SOLVED Cisco phone 7970 cannot register

[wzheng]

Hello,

I try to register my Cisco 7960 phone but it shows "registering" for a long time. Then after a while, it shows "Your current options". and the telephone icon near the phone number has a cross.

This is the console log from the phone:

894: NOT 03:13:48.836857 DSP: CODEC[0] G.711 direction:2 cost:29 budget:100 available
895: NOT 03:13:48.837879 DSP: CODEC[1] G.729A or G.729AB direction:2 cost:41 budget:100 available
896: NOT 03:13:48.838240 DSP: CODEC[2] G.729 or G.729B direction:2 cost:41 budget:100 available
897: NOT 03:13:48.838552 DSP: CODEC[3] LINEAR 8 or 16kHz direction:2 cost:29 budget:100 available
898: NOT 03:13:48.838873 DSP: CODEC[4] G.722 direction:2 cost:32767 budget:100 NOT available
899: NOT 03:13:48.839183 DSP: CODEC[5] iLBC direction:2 cost:65534 budget:100 NOT available
900: NOT 03:13:48.839494 DSP: STREAM- GetCapableCodecList requestType:2 bitmap:0xf
901: ERR 03:14:47.816379 JVM: %REG send failure: REGISTER
902: NOT 03:14:48.867237 DSP: CODEC[0] G.711 direction:2 cost:29 budget:100 available
903: NOT 03:14:48.868207 DSP: CODEC[1] G.729A or G.729AB direction:2 cost:41 budget:100 available
904: NOT 03:14:48.869136 DSP: CODEC[2] G.729 or G.729B direction:2 cost:41 budget:100 available
905: NOT 03:14:48.870182 DSP: CODEC[3] LINEAR 8 or 16kHz direction:2 cost:29 budget:100 available
906: NOT 03:14:48.871097 DSP: CODEC[4] G.722 direction:2 cost:32767 budget:100 NOT available
907: NOT 03:14:48.872024 DSP: CODEC[5] iLBC direction:2 cost:65534 budget:100 NOT available
908: NOT 03:14:48.872943 DSP: STREAM- GetCapableCodecList requestType:2 bitmap:0xf
909: DBG 03:15:07.750972 JVM: HTTP JNI| RpExternalCgi() 1st call for fConnectionId=0
910: DBG 03:15:07.751956 JVM: HTTP JNI| bypassing authentication for Serviceability
911: DBG 03:15:07.752264 JVM: HTTP JNI| *** HTTP_CGI_AUTHENTICATION_COMPLETE ***
912: DBG 03:15:07.752575 JVM: HTTP JNI| create_http_rsp_file() 0 [/usr/cache/PUSH_RESP_0_0.x] []
913: DBG 03:15:07.756306 JVM: HTTP JNI| *** process /CGI/Java ***
914: DBG 03:15:07.757129 JVM: HTTP JNI| process_servlet()
915: DBG 03:15:07.758232 JVM: HTTP JNI| calling ServletContainerClass_requestServlet()
916: INF 03:15:07.779439 JVM: HTTP JNI| call to ServletContainerClass_requestServlet successful
917: DBG 03:15:08.169546 JVM: HTTP JNI| starting updateServletRequest()
918: DBG 03:15:08.171150 JVM: HTTP JNI| UIUpdatePush: 0 3 0
919: DBG 03:15:08.172648 JVM: HTTP JNI| starting updateServletRequest()
920: DBG 03:15:08.173538 JVM: HTTP JNI| UIUpdatePush: 0 3 0
921: DBG 03:15:08.190394 JVM: HTTP JNI| *** *** *** PUSH_DONE: res file:/usr/cache/PUSH_RESP_0_0.x
922: DBG 03:15:08.191889 JVM: HTTP JNI| Rsp file</usr/cache/PUSH_RESP_0_0.x> has size:9173 max buff size is:1024
923: DBG 03:15:11.050984 JVM: HTTP JNI| RpExternalCgi() 1st call for fConnectionId=0
924: DBG 03:15:11.051959 JVM: HTTP JNI| bypassing authentication for Serviceability
925: DBG 03:15:11.052802 JVM: HTTP JNI| *** HTTP_CGI_AUTHENTICATION_COMPLETE ***
926: DBG 03:15:11.053760 JVM: HTTP JNI| create_http_rsp_file() 0 [/usr/cache/PUSH_RESP_0_1.x] [/usr/cache/PUSH_RESP_0_0.x]
927: DBG 03:15:11.055902 JVM: HTTP JNI| *** process /CGI/Java ***
928: DBG 03:15:11.056734 JVM: HTTP JNI| process_servlet()
929: DBG 03:15:11.057926 JVM: HTTP JNI| calling ServletContainerClass_requestServlet()
930: INF 03:15:11.174121 JVM: HTTP JNI| call to ServletContainerClass_requestServlet successful



This the debug message from PIAF:



v=0
o=Cisco-SIPUA 19000 0 IN IP4 192.168.1.115
s=SIP Call
t=0 0
m=audio 0 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (17 headers 11 lines) ---
Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS
Reliably Transmitting (no NAT) to 192.168.1.115:5060:
OPTIONS sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.108:5060;branch=z9hG4bK00effa34
Max-Forwards: 70
From: "Unknown" <sip:[email protected]>;tag=as3cd3df4b
To: <sip:[email protected]:5060;transport=udp>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.10.1(1.8.28.0)
Date: Thu, 05 Jun 2014 23:32:01 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:192.168.1.115:49209 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.108:5060;branch=z9hG4bK00effa34
From: "Unknown" <sip:[email protected]>;tag=as3cd3df4b
To: <sip:[email protected]:5060;transport=udp>;tag=0018ba32c719002c41bdf93f-84888db0
Call-ID: [email protected]:5060
Date: Wed, 16 Dec 2009 08:28:49 GMT
CSeq: 102 OPTIONS
Server: Cisco-CP7970G/8.5.3
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Allow-Events: kpml,dialog,refer
Accept: application/sdp,multipart/mixed,multipart/alternative
Accept-Encoding: identity
Accept-Language: en
Supported: replaces,join,norefersub
Content-Length: 237
Content-Type: application/sdp
Content-Disposition: session;handling=optional

v=0
o=Cisco-SIPUA 9177 0 IN IP4 192.168.1.115
s=SIP Call
t=0 0
m=audio 0 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (17 headers 11 lines) ---
Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS



I attached the cnf.xml file of the phone. I delete many of setting in the files to eliminate the possible errors. Does anyone know what the problem might be. Thanks.



Strange enough, PIAF sip show peers show the status like registered even though when it's registering.


pbx*CLI> sip show peers
Name/username Host Dyn Forcerport ACL Port Status
201/201 192.168.1.115 D A 5060 OK (34 ms)
202/202 192.168.1.118 D A 4358 OK (56 ms)

I can call from my softphone 202 to 201. But I cannot pick up the phone by hitting answer of pressing the speaker button. The phone just keeps ringing. I've attached some logs during the calling.

And the time is always starting from 12/16/09 no matter how I set up the NTP before.I don't know why the locale info cannot be updated.



Regards,
Bill

 

[rossiv]

The time starting from 12/16/09 sounds like the firmware compilation date. Sold my 7970, but on each boot up it always showed that time.

Do you have NAT=no and Qualify=No (or never?) in the extension config?

What firmware version are you running?

 

[John Bonzey]

I have used a 7970 in my office tied to my asterisk system for over 3 years. I could not be happier with the phone and have no issues at all with it. I know a lot of people seem to have issues getting it working but once you figure it out you will be very glad you did - it will be rock solid. Conference, transfer, message indicator, distinctive rings, wallpaper photos on screen etc - all works great. It has been a while since I messed with it but have attached my cmf.xml file. All you will need to do is change anything that has 192.168.1.110 to your asterisk ip address which i think is 192.168.1.108 reading your notes. You will also need to change <authName>1680</authName> and <authPassword>your_password fields to your peer settings (looks like you know this already though).

I'm assuming you already have your t*f*t*p server on the asterisk box activated with the cmf.xml file in the tftpboot directory. Also, I am running SIP70.8-5-2SR1S for firmware. Not sure about your newer firmware but I know this version works perfectly and earlier versions had issues. You will need to change <loadInformation> field to match your firmware in the cmf.xml file as well.

Other than that it should just work and register. When my phone gets those x's on the screen it is usually because it is not loading the file correctly from the t*f*t*p server.

For what is't worth I have my extension setting to qualify=yes, NAT=no, context=from-internal, can reinvite=no, transport=udp only


Good luck

 

[wzheng]

Thanks guy.

Hi Ross, the firmware version is 854. I didn't try 931 yet even I have the firmware. Not sure about qualify, but the NAT is no. I will check the setting later.


Hi John, thanks for the sample. It will be very helpful. I am going to try it over the weekend. I will downgrade to the 852 firmware version if it doesn't work.


BTW, in your config file, I see this:

<ntp>
<name>64.90.182.55</name>
<ntpMode>Unicast</ntpMode>
</ntp>


I guess you got the IP info from http://tf.nist.gov/tf-cgi/servers.cgi
I used to use that IP but I found out that they turn the server off some time ago. You will notice your phone lose time information on the screen and you cannot see when people called you(it will show 00:00). So the best strategy is use the domian name " time.nist.gov" as it said:


The global address time.nist.gov is resolved to all of the server addresses below in a round-robin sequence to equalize the load across all of the servers.

Thanks again, guys. I will let you know the results later.

 

[John Bonzey]

thanks I saw the server was off a few minutes ago. For some reason the phone seems to hold perfect time anyway - not sure why. I'll make the change to the domain name - Thanks !!!!!