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Cisco IP Phone won't dial out with GV

Started by Muxor, January 29, 2020, 09:10:44 AM

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Muxor

January 29, 2020, 09:10:44 AM Last Edit: February 04, 2020, 02:04:23 PM by Muxor
I have an OBi200 and a Cisco 8865 Phone.  I gave up on trying to get the phone to register and I can successfully receive inbound calls.  If I try to make outbound calls I see the invite message but I always get a 503 service unavailable.  My dial plan looks correct and if I change the dial plan to use the analog phone it will work.  Also I tried using a Zoiper soft phone on my laptop with the same setup on my OBi200 it does dial out.  Based on this I believe my OBi200 is setup correctly or close.  I will paste the syslog output below.

If anyone has any ideas on why this is failing or another place for me to look I'd appreciate any help.

Working Zoiper - No registration

Jan 27 13:12:50 172.16.1.77  sendto ac100172:60305(381)
Jan 27 13:12:50 172.16.1.77  SIP/2.0 100 Trying#015#012Call-ID: BaRXtUdlKaiKtbYWxn8eFw..#015#012CSeq: 1 INVITE#015#012Content-Length: 0#015#012From: "Test User"<sip:1001@172.16.1.77:5063;transport=UDP>;tag=8c0e1802#015#012To: <sip:3115551234@172.16.1.77:5063>;tag=SP47fff616ffff9450f#015#012Via: SIP/2.0/UDP 172.16.1.114:60305;branch=z9hG4bK-524287-1---76c780f3dac5979f;received=172.16.1.114;rport=60305#015#012Server: OBIHAI/OBi200-3.2.2.5921#015#012#015
Jan 27 13:12:50 172.16.1.77  CCTL:NewCallOn Term 10[0] 1001->+13115551234,+13115551234
Jan 27 13:12:51 172.16.1.77  RTP:DtmfTxMtd:1(1),0
Jan 27 13:12:51 172.16.1.77  SIP/DLG:Start Early ICE on 18x
Jan 27 13:12:51 172.16.1.77  sendto ac100172:60305(507)
Jan 27 13:12:51 172.16.1.77  SIP/2.0 180 Ringing#015#012Call-ID: BaRXtUdlKaiKtbYWxn8eFw..#015#012CSeq: 1 INVITE#015#012Content-Length: 0#015#012From: "Test User"<sip:1001@172.16.1.77:5063;transport=UDP>;tag=8c0e1802#015#012To: <sip:3115551234@172.16.1.77:5063>;tag=SP47fff616ffff9450f#015#012Via: SIP/2.0/UDP 172.16.1.114:60305;branch=z9hG4bK-524287-1---76c780f3dac5979f;received=172.16.1.114;rport=60305#015#012Server: OBIHAI/OBi200-3.2.2.5921#015#012Contact: <sip:1001@172.16.1.77:5063>#015#012Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE#015#012#015
Jan 27 13:12:51 172.16.1.77  RTP:Start->4a7d2727:19305(80 0);0;0;0:0:0;0(47)
Jan 27 13:12:51 172.16.1.77  RTP:Set actpass 2
Jan 27 13:12:51 172.16.1.77  DTLS:Setup active
Jan 27 13:12:51 172.16.1.77  DTLS:Handshake Success
Jan 27 13:12:52 172.16.1.77  RTP:PeerRflxAddr=4a7d2727:19305
Jan 27 13:13:05 172.16.1.77  RTP:DtmfTxMtd:1(1),0
Jan 27 13:13:05 172.16.1.77  sendto ac100172:60305(796)
Jan 27 13:13:05 172.16.1.77  SIP/2.0 200 OK#015#012Call-ID: BaRXtUdlKaiKtbYWxn8eFw..#015#012CSeq: 1 INVITE#015#012Content-Length: 261#015#012From: "Test User"<sip:1001@172.16.1.77:5063;transport=UDP>;tag=8c0e1802#015#012To: <sip:3115551234@172.16.1.77:5063>;tag=SP47fff616ffff9450f#015#012Via: SIP/2.0/UDP 172.16.1.114:60305;branch=z9hG4bK-524287-1---76c780f3dac5979f;received=172.16.1.114;rport=60305#015#012Server: OBIHAI/OBi200-3.2.2.5921#015#012Contact: <sip:1001@172.16.1.77:5063>#015#012Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE#015#012Content-Type: application/sdp#015#012#015#012v=0#015#012o=- 749870 1 IN IP4 172.16.1.77#015#012s=-#015#012c=IN IP4 172.16.1.77#015#012t=0 0#015#012m=audio 17102 RTP/AVP 0 101 2 8#015#012a=rtpmap:0 PCMU/8000#015#012a=rtpmap:101 telephone-event/8000#015#012a=rtpmap:2 G726-32/8000#015#012a=rtpmap:8 PCMA/8000#015#012a=sendrecv#015#012a=ptime:20#015#012a=xg726bitorder:big-endian#015
Jan 27 13:13:05 172.16.1.77  RxFrom:ac100172:60305
Jan 27 13:13:05 172.16.1.77  ACK sip:1001@172.16.1.77:5063 SIP/2.0#015#012Via: SIP/2.0/UDP 172.16.1.114:60305;branch=z9hG4bK-524287-1---f27f469e6a64ba4c;rport#015#012Max-Forwards:70#015#012Contact: <sip:1001@98.30.197.85:60305;transport=UDP>#015#012To: <sip:3115551234@172.16.1.77:5063>;tag=SP47fff616ffff9450f#015#012From: "Test User"<sip:1001@172.16.1.77:5063;transport=UDP>;tag=8c0e1802#015#012Call-ID: BaRXtUdlKaiKtbYWxn8eFw..#015#012CSeq: 1 ACK#015#012User-Agent: Z 5.3.7 rv2.9.30#015#012Content-Length: 0#015#012#015
Jan 27 13:13:05 172.16.1.77  RTP:Start->621ec555:8000(80 0);0;0;0:0:0;0(45)
Jan 27 13:13:08 172.16.1.77  RTCP:RxPkt[52]=201<--ac100172:8001
Jan 27 13:13:20 172.16.1.77  message repeated 3 times: [ RTCP:RxPkt[52]=201<--ac100172:8001]
Jan 27 13:13:21 172.16.1.77  SRTP:BadMAC
Jan 27 13:13:21 172.16.1.77  message repeated 3 times: [ SRTP:BadMAC]
Jan 27 13:13:21 172.16.1.77  sendto 621ec555:60305(458)
Jan 27 13:13:21 172.16.1.77  BYE sip:1001@98.30.197.85:60305;transport=UDP SIP/2.0#015#012Call-ID: BaRXtUdlKaiKtbYWxn8eFw..#015#012Content-Length: 0#015#012CSeq: 8001 BYE#015#012From: <sip:3115551234@172.16.1.77>;tag=SP47fff616ffff9450f#015#012Max-Forwards: 70#015#012To: "Test User" <sip:1001@172.16.1.77>;tag=8c0e1802#015#012Via: SIP/2.0/UDP 172.16.1.77:5063;branch=z9hG4bK-7b1d34f9;rport#015#012User-Agent: OBIHAI/OBi200-3.2.2.5921#015#012X-RTP-Stat: PS=775,OS=133300,PR=780,OR=134001,PL=0,JI=0,DU=15,EN=G711U,DE=N/A,MOS=0.00#015#012#015

No working with Cisco - No registration

Jan 27 13:23:20 172.16.1.77  GAPI:Refreshing tk
Jan 27 13:23:20 172.16.1.77  TCP:Connect OK(lhttpc)45
Jan 27 13:23:20 172.16.1.77  Trying to connect ssl
Jan 27 13:23:20 172.16.1.77  TC:ssl connected
Jan 27 13:23:20 172.16.1.77  LHC:Response OK
Jan 27 13:23:20 172.16.1.77  GAPI:refresh tk ok
Jan 27 13:23:21 172.16.1.77  PARAM Cache Write Back(512 bytes)
Jan 27 13:23:40 172.16.1.77  RxFrom:ac100144:5060
Jan 27 13:23:40 172.16.1.77  INVITE sip:3115551234@172.16.1.77:5063 SIP/2.0#015#012Via: SIP/2.0/UDP 172.16.1.68:5060;branch=z9hG4bK-aa386a62#015#012From: <sip:1001@172.16.1.77>;tag=185cbf2176643d27o0#015#012To: "3115551234" <sip:3115551234@172.16.1.77>#015#012Call-ID: 62d866de-dbf3f093@172.16.1.68#015#012CSeq: 101 INVITE#015#012Max-Forwards: 70#015#012Contact: <sip:1001@172.16.1.68:5060>#015#012Expires: 240#015#012User-Agent: Cisco-CP-8865-3PCC/11.3.1#015#012Content-Length: 1526#015#012Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE#015#012Supported: replaces, sec-agree#015#012Content-Type: application/sdp#015#012#015#012v=0#015#012o=- 927573 927573 IN IP4 172.16.1.68#015#012s=-#015#012c=IN IP4 172.16.1.68#015#012b=AS:4064#015#012t=0 0#015#012m=audio 16392 RTP/AVP 0 8 9 18 97 105 96 103 99 101 107 108#015#012b=TIAS:64000#015#012a=rtpmap:0 PCMU/8000#015#012a=rtpmap:8 PCMA/8000#015#012a=rtpmap:9 G722/8000#015#012a=rtpmap:18 G729a/8000#015#012a=fmtp:18 annexb=yes#015#012a=rtpmap:97 iLBC/8000#015#012a=fmtp:97 mode=20#015#012a=rtpmap:105 iLBC/8000#015#012a=fmtp:105 mode=30#015#012a=rtpmap:96 AMR-WB/16000#015#012a=fmtp:96 channels=1;octet-align=0;mode-change-capability=2;mode-change-neighbor=0;crc=0;robust-sorting=0;interleaving=0;max-red=0#015#012a=rtpmap:103 AMR-WB/16000#015#012a=fmtp:103 channels=1;octet-align=1;mode-change-capability=2;mode-change-neighbor=0;crc=0;robust-sorting=0;interleaving=0;max-red=0#015#012a=rtpmap:99 OPUS/48000/2#015#012a=fmtp:99 maxplaybackrate=16000;sprop-maxcapturerate=16000;maxaveragebitrate=64000;stereo=0;sprop-stereo=0;usedtx=0#015#012a=rtpmap:101 telephone-event/8000#015#012a=fmtp:101 0-15#015#012a=rtpmap:107 telephone-event/16000#015#012a=fmtp:107 0-15#015#012a=rtpmap:108 telephone-event/48000#015#012a=fmtp:108 0-15#015#012a=ptime:20#015#012a=sendrecv#015#012m=video 16406 RTP/AVP 112 111 110#015#012b=TIAS:4000000#015#012a=rtpmap:112 H264/90000#015#012a=fmtp:112 profile-level-id=640c16;packetization-mode=1;max-fs=3600;max-mbps=108000#015#012a=rtpmap:111 H264/90000#015#012a=fmtp:111 profile-level-id=428016;packetization-mode=1;max-fs=3600;max-mbps=108000#015#012a=rtpmap:110 H264/90000#015#012a=fmtp:110 profile-level-id=428016;packetization-mode=0;max-fs=3600;max-mbps=108000#015#012a=imageattr:* recv [x=800,y=480,q=0.60] [x=1280,y=720,q=0.50]#015#012a=rtcp-fb:* nack pli#015#012a=rtcp-fb:* ccm fir#015#012a=rtcp-fb:* ccm tmmbr#015#012a=sendrecv#015
Jan 27 13:23:40 172.16.1.77  sendto ac100144:5060(352)
Jan 27 13:23:40 172.16.1.77  SIP/2.0 100 Trying#015#012Call-ID: 62d866de-dbf3f093@172.16.1.68#015#012CSeq: 101 INVITE#015#012Content-Length: 0#015#012From: <sip:1001@172.16.1.77>;tag=185cbf2176643d27o0#015#012To: "3115551234" <sip:3115551234@172.16.1.77>;tag=SP4edffda0dfdbf8279#015#012Via: SIP/2.0/UDP 172.16.1.68:5060;branch=z9hG4bK-aa386a62;received=172.16.1.68;rport=5060#015#012Server: OBIHAI/OBi200-3.2.2.5921#015#012#015
Jan 27 13:23:40 172.16.1.77  CCTL:NewCallOn Term 10[0] 1001->+13115551234,+13115551234
Jan 27 13:23:41 172.16.1.77  sendto ac100144:5060(365)
Jan 27 13:23:41 172.16.1.77  SIP/2.0 503 Service Unavailable#015#012Call-ID: 62d866de-dbf3f093@172.16.1.68#015#012CSeq: 101 INVITE#015#012Content-Length: 0#015#012From: <sip:1001@172.16.1.77>;tag=185cbf2176643d27o0#015#012To: "3115551234" <sip:3115551234@172.16.1.77>;tag=SP4edffda0dfdbf8279#015#012Via: SIP/2.0/UDP 172.16.1.68:5060;branch=z9hG4bK-aa386a62;received=172.16.1.68;rport=5060#015#012Server: OBIHAI/OBi200-3.2.2.5921#015#012#015
Jan 27 13:23:41 172.16.1.77  RTP:Del Channel
Jan 27 13:23:41 172.16.1.77  RTP:Del Channel
Jan 27 13:23:41 172.16.1.77  RxFrom:ac100144:5060
Jan 27 13:23:41 172.16.1.77  ACK sip:3115551234@172.16.1.77:5063 SIP/2.0#015#012Via: SIP/2.0/UDP 172.16.1.68:5060;branch=z9hG4bK-aa386a62#015#012From: <sip:1001@172.16.1.77>;tag=185cbf2176643d27o0#015#012To: "3115551234" <sip:3115551234@172.16.1.77>;tag=SP4edffda0dfdbf8279#015#012Call-ID: 62d866de-dbf3f093@172.16.1.68#015#012CSeq: 101 ACK#015#012Max-Forwards: 70#015#012Contact: <sip:1001@172.16.1.68:5060>#015#012User-Agent: Cisco-CP-8865-3PCC/11.3.1#015#012Content-Length: 0#015#012#015

Muxor

#1
February 04, 2020, 02:10:56 PM
After a lot of digging I found the answer and it's working now.  I'll post the solution in case someone else is having the same issue.  The key here is where to look for the error.  I was running Wireshark and looking at the OBi syslog but kept only seeing a 503 message with no detail.  After more digging I found a post from azrobert where he suggested checking the call log.  When I did I found there was an error of 488 not acceptable here.

Following that error I found that this message happens when the phone is requesting an invalid codec.  You can see this here in this OBitalk forum as well as on Cisco's forums.
https://community.cisco.com/t5/ip-telephony-and-phones/sip-2-0-488-not-acceptable-here/td-p/1445481

I found the fix here on this forum in this thread:
http://www.obitalk.com/forum/index.php?topic=13489.msg89474#msg89474

Even though the Cisco phone is setup for auto codec by default the request is somehow triggering the 488 message from the OBi200.  The annoying part is the 488 only shows up in the call log and not in the syslog but oh well.

Once I set the Cisco phone to only use the preferred codec it started working.  My current setup on the Cisco phone is preferred codec g711u and use preferred codec only = yes.

ElectroBlvd

#2
September 11, 2020, 05:52:26 PM
I have a Cisco CP-8865 phone that I have been wanting to use for the past year. So I purchased a Obi200 and received it today. The Obi200 is all set up for Google Voice and I can use it with a standard analog phone. I want to use my CP-8865 though and can't find any information on how to set this up. Would you mind posting some kind of walk through?
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