[Freeswitch-users] Music On Hold for SNOM phones with freeswitch

Christian van der Leeden fanta at me.com
Thu Apr 15 06:44:05 PDT 2010


Hi,

	I'd like to configure freeswitch so it acts as music on hold server (replying to a
SIP Invite with music) for a SNOM phone. The SNOM phone is registered on my
Patton SmartNode. I've tried to point it to 9999 at 192.168.140.7 (ip of my server)
but freeswitch responds with 
just installed freeswitch.org on my linux box. I'd like to use it play music
on hold to a SNOM VoIP phone (it is registered on my patton SmartNode, not
with the freeswitch server).

I receive a 407 Proxy authentication required.

This is on a out of the box "quick install" from the git trunk.

What would I have to do, so freeswitch would answer to the invite?

Christian
P.S.: attached the session log

INVITE sip:9999 at 192.168.140.7;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.140.42:1024;branch=z9hG4bK-d0e7ezmhzuzs;rport
From: "Zentrale" <sip:central at patton.logicunited.com>;tag=d3h27cphyo
To: <sip:9999 at 192.168.140.7;user=phone>
Call-ID: 3c28f4234780-u9oqvzvds56e
CSeq: 1 INVITE
Max-Forwards: 70
Contact: <sip:central at 192.168.140.42:1024>;reg-id=1
P-Key-Flags: resolution="31x13", keys="4"
User-Agent: snom370/7.3.30
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Session-Expires: 3600;refresher=uas
Min-SE: 90
Content-Type: application/sdp
Content-Length: 452

v=0
o=root 30256492 30256492 IN IP4 192.168.140.42
s=call
c=IN IP4 192.168.140.42
t=0 0
m=audio 54834 RTP/AVP 0 8 9 2 3 18 4 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:TpHIcA93pqErUArxCuWRC1tcd7mp8LTs73clcwnE
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:2 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

Received from udp:192.168.140.7:5060 at 15/4/2010 15:39:52:647 (362 bytes):

SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.140.42:1024;branch=z9hG4bK-d0e7ezmhzuzs;rport=1024
From: "Zentrale" <sip:central at patton.logicunited.com>;tag=d3h27cphyo
To: <sip:9999 at 192.168.140.7;user=phone>
Call-ID: 3c28f4234780-u9oqvzvds56e
CSeq: 1 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-d6ee682 2010-04-13 13:38:47 -0700
Content-Length: 0


Received from udp:192.168.140.7:5060 at 15/4/2010 15:39:52:802 (853 bytes):

SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.140.42:1024;branch=z9hG4bK-d0e7ezmhzuzs;rport=1024
From: "Zentrale" <sip:central at patton.logicunited.com>;tag=d3h27cphyo
To: <sip:9999 at 192.168.140.7;user=phone>;tag=4jg7ajj84BB7K
Call-ID: 3c28f4234780-u9oqvzvds56e
CSeq: 1 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-d6ee682 2010-04-13 13:38:47 -0700
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
Allow-Events: talk, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Proxy-Authenticate: Digest realm="patton.logicunited.com", nonce="82f5fa24-ae3a-4def-a67f-b097f599bc34", algorithm=MD5, qop="auth"
Content-Length: 0



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