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/2.0 Via: SIP/2.0/UDP 1.2.3.4:5080;rport;branch=z9hG4bKe57KcBtmmr9cB Max-Forwards: 69 From: "0<sourcenumber>" <sip:<accountname>@sip.mysipprovider>;tag=DvXrSpNBaFc1e To: <sip:0<desintationnumber>@sip.mysipprovider> Call-ID: a553eda8-7fa4-1236-bd95-040118c15f01 CSeq: 118282234 INVITE Contact: <sip:gw+gatewayname@1.2.3.4:5080;transport=udp;gw=gatewayname> User-Agent: FreeSWITCH-mod_sofia/1.6.20-37-987c9b9~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Proxy-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="4702ba8b72c43939d5fa88528eafaf2b", qop=auth, nc=00000001 Content-Type: application/sdp Content-Disposition: session Content-Length: 246 X-FS-Support: update_display,send_info Remote-Party-ID: "0<sourcenumber>" <sip:0<sourcenumber>@sip.mysipprovider>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1517205769 1517205770 IN IP4 1.2.3.4 s=FreeSWITCH c=IN IP4 1.2.3.4 t=0 0 m=audio 30824 RTP/AVP 0 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=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 Trying Via: SIP/2.0/UDP 1.2.3.4:5080;rport=5080;branch=z9hG4bKe57KcBtmmr9cB From: "0<sourcenumber>" <sip:<accountname>@sip.mysipprovider>;tag=DvXrSpNBaFc1e To: <sip:0<desintationnumber>@sip.mysipprovider> Call-ID: a553eda8-7fa4-1236-bd95-040118c15f01 CSeq: 118282234 INVITE User-Agent: mysipprovider Content-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:START 2018-01-29 14:36:34.066706 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47731 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:34.186741 [DEBUG] switch_core_io.c:448 Setting BUG Codec PCMU:0 2018-01-29 14:36:34.266726 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47732 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:34.366714 [INFO] switch_rtp.c:7268 Auto Changing audio port from 10.10.131.204:11782 to <myphonesip>:11782 2018-01-29 14:36:34.466703 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47733 size=148. Stream 5:CLEAR:START 2018-01-29 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=5 2018-01-29 14:36:34.666721 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47734 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:34.866711 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47735 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:35.066711 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47736 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:35.266717 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47737 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:35.466732 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47738 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:35.666724 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47739 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:35.866711 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47740 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:36.066719 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47741 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:36.266719 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47742 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:36.466719 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47743 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:36.666711 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47744 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:36.866721 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47745 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:37.066709 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47746 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:37.266699 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47747 size=148. Stream 5:CLEAR:START recv 705 bytes from udp/[<providerip>]:5060 at 14:36:37.863874: ------------------------------------------------------------------------ SIP/2.0 488 Not Acceptable Here Via: SIP/2.0/UDP 1.2.3.4:5080;rport=5080;branch=z9hG4bKe57KcBtmmr9cB Max-Forwards: 69 From: "0<sourcenumber>" <sip:<accountname>@sip.mysipprovider>;tag=DvXrSpNBaFc1e To: <sip:0<desintationnumber>@sip.mysipprovider>;tag=v2D22KNm5y4KS Call-ID: a553eda8-7fa4-1236-bd95-040118c15f01 CSeq: 118282234 INVITE User-Agent: mysipprovider Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION" Content-Length: 0 P-Charging-Vector: icid-value=0954708e-d24b-489c-a16a-0244d89313f8 ------------------------------------------------------------------------
http://lists.freeswitch.org/pipermail/freeswitch-users/2017-January/124459.h... On Mon, Jan 29, 2018 at 8:46 AM, Rick Jarvis <rick@magicmail.mooo.com> wrote:
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/2.0 Via: SIP/2.0/UDP 1.2.3.4:5080;rport;branch=z9hG4bKe57KcBtmmr9cB Max-Forwards: 69 From: "0<sourcenumber>" <sip:<accountname>@sip.mysipprovider>;tag= DvXrSpNBaFc1e To: <sip:0<desintationnumber>@sip.mysipprovider> Call-ID: a553eda8-7fa4-1236-bd95-040118c15f01 CSeq: 118282234 INVITE Contact: <sip:gw+gatewayname@1.2.3.4:5080;transport=udp;gw=gatewayname> User-Agent: FreeSWITCH-mod_sofia/1.6.20-37-987c9b9~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Proxy-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=" 4702ba8b72c43939d5fa88528eafaf2b", qop=auth, nc=00000001 Content-Type: application/sdp Content-Disposition: session Content-Length: 246 X-FS-Support: update_display,send_info Remote-Party-ID: "0<sourcenumber>" <sip:0<sourcenumber>@sip. mysipprovider>;party=calling;screen=yes;privacy=off
v=0 o=FreeSWITCH 1517205769 1517205770 IN IP4 1.2.3.4 s=FreeSWITCH c=IN IP4 1.2.3.4 t=0 0 m=audio 30824 RTP/AVP 0 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=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 Trying Via: SIP/2.0/UDP 1.2.3.4:5080;rport=5080;branch=z9hG4bKe57KcBtmmr9cB From: "0<sourcenumber>" <sip:<accountname>@sip.mysipprovider>;tag= DvXrSpNBaFc1e To: <sip:0<desintationnumber>@sip.mysipprovider> Call-ID: a553eda8-7fa4-1236-bd95-040118c15f01 CSeq: 118282234 INVITE User-Agent: mysipprovider Content-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:START 2018-01-29 14:36:34.066706 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47731 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:34.186741 [DEBUG] switch_core_io.c:448 Setting BUG Codec PCMU:0 2018-01-29 14:36:34.266726 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47732 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:34.366714 [INFO] switch_rtp.c:7268 Auto Changing audio port from 10.10.131.204:11782 to <myphonesip>:11782 2018-01-29 14:36:34.466703 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47733 size=148. Stream 5:CLEAR:START 2018-01-29 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=5 2018-01-29 14:36:34.666721 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47734 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:34.866711 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47735 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:35.066711 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47736 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:35.266717 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47737 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:35.466732 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47738 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:35.666724 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47739 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:35.866711 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47740 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:36.066719 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47741 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:36.266719 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47742 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:36.466719 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47743 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:36.666711 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47744 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:36.866721 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47745 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:37.066709 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47746 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:37.266699 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47747 size=148. Stream 5:CLEAR:START recv 705 bytes from udp/[<providerip>]:5060 at 14:36:37.863874: ------------------------------------------------------------ ------------ SIP/2.0 488 Not Acceptable Here Via: SIP/2.0/UDP 1.2.3.4:5080;rport=5080;branch=z9hG4bKe57KcBtmmr9cB Max-Forwards: 69 From: "0<sourcenumber>" <sip:<accountname>@sip.mysipprovider>;tag= DvXrSpNBaFc1e To: <sip:0<desintationnumber>@sip.mysipprovider>;tag=v2D22KNm5y4KS Call-ID: a553eda8-7fa4-1236-bd95-040118c15f01 CSeq: 118282234 INVITE User-Agent: mysipprovider Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION" Content-Length: 0 P-Charging-Vector: icid-value=0954708e-d24b-489c-a16a-0244d89313f8
------------------------------------------------------------ ------------
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Actually I might need to see more of the debug logs, because the outline I posted was a behavior change that results in a 488 on t.38 reinvite, but clearly the remote end is telling you that it has no codecs in common. /b On Mon, Jan 29, 2018 at 8:49 AM, Brian West <brian@freeswitch.com> wrote:
http://lists.freeswitch.org/pipermail/freeswitch-users/ 2017-January/124459.html
On Mon, Jan 29, 2018 at 8:46 AM, Rick Jarvis <rick@magicmail.mooo.com> wrote:
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/2.0 Via: SIP/2.0/UDP 1.2.3.4:5080;rport;branch=z9hG4bKe57KcBtmmr9cB Max-Forwards: 69 From: "0<sourcenumber>" <sip:<accountname>@sip.mysippr ovider>;tag=DvXrSpNBaFc1e To: <sip:0<desintationnumber>@sip.mysipprovider> Call-ID: a553eda8-7fa4-1236-bd95-040118c15f01 CSeq: 118282234 INVITE Contact: <sip:gw+gatewayname@1.2.3.4:5080;transport=udp;gw=gatewayname
User-Agent: FreeSWITCH-mod_sofia/1.6.20-37-987c9b9~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Proxy-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="4702ba8b72c43939d5fa88528eafaf2b", qop=auth, nc=00000001 Content-Type: application/sdp Content-Disposition: session Content-Length: 246 X-FS-Support: update_display,send_info Remote-Party-ID: "0<sourcenumber>" <sip:0<sourcenumber>@sip.mysip provider>;party=calling;screen=yes;privacy=off
v=0 o=FreeSWITCH 1517205769 1517205770 IN IP4 1.2.3.4 s=FreeSWITCH c=IN IP4 1.2.3.4 t=0 0 m=audio 30824 RTP/AVP 0 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=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 Trying Via: SIP/2.0/UDP 1.2.3.4:5080;rport=5080;branch=z9hG4bKe57KcBtmmr9cB From: "0<sourcenumber>" <sip:<accountname>@sip.mysippr ovider>;tag=DvXrSpNBaFc1e To: <sip:0<desintationnumber>@sip.mysipprovider> Call-ID: a553eda8-7fa4-1236-bd95-040118c15f01 CSeq: 118282234 INVITE User-Agent: mysipprovider Content-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:START 2018-01-29 14:36:34.066706 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47731 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:34.186741 [DEBUG] switch_core_io.c:448 Setting BUG Codec PCMU:0 2018-01-29 14:36:34.266726 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47732 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:34.366714 [INFO] switch_rtp.c:7268 Auto Changing audio port from 10.10.131.204:11782 to <myphonesip>:11782 2018-01-29 14:36:34.466703 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47733 size=148. Stream 5:CLEAR:START 2018-01-29 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=5 2018-01-29 14:36:34.666721 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47734 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:34.866711 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47735 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:35.066711 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47736 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:35.266717 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47737 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:35.466732 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47738 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:35.666724 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47739 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:35.866711 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47740 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:36.066719 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47741 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:36.266719 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47742 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:36.466719 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47743 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:36.666711 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47744 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:36.866721 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47745 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:37.066709 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47746 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:37.266699 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47747 size=148. Stream 5:CLEAR:START recv 705 bytes from udp/[<providerip>]:5060 at 14:36:37.863874: ------------------------------------------------------------ ------------ SIP/2.0 488 Not Acceptable Here Via: SIP/2.0/UDP 1.2.3.4:5080;rport=5080;branch=z9hG4bKe57KcBtmmr9cB Max-Forwards: 69 From: "0<sourcenumber>" <sip:<accountname>@sip.mysippr ovider>;tag=DvXrSpNBaFc1e To: <sip:0<desintationnumber>@sip.mysipprovider>;tag=v2D22KNm5y4KS Call-ID: a553eda8-7fa4-1236-bd95-040118c15f01 CSeq: 118282234 INVITE User-Agent: mysipprovider Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION" Content-Length: 0 P-Charging-Vector: icid-value=0954708e-d24b-489c-a16a-0244d89313f8
------------------------------------------------------------ ------------
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Email: brian@freeswitch.com
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-- Brian West | Co-founder and Developer Need Commercial support? email sales@freeswitch.com FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g> Email: brian@freeswitch.com Mobile: 918-424-9378 Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/> [image: color-facebook-96.png] <https://www.facebook.com/freeswitch/>[image: color-twitter-96.png] <https://twitter.com/freeswitch?ref_src=twsrc%5Egoogle%7Ctwcamp%5Eserp%7Ctwgr%5Eauthor>
Thanks Brian… what am I not getting here? This is for voice calls, and that link says it’s fixed in 1.6.15 (I’m running 1.6.20) I put the setting into vars.xml anyway, but to no avail?...
On 29 Jan 2018, at 14:49, Brian West <brian@freeswitch.com> wrote:
http://lists.freeswitch.org/pipermail/freeswitch-users/2017-January/124459.h... <http://lists.freeswitch.org/pipermail/freeswitch-users/2017-January/124459.html>
On Mon, Jan 29, 2018 at 8:46 AM, Rick Jarvis <rick@magicmail.mooo.com <mailto:rick@magicmail.mooo.com>> wrote: 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/2.0 Via: SIP/2.0/UDP 1.2.3.4:5080;rport;branch=z9hG4bKe57KcBtmmr9cB Max-Forwards: 69 From: "0<sourcenumber>" <sip:<accountname>@sip.mysipprovider>;tag=DvXrSpNBaFc1e To: <sip:0<desintationnumber>@sip.mysipprovider <>> Call-ID: a553eda8-7fa4-1236-bd95-040118c15f01 CSeq: 118282234 INVITE Contact: <sip:gw+gatewayname@1.2.3.4:5080;transport=udp;gw=gatewayname <>> User-Agent: FreeSWITCH-mod_sofia/1.6.20-37-987c9b9~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Proxy-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="4702ba8b72c43939d5fa88528eafaf2b", qop=auth, nc=00000001 Content-Type: application/sdp Content-Disposition: session Content-Length: 246 X-FS-Support: update_display,send_info Remote-Party-ID: "0<sourcenumber>" <sip:0<sourcenumber>@sip.mysipprovider>;party=calling;screen=yes;privacy=off <>
v=0 o=FreeSWITCH 1517205769 1517205770 IN IP4 1.2.3.4 s=FreeSWITCH c=IN IP4 1.2.3.4 t=0 0 m=audio 30824 RTP/AVP 0 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=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 Trying Via: SIP/2.0/UDP 1.2.3.4:5080;rport=5080;branch=z9hG4bKe57KcBtmmr9cB From: "0<sourcenumber>" <sip:<accountname>@sip.mysipprovider>;tag=DvXrSpNBaFc1e To: <sip:0<desintationnumber>@sip.mysipprovider <>> Call-ID: a553eda8-7fa4-1236-bd95-040118c15f01 CSeq: 118282234 INVITE User-Agent: mysipprovider Content-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:START 2018-01-29 14:36:34.066706 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47731 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:34.186741 [DEBUG] switch_core_io.c:448 Setting BUG Codec PCMU:0 2018-01-29 14:36:34.266726 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47732 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:34.366714 [INFO] switch_rtp.c:7268 Auto Changing audio port from 10.10.131.204:11782 <http://10.10.131.204:11782/> to <myphonesip>:11782 2018-01-29 14:36:34.466703 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47733 size=148. Stream 5:CLEAR:START 2018-01-29 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=5 2018-01-29 14:36:34.666721 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47734 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:34.866711 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47735 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:35.066711 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47736 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:35.266717 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47737 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:35.466732 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47738 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:35.666724 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47739 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:35.866711 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47740 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:36.066719 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47741 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:36.266719 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47742 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:36.466719 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47743 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:36.666711 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47744 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:36.866721 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47745 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:37.066709 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47746 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:37.266699 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47747 size=148. Stream 5:CLEAR:START recv 705 bytes from udp/[<providerip>]:5060 at 14:36:37.863874: ------------------------------------------------------------------------ SIP/2.0 488 Not Acceptable Here Via: SIP/2.0/UDP 1.2.3.4:5080;rport=5080;branch=z9hG4bKe57KcBtmmr9cB Max-Forwards: 69 From: "0<sourcenumber>" <sip:<accountname>@sip.mysipprovider>;tag=DvXrSpNBaFc1e To: <sip:0<desintationnumber>@sip.mysipprovider>;tag=v2D22KNm5y4KS <> Call-ID: a553eda8-7fa4-1236-bd95-040118c15f01 CSeq: 118282234 INVITE User-Agent: mysipprovider Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION" Content-Length: 0 P-Charging-Vector: icid-value=0954708e-d24b-489c-a16a-0244d89313f8
------------------------------------------------------------------------
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Official FreeSWITCH Sites http://www.freeswitch.org <http://www.freeswitch.org/> http://confluence.freeswitch.org <http://confluence.freeswitch.org/> http://www.cluecon.com <http://www.cluecon.com/>
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Brian West | Co-founder and Developer Need Commercial support? email sales@freeswitch.com <mailto:sales@freeswitch.com> FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g> Email: brian@freeswitch.com <mailto:brian@freeswitch.com> Mobile: 918-424-9378 Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/> <https://www.facebook.com/freeswitch/> <https://twitter.com/freeswitch?ref_src=twsrc%5Egoogle%7Ctwcamp%5Eserp%7Ctwgr%5Eauthor>_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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Brian, can you clarify please? I have users on this one! Incidentally, it’s actually ringing the destination number once before cutting off, which is odd… Should I just downgrade to 1.4 again?
On 29 Jan 2018, at 15:00, Rick Jarvis <rick@magicmail.mooo.com> wrote:
Thanks Brian… what am I not getting here? This is for voice calls, and that link says it’s fixed in 1.6.15 (I’m running 1.6.20)
I put the setting into vars.xml anyway, but to no avail?...
On 29 Jan 2018, at 14:49, Brian West <brian@freeswitch.com <mailto:brian@freeswitch.com>> wrote:
http://lists.freeswitch.org/pipermail/freeswitch-users/2017-January/124459.h... <http://lists.freeswitch.org/pipermail/freeswitch-users/2017-January/124459.html>
On Mon, Jan 29, 2018 at 8:46 AM, Rick Jarvis <rick@magicmail.mooo.com <mailto:rick@magicmail.mooo.com>> wrote: 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/2.0 Via: SIP/2.0/UDP 1.2.3.4:5080;rport;branch=z9hG4bKe57KcBtmmr9cB Max-Forwards: 69 From: "0<sourcenumber>" <sip:<accountname>@sip.mysipprovider>;tag=DvXrSpNBaFc1e To: <sip:0<desintationnumber>@sip.mysipprovider <>> Call-ID: a553eda8-7fa4-1236-bd95-040118c15f01 CSeq: 118282234 INVITE Contact: <sip:gw+gatewayname@1.2.3.4:5080;transport=udp;gw=gatewayname <>> User-Agent: FreeSWITCH-mod_sofia/1.6.20-37-987c9b9~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Proxy-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="4702ba8b72c43939d5fa88528eafaf2b", qop=auth, nc=00000001 Content-Type: application/sdp Content-Disposition: session Content-Length: 246 X-FS-Support: update_display,send_info Remote-Party-ID: "0<sourcenumber>" <sip:0<sourcenumber>@sip.mysipprovider>;party=calling;screen=yes;privacy=off <>
v=0 o=FreeSWITCH 1517205769 1517205770 IN IP4 1.2.3.4 s=FreeSWITCH c=IN IP4 1.2.3.4 t=0 0 m=audio 30824 RTP/AVP 0 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=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 Trying Via: SIP/2.0/UDP 1.2.3.4:5080;rport=5080;branch=z9hG4bKe57KcBtmmr9cB From: "0<sourcenumber>" <sip:<accountname>@sip.mysipprovider>;tag=DvXrSpNBaFc1e To: <sip:0<desintationnumber>@sip.mysipprovider <>> Call-ID: a553eda8-7fa4-1236-bd95-040118c15f01 CSeq: 118282234 INVITE User-Agent: mysipprovider Content-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:START 2018-01-29 14:36:34.066706 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47731 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:34.186741 [DEBUG] switch_core_io.c:448 Setting BUG Codec PCMU:0 2018-01-29 14:36:34.266726 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47732 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:34.366714 [INFO] switch_rtp.c:7268 Auto Changing audio port from 10.10.131.204:11782 <http://10.10.131.204:11782/> to <myphonesip>:11782 2018-01-29 14:36:34.466703 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47733 size=148. Stream 5:CLEAR:START 2018-01-29 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=5 2018-01-29 14:36:34.666721 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47734 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:34.866711 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47735 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:35.066711 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47736 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:35.266717 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47737 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:35.466732 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47738 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:35.666724 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47739 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:35.866711 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47740 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:36.066719 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47741 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:36.266719 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47742 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:36.466719 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47743 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:36.666711 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47744 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:36.866721 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47745 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:37.066709 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47746 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:37.266699 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47747 size=148. Stream 5:CLEAR:START recv 705 bytes from udp/[<providerip>]:5060 at 14:36:37.863874: ------------------------------------------------------------------------ SIP/2.0 488 Not Acceptable Here Via: SIP/2.0/UDP 1.2.3.4:5080;rport=5080;branch=z9hG4bKe57KcBtmmr9cB Max-Forwards: 69 From: "0<sourcenumber>" <sip:<accountname>@sip.mysipprovider>;tag=DvXrSpNBaFc1e To: <sip:0<desintationnumber>@sip.mysipprovider>;tag=v2D22KNm5y4KS <> Call-ID: a553eda8-7fa4-1236-bd95-040118c15f01 CSeq: 118282234 INVITE User-Agent: mysipprovider Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION" Content-Length: 0 P-Charging-Vector: icid-value=0954708e-d24b-489c-a16a-0244d89313f8
------------------------------------------------------------------------
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Brian West | Co-founder and Developer Need Commercial support? email sales@freeswitch.com <mailto:sales@freeswitch.com> FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g> Email: brian@freeswitch.com <mailto:brian@freeswitch.com> Mobile: 918-424-9378 Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/> <https://www.facebook.com/freeswitch/> <https://twitter.com/freeswitch?ref_src=twsrc%5Egoogle%7Ctwcamp%5Eserp%7Ctwgr%5Eauthor>_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com
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Downgrading is never a fix for something like this. 1.4 has been EOL for a few years now. You should properly collect the data and file a proper JIRA. /b On Mon, Jan 29, 2018 at 10:15 AM, Rick Jarvis <rick@magicmail.mooo.com> wrote:
Brian, can you clarify please? I have users on this one!
Incidentally, it’s actually ringing the destination number once before cutting off, which is odd…
Should I just downgrade to 1.4 again?
On 29 Jan 2018, at 15:00, Rick Jarvis <rick@magicmail.mooo.com> wrote:
Thanks Brian… what am I not getting here? This is for voice calls, and that link says it’s fixed in 1.6.15 (I’m running 1.6.20)
I put the setting into vars.xml anyway, but to no avail?...
On 29 Jan 2018, at 14:49, Brian West <brian@freeswitch.com> wrote:
http://lists.freeswitch.org/pipermail/freeswitch-users/ 2017-January/124459.html
On Mon, Jan 29, 2018 at 8:46 AM, Rick Jarvis <rick@magicmail.mooo.com> wrote:
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/2.0 Via: SIP/2.0/UDP 1.2.3.4:5080;rport;branch=z9hG4bKe57KcBtmmr9cB Max-Forwards: 69 From: "0<sourcenumber>" <sip:<accountname>@sip.mysippr ovider>;tag=DvXrSpNBaFc1e To: <sip:0<desintationnumber>@sip.mysipprovider> Call-ID: a553eda8-7fa4-1236-bd95-040118c15f01 CSeq: 118282234 INVITE Contact: <sip:gw+gatewayname@1.2.3.4:5080;transport=udp;gw=gatewayname
User-Agent: FreeSWITCH-mod_sofia/1.6.20-37-987c9b9~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Proxy-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="4702ba8b72c43939d5fa88528eafaf2b", qop=auth, nc=00000001 Content-Type: application/sdp Content-Disposition: session Content-Length: 246 X-FS-Support: update_display,send_info Remote-Party-ID: "0<sourcenumber>" <sip:0<sourcenumber>@sip.mysip provider>;party=calling;screen=yes;privacy=off
v=0 o=FreeSWITCH 1517205769 1517205770 IN IP4 1.2.3.4 s=FreeSWITCH c=IN IP4 1.2.3.4 t=0 0 m=audio 30824 RTP/AVP 0 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=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 Trying Via: SIP/2.0/UDP 1.2.3.4:5080;rport=5080;branch=z9hG4bKe57KcBtmmr9cB From: "0<sourcenumber>" <sip:<accountname>@sip.mysippr ovider>;tag=DvXrSpNBaFc1e To: <sip:0<desintationnumber>@sip.mysipprovider> Call-ID: a553eda8-7fa4-1236-bd95-040118c15f01 CSeq: 118282234 INVITE User-Agent: mysipprovider Content-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:START 2018-01-29 14:36:34.066706 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47731 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:34.186741 [DEBUG] switch_core_io.c:448 Setting BUG Codec PCMU:0 2018-01-29 14:36:34.266726 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47732 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:34.366714 [INFO] switch_rtp.c:7268 Auto Changing audio port from 10.10.131.204:11782 to <myphonesip>:11782 2018-01-29 14:36:34.466703 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47733 size=148. Stream 5:CLEAR:START 2018-01-29 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=5 2018-01-29 14:36:34.666721 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47734 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:34.866711 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47735 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:35.066711 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47736 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:35.266717 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47737 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:35.466732 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47738 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:35.666724 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47739 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:35.866711 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47740 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:36.066719 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47741 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:36.266719 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47742 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:36.466719 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47743 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:36.666711 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47744 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:36.866721 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47745 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:37.066709 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47746 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:37.266699 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47747 size=148. Stream 5:CLEAR:START recv 705 bytes from udp/[<providerip>]:5060 at 14:36:37.863874: ------------------------------------------------------------ ------------ SIP/2.0 488 Not Acceptable Here Via: SIP/2.0/UDP 1.2.3.4:5080;rport=5080;branch=z9hG4bKe57KcBtmmr9cB Max-Forwards: 69 From: "0<sourcenumber>" <sip:<accountname>@sip.mysippr ovider>;tag=DvXrSpNBaFc1e To: <sip:0<desintationnumber>@sip.mysipprovider>;tag=v2D22KNm5y4KS Call-ID: a553eda8-7fa4-1236-bd95-040118c15f01 CSeq: 118282234 INVITE User-Agent: mysipprovider Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION" Content-Length: 0 P-Charging-Vector: icid-value=0954708e-d24b-489c-a16a-0244d89313f8
------------------------------------------------------------ ------------
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Brian West | Co-founder and Developer Need Commercial support? email sales@freeswitch.com FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g> Email: brian@freeswitch.com Mobile: 918-424-9378 <(918)%20424-9378> Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/> [image: color-facebook-96.png] <https://www.facebook.com/freeswitch/>[image: color-twitter-96.png] <https://twitter.com/freeswitch?ref_src=twsrc%5Egoogle%7Ctwcamp%5Eserp%7Ctwgr%5Eauthor> _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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Hi Rick From the invite it looks like it is only offering G.711ulaw or PCMU as its also referred to, and not PCMA. I would guess that the other end does not support PCMU and so the call is failing, the message your receiving from your sip provider is almost certainly due to codecs. Check what you have set for global_codec_prefs and also ensure that there is a comma between the different codecs both in global_codec_prefs and also outbound_codec_prefs in your vars.xml Also check the settings you have in the relevant sofia profile, as this is where the codecs are actually set, if they don’t reference the settings in vars.xml then the could be your issue. In each sofia profile, there should be inbound-codec-prefs and outbound-codec-prefs, the config may have some hardcoding here. Regards
On 29 Jan 2018, at 14:46, Rick Jarvis <rick@magicmail.mooo.com> wrote:
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:0<desintationnumber>@sip.mysipprovider> SIP/2.0 Via: SIP/2.0/UDP 1.2.3.4:5080;rport;branch=z9hG4bKe57KcBtmmr9cB Max-Forwards: 69 From: "0<sourcenumber>" <sip:<accountname>@sip.mysipprovider>;tag=DvXrSpNBaFc1e To: <sip:0<desintationnumber>@sip.mysipprovider <sip:0<desintationnumber>@sip.mysipprovider>> Call-ID: a553eda8-7fa4-1236-bd95-040118c15f01 CSeq: 118282234 INVITE Contact: <sip:gw+gatewayname@1.2.3.4:5080;transport=udp;gw=gatewayname <sip:gw+gatewayname@1.2.3.4:5080;transport=udp;gw=gatewayname>> User-Agent: FreeSWITCH-mod_sofia/1.6.20-37-987c9b9~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Proxy-Authorization: Digest username="<accountname>", realm="sip.mysipprovider", nonce="81346aeb-453b-466e-ba50-5d027c1a55e3", cnonce="pVaZvn+kEjaVvQQBGMFfAQ", algorithm=MD5, uri="sip:0<desintationnumber>@sip.mysipprovider <sip:0<desintationnumber>@sip.mysipprovider>", response="4702ba8b72c43939d5fa88528eafaf2b", qop=auth, nc=00000001 Content-Type: application/sdp Content-Disposition: session Content-Length: 246 X-FS-Support: update_display,send_info Remote-Party-ID: "0<sourcenumber>" <sip:0<sourcenumber>@sip.mysipprovider>;party=calling;screen=yes;privacy=off <sip:0<sourcenumber>@sip.mysipprovider>;party=calling;screen=yes;privacy=off>
v=0 o=FreeSWITCH 1517205769 1517205770 IN IP4 1.2.3.4 s=FreeSWITCH c=IN IP4 1.2.3.4 t=0 0 m=audio 30824 RTP/AVP 0 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=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 Trying Via: SIP/2.0/UDP 1.2.3.4:5080;rport=5080;branch=z9hG4bKe57KcBtmmr9cB From: "0<sourcenumber>" <sip:<accountname>@sip.mysipprovider>;tag=DvXrSpNBaFc1e To: <sip:0<desintationnumber>@sip.mysipprovider <sip:0<desintationnumber>@sip.mysipprovider>> Call-ID: a553eda8-7fa4-1236-bd95-040118c15f01 CSeq: 118282234 INVITE User-Agent: mysipprovider Content-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:START 2018-01-29 14:36:34.066706 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47731 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:34.186741 [DEBUG] switch_core_io.c:448 Setting BUG Codec PCMU:0 2018-01-29 14:36:34.266726 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47732 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:34.366714 [INFO] switch_rtp.c:7268 Auto Changing audio port from 10.10.131.204:11782 to <myphonesip>:11782 2018-01-29 14:36:34.466703 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47733 size=148. Stream 5:CLEAR:START 2018-01-29 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=5 2018-01-29 14:36:34.666721 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47734 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:34.866711 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47735 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:35.066711 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47736 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:35.266717 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47737 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:35.466732 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47738 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:35.666724 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47739 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:35.866711 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47740 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:36.066719 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47741 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:36.266719 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47742 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:36.466719 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47743 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:36.666711 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47744 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:36.866721 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47745 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:37.066709 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47746 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:37.266699 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47747 size=148. Stream 5:CLEAR:START recv 705 bytes from udp/[<providerip>]:5060 at 14:36:37.863874: ------------------------------------------------------------------------ SIP/2.0 488 Not Acceptable Here Via: SIP/2.0/UDP 1.2.3.4:5080;rport=5080;branch=z9hG4bKe57KcBtmmr9cB Max-Forwards: 69 From: "0<sourcenumber>" <sip:<accountname>@sip.mysipprovider>;tag=DvXrSpNBaFc1e To: <sip:0<desintationnumber>@sip.mysipprovider>;tag=v2D22KNm5y4KS <sip:0<desintationnumber>@sip.mysipprovider>;tag=v2D22KNm5y4KS> Call-ID: a553eda8-7fa4-1236-bd95-040118c15f01 CSeq: 118282234 INVITE User-Agent: mysipprovider Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION" Content-Length: 0 P-Charging-Vector: icid-value=0954708e-d24b-489c-a16a-0244d89313f8
------------------------------------------------------------------------
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Thanks for this - it’s working now, although I’m sure I just changed it back to what it was before when it didn’t work! Strange (clearly user error!), but thanks again!
On 29 Jan 2018, at 16:37, Jospeh Waite <joelists@tm.net.uk> wrote:
Hi Rick
From the invite it looks like it is only offering G.711ulaw or PCMU as its also referred to, and not PCMA.
I would guess that the other end does not support PCMU and so the call is failing, the message your receiving from your sip provider is almost certainly due to codecs.
Check what you have set for global_codec_prefs and also ensure that there is a comma between the different codecs both in global_codec_prefs and also outbound_codec_prefs in your vars.xml
Also check the settings you have in the relevant sofia profile, as this is where the codecs are actually set, if they don’t reference the settings in vars.xml then the could be your issue.
In each sofia profile, there should be inbound-codec-prefs and outbound-codec-prefs, the config may have some hardcoding here.
Regards
On 29 Jan 2018, at 14:46, Rick Jarvis <rick@magicmail.mooo.com <mailto:rick@magicmail.mooo.com>> wrote:
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:0<desintationnumber>@sip.mysipprovider> SIP/2.0 Via: SIP/2.0/UDP 1.2.3.4:5080;rport;branch=z9hG4bKe57KcBtmmr9cB Max-Forwards: 69 From: "0<sourcenumber>" <sip:<accountname>@sip.mysipprovider>;tag=DvXrSpNBaFc1e To: <sip:0<desintationnumber>@sip.mysipprovider <sip:0<desintationnumber>@sip.mysipprovider>> Call-ID: a553eda8-7fa4-1236-bd95-040118c15f01 CSeq: 118282234 INVITE Contact: <sip:gw+gatewayname@1.2.3.4:5080;transport=udp;gw=gatewayname <sip:gw+gatewayname@1.2.3.4:5080;transport=udp;gw=gatewayname>> User-Agent: FreeSWITCH-mod_sofia/1.6.20-37-987c9b9~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Proxy-Authorization: Digest username="<accountname>", realm="sip.mysipprovider", nonce="81346aeb-453b-466e-ba50-5d027c1a55e3", cnonce="pVaZvn+kEjaVvQQBGMFfAQ", algorithm=MD5, uri="sip:0<desintationnumber>@sip.mysipprovider <sip:0<desintationnumber>@sip.mysipprovider>", response="4702ba8b72c43939d5fa88528eafaf2b", qop=auth, nc=00000001 Content-Type: application/sdp Content-Disposition: session Content-Length: 246 X-FS-Support: update_display,send_info Remote-Party-ID: "0<sourcenumber>" <sip:0<sourcenumber>@sip.mysipprovider>;party=calling;screen=yes;privacy=off <sip:0<sourcenumber>@sip.mysipprovider>;party=calling;screen=yes;privacy=off>
v=0 o=FreeSWITCH 1517205769 1517205770 IN IP4 1.2.3.4 s=FreeSWITCH c=IN IP4 1.2.3.4 t=0 0 m=audio 30824 RTP/AVP 0 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=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 Trying Via: SIP/2.0/UDP 1.2.3.4:5080;rport=5080;branch=z9hG4bKe57KcBtmmr9cB From: "0<sourcenumber>" <sip:<accountname>@sip.mysipprovider>;tag=DvXrSpNBaFc1e To: <sip:0<desintationnumber>@sip.mysipprovider <sip:0<desintationnumber>@sip.mysipprovider>> Call-ID: a553eda8-7fa4-1236-bd95-040118c15f01 CSeq: 118282234 INVITE User-Agent: mysipprovider Content-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:START 2018-01-29 14:36:34.066706 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47731 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:34.186741 [DEBUG] switch_core_io.c:448 Setting BUG Codec PCMU:0 2018-01-29 14:36:34.266726 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47732 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:34.366714 [INFO] switch_rtp.c:7268 Auto Changing audio port from 10.10.131.204:11782 to <myphonesip>:11782 2018-01-29 14:36:34.466703 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47733 size=148. Stream 5:CLEAR:START 2018-01-29 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=5 2018-01-29 14:36:34.666721 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47734 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:34.866711 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47735 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:35.066711 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47736 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:35.266717 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47737 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:35.466732 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47738 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:35.666724 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47739 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:35.866711 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47740 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:36.066719 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47741 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:36.266719 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47742 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:36.466719 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47743 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:36.666711 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47744 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:36.866721 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47745 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:37.066709 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47746 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:37.266699 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47747 size=148. Stream 5:CLEAR:START recv 705 bytes from udp/[<providerip>]:5060 at 14:36:37.863874: ------------------------------------------------------------------------ SIP/2.0 488 Not Acceptable Here Via: SIP/2.0/UDP 1.2.3.4:5080;rport=5080;branch=z9hG4bKe57KcBtmmr9cB Max-Forwards: 69 From: "0<sourcenumber>" <sip:<accountname>@sip.mysipprovider>;tag=DvXrSpNBaFc1e To: <sip:0<desintationnumber>@sip.mysipprovider>;tag=v2D22KNm5y4KS <sip:0<desintationnumber>@sip.mysipprovider>;tag=v2D22KNm5y4KS> Call-ID: a553eda8-7fa4-1236-bd95-040118c15f01 CSeq: 118282234 INVITE User-Agent: mysipprovider Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION" Content-Length: 0 P-Charging-Vector: icid-value=0954708e-d24b-489c-a16a-0244d89313f8
------------------------------------------------------------------------
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Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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I'm pretty sure that: https://freeswitch.org/jira/plugins/servlet/mobile#issue/FS-8321 May be cause that b - leg has less codecs in sdp. On 29 Jan 2018 17:54, "Rick Jarvis" <rick@magicmail.mooo.com> wrote:
Thanks for this - it’s working now, although I’m sure I just changed it back to what it was before when it didn’t work! Strange (clearly user error!), but thanks again!
On 29 Jan 2018, at 16:37, Jospeh Waite <joelists@tm.net.uk> wrote:
Hi Rick
From the invite it looks like it is only offering G.711ulaw or PCMU as its also referred to, and not PCMA.
I would guess that the other end does not support PCMU and so the call is failing, the message your receiving from your sip provider is almost certainly due to codecs.
Check what you have set for global_codec_prefs and also ensure that there is a comma between the different codecs both in global_codec_prefs and also outbound_codec_prefs in your vars.xml
Also check the settings you have in the relevant sofia profile, as this is where the codecs are actually set, if they don’t reference the settings in vars.xml then the could be your issue.
In each sofia profile, there should be inbound-codec-prefs and outbound-codec-prefs, the config may have some hardcoding here.
Regards
On 29 Jan 2018, at 14:46, Rick Jarvis <rick@magicmail.mooo.com> wrote:
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/2.0 Via: SIP/2.0/UDP 1.2.3.4:5080;rport;branch=z9hG4bKe57KcBtmmr9cB Max-Forwards: 69 From: "0<sourcenumber>" <sip:<accountname>@sip.mysipprovider>;tag= DvXrSpNBaFc1e To: <sip:0<desintationnumber>@sip.mysipprovider> Call-ID: a553eda8-7fa4-1236-bd95-040118c15f01 CSeq: 118282234 INVITE Contact: <sip:gw+gatewayname@1.2.3.4:5080;transport=udp;gw=gatewayname> User-Agent: FreeSWITCH-mod_sofia/1.6.20-37-987c9b9~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Proxy-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=" 4702ba8b72c43939d5fa88528eafaf2b", qop=auth, nc=00000001 Content-Type: application/sdp Content-Disposition: session Content-Length: 246 X-FS-Support: update_display,send_info Remote-Party-ID: "0<sourcenumber>" <sip:0<sourcenumber>@sip. mysipprovider>;party=calling;screen=yes;privacy=off
v=0 o=FreeSWITCH 1517205769 1517205770 IN IP4 1.2.3.4 s=FreeSWITCH c=IN IP4 1.2.3.4 t=0 0 m=audio 30824 RTP/AVP 0 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=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 Trying Via: SIP/2.0/UDP 1.2.3.4:5080;rport=5080;branch=z9hG4bKe57KcBtmmr9cB From: "0<sourcenumber>" <sip:<accountname>@sip.mysipprovider>;tag= DvXrSpNBaFc1e To: <sip:0<desintationnumber>@sip.mysipprovider> Call-ID: a553eda8-7fa4-1236-bd95-040118c15f01 CSeq: 118282234 INVITE User-Agent: mysipprovider Content-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:START 2018-01-29 14:36:34.066706 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47731 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:34.186741 [DEBUG] switch_core_io.c:448 Setting BUG Codec PCMU:0 2018-01-29 14:36:34.266726 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47732 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:34.366714 [INFO] switch_rtp.c:7268 Auto Changing audio port from 10.10.131.204:11782 to <myphonesip>:11782 2018-01-29 14:36:34.466703 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47733 size=148. Stream 5:CLEAR:START 2018-01-29 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=5 2018-01-29 14:36:34.666721 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47734 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:34.866711 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47735 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:35.066711 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47736 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:35.266717 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47737 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:35.466732 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47738 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:35.666724 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47739 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:35.866711 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47740 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:36.066719 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47741 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:36.266719 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47742 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:36.466719 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47743 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:36.666711 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47744 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:36.866721 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47745 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:37.066709 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47746 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:37.266699 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47747 size=148. Stream 5:CLEAR:START recv 705 bytes from udp/[<providerip>]:5060 at 14:36:37.863874: ------------------------------------------------------------ ------------ SIP/2.0 488 Not Acceptable Here Via: SIP/2.0/UDP 1.2.3.4:5080;rport=5080;branch=z9hG4bKe57KcBtmmr9cB Max-Forwards: 69 From: "0<sourcenumber>" <sip:<accountname>@sip.mysipprovider>;tag= DvXrSpNBaFc1e To: <sip:0<desintationnumber>@sip.mysipprovider>;tag=v2D22KNm5y4KS Call-ID: a553eda8-7fa4-1236-bd95-040118c15f01 CSeq: 118282234 INVITE User-Agent: mysipprovider Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION" Content-Length: 0 P-Charging-Vector: icid-value=0954708e-d24b-489c-a16a-0244d89313f8
------------------------------------------------------------ ------------
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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
participants (4)
-
Brian West
-
Jospeh Waite
-
Rick Jarvis
-
Volodymyr Fedorov