[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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