cen tell in the sip headers between Xlite
 and Asterisk 
 
  ( I have removed IPs and numbers and replaces them with text)
 
 
 
 First Xlite:  (this works)
 
 -snip
 SEND  provider.ip.ip.ip:5060
 REGISTER sip:provider.com SIP/2.0
 Via: SIP/2.0/UDP
 ip.ip.ip.ip:5060;rport;branch=z9hG4bK06595964B0AE46CF9271267AD534E632
 From: pstn-number sip:[EMAIL PROTECTED]
 To: pstn-number sip:[EMAIL PROTECTED]
 Contact: pstn-number sip:[EMAIL PROTECTED]:5060
 Call-ID: [EMAIL PROTECTED]
 CSeq: 8823 REGISTER
 Expires: 1800
 Authorization: Digest
 username=pstn-number,realm=provider.com,nonce=MTA3NzAyOTUwMjk2NWI
 2Y
 jg4MjcxOGNlZWRkODRhYzg4NmEyZWE5NTYwN2Y0,response=f833201fd4a8719ea9a
 2e
 c505debbd56,uri=sip:provider.com,opaque=dd5d790f90d0307c7390cdb8f6
 e9
 4cc8,qop=auth,cnonce=4B86525A67C646469656D90AD4C1273C,nc=0002
 Max-Forwards: 70 User-Agent: X-Lite build 1101   
 Content-Length: 0
 
 
 RECEIVE  provider.ip.ip.ip:5060
 SIP/2.0 200 OK
 
 - end snip -
 
 This is Asterisk (does not work)
 
 --snip
 Reliably Transmitting:
 REGISTER sip:provider.com SIP/2.0
 Via: SIP/2.0/UDP ip.ip.ip:5060;branch=z9hG4bK56158c1f
 From: sip:[EMAIL PROTECTED];tag=as017cdd56
 To: sip:[EMAIL PROTECTED]
 Call-ID: [EMAIL PROTECTED]
 CSeq: 104 REGISTER
 User-Agent: Asterisk PBX
 Expires: 1200
 Contact: sip:[EMAIL PROTECTED]
 Event: registration
 Content-length: 0
 
  (no NAT) to provider.ip.ip.ip:5060
 pbx1*CLI
 
 Sip read:
 SIP/2.0 403 Forbidden
 
 
 --- end snip ---
 
 The difference as I can tell is in the From: and to: lines
 
 xlite says From: number [EMAIL PROTECTED]
 
 asterisk only says From: [EMAIL PROTECTED]
 
 
 How do I tell my Asterisk to send the registration as xlite ?
 
 /Mike
 
 
 ___
 Asterisk-Users mailing list
 [EMAIL PROTECTED]
 http://lists.digium.com/mailman/listinfo/asterisk-users
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
 
 ___
 Asterisk-Users mailing list
 [EMAIL PROTECTED]
 http://lists.digium.com/mailman/listinfo/asterisk-users
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users


___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


[Asterisk-Users] mysql_freinds

2004-02-18 Thread Micke Andersson

Hi,

I was wondering if you use mysql_friends,  if a friend is behind nat ?

What I understand all the availible fields are not used in the sql table
?

Or did I get this wrong ?

/Regards  Mike

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


RE: [Asterisk-Users] SIP REGISTER

2004-02-18 Thread Micke Andersson
John Fraizer  wrote on the Wednesday, February 18, 2004 6:16 PM 

 
 You've got a syntax problem.  It SHOULD be:
 
 register = pstn-number:[EMAIL PROTECTED]/pstn-number
 

Tried that too, no go..

I thought the syntax were:

 register = username:passwd:[EMAIL PROTECTED]/local number

/Mike

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


[Asterisk-Users] SIP REGISTER

2004-02-17 Thread Micke Andersson

Hiyas..

I have a little problem ..

I try to register my Asterisk at a sip provider.. but it just wont work.

It works fine with eg xlite or Grandstream.. .but not with Asterisk.


I think it is in the Register process:

This is the difference I cen tell in the sip headers between Xlite and
Asterisk

 ( I have removed IPs and numbers and replaces them with text)



First Xlite:  (this works)

-snip
SEND  provider.ip.ip.ip:5060
REGISTER sip:provider.com SIP/2.0
Via: SIP/2.0/UDP
ip.ip.ip.ip:5060;rport;branch=z9hG4bK06595964B0AE46CF9271267AD534E632
From: pstn-number sip:[EMAIL PROTECTED]
To: pstn-number sip:[EMAIL PROTECTED]
Contact: pstn-number sip:[EMAIL PROTECTED]:5060
Call-ID: [EMAIL PROTECTED]
CSeq: 8823 REGISTER
Expires: 1800
Authorization: Digest
username=pstn-number,realm=provider.com,nonce=MTA3NzAyOTUwMjk2NWI2Y
jg4MjcxOGNlZWRkODRhYzg4NmEyZWE5NTYwN2Y0,response=f833201fd4a8719ea9a2e
c505debbd56,uri=sip:provider.com,opaque=dd5d790f90d0307c7390cdb8f6e9
4cc8,qop=auth,cnonce=4B86525A67C646469656D90AD4C1273C,nc=0002
Max-Forwards: 70
User-Agent: X-Lite build 1101
Content-Length: 0


RECEIVE  provider.ip.ip.ip:5060
SIP/2.0 200 OK

- end snip -

This is Asterisk (does not work)

--snip
Reliably Transmitting:
REGISTER sip:provider.com SIP/2.0
Via: SIP/2.0/UDP ip.ip.ip:5060;branch=z9hG4bK56158c1f
From: sip:[EMAIL PROTECTED];tag=as017cdd56
To: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 104 REGISTER
User-Agent: Asterisk PBX
Expires: 1200
Contact: sip:[EMAIL PROTECTED]
Event: registration
Content-length: 0

 (no NAT) to provider.ip.ip.ip:5060
pbx1*CLI 

Sip read: 
SIP/2.0 403 Forbidden


--- end snip ---

The difference as I can tell is in the From: and to: lines

xlite says From: number [EMAIL PROTECTED]

asterisk only says From: [EMAIL PROTECTED]


How do I tell my Asterisk to send the registration as xlite ?

/Mike


___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


[Asterisk-Users] (no subject)

2004-02-16 Thread Micke Andersson
Hi all.

If I want to use the * only as a GW to PSTN  and allow only one external
proxy to place calls. how is the smartest way to do this ?

I dont want the world to be able to do invites   only a specific IP,
in this case my proxy that handles all the users.

/Mike

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


RE: [Asterisk-Users] Problem with fax detection

2003-11-26 Thread Micke Andersson
 
 How should I solve this ?


Did nobody recognize this problem ?

I se it as a major bug, or am I doing something wrong ?

/Mike


___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users


[Asterisk-Users] Problem with fax detection

2003-11-25 Thread Micke Andersson

Hi

I have a problem with the fax detection.

I want to be able to turn that of on all zap channels.

the * is in between my E1 and my PBX and when I try to make a fax call out
on the E1 the * detects the fax tone and hangsup the outgoing zap channel.

How should I solve this ?

/Mike

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users


[Asterisk-Users] Cisco 7960

2003-10-24 Thread Micke Andersson

I need some help with upgrading a 7960.

Any of you guys familiar with that ?

I friend of mine have a couple of 7960 , and would like to get 'em to work.

/Mike

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users


RE: [Asterisk-Users] Version 1 vs Version 2

2003-10-04 Thread Micke Andersson
 
 
 The differences in VM2 and the ability to create VM contexts 
 for things like virtual PBX's on one box, VM2 allows you to 
 modify the email that gets sent when a voicemail is recieved 
 and a few more config features..

How do you modify the emails ?

Are there other configfiles ?

/M

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users


RE: [Asterisk-Users] CLASS feature syntax

2003-08-25 Thread Micke Andersson
 
ct: Re: [Asterisk-Users] CLASS feature syntax
 
 http://www.nanpa.com/number_resource_info/vsc_assignments.html
 http://www.nanpa.com/number_resource_info/vsc_definitions.html

Does anybody know if there is a similar webpage for European standards ? Or
other countries standards ?

/Mike

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users


[Asterisk-Users] HP300 phone

2003-08-17 Thread Micke Andersson


Hiyas..

Have any of you tried the HP300 phone and got it to work with asterisk ?
(sip)

/Mike

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users