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<font face="Helvetica, Arial, sans-serif">Hello,<br>
<br>
When I want to send a call from asterisk-server 1 to asterisk-server 2,
it fails.<br>
<br>
On Asterisk server 1 :<br>
<br>
register => user:passwd@asterisk1 ; Test TRUNK<br>
[trunk2]<br>
type=peer<br>
host=asterisk1<br>
username=user<br>
;defaultuser=user<br>
secret=passwd<br>
disallow=all<br>
allow=alaw<br>
allow=gsm<br>
qualify=yes<br>
canreinvite=no<br>
dtmfmode=rfc2833<br>
<br>
Dialplan on Asterisk server 1 :<br>
<br>
exten => </font><font face="Helvetica, Arial, sans-serif">3291</font><font
face="Helvetica, Arial, sans-serif">,1,NoOp()<br>
exten => </font><font face="Helvetica, Arial, sans-serif">3291</font><font
face="Helvetica, Arial, sans-serif">,n,Set(${CALLERID(all)}="3291"
<3291>)<br>
exten => </font><font face="Helvetica, Arial, sans-serif">3291</font><font
face="Helvetica, Arial, sans-serif">,n,Dial(SIP/trunk2/</font><font
face="Helvetica, Arial, sans-serif">3291</font><font
face="Helvetica, Arial, sans-serif">)<br>
<br>
<br>
On Asterisk server 2 I see the following when I make a call with a
Grandstream IP-phone, registered at Asterisk server 1 :<br>
<br>
[Mar 16 20:32:44] WARNING[1680]: chan_sip.c:12872 check_auth: username
mismatch, have <test7>, digest has <user><br>
[Mar 16 20:32:44] NOTICE[1680]: chan_sip.c:20235 handle_request_invite:
Failed to authenticate device "T 7"
<a class="moz-txt-link-rfc2396E" href="sip:test7@192.168.1.150"><sip:test7@192.168.1.150></a>;tag=as5a2d92df<br>
<br>
<br>
<br>
How come the credentials of the Grandstream IP-phone are used and not
the username + password of [trunk2] ??<br>
<br>
<br>
Kind regards,<br>
Jonas.<br>
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