[Freeswitch-users] Code 88
Jospeh Waite
joelists at tm.net.uk
Mon Jan 29 16:37:55 UTC 2018
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 at 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 at 1.2.3.4:5080;transport=udp;gw=gatewayname <sip:gw+gatewayname at 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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