[Freeswitch-users] Code 88

Volodymyr Fedorov lexxua at gmail.com
Mon Jan 29 17:30:36 UTC 2018


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 at 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 at 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 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/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 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", 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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>
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>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
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> http://confluence.freeswitch.org
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