[Asterisk-Users] Sip problems

[jorge at redcetus.com]

Hi,

I have been trying to configure one Asterisk to use a Sip provider.

My sip.conf is:

register => user:passwd at www.xxx.yyy.zzz

[www.xxx.yyy.zzz]
type=friend
secret=passwd
username=user
host=www.xxx.yyy.zzz
insecure=very
disallow=all
allow=g729;gsm;ulaw;alaw
reinvite=no

[sipphone]
;dtmfmode=info
host=dynamic
language=es
nat=yes
secret=mysecret
type=friend
username=sipphone
allow=g729;ilbc;gsm;ulaw;alaw
regseconds=0
cancallforward=yes


The problem is:

The outgoing call doesn't works, SIP responses 403 in my sip phone

the sip debug say


Sip read:

SIP/2.0 403 Insufficient Balance
Via: SIP/2.0/UDP aaa.bbb.ccc.ddd:5060;branch=z9hG4bK6d2b3580
Record-Route: <sip:www.xxx.yyy.zzz;ftag=as752586f3;lr>
From: 1090 <sip:1090 at aaa.bbb.ccc.ddd>;tag=as752586f3
To: <sip:5642216156 at www.xxx.yyy.zzz>;tag=65a531031f6d1fcdde9ff201087cff4e
Call-ID: 31d3fdd97a098fd87fed1b6d4e83d8dc at aaa.bbb.ccc.ddd
CSeq: 102 INVITE
Server: Sippy


chan_sip.c:6864 handle_response: Forbidden - wrong password on authentication
for INVITE

I have installed a sip phone direct to provider, and outgoing call works.

I will be happy about any suggestions.

Thanks in advance!


Jorge Verastegui
redcetus.com
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