[Freeswitch-users] Code 88

Rick Jarvis rick at magicmail.mooo.com
Mon Jan 29 16:15:01 UTC 2018 

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 at 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 at freeswitch.com <mailto:brian at freeswitch.com>> wrote:
>> 
>> http://lists.freeswitch.org/pipermail/freeswitch-users/2017-January/124459.html <http://lists.freeswitch.org/pipermail/freeswitch-users/2017-January/124459.html>
>> 
>> On Mon, Jan 29, 2018 at 8:46 AM, Rick Jarvis <rick at magicmail.mooo.com <mailto: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 <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
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