Hi Allan! Did you ever manage to solve this issue? I saw the same behavior yesterday while was trying to accept incoming calls from CS2K FreeSWITCH Version 1.3.13b+git~20130214T200725Z~b14fd4a5a4 (git b14fd4a 2013-02-14 20:07:25Z) Best regards, Kirill ######################################################################### *Allan Piske*zyryznet at gmail.com<mailto:freeswitch-users%40lists.freeswitch.org?Subject=%5BFreeswitch-users%5D%20hangup%20error%20missing%20remote%20port&In-Reply-To=> /Sun Sep 4 08:44:25 MSD 2011/ Hi, I'm having some trouble discovering the source and solution to one problem. I setup a FS box for receiving faxes, and a already have another FS box as SBC. SPA3102 ---> opensips ---> FS_SBC --> opensips --> FS_FAX this works ok, call gets answered, i hear the fax tone, the SPA3102 changes to T38 and everyone is happy. Nortel CS2000 ------> FS_SBC -------> opensips(regitrar,rtpproxy,etc) ---> FS_FAX but his doen't. This little devil CS2000 sends every invite with m=image header along with m=audio. With aparently FS doesn't like. like that: freeswitch at internal <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>> recv 1215 bytes from udp/[10.143.82.250]:5060 at 08:26:08.226858: ------------------------------------------------------------------------ INVITE sip:4730365302 at 10.143.82.253 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>:5060;transport=UDP;user=phone SIP/2.0 Record-Route: <sip:10.143.82.250;lr=on;ftag=-45026-10d9cdb-2668a317-10d9cdb> f: <sip:4784064435 at 10.150.65.16 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>:5060 ;user=phone>;tag=-45026-10d9cdb-2668a317-10d9cdb t: <sip:4730365302 at 10.143.82.250 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>:5060;user=phone> i: 416cf3e01041960a13c410d9cdb1d2cba42b82409c10f14bb8-0322-4481 CSeq: 1 INVITE User-agent: CS2000_NGSS/9.0 P-Asserted-Identity: <sip:4784064435 at 10.150.65.16 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>;user=phone> Allow: ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK,UPDATE Via: SIP/2.0/UDP 10.143.82.250;branch=z9hG4bK1d78.5262ccd1.0 v: SIP/2.0/UDP PAE1CS2K:5060;maddr=10.150.65.16;branch=z9hG4bK-10d9cdb-1d2cba43-57ebb21d Max-Forwards: 69 m: <sip:10.150.65.16:5060;transport=UDP> k: 100rel c: application/sdp l: 420 v=0 o=PVG 1315110636830 1315110636830 IN IP4 10.152.204.202 s=- p=+1 6135555555 c=IN IP4 10.152.204.202 t=0 0 m=audio 55920 RTP/AVP 18 8 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=fmtp:18 annexb=no m=image 64112 udptl t38 a=T38FaxVersion:0 a=T38FaxMaxBuffer:1100 a=T38FaxMaxDatagram:612 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxUdpEC:t38UDPRedundancy But then we have a problem ... My beloved FS_SBC changes the SDP before sending it to the next hop .. as it should becase they are on diferent subnets and set to proxy_media 2011-09-04 01:26:08.231102 [DEBUG] sofia_glue.c:1759 sofia/tpa/4730365302 Patched SDP --- v=0 o=PVG 1315110636830 1315110636830 IN IP4 10.152.204.202 s=- p=+1 6135555555 c=IN IP4 10.152.204.202 t=0 0 m=audio 55920 RTP/AVP 18 8 101 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 m=image 64112 udptl t38 a=T38FaxVersion:0 a=T38FaxMaxBuffer:1100 a=T38FaxMaxDatagram:612 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxUdpEC:t38UDPRedundancy +++ v=0 o=FreeSWITCH 2718891449 2718891450 IN IP4 A.B.C.D s=FreeSWITCH p=+1 6135555555 c=IN IP4 A.B.C.D t=0 0 m=audio 19262 RTP/AVP 18 8 101 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 m=image 19262 udptl t38 a=T38FaxVersion:0 a=T38FaxMaxBuffer:1100 a=T38FaxMaxDatagram:612 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxUdpEC:t38UDPRedundancy so we have now 2 m= fields with the same port !!!!! then when it gets answered on the FS_FAX box .. we got disconnection on the FS_SBC box 2011-09-04 01:26:08.257064 [DEBUG] sofia.c:4761 Channel sofia/tpa/4730365302 entering state [completing][200] 2011-09-04 01:26:08.257064 [DEBUG] sofia.c:4772 Remote SDP: v=0 o=FreeSWITCH 1315079202 1315079203 IN IP4 189.45.192.19 s=FreeSWITCH c=IN IP4 189.45.192.19 t=0 0 m=audio 31436 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 m=image 0 udptl 19 2011-09-04 01:26:08.258409 [DEBUG] sofia.c:4761 Channel sofia/tpa/4730365302 entering state [ready][200] 2011-09-04 01:26:08.258409 [DEBUG] switch_channel.c:2830 (sofia/tpa/4730365302) Callstate Change RINGING -> ACTIVE 2011-09-04 01:26:08.258409 [DEBUG] switch_channel.c:2842 Send signal sofia/voxip/4784064435 at 10.150.65.16 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>:5060 [BREAK] 2011-09-04 01:26:08.258409 [NOTICE] sofia.c:5318 Channel [sofia/tpa/4730365302] has been answered 2011-09-04 01:26:08.258409 [DEBUG] sofia_glue.c:2774 Set Codec sofia/tpa/4730365302 PROXY/8000 20 ms 160 samples 0 bits 2011-09-04 01:26:08.258409 [DEBUG] sofia_glue.c:3079 PROXY AUDIO RTP [sofia/tpa/4730365302] A.B.C.D:19262->W.X.Y.Z:0 codec: 0 ms: 20 2011-09-04 01:26:08.258409 [ERR] sofia_glue.c:3512 AUDIO RTP REPORTS ERROR: [Missing remote port] 2011-09-04 01:26:08.258409 [DEBUG] switch_channel.c:2563 (sofia/tpa/4730365302) Callstate Change ACTIVE -> HANGUP 2011-09-04 01:26:08.258409 [NOTICE] sofia_glue.c:3513 Hangup sofia/tpa/4730365302 [CS_CONSUME_MEDIA] [DESTINATION_OUT_OF_ORDER] 2011-09-04 01:26:08.258409 [DEBUG] switch_channel.c:2579 Send signal sofia/tpa/4730365302 [KILL] 2011-09-04 01:26:08.258409 [DEBUG] switch_core_session.c:1116 Send signal sofia/tpa/4730365302 [BREAK] 2011-09-04 01:26:08.259605 [DEBUG] switch_ivr_originate.c:3299 Originate Resulted in Error Cause: 27 [DESTINATION_OUT_OF_ORDER] 2011-09-04 01:26:08.259605 [INFO] mod_dptools.c:2647 Originate Failed. Cause: DESTINATION_OUT_OF_ORDER port = 0 !!! is that maybe a fault in opensips in the middle because the both m= fields with the same port ? , or one of the FS boxes ? something is broken here, sadly :( Appreciate any help. Allan