Displaying 11 results from an estimated 11 matches for "budgetphon".
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budgetphone
2005 Jul 10
3
Incoming calls from BudgetPhone.nl
(this time with subject....)
Hello,
I?m trying to get Asterisk to accept incoming calls from budgetphone.nl.
When I dial my budgetphone nr on a PSTN KPN line it immediately gives a busy
tone.
I tried X-lite, which worked perfect, so my modem (with nat) probably is not
the problem.
I did a sip debug and got the following output.
Because I?m new to Asterisk I can?t get the error why this is not working...
2005 Jul 10
0
(no subject)
I'm trying to get Asterisk to accept incoming calls from budgetphone.nl.
When I dial my budgetphone nr on a PSTN KPN line it immediately gives a busy
tone.
I tried X-lite, which worked perfect, so my modem (with nat) probably is not
the problem.
I did a sip debug and got the following output.
Because I'm new to Asterisk I can't get the error why this is...
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...
2005 May 17
0
Can't connect to SIP provider
Hello all,
I've been trying everything I could find, but I can't seem to get my *
server connected to my SIP provider (budgetphone.nl). Here's my sip.conf:
[budgetphone]
port=5060
bindaddr=0.0.0.0
context=from-budgetphone
register => 31307110000:secret@budgetphone.nl/500
type=friend
host=budgetphone.nl
fromuser=31307110000
secret=secret
fromdomain=budgetphone.nl
username=31307110000
And my extensions.conf:
[from-budge...
2005 May 10
0
outbound PSTN numbers over SIP failing
...1 VERBOSE[1563]: -- Executing
[1;36;40mSetCIDNum[0;37;40m("[1;35;40mSIP/200-8f7f[0;37;40m",
"[1;35;40m31437110323[0;37;40m") in new stack
May 8 10:47:11 VERBOSE[1563]: -- Executing
[1;36;40mDial[0;37;40m("[1;35;40mSIP/200-8f7f[0;37;40m",
"[1;35;40mSIP/XXXXXXXXXX@budgetphone.nl|30|r[0;37;40m") in new stack
May 8 10:47:11 DEBUG[1563]: SIMPLE DIAL (NO URL)
May 8 10:47:11 DEBUG[1563]: Outgoing Call for XXXXXXXXXX
May 8 10:47:11 DEBUG[1563]: XXXXXXXXXX is not a local user
May 8 10:47:11 VERBOSE[1563]: -- Called XXXXXXXXXX@budgetphone.nl
May 8 10:47:11 DEBUG...
2007 Jul 17
0
help with sip configuration for sipgate.de on asterisk 1.4
...lso make calls, mainly only sip, i don't mean
trixbox or freepbx. anyhow my main problem is the sipgate.de thing, below are my sip.conf and
extension.conf thx for any help
sip.conf:
[general]
language=en
disable=all
register => user:pass at voip.eutelia.it/number
register => user:pass at budgetphone.nl/number
register => user:pass at sipgate.de/number
register => user:pass at sip.voiparound.com/number
register => user:pass at sip.webcalldirect.com/
register => user:pass at ixcall.net/number
register => user:pass at freedigits.net/number
register => user:pass at sip.messagene...
2004 Oct 11
4
outgoing calls
Hi,
here what i have:
[2001]--[Asterisk]---[ISDN-Trunk]---[PBX]--[8004]
Eicon Diva 4BRI Card to a PBX. Asterisk is running in version 1.0.0 on
RedHat Enterprise Linux 3AS with kernel 2.4.21-4.EL.
Dialing from Astersik extension 2001 to PBX extension 8004 via ISDN Trunk gives me the following error,
-- Executing Dial("SIP/2001-8a8e", "Modem/ttyI0:998004|20|r") in new stack
2005 Jun 05
1
Unable to create channel of type SIP-please help
...SetCallerID(${T2Y1})
exten => s,2,SetCIDName(${MYNAME})
exten => s,3,Dial(SIP/${ARG1}@t2y,${ARG2})
exten => s,4,Playback(new/acnt-or-cir-busy-now)
exten => s,5,Hangup
I do have a [t2y] reference in my SIP.conf file
[t2y]
type=friend
secret=PASSWORD
username=USERNAME
host=budgetphone.nl
dtmfmode=rfc2833
fromuser= USERNAME
fromdomain=budgetphone.nl
username= USERNAME
insecure=very
nat=yes
qualify=no
register=yes
I tried changing to change the reference t2y in the context in
extensions.conf to @budgetphone.nl and @sip.budgetphone.nl, but then I
get a authorizatio...
2006 Jun 14
0
Strange problem with MusicOnHold - works outgoing - works with extension - but not incoming!
...P/voipbuster-ac66
-- Started music on hold, class 'default', on SIP/voipbuster-ac66
-- Stopped music on hold on SIP/voipbuster-ac66
But.... If somebody rings me and I put him on hold he hears nothing:
Debug output for incoming call:
-- Executing SetCallerID("SIP/gw02-mci.budgetphone.nl-42ba1908",
""prive" <xxxxxx>") in new stack
-- Executing Dial("SIP/gw02-mci.budgetphone.nl-42ba1908",
"SIP/sipura&SIP/gs4") in new stack
-- Called sipura
-- Called gs4
-- SIP/sipura-7685 is ringing
-- SIP/gs4-4a86 is rin...
2005 Feb 16
4
Dutch VOIP-PSTN provider
Hi,
I read a lot about US providers that can terminate a PSTN
number for you and offer IAX or SIP connectivity.
Does anyone know such a company in The Netherlands ?
I read about Unet. Anyone with experience with them ?
Any information is welcome.
--
Michiel van Baak
http://lunteren.vanbaak.info
michiel@vanbaak.info
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D
2005 Jan 31
5
RE: Answering Machine Function?
-----Original Message-----
<snip>
Is this possible with asterisk? Anyone have a sample dialplan?
-other than the problem outlined below I would try something like
S,1,wait(20)
S,2,voicemail(uwhatever)
S,3,hangup
That should ignore the call for 20 seconds and then leave a message in the
unavailable greeting for 'whatever' then hangup
That leaves another problem -