[Asterisk-Users] SER authenitification failure on ASTERISK
Juraj Varchol
varchol at gmail.com
Wed Nov 16 11:51:47 MST 2005
Hey guys,
I'm quite new in this SER/Asterisk environment and I'm having some
problems with authentification. Now i'm running SER and Asterisk on 2
boxes. It is working according to the scheme
SIP user <------> SER Server <--------> Asterisk <----->IAX users/PSTN/...
SIP calls are managed by SER Server and the rest is forwarded to
Asterisk for further processing. Also acording to dialing plan on
Asterisk, calls to SIP users are fwd to SER Server where are
redirected to their actual location.
Outgoing calls to SER Server are set up:
**extensions.conf**
exten => _77.,1,Dial(SIP/${EXTEN:2}@sip_proxy-out,10)
exten => _77.,2,Hangup
**sip.conf** - where only for outgoing calls is set up context
[sip_proxy-out]
type=peer
username= asterisk_user-name_on_SER_Server
secret=asterisk_passwd_on_SER_Server
allow=all
host=SER_Server_IP or URL
----> this way it does work even sometimes UA on SER shows as incoming
call from "asterisk_user-name_on_SER_Server" instead of callerID from
Asterisk ... but that would not be such a problem
... also is important to mention that
"asterisk_user-name_on_SER_Server" is registred on SER Server. (* if
there is better way how to deal with it i would b very glad to at
least get pointed the right direction ... ;))
!!!!THE PROBLEM comes the other way when SER users want to call
extension that is managed by ASTERISK.
I get error message in both cases that r described at the end. I'm
looking for some solution that would not include registering all SER
users on Asterisk or just omit authentification for all users coming
from SER_Server
Thanx in advance,
Juraj
**error message**
Nov 16 19:13:53 NOTICE[9339]: chan_sip.c:7355 handle_request: Failed
to authenticate user <sip:SER_USER at SER_SERVER_IP:5060>
;tag=93af6721-13c4-437b76e2-5f0d59d-2fee
*****both options i tried but no luck**************
my setup on SER Server is
**ser.cfg**
....
if (uri=~"^sip:[4-9]*@(IP_SER_SERVER|(proxy\.)?URL_SER_SERVER)") {
rewritehostport("IP_ASTERISK_SERVER:5060");
forward(uri:host, uri:port);
break;
};
...
**extensions.cong** --- i tried to redirect incoming call right away
to normal dialing plan...
[inpbx]
...
exten => 456,2,Dial(IAX2/User_1,15)
exten => 456,3,Voicemail(u44 at test)
...
exten => 456,6,Hangup
...
exten => 567,2,Dial(IAX2/User_2,15)
....
exten => 567,6,Hangup
...
OR modification 2 ... when we want to reach Asterisk Server must by
dialed prefix (for example 44_) + extensions by SIP user.
**extension.conf**
[default]
exten => _44.,1,Dial(SIP/${EXTEN:2}@sip_proxy,15,rm) ;here i strip
it from the extension
exten => _44.,2,Hangup
**sip.conf**
[sip_proxy]
type=friend
context=inpbx
;insecure=very
host=URL_SER_SERVER
;mailbox=user at test
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