Music On Hold for SNOM phones with freeswitch
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@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@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@patton.logicunited.com>;tag=d3h27cphyo To: <sip:9999@192.168.140.7;user=phone> Call-ID: 3c28f4234780-u9oqvzvds56e CSeq: 1 INVITE Max-Forwards: 70 Contact: <sip:central@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@patton.logicunited.com>;tag=d3h27cphyo To: <sip:9999@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@patton.logicunited.com>;tag=d3h27cphyo To: <sip:9999@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
Hi, thanks for the hint, I was addressing the internal profile, which needs authentication, so I am now going to port 5080 (profile external, dialplan public) and added the XML for the hold music extension to the public.xml dialplan.
From SIP Trace this works fine, have to try it out tomorrow when in the office to see if the phone does play music on hold :-)
Best Regards Christian On 15.04.2010, at 15:50, Brian West wrote:
Chance are you'll need to turn off authentication on the profile in question.
/b
On Apr 15, 2010, at 8:44 AM, Christian van der Leeden wrote:
I receive a 407 Proxy authentication required.
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participants (2)
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Brian West
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Christian van der Leeden