I was having trouble doing call forwarding from my SIP phone that is connected to FreeSWITCH. It turns out that my SIP phone is actually sending 302 Moved Temporarily responses, but my SIP gateway does not support 302 Moved Temporarily or SIP REFER messages. How do I get FreeSWITCH to forward calls without sending 302 Moved Temporarily or SIP REFER messages?
Here is the SIP debug from our gateway: Received: INVITE sip:+19725357...@ipipgw.ipdimensions.com:5060;user=phone;transport=UDP;maddr=168.75.202.246 SIP/2.0 v: SIP/2.0/UDP 65.243.172.245:5060;branch=z9hG4bKe19865e46222056ca70435e66fde4127.19be3eb0 Record-Route: <sip:65.243.172.245:5060;lr> v: SIP/2.0/UDP 63.77.76.236:5060;branch=z9hG4bK3f49bc4eb4ac163ffa354de0e6384d30.12e7ffbd;received=63.77.76.236 record-route: <sip:63.77.76.236;lr> f: <sip:+19729555...@199.173.101.208:5060;user=phone>;tag=dc7-13c4-3ec95a-3ad03068-3ec95a t: <sip:+19725357...@63.77.76.236:5060;user=phone> i: a1f37fb0d065adc713c43ec95af54289baa8ec2034c293850-0569-7989 CSeq: 1 INVITE Max-Forwards: 18 k: 100rel, replaces allow: ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK v: SIP/2.0/UDP DAL4:5060;maddr=199.173.101.208;branch=z9hG4bK-3ec95a-f54289ba-139ab2d1;received=199.173.101.208 m: <sip:199.173.101.208:5060;transport=UDP> c: application/SDP l: 210 P-Asserted-Identity: <sip:9729555...@63.77.76.236;user=phone> Privacy: none v=0 o=- 540754816 540754816 IN IP4 199.173.111.141 s=- c=IN IP4 199.173.111.141 t=0 0 m=audio 30056 RTP/AVP 18 0 8 101 a=ptime:20 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 Nov 24 15:08:00.367 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 100 Trying Via: SIP/2.0/UDP 65.243.172.245:5060;branch=z9hG4bKe19865e46222056ca70435e66fde4127.19be3eb0,SIP/2.0/UDP 63.77.76.236:5060;branch=z9hG4bK3f49bc4eb4ac163ffa354de0e6384d30.12e7ffbd;received=63.77.76.236,SIP/2.0/UDP DAL4:5060;maddr=199.173.101.208;branch=z9hG4bK-3ec95a-f54289ba-139ab2d1;received=199.173.101.208 From: <sip:+19729555...@199.173.101.208:5060;user=phone>;tag=dc7-13c4-3ec95a-3ad03068-3ec95a To: <sip:+19725357...@63.77.76.236:5060;user=phone> Date: Tue, 24 Nov 2009 21:08:00 GMT Call-ID: a1f37fb0d065adc713c43ec95af54289baa8ec2034c293850-0569-7989 CSeq: 1 INVITE Allow-Events: telephone-event Server: Cisco-SIPGateway/IOS-12.x Content-Length: 0 Nov 24 15:08:00.367 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Sent: INVITE sip:19725357...@168.75.202.212:5062 SIP/2.0 Via: SIP/2.0/UDP 168.75.202.246:5060;branch=z9hG4bK6821870 From: <sip:19729555...@168.75.202.246>;tag=14E93594-2488 To: <sip:19725357...@168.75.202.212> Date: Tue, 24 Nov 2009 21:08:00 GMT Call-ID: 4802bacc-d87411de-ac70d9df-3419a...@168.75.202.246 Supported: timer,resource-priority,replaces Min-SE: 1800 Cisco-Guid: 1208058493-3631485406-2892683743-874095366 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER CSeq: 101 INVITE Timestamp: 1259096880 Contact: <sip:19729555...@168.75.202.246:5060> Expires: 180 Allow-Events: telephone-event Max-Forwards: 17 P-Asserted-Identity: <sip:19729555...@168.75.202.246> Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 314 v=0 o=CiscoSystemsSIP-GW-UserAgent 2925 1780 IN IP4 168.75.202.246 s=SIP Call c=IN IP4 199.173.111.141 t=0 0 m=audio 30056 RTP/AVP 18 0 8 101 c=IN IP4 199.173.111.141 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 Nov 24 15:08:00.367 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 100 Trying Via: SIP/2.0/UDP 168.75.202.246:5060;branch=z9hG4bK6821870 From: <sip:19729555...@168.75.202.246>;tag=14E93594-2488 To: <sip:19725357...@168.75.202.212> Call-ID: 4802bacc-d87411de-ac70d9df-3419a...@168.75.202.246 CSeq: 101 INVITE Timestamp: 1259096880 0.000342 User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15654M Content-Length: 0 Nov 24 15:08:00.419 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 302 Moved Temporarily Via: SIP/2.0/UDP 168.75.202.246:5060;branch=z9hG4bK6821870 From: <sip:19729555...@168.75.202.246>;tag=14E93594-2488 To: <sip:19725357...@168.75.202.212>;tag=49aF8vtgHme2c Call-ID: 4802bacc-d87411de-ac70d9df-3419a...@168.75.202.246 CSeq: 101 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15654M 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=16;text="NORMAL_CLEARING" Content-Length: 0 P-Asserted-Identity: "19725357722" <sip:19725357...@168.75.202.212> Nov 24 15:08:00.427 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Sent: ACK sip:19725357...@168.75.202.212:5062 SIP/2.0 Via: SIP/2.0/UDP 168.75.202.246:5060;branch=z9hG4bK6821870 From: <sip:19729555...@168.75.202.246>;tag=14E93594-2488 To: <sip:19725357...@168.75.202.212>;tag=49aF8vtgHme2c Date: Tue, 24 Nov 2009 21:08:00 GMT Call-ID: 4802bacc-d87411de-ac70d9df-3419a...@168.75.202.246 Max-Forwards: 70 CSeq: 101 ACK Allow-Events: telephone-event Content-Length: 0 _________________________________________________________________ Windows 7: I wanted simpler, now it's simpler. 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