Hi,
I have the following setup:
E100P SER <----> * <-----> PBX
This works just fine, except when there are users on both boxes (ie. SER and asterisk), whose usernames are the same, although the realm is different.
An example:
user '[EMAIL PROTECTED]' wants to call some extension in the PBX, but as user '[EMAIL PROTECTED]' exists too, * tries to authenticate the user, which it shouldn't do, at least I guess so.
Shouldn't asterisk differentiate between the realms ie. [EMAIL PROTECTED] != [EMAIL PROTECTED] ?
Find attached, the relevant part of the logged sip communication and the sip.conf.
If you have any hints, please let me know. Thanks in advance,
best regards, Kurt
<example sip.log>
Sip read:
INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0
Max-Forwards: 10
Record-Route: <sip:[EMAIL PROTECTED];ftag=000cce3a7be800087fd8099f-62cc5396;lr=on>
Via: SIP/2.0/UDP 83.136.32.160;branch=z9hG4bK1cba.998c27f1.0
Via: SIP/2.0/UDP 131.130.220.101:5060;branch=z9hG4bK03521c86
From: "Kurt Bauer" <sip:[EMAIL PROTECTED]>;tag=000cce3a7be800087fd8099f-62cc5396
To: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
Date: Mon, 06 Sep 2004 10:01:57 GMT
CSeq: 102 INVITE
User-Agent: CSCO/7
Contact: <sip:[EMAIL PROTECTED]:5060>
Expires: 180
Content-Type: application/sdp
Content-Length: 253
v=0 o=Cisco-SIPUA 23148 13380 IN IP4 131.130.220.101 s=SIP Call c=IN IP4 131.130.220.101 t=0 0 m=audio 30596 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15
15 headers, 11 lines
Using latest request as basis request
Sending to 83.136.32.160 : 5060 (non-NAT)
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 131.130.220.101:30596
Found description format PCMU
Found description format PCMA
Found description format G729
Found description format telephone-event
Capabilities: us - 0x10f(G723|GSM|ULAW|ALAW|G729A), peer - audio=0x10c(ULAW|ALAW|G729A)/video=0x0(EMPTY), combined - 0x10c(ULAW|ALAW|G729A)
Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x0(EMPTY)
Reliably Transmitting (no NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 83.136.32.160;branch=z9hG4bK1cba.998c27f1.0
Via: SIP/2.0/UDP 131.130.220.101:5060;branch=z9hG4bK03521c86
From: "Kurt Bauer" <sip:[EMAIL PROTECTED]>;tag=000cce3a7be800087fd8099f-62cc5396
To: <sip:[EMAIL PROTECTED]>;tag=as6191c2dd
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:[EMAIL PROTECTED]>
Proxy-Authenticate: Digest realm="troubadix.univie.ac.at", nonce="5276f268"
Content-Length: 0
to 83.136.32.160:5060
Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms
Found user 'kb'
troubadix*CLI>
Sip read:
ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP 83.136.32.160;branch=z9hG4bK1cba.998c27f1.0
From: "Kurt Bauer" <sip:[EMAIL PROTECTED]>;tag=000cce3a7be800087fd8099f-62cc5396
Call-ID: [EMAIL PROTECTED]
To: <sip:[EMAIL PROTECTED]>;tag=as6191c2dd
CSeq: 102 ACK
User-Agent: Sip EXpress router(0.8.12-tcp_nonb-tls (i386/linux))
Content-Length: 0
8 headers, 0 lines
</example sip.log>
-->note the "SIP/2.0 407 Proxy Authentication Required"
<sip.conf> ; ; SIP Configuration for Asterisk ; [general]
port=5060 bindaddr=0.0.0.0 realm=troubadix.univie.ac.at disallow=all allow=ulaw allow=alaw allow=g729 allow=g723.1 allow=gsm
[at43_in] type=peer host=sip.at43.at context=at43 insecure=very deny=0.0.0.0/0.0.0.0 permit=83.136.32.160/255.255.255.255
</sip.conf>
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