Displaying 20 results from an estimated 3000 matches similar to: "It's true - Nikotel charge for not-completed calls"
2003 Nov 23
1
SIP Asterisk -> Nikotel disconnects after 1 Minute
Hello list!
I'm using "Asterisk CVS-11/22/03-04:28:51" and try to route my normal
(classic) phone calls via nikotel (www.nikotel.com). I can talk about 1
minute and get then disconnected. Here my current configuration parts
which affect nikotel:
register => chabrol:PASSWORD_REMOVED@nikotel/500
[nikotel]
type=friend
secret=PASSWORD_REMOVED
username=chabrol
fromuser=chabrol
2005 Jan 10
2
Some questions (maybe Nikotel related)
Hi list,
I have some nontrivial questions. I am no telecommunication guru and I
will explain it with my simple words. I hope someone can help me with
these issues (with Asterisk 1.0.3):
- If I call outside (with Nikotel to German Telekom) there is a remote
hangup after 2 minutes. I've seen other people posting this but nothing
helped. I luckily managed to get around this issue with the
2003 Jul 07
0
Follow-up -- Using Asterisk with Nikotel
Hi
thanks to everybody who has been assisting me in solving the various
problems I had to dial out from Asterisk to a PSTN number with SIP using
Nikotel's VoIP service.
I have drafted a mini-how-to which is available at
http://www.akabeni.com/benjk/Using_Asterisk_with_Nikotel.pdf
This is a first draft, I will amend this further, in particular the
"verify and debug" section
2003 Oct 03
9
No Ringback on Iconnect
When I place a call using Iconnecthere as my sip provider, I hear no
ringback when making a call. Does anyone else have this problem or
offer any suggestions? Thanks, Kevin
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031003/b048f72f/attachment.htm
2004 May 07
5
SIP: Trouble with "Moved temporarily" (302)
Hi folks,
this does look like a bug to me: Asterisk replaces the @63.214.186.6 by
@context which obviously leads to a failure. Any comments, do I have a
configuration issue on my side that I missed?
Cheers, Philipp
-- Executing Dial("SIP/philipp-bd5f", "SIP/992365264680@nikotel-
out|90") in new stack
-- Called 99xxxxxxxxxx@nikotel-out
-- Got SIP response 302
2003 Oct 14
0
No Ringback on Iconnect or Nikotel
When I place a call using Iconnecthere or Nikotel as my sip provider, I
hear no ringback tone when making a call. I have tried adding the 'r'
command to the dial sting with no help. Does anyone else have this
problem or offer any suggestions? Thanks, Kevin
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
2003 Nov 13
3
multi call iconenct?
Is there a service like iconnect that does allow dialing out more then
one concurrent connection?
Asterisk works great with iConnectHere, but they only allow one call at
a time.
I don't want to setup an account for each concurrent call, because it
will make iConnect an expensive service, and besides, all of our calls
combined doesn't reach 1000 minutes per month.
Any ideas?
2005 Aug 17
0
Nikotel issues
Hi!
I've read in the archives that there are problems concerning Nikotel calls
being disconnected after two minutes. I had the same problem yesterday. Is
there a fix? There was only a "giving up" statement after the last e-mail in
the archive, I'm about to do that too.
Here's my sip.conf entry for Nikotel (left out the register stuff 'cause it's
working):
2003 Jul 03
0
How do I make Asterisk login at/use VoIP provider?
Hi
please excuse if this seems obvious, but I am new to this and the SIP
section in the Asterisk handbook do not give any clues nor do the SIP
examples in there seem to represent real-world situations.
I am using Nikotel as a VoIP provider (for now) and I would like to
configure Asterisk to sign on with Nikotel so that I can use the
telephones connected to Asterisk to make calls using the
2003 Oct 17
4
Extension syntax specification - please help!
John Todd have started creating a document called Readme.channels that will document the
syntax of extensions in all channels. I have uploaded his draft to the Wiki, so that all
of you can help find the syntax, it's not so easy to grasp from reading the source. It
would really be handy to have it all in one place, within the source distribution (and
of course in the Wiki...)
2005 Mar 04
2
budgetphone
Hi all,
I registered a SIP account at budgetphone.nl/talkin2ya.nl
Receiving calls works like a charm, I even redirected my
normal PSTN number to the number I got from them so
everything ends up in my * server.
Before I ask them to take over my normal phone number I
wanted to test all of it, so I ordered some calling minutes
to test. Now I cannot get outbound calling to work with
them. Anyone here
2003 Jul 05
2
Please help -- Syntax for dialing VoIP provider
Hi
thanks to everybody who responded to my earlier post. I have looked at
all the material and links provided and tried everything in there, but
it simply won't work for me.
My SIP phones register with Asterisk, but they cannot be called
(everybody is busy at this time) nor can they call anything (error code
4, whatever that means) not even internal (yes I did give them
appropriate
2003 Sep 30
1
SIP Registration Difficulties
I have SIP registrations working correctly for FWD and Sipphone, but it
is impossible to connect to Sipcall or Nikotel, I saw that someone on
the list has problems with ICH.
To try and sort out the problem I tried to register to Sipcall with
Linphone and sent the dialogs to tech support of the equipment provider.
Here is their answer:-
The reason the registration fails is because not
2006 Jan 21
1
Caller ID and Sipura Router
Could anyone please help me with that:
I have an analog telephone connected to Asterisk using a Sipura 2002 ATA. When
calling the extension, the caller ID presented is always the number of that
extension rather than the number of the calling one.
While I learned that this is the new standard behaviour (?) of Asterisk, I
want to show the original caller ID.
I tried the options o and f in the
2004 Sep 13
2
allowing/disallowing codecs in dialplan?
Hi all,
Is there a possibility to set the codecs Asterisk will choose in the dialplan
("exten=>" statements or their contexts) instead of sip.conf?
My problem is that I connect my SIP phone with several providers (Nikotel,
Sipgate, Stanaphone) for icoming and outgoing calls. Not all of these providers
offer the same set of codecs. I'd like Asterisk to use the same codec for the
2003 Jul 22
3
SIP Call Forwarding/Transfer support ?
Hi All,
I was wondering, in my effort to show how Asterisk can replace Call Manager, if there is support for call transfers/forwarding from the users Cisco 7940 SIP phone to either another SIP client or through the AS5300 on to the PSTN. I do see some stuff in the docs but seems to be specific to a local PRI board in the PC of which I don't have.
Any experiences/comments most appreciated.
2003 Jul 07
0
Asterisk crashing after Voicemail box creation
Hi
I have just been struggling for four days to get SIP working and now as
I created a voicemail box, Asterisk has become very unstable and it
can't bridge SIP phone to SIP provider calls anymore.
Calling internally from one SIP phone to another works fine.
Calling internally from a SIP phone to an analog phone on a Zap channel
and vice versa works fine.
Incoming PSTN calls delivered to
2003 May 23
3
iConnectHere - calls dropping out?
Hi all,
This is my first post here - I started with Asterisk a few days ago and have
"fallen in love" - fantastic product. I've only got softphones connected at
the moment - I'll probably order the FXO/FXS cards in about a month (and
then think about getting some hardware SIP phones). Our current phone system
is quite a few years old and isn't growing with us (when a single
2004 Apr 28
2
Asterisk and Iconnecthere pause
Hi, I just got a SPA-2000 in and was finally able to complete my asterisk
setup. I'm making my outgoing calls through iconnecthere from the
asterisk server however I'm running into a problem when placing calls. I
can connect fine but when the person (or answering machine) picks up I
hear them talk for a about half a second then there is a half a second
pause or muted period and then the
2003 Mar 03
3
iconnecthere 480 error: is there a workaround?
I am going to have to find a fix for this problem or I'm going to have
to quit using iconnect.
About one call in 10 or so, iconnect's gateway gives me an error
(console output appended below).
So upon receiving the error, which as a 4XX error means, "Fatal,"
asterisk gives up and drops the call. But not iconnect!! The phone at
the other end starts ringing, and rings