I had a box which was running FS 1.4, so I upgraded it to 1.6. Now I’m getting incompatible destination when dialing out through my provider, despite having changed nothing else. I’ve tried changing the outbound codec prefs to just PCMU,PCMA in case that was it, but still no joy.Here’s a SIP trace, can we tell anything from this, or are there any other likely explanations given the upgrade?INVITE sip:0<desintationnumber>@sip.mysipprovider SIP/2.0Via: SIP/2.0/UDP 1.2.3.4:5080;rport;branch=z9hG4bKe57KcBtmmr9cB Max-Forwards: 69From: "0<sourcenumber>" <sip:<accountname>@sip.mysipprovider>;tag= DvXrSpNBaFc1e Call-ID: a553eda8-7fa4-1236-bd95-040118c15f01 CSeq: 118282234 INVITEUser-Agent: FreeSWITCH-mod_sofia/1.6.20-37-987c9b9~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFYSupported: timer, path, replacesAllow-Events: talk, hold, conference, referProxy-Authorization: Digest username="<accountname>", realm="sip.mysipprovider", nonce="81346aeb-453b-466e-ba50-5d027c1a55e3", cnonce="pVaZvn+ kEjaVvQQBGMFfAQ", algorithm=MD5, uri="sip:0<desintationnumber>@ sip.mysipprovider ", response="4702ba8b72c43939d5fa88528eafaf 2b", qop=auth, nc=00000001 Content-Type: application/sdpContent-Disposition: sessionContent-Length: 246X-FS-Support: update_display,send_infoRemote-Party-ID: "0<sourcenumber>" <sip:0<sourcenumber>@sip.mysipprovider>;party=calling; screen=yes;privacy=off
v=0o=FreeSWITCH 1517205769 1517205770 IN IP4 1.2.3.4s=FreeSWITCHc=IN IP4 1.2.3.4t=0 0m=audio 30824 RTP/AVP 0 101 13a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=rtpmap:13 CN/8000a=ptime:20------------------------------------------------------------ ------------ 2018-01-29 14:36:33.926697 [DEBUG] sofia.c:7084 Channel sofia/external/0<desintationnumber> entering state [calling][0] recv 338 bytes from udp/[<providerip>]:5060 at 14:36:34.354355:------------------------------------------------------------ ------------ SIP/2.0 100 TryingVia: SIP/2.0/UDP 1.2.3.4:5080;rport=5080;branch=z9hG4bKe57KcBtmmr9cB From: "0<sourcenumber>" <sip:<accountname>@sip.mysipprovider>;tag= DvXrSpNBaFc1e Call-ID: a553eda8-7fa4-1236-bd95-040118c15f01 CSeq: 118282234 INVITEUser-Agent: mysipproviderContent-Length: 0
------------------------------------------------------------ ------------ 2018-01-29 14:36:33.966698 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47730 size=148. Stream 5:CLEAR:START2018-01-29 14:36:34.066706 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47731 size=148. Stream 5:CLEAR:START2018-01-29 14:36:34.186741 [DEBUG] switch_core_io.c:448 Setting BUG Codec PCMU:02018-01-29 14:36:34.266726 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47732 size=148. Stream 5:CLEAR:START2018-01-29 14:36:34.366714 [INFO] switch_rtp.c:7268 Auto Changing audio port from 10.10.131.204:11782 to <myphonesip>:117822018-01-29 14:36:34.466703 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47733 size=148. Stream 5:CLEAR:START2018-01-29 14:36:34.666721 [DEBUG] switch_rtp.c:1463 [ zrtp engine]: WARNING! HELLO have been resent 5 times without a response. Raising ZRTP_EVENT_NO_ZRTP_QUICK event. ID=52018-01-29 14:36:34.666721 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47734 size=148. Stream 5:CLEAR:START2018-01-29 14:36:34.866711 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47735 size=148. Stream 5:CLEAR:START2018-01-29 14:36:35.066711 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47736 size=148. Stream 5:CLEAR:START2018-01-29 14:36:35.266717 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47737 size=148. Stream 5:CLEAR:START2018-01-29 14:36:35.466732 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47738 size=148. Stream 5:CLEAR:START2018-01-29 14:36:35.666724 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47739 size=148. Stream 5:CLEAR:START2018-01-29 14:36:35.866711 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47740 size=148. Stream 5:CLEAR:START2018-01-29 14:36:36.066719 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47741 size=148. Stream 5:CLEAR:START2018-01-29 14:36:36.266719 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47742 size=148. Stream 5:CLEAR:START2018-01-29 14:36:36.466719 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47743 size=148. Stream 5:CLEAR:START2018-01-29 14:36:36.666711 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47744 size=148. Stream 5:CLEAR:START2018-01-29 14:36:36.866721 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47745 size=148. Stream 5:CLEAR:START2018-01-29 14:36:37.066709 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47746 size=148. Stream 5:CLEAR:START2018-01-29 14:36:37.266699 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47747 size=148. Stream 5:CLEAR:STARTrecv 705 bytes from udp/[<providerip>]:5060 at 14:36:37.863874:------------------------------------------------------------ ------------ SIP/2.0 488 Not Acceptable HereVia: SIP/2.0/UDP 1.2.3.4:5080;rport=5080;branch=z9hG4bKe57KcBtmmr9cB Max-Forwards: 69From: "0<sourcenumber>" <sip:<accountname>@sip.mysipprovider>;tag= DvXrSpNBaFc1e Call-ID: a553eda8-7fa4-1236-bd95-040118c15f01 CSeq: 118282234 INVITEUser-Agent: mysipproviderAccept: application/sdpAllow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFYSupported: timer, path, replacesAllow-Events: talk, hold, conference, referReason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION" Content-Length: 0P-Charging-Vector: icid-value=0954708e-d24b-489c-a16a-0244d89313f8
------------------------------------------------------------ ------------
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