The default behaviour for INVITEs without SDP is to reject the call because media cannot be negotiated. It's possible to accept these INVITEs by setting the following parameter on the SIP profile: <param name="enable-3pcc" value="true"/> See these previous discussions from the archive: http://www.mail-archive.com/freeswitch-users@lists.freeswitch.org/msg03504.h... http://www.mail-archive.com/freeswitch-users@lists.freeswitch.org/msg11825.h... Warm regards, -Steve On 10 November 2010 15:22, SIC FS LIST <sicfslist@gmail.com> wrote:
Hello, I have a working FS instance I am using as a redirect server (to serve up LNP requests). It works fine on calls with invites that have SDP and does not work with invites without SDP. I enabled 3pcc to true thinking that would fix the issue. Version info is FreeSWITCH Version 1.0.6 (hacked-20100921T052029Z). With the console log level set to debug the only thing I see is this message (just before returning a 480): freeswitch@lnpdal0001> 2010-11-10 08:58:50.949818 [CRIT] switch_core_state_machine.c:382 e390f5b0-ecda-11df-906a-13020c8dafd9 Timeout waiting for next instruction in CS_NEW! I also do not ever see FS to an xml http req (which is how we control the dialplan). freeswitch@lnpdal0001> recv 495 bytes from udp/[X.X.X.X]:5060 at 14:57:50.934431: ------------------------------------------------------------------------ INVITE sip:19033226103@X.X.X.X:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bK11288997ec8ll7466dfINV1db35e674cd968d5 Max-Forwards: 70 Contact: <sip:X.X.X.X:5060> To: <sip:19033226103@X.X.X.X:5060> From: <sip:X.X.X.X:5060>;tag=1db35e67-co6935-INS001 Call-ID: 993186-34928005397-80464@ens.com CSeq: 693501 INVITE Date: Tue, 09 Nov 2010 15:29:25 GMT Supported: 100rel User-Agent: ENSR3.0.63.0-IS1-RMRG2101-RG21-CPO11152 Content-Length: 0 ------------------------------------------------------------------------ send 336 bytes to udp/[X.X.X.X]:5060 at 14:57:50.935324: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bK11288997ec8ll7466dfINV1db35e674cd968d5 From: <sip:X.X.X.X:5060>;tag=1db35e67-co6935-INS001 To: <sip:19033226103@X.X.X.X:5060> Call-ID: 993186-34928005397-80464@ens.com CSeq: 693501 INVITE User-Agent: lnpdal0001.sipinterchange.com Content-Length: 0 ------------------------------------------------------------------------ send 714 bytes to udp/[X.X.X.X]:5060 at 14:58:50.951429: ------------------------------------------------------------------------ SIP/2.0 480 Temporarily Unavailable Via: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bK11288997ec8ll7466dfINV1db35e674cd968d5 From: <sip:X.X.X.X:5060>;tag=1db35e67-co6935-INS001 To: <sip:19033226103@X.X.X.X:5060>;tag=t7ycB64mX33mQ Call-ID: 993186-34928005397-80464@ens.com CSeq: 693501 INVITE User-Agent: lnpdal0001.sipinterchange.com Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, refer Reason: Q.850;cause=81;text="INVALID_CALL_REFERENCE" Content-Length: 0 Remote-Party-ID: "19033226103" <sip:19033226103@X.X.X.X>;party=calling;privacy=off;screen=no ------------------------------------------------------------------------ Here is the sofia profile: <profile name="external"> <aliases> <alias name="outbound"/> <alias name="nat"/> <!-- for backwards compatibility --> </aliases> <domains> <domain name="all" alias="false" parse="true"/> </domains> <settings> <param name="user-agent-string" value="lnpdal0001"/> <param name="debug" value="0"/> <param name="sip-trace" value="no"/> <param name="context" value="public"/> <param name="enable-100rel" value="false"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="$${global_codec_prefs}"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="X.X.X.X"/> <param name="sip-ip" value="X.X.X.X"/> <param name="manage-presence" value="false"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="bind-params" value="transport=udp"/> <param name="tls" value="$${external_ssl_enable}"/> <!-- <param name="pass-rfc2833" value="true"/> --> <!-- <param name="inbound-proxy-media" value="true"/> --> <param name="inbound-bypass-media" value="true"/> <param name="inbound-late-negotiation" value="true"/> <param name="accept-blind-reg" value="false"/> <param name="accept-blind-auth" value="true"/> <param name="nonce-ttl" value="60"/> <param name="disable-transcoding" value="true"/> <param name="auth-calls" value="false"/> <param name="inbound-reg-force-matching-username" value="false"/> <param name="auth-all-packets" value="false"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> <param name="challenge-realm" value="auto_to"/> <param name="enable-3pcc" value="true"/> </settings> </profile> Any thoughts / ideas / help would be greatly appreciated. Thanks! SDR _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org