?
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
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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,
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
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
___
Hi all.
I am going to do a simple voting application for a radiostation.
The idea is to have listeners call in to vote on songs.
What I want to do is to take a phonenumer for each song and present the
result on a simple webpage.
Eg.
To vote on song number one, call 555-
To vote on
Can somebody help me with cheap terminiation in the UK ? With different
areacodes for in/out going traffic.
Please contact me OFFLIST
/Regards Mike
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Hi all,
I was wondering how the easiest way to restrict the users ability to set
caller ID would be ?
I have some users that uses IAX to connect with me. multiple numers via
iax.
on outgoing calls I would like the user to only be able to set his
range of numbers on the outgoing calls.
Is
HI all.
I was wondering... Isn't it a good thing to store the IP of the client
making the call ?
Or does asterisk store that in some other place ?
/Mike
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[EMAIL PROTECTED] wrote:
HI all.
I was wondering... Isn't it a good thing to store the IP of the
client making the call ?
Or does asterisk store that in some other place ?
/Mike
I'm gonna reply to myself here and add another quiestion.
When an incoming call is being transfered.
Hi all,
Can you guys recomend a good terminiation partner in Holland ?
/Mike
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Hi all.
Can you reccomend a SIP / IAX provider in the Netherlands ?
I need a few Numbers, and of course cheap rates :)
/Regards Mike
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To
HI all
I have a little question, and since there is a alot of Cisco Gurus
somebody might be able to help me.
I think It is an easy problem.
My PSTN proviver strips the first digit in the callerid on all incoming
calls.
So when the call reaches my Asterisk I am missing a 0 in the CLID
I
Hi.
If I use a Cisco as a PSTN termination GW and need to route all incoming
isdn calls to my asterisk and all outgoing calls from asterisk via the
cisco out to pstn, how do I do that ?
in the cisco I have this:
dial-peer voice 1 pots
destination-pattern [0-9]T
no digit-strip
Hi all.
I have a strange problem, I've got a AS5350 hooked up to a telco using
two trunked E1's
The 5350 should only act as a GW to a sipproxyserver.
THe thing is it seems to be only oneway audio?
There are no firewall at all, and the audio still only get one-way
When I call from pstn --
Hi,
I have a problem with call forwarding.
When I call forward I need to forward the call with the callerid on the
called phone, not the callers.
How do I do this in a smart way ?
-- Called 89
-- Got SIP response 302 Moved Temporarily back from x.x.x.248
-- Now
Hi all
Does anybody know if it is possible to change the tones on a 7960 ?
I guess there must be some way to edit the dial/busy/congestion tones ?
/Mike
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Hi all.
I was wondering about how to set different tones, in the Asterisk I use
indications.conf, in the Cisco ATA-186 I use the webinterface.
How do I set tones in the Grandstream, handytone, Cisco 7960 ?
The US tones does not apply to all countries. (Unfortunatley)
/Mike
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.
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
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Does anybody know or have good examples of using all functions in a 7960
(SIP)
/Mike
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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
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
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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
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
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
How should I solve this ?
Did nobody recognize this problem ?
I se it as a major bug, or am I doing something wrong ?
/Mike
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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
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
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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
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
Hiyas..
Have any of you tried the HP300 phone and got it to work with asterisk ?
(sip)
/Mike
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