[asterisk-users] asterisk server to server using sip question

[Jerry Geis]

Steve,

Yes Box1 does have multiple other peers. Do you know of something to try?

Jerry

> / -----Original Message-----
/>/ From: asterisk-users-bounces at lists.digium.com 
<http://lists.digium.com/mailman/listinfo/asterisk-users> 
[mailto:asterisk-users-
/>/ bounces at lists.digium.com 
<http://lists.digium.com/mailman/listinfo/asterisk-users>] On Behalf Of 
Jerry Geis
/>/ Sent: Thursday, September 14, 2006 8:14 AM
/>/ To: asterisk-users at lists.digium.com 
<http://lists.digium.com/mailman/listinfo/asterisk-users>
/>/ Subject: [asterisk-users] asterisk server to server using sip question
/>/ />/ I have 2 asterisk servers. I am trying to connect them with SIP and
/>/ getting an error.
/>/ My first box I define sip.conf as:
/>/ />/ [devcentos64_to_bt610tMM]
/>/ type=friend
/>/ username=devcentos64_to_bt610tMM
/>/ secret=password
/>/ disallow=all
/>/ allow=ulaw
/>/ allow=alaw
/>/ allow=gsm
/>/ host=192.168.1.159
/>/ context=default
/>/ />/ my second box I define sip.conf as:
/>/ [devcentos64_to_bt610tMM]
/>/ type=friend
/>/ username=devcentos64_to_bt610tMM
/>/ secret=password
/>/ disallow=all
/>/ allow=ulaw
/>/ allow=alaw
/>/ allow=gsm
/>/ host=192.168.1.10
/>/ context=default
/>/ />/ So Box2 points to Box1 and Box1 points to Box2 by the host= fields.
/>/ />/ I am getting the following error:
/>/     -- Attempting call on SIP/devcentos64_to_bt610tmm/1041 for
/>/ smvoice_callprogress at smvoice-dialout 
<http://lists.digium.com/mailman/listinfo/asterisk-users>:1 (Retry 1)
/>/ Sep 14 08:10:52 WARNING[2512]: chan_sip.c:9715 handle_response_invite:
/>/ Forbidden - wrong password on authentication for INVITE to '"Jerry
/Geis
> / 204" <sip:3175661677 at 192.168.1.10 
> <http://lists.digium.com/mailman/listinfo/asterisk-users>>;tag=as07330b38' 
>
/>/        > Channel SIP/devcentos64_to_bt610tmm-007afe00 was never
/answered.
> / Sep 14 08:10:52 WARNING[4639]: cdr.c:550 ast_cdr_disposition: Cause
/not
> / handled
/>/ />/ />/ Why is that??? My passwords match. I am using asterisk.1.2.11
/>/ Or what is the correct way to connect asterisk SIP server to asterisk
/>/ SIP server.
/>/ />/ Jerry
/>/ />/ /

Do you have other peers on the same boxes pointing to each other?