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 knows how to set it up ?
Some more info:
Asterisk CVS-HEAD as of 15-02-2005
My sip.conf
[general]
context=from-sip
realm=vanbaak
port=5060
bindaddr=0.0.0.0
srvlookup=yes
maxexpirey=3600
defaultexpirey=120
musicclass=default
allow=all
language=en
relaxdtmf=yes
rtptimeout=60
rtpholdtimeout=300
;trustrpid = no
;progressinband=no
useragent=Asterisk
nat=no
externip=XXX.XXX.XXX.XXX
localnet=192.168.2.0/255.255.255.0
promiscredir = no
register => 7304502:my_sipgate_pass@sipgate.de/7304502
register => 31557110304:my_budgetphone_pass@budgetphone.nl/557110304
register => mvanbaak:my_nikotel_pass@calamar0.nikotel.com
[7304502]
type=friend
context=from-sipgate
host=sipgate.de
username=7304502
secret=my_sipgate_pass
nat=yes
canreinvite=no
insecure=very
[31557110304]
type=friend
context=from-budgetphone
host=sip.budgetphone.nl
username=31557110304
secret=my_budgetphone_pass
qualify=yes
nat=yes
canreinvite=no
insecure=very
[nikotel]
secret=my_nikotel_pass
username=mvanbaak
fromuser=mvanbaak
type=peer
context=from-nikotel
host=calamar0.nikotel.com
canreinvite=no
nat=yes
...some more entries for sip phones/softphones follow, they
all work...
the dial statement in my extensions.conf
[outgoing-budgetphone]
exten => _0XXXXXXXXX,1,SetAccount(outgoing-budgetphone)
exten => _0XXXXXXXXX,2,SetCallerID(31557110304)
exten => _0XXXXXXXXX,3,Dial(SIP/31557110304/${EXTEN})
exten => _0XXXXXXXXX,4,Congestion
exten => _0XXXXXXXXX,104,Busy
And this is wat I get on the CLI when I call my cellphone:
-- Executing SetAccount("SIP/michiel-d5bd",
"outgoing-budgetphone") in new stack
-- Executing SetCallerID("SIP/michiel-d5bd",
"31557110304") in new stack
-- Executing Dial("SIP/michiel-d5bd",
"SIP/31557110304/06XXXXXXXXX") in new stack
-- Called 31557110304/06XXXXXXXXX
Mar 4 18:51:11 WARNING[4529]: chan_sip.c:6830 handle_response: Forbidden -
wrong password on authentication for INVITE to '"31557110304"
<sip:31557110304@80.126.97.99>;tag=as0ccbacfe'
-- SIP/31557110304-5857 is circuit-busy
== Everyone is busy/congested at this time
-- Executing Busy("SIP/michiel-d5bd", "") in new stack
== Spawn extension (internal, 06XXXXXXXXX, 104) exited non-zero on
'SIP/michiel-d5bd'
-- Got SIP response 483 "Too many hops" back from 81.23.228.150
I tripple checked my password, and I am sure it is correct.
What to do ?
--
Michiel van Baak
http://lunteren.vanbaak.info
michiel@vanbaak.info
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D
"Two of the most famous products of Berkeley are LSD and BSD. I don't
think that this is a coincidence."
Yes, I've had this before with sipgate.
Try using either "31557110304" or "557110304" in both places
in:
register => 31557110304:my_budgetphone_pass@budgetphone.nl/557110304
And use use this number as a context for incoming calls
What also might work: the incoming number (557110304) in
register => 31557110304:my_budgetphone_pass@budgetphone.nl/557110304
Should be the same as a context name:
[31557110304]
type=friend
context=from-budgetphone
host=sip.budgetphone.nl
username=31557110304
secret=my_budgetphone_pass
qualify=yes
nat=yes
canreinvite=no
insecure=very
Hope it helps.
Regards,
Roman Zhovtulya
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Michiel
van Baak
Sent: Freitag, 4. M?rz 2005 18:53
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] 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 knows how to set it up ?
Some more info:
Asterisk CVS-HEAD as of 15-02-2005
My sip.conf
[general]
context=from-sip
realm=vanbaak
port=5060
bindaddr=0.0.0.0
srvlookup=yes
maxexpirey=3600
defaultexpirey=120
musicclass=default
allow=all
language=en
relaxdtmf=yes
rtptimeout=60
rtpholdtimeout=300
;trustrpid = no
;progressinband=no
useragent=Asterisk
nat=no
externip=XXX.XXX.XXX.XXX
localnet=192.168.2.0/255.255.255.0
promiscredir = no
register => 7304502:my_sipgate_pass@sipgate.de/7304502
register => 31557110304:my_budgetphone_pass@budgetphone.nl/557110304
register => mvanbaak:my_nikotel_pass@calamar0.nikotel.com
[7304502]
type=friend
context=from-sipgate
host=sipgate.de
username=7304502
secret=my_sipgate_pass
nat=yes
canreinvite=no
insecure=very
[31557110304]
type=friend
context=from-budgetphone
host=sip.budgetphone.nl
username=31557110304
secret=my_budgetphone_pass
qualify=yes
nat=yes
canreinvite=no
insecure=very
[nikotel]
secret=my_nikotel_pass
username=mvanbaak
fromuser=mvanbaak
type=peer
context=from-nikotel
host=calamar0.nikotel.com
canreinvite=no
nat=yes
...some more entries for sip phones/softphones follow, they
all work...
the dial statement in my extensions.conf
[outgoing-budgetphone]
exten => _0XXXXXXXXX,1,SetAccount(outgoing-budgetphone)
exten => _0XXXXXXXXX,2,SetCallerID(31557110304)
exten => _0XXXXXXXXX,3,Dial(SIP/31557110304/${EXTEN})
exten => _0XXXXXXXXX,4,Congestion
exten => _0XXXXXXXXX,104,Busy
And this is wat I get on the CLI when I call my cellphone:
-- Executing SetAccount("SIP/michiel-d5bd",
"outgoing-budgetphone")
in new stack
-- Executing SetCallerID("SIP/michiel-d5bd",
"31557110304") in new
stack
-- Executing Dial("SIP/michiel-d5bd",
"SIP/31557110304/06XXXXXXXXX")
in new stack
-- Called 31557110304/06XXXXXXXXX
Mar 4 18:51:11 WARNING[4529]: chan_sip.c:6830 handle_response:
Forbidden - wrong password on authentication for INVITE to
'"31557110304"
<sip:31557110304@80.126.97.99>;tag=as0ccbacfe'
-- SIP/31557110304-5857 is circuit-busy
== Everyone is busy/congested at this time
-- Executing Busy("SIP/michiel-d5bd", "") in new stack
== Spawn extension (internal, 06XXXXXXXXX, 104) exited non-zero on
'SIP/michiel-d5bd'
-- Got SIP response 483 "Too many hops" back from 81.23.228.150
I tripple checked my password, and I am sure it is correct.
What to do ?
--
Michiel van Baak
http://lunteren.vanbaak.info
michiel@vanbaak.info
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D
"Two of the most famous products of Berkeley are LSD and BSD. I don't
think that this is a coincidence."
_______________________________________________
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Dear all, I'm trying to get asterisk to register to budgetphone.nl. In several threads I saw people who got this to work: http://www.voip-info.org/wiki-Talkin2ya http://lists.digium.com/pipermail/asterisk-users/2005-March/092850.html But I've spent a whole saturday on it now and didn't get any further. I also have a granstrema handytone 486. This thing manages to register. I've tried to look into the differences between the sip messages with ethereal. In ethereal I see the following sip conversation for the handytone: 192.168.0.60->81.23.228.150 SIP Request: REGISTER sip:budgetphone.nl 81.23.228.150->192.168.0.60 SIP Status: 401 Unauthorized (0 bindings) 192.168.0.60->81.23.228.150 SIP Request: REGISTER sip:budgetphone.nl 81.23.228.150->192.168.0.60 SIP Status: 200 OK (1 bindings) In the first message the handytone tries to register, but it gets a request for authentication (second packet) with a challenge. The third packet is a retry to register, but this time with the response to the challenge. The fourth packet is then the confirmation that all went well. When I do the same with asterisk I get the following 192.168.0.35->81.23.228.150 SIP Request: REGISTER sip:budgetphone.nl 81.23.228.150->192.168.0.35 SIP Status: 401 Unauthorized (0 bindings) 192.168.0.35->81.23.228.150 SIP Request: REGISTER sip:budgetphone.nl 192.168.0.35->81.23.228.150 SIP Request: REGISTER sip:budgetphone.nl 192.168.0.35->81.23.228.150 SIP Request: REGISTER sip:budgetphone.nl ... Asterisk gives a response to the challenge, but never gets an answer back. What is going wrong? Hope someone can shed some light here.. ; SIP Configuration for Asterisk ; [general] context=default ; Default context for incoming calls recordhistory=yes ; Record SIP history by default srvlookup=yes ; Enable DNS SRV lookups on outbound calls language=en ; Default language setting for all users/peers nat=no defaultexpirey=1200 disallow=all allow=g729 allow=gsm allow=ulaw allow=alaw register => 31437110310:PASSWD@budgetphone.nl/31437110310 [31437110310] type=friend context=from-budgetphone host=budgetphone.nl callerid="John Doe" fromuser=31437110310 fromdomain=budgetphone.nl username=31437110310 insecure=very secret=PASSWD qualify=no canreinvite=no nat=yes port=5060 -------------------------------------------------------