Switching from Trixbox to FreePBX because I need long term support. Issue has come up when calling an extension - no answer. Works in Trixbox but not FreePBX. Wireshark trace shows a “401 unauthorized” response to an INVITE whereas with Trixbox I receive the response “407 Proxy Identification Required”. Below is a trace with the invite message and the subsequent response. FreePBX is ver 2.10 running on Centos v5.8 Asterisk ver 1.8.16.0. TrixBox is ver 2.6.2.3 on Centos 5.3 using Asterisk v1.4.26.2.
I should also mention that the extension 1011 is a custom SIP device which doesn’t register with the server so when I run SIP SHOW PEERS on the CLI the status of the extensions are UNKNOWN. Any suggestions on what I could try?
FreePBX trace start
INVITE sip:[email protected]:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.0.0.212;branch=z9hG4bK56992aec14CDFCAB
From: “210” sip:[email protected];tag=C0EB91EE-CF399CAD
To: sip:[email protected];user=phone
CSeq: 1 INVITE
Call-ID: [email protected]
Contact: sip:[email protected]
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSpectraLink-SL_8440-UA/4.0.0.27539
Accept-Language: en
Supported: 100rel,replaces
Allow-Events: conference,talk,hold
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 342
v=0
o=- 54166695 54166695 IN IP4 10.0.0.212
s=Polycom IP Phone
c=IN IP4 10.0.0.212
t=0 0
a=sendrecv
m=audio 2262 RTP/AVP 9 102 0 8 18 127
a=rtpmap:9 G722/8000
a=rtpmap:102 G7221/16000
a=fmtp:102 bitrate=32000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:127 telephone-event/8000
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.0.0.212;branch=z9hG4bK56992aec14CDFCAB;received=10.0.0.212
From: “210” sip:[email protected];tag=C0EB91EE-CF399CAD
To: sip:[email protected];user=phone;tag=as12fe44d3
Call-ID: [email protected]
CSeq: 1 INVITE
Server: FPBX-2.10.1(1.8.16.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="7706eaa9"
Content-Length: 0
FreePBX Trace end
Trixbox Trace start
INVITE sip:[email protected]:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.0.0.212;branch=z9hG4bKf9eb6c8b3C75B1CA
From: “210” sip:[email protected];tag=F6062C8D-49522ACC
To: sip:[email protected];user=phone
CSeq: 1 INVITE
Call-ID: [email protected]
Contact: sip:[email protected]
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSpectraLink-SL_8440-UA/4.0.0.27539
Accept-Language: en
Supported: 100rel,replaces
Allow-Events: conference,talk,hold
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 342
v=0
o=- 54166997 54166997 IN IP4 10.0.0.212
s=Polycom IP Phone
c=IN IP4 10.0.0.212
t=0 0
a=sendrecv
m=audio 2264 RTP/AVP 9 102 0 8 18 127
a=rtpmap:9 G722/8000
a=rtpmap:102 G7221/16000
a=fmtp:102 bitrate=32000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:127 telephone-event/8000
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.0.0.212;branch=z9hG4bKf9eb6c8b3C75B1CA;received=10.0.0.212
From: “210” sip:[email protected];tag=F6062C8D-49522ACC
To: sip:[email protected];user=phone;tag=as4fd12b06
Call-ID: [email protected]
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="7de61d1d"
Content-Length: 0
TrixBox trace ends