[asterisk-users] Problem registering Cisco 7970 phone with Asterisk 1.4 running FreePBX
John Constalgie
caduceus_abode at hotmail.com
Mon Dec 3 15:28:29 CST 2007
Hi Edwin,
I did what you said for the SEP file ( updated SEP xml file : http://cid-ff3ef0764138e401.skydrive.live.com/self.aspx/Public/SEP001E4A5F1270.cnf.xml )
By the way, I was reading up online that I could change the "qualify=yes" setting to "no" in sip_additional.conf to make my phone unmonitored. My sip_additional.conf is now :
[2001]type=friendpassword=2001 record_out=Adhocrecord_in=Adhocqualify=noport=5060pickupgroup=nat=yesmailbox=2001 at devicehost=dynamicdtmfmode=rfc2833disallow=dial=SIP/2001context=from-sipcanreinvite=nocallgroup=callerid=device <2001>allow=ulawaccountcode=call-limit=50
Now, when I try to register my phone, it is still stuck in registering, but on my Asterisk command line, I see these SIP debug messages :
<--- SIP read from 10.16.121.170:49184 --->REGISTER sip:172.19.125.13 SIP/2.0Via: SIP/2.0/UDP 10.16.121.170:5060;branch=z9hG4bKc49e333fFrom: <sip:2001 at 172.19.125.13>;tag=001e4a5f127000026843af34-83d1cb71To: <sip:2001 at 172.19.125.13>Call-ID: 001e4a5f-12700002-424e3c02-7ef59263 at 10.16.121.170Max-Forwards: 70Date: Fri, 02 Nov 2007 21:53:19 GMTCSeq: 101 REGISTERUser-Agent: Cisco-CP7970G/8.3.0Contact: <sip:2001 at 10.16.121.170:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-0000-0000-001e4a5f1270>";+u.sip!model.ccm.cisco.com="30006"Supported: (null),X-cisco-xsi-6.0.2Content-Length: 0Reason: SIP;cause=200;text="cisco-alarm:20 Name=SEP001E4A5F1270 Load=SIP70.8-3-3SR2S Last=phone-keypad"Expires: 3600
<------------->--- (14 headers 0 lines) ---Using latest REGISTER request as basis requestSending to 10.16.121.170 : 5060 (no NAT)
<--- Transmitting (NAT) to 10.16.121.170:49184 --->SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.16.121.170:5060;branch=z9hG4bKc49e333f;received=10.16.121.170From: <sip:2001 at 172.19.125.13>;tag=001e4a5f127000026843af34-83d1cb71To: <sip:2001 at 172.19.125.13>Call-ID: 001e4a5f-12700002-424e3c02-7ef59263 at 10.16.121.170CSeq: 101 REGISTERUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContact: <sip:2001 at 172.19.125.13>Content-Length: 0
<------------>d2armyFreePBX*CLI> <--- Transmitting (NAT) to 10.16.121.170:49184 --->SIP/2.0 200 OKVia: SIP/2.0/UDP 10.16.121.170:5060;branch=z9hG4bKc49e333f;received=10.16.121.170From: <sip:2001 at 172.19.125.13>;tag=001e4a5f127000026843af34-83d1cb71To: <sip:2001 at 172.19.125.13>;tag=as473e804fCall-ID: 001e4a5f-12700002-424e3c02-7ef59263 at 10.16.121.170CSeq: 101 REGISTERUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesExpires: 3600Contact: <sip:2001 at 10.16.121.170:5060;transport=udp>;expires=3600Date: Mon, 03 Dec 2007 08:14:28 GMTContent-Length: 0
<------------>Scheduling destruction of SIP dialog '001e4a5f-12700002-424e3c02-7ef59263 at 10.16.121.170' in 32000 ms (Method: REGISTER)d2armyFreePBX*CLI> <--- SIP read from 10.16.121.170:49184 --->REGISTER sip:172.19.125.13 SIP/2.0Via: SIP/2.0/UDP 10.16.121.170:5060;branch=z9hG4bKc49e333fFrom: <sip:2001 at 172.19.125.13>;tag=001e4a5f127000026843af34-83d1cb71To: <sip:2001 at 172.19.125.13>Call-ID: 001e4a5f-12700002-424e3c02-7ef59263 at 10.16.121.170Max-Forwards: 70Date: Fri, 02 Nov 2007 21:53:19 GMTCSeq: 101 REGISTERUser-Agent: Cisco-CP7970G/8.3.0Contact: <sip:2001 at 10.16.121.170:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-0000-0000-001e4a5f1270>";+u.sip!model.ccm.cisco.com="30006"Supported: (null),X-cisco-xsi-6.0.2Content-Length: 0Reason: SIP;cause=200;text="cisco-alarm:20 Name=SEP001E4A5F1270 Load=SIP70.8-3-3SR2S Last=phone-keypad"Expires: 3600
<------------->--- (14 headers 0 lines) ---Using latest REGISTER request as basis requestSending to 10.16.121.170 : 49184 (NAT)
<--- Transmitting (NAT) to 10.16.121.170:49184 --->SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.16.121.170:5060;branch=z9hG4bKc49e333f;received=10.16.121.170From: <sip:2001 at 172.19.125.13>;tag=001e4a5f127000026843af34-83d1cb71To: <sip:2001 at 172.19.125.13>Call-ID: 001e4a5f-12700002-424e3c02-7ef59263 at 10.16.121.170CSeq: 101 REGISTERUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContact: <sip:2001 at 172.19.125.13>Content-Length: 0
<------------>d2armyFreePBX*CLI> <--- Transmitting (NAT) to 10.16.121.170:49184 --->SIP/2.0 200 OKVia: SIP/2.0/UDP 10.16.121.170:5060;branch=z9hG4bKc49e333f;received=10.16.121.170From: <sip:2001 at 172.19.125.13>;tag=001e4a5f127000026843af34-83d1cb71To: <sip:2001 at 172.19.125.13>;tag=as473e804fCall-ID: 001e4a5f-12700002-424e3c02-7ef59263 at 10.16.121.170CSeq: 101 REGISTERUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesExpires: 3600Contact: <sip:2001 at 10.16.121.170:5060;transport=udp>;expires=3600Date: Mon, 03 Dec 2007 08:14:29 GMTContent-Length: 0
<------------>Scheduling destruction of SIP dialog '001e4a5f-12700002-424e3c02-7ef59263 at 10.16.121.170' in 32000 ms (Method: REGISTER)d2armyFreePBX*CLI> <--- SIP read from 10.16.121.170:49184 --->REGISTER sip:172.19.125.13 SIP/2.0Via: SIP/2.0/UDP 10.16.121.170:5060;branch=z9hG4bKc49e333fFrom: <sip:2001 at 172.19.125.13>;tag=001e4a5f127000026843af34-83d1cb71To: <sip:2001 at 172.19.125.13>Call-ID: 001e4a5f-12700002-424e3c02-7ef59263 at 10.16.121.170Max-Forwards: 70Date: Fri, 02 Nov 2007 21:53:19 GMTCSeq: 101 REGISTERUser-Agent: Cisco-CP7970G/8.3.0Contact: <sip:2001 at 10.16.121.170:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-0000-0000-001e4a5f1270>";+u.sip!model.ccm.cisco.com="30006"Supported: (null),X-cisco-xsi-6.0.2Content-Length: 0Reason: SIP;cause=200;text="cisco-alarm:20 Name=SEP001E4A5F1270 Load=SIP70.8-3-3SR2S Last=phone-keypad"Expires: 3600
<------------->--- (14 headers 0 lines) ---Using latest REGISTER request as basis requestSending to 10.16.121.170 : 49184 (NAT)
<--- Transmitting (NAT) to 10.16.121.170:49184 --->SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.16.121.170:5060;branch=z9hG4bKc49e333f;received=10.16.121.170From: <sip:2001 at 172.19.125.13>;tag=001e4a5f127000026843af34-83d1cb71To: <sip:2001 at 172.19.125.13>Call-ID: 001e4a5f-12700002-424e3c02-7ef59263 at 10.16.121.170CSeq: 101 REGISTERUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContact: <sip:2001 at 172.19.125.13>Content-Length: 0
<------------>d2armyFreePBX*CLI> <--- Transmitting (NAT) to 10.16.121.170:49184 --->SIP/2.0 200 OKVia: SIP/2.0/UDP 10.16.121.170:5060;branch=z9hG4bKc49e333f;received=10.16.121.170From: <sip:2001 at 172.19.125.13>;tag=001e4a5f127000026843af34-83d1cb71To: <sip:2001 at 172.19.125.13>;tag=as473e804fCall-ID: 001e4a5f-12700002-424e3c02-7ef59263 at 10.16.121.170CSeq: 101 REGISTERUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesExpires: 3600Contact: <sip:2001 at 10.16.121.170:5060;transport=udp>;expires=3600Date: Mon, 03 Dec 2007 08:14:30 GMTContent-Length: 0
<------------>Scheduling destruction of SIP dialog '001e4a5f-12700002-424e3c02-7ef59263 at 10.16.121.170' in 32000 ms (Method: REGISTER)d2armyFreePBX*CLI> <--- SIP read from 10.16.121.170:49184 --->REGISTER sip:172.19.125.13 SIP/2.0Via: SIP/2.0/UDP 10.16.121.170:5060;branch=z9hG4bKc49e333fFrom: <sip:2001 at 172.19.125.13>;tag=001e4a5f127000026843af34-83d1cb71To: <sip:2001 at 172.19.125.13>Call-ID: 001e4a5f-12700002-424e3c02-7ef59263 at 10.16.121.170Max-Forwards: 70Date: Fri, 02 Nov 2007 21:53:19 GMTCSeq: 101 REGISTERUser-Agent: Cisco-CP7970G/8.3.0Contact: <sip:2001 at 10.16.121.170:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-0000-0000-001e4a5f1270>";+u.sip!model.ccm.cisco.com="30006"Supported: (null),X-cisco-xsi-6.0.2Content-Length: 0Reason: SIP;cause=200;text="cisco-alarm:20 Name=SEP001E4A5F1270 Load=SIP70.8-3-3SR2S Last=phone-keypad"Expires: 3600
<------------->--- (14 headers 0 lines) ---Using latest REGISTER request as basis requestSending to 10.16.121.170 : 49184 (NAT)
<--- Transmitting (NAT) to 10.16.121.170:49184 --->SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.16.121.170:5060;branch=z9hG4bKc49e333f;received=10.16.121.170From: <sip:2001 at 172.19.125.13>;tag=001e4a5f127000026843af34-83d1cb71To: <sip:2001 at 172.19.125.13>Call-ID: 001e4a5f-12700002-424e3c02-7ef59263 at 10.16.121.170CSeq: 101 REGISTERUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContact: <sip:2001 at 172.19.125.13>Content-Length: 0
<------------>d2armyFreePBX*CLI> <--- Transmitting (NAT) to 10.16.121.170:49184 --->SIP/2.0 200 OKVia: SIP/2.0/UDP 10.16.121.170:5060;branch=z9hG4bKc49e333f;received=10.16.121.170From: <sip:2001 at 172.19.125.13>;tag=001e4a5f127000026843af34-83d1cb71To: <sip:2001 at 172.19.125.13>;tag=as473e804fCall-ID: 001e4a5f-12700002-424e3c02-7ef59263 at 10.16.121.170CSeq: 101 REGISTERUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesExpires: 3600Contact: <sip:2001 at 10.16.121.170:5060;transport=udp>;expires=3600Date: Mon, 03 Dec 2007 08:14:32 GMTContent-Length: 0
This keeps repeating again and again, and Asterisk keeps retransmitting to the NAT for some reason even if the phone is off.
The command "sip show peers" gives me :
Name/username Host Dyn Nat ACL Port Status 2001/2001 10.16.121.170 D N 49184 Unmonitored 1 sip peers [Monitored: 0 online, 0 offline Unmonitored: 1 online, 0 offline]
On the FreePBX web GUI, I see 1 IP Phone online under FreePBX connections.
On the 7970 phone side, I see:
LINE REGISTRATION TABLEProxy Registration: ENABLED, state: REGISTERINGline APR state timer expires proxy:port---- --- ------------- ---------- ---------- ----------------------------1 .1x REGISTERING 3600 23 172.19.125.13:50602 ... NONE 0 0 undefined:03 ... NONE 0 0 undefined:04 ... NONE 0 0 undefined:05 ... NONE 0 0 undefined:06 ... NONE 0 0 undefined:07 ... NONE 0 0 undefined:08 ... NONE 0 0 undefined:01-BU .1x IDLE 0 0 undefined:0
Note: APR is Authenticated, Provisioned, Registered
Is this a problem with the phone side instead of Asterisk?
I hope I am getting closer to getting this working!
Thanks
John
> Date: Mon, 3 Dec 2007 11:34:10 -0800> From: elam at officegeneral.com> To: asterisk-users at lists.digium.com> Subject: Re: [asterisk-users] Problem registering Cisco 7970 phone with Asterisk 1.4 running FreePBX> > John Constalgie wrote:> > > > > > My updated SEP<MAC> file for this hard phone is at > > http://cid-ff3ef0764138e401.skydrive.live.com/self.aspx/Public/SEP001E4A5F1270.cnf.xml > > try set the backup, emergency, and outbound proxies to blank> under sipProxies section:> > <sipProxies>> <backupProxy></backupProxy>> <backupProxyPort></backupProxyPort>> <emergencyProxy></emergencyProxy>> <emergencyProxyPort></emergencyProxyPort>> <outboundProxy></outboundProxy>> <outboundProxyPort></outboundProxyPort>> <registerWithProxy>true</registerWithProxy>> </sipProxies>> > > > -- > Edwin Lam <edwin.lam at officegeneral.com>> Systems Engineer, Office General, Inc.> Ph: +1 415 439 4988 Fax: +1 415 283 3370> http://pgpkeys.mit.edu:11371/pks/lookup?op=get&search=0xD6506D20> > > _______________________________________________> --Bandwidth and Colocation Provided by http://www.api-digital.com--> > asterisk-users mailing list> To UNSUBSCRIBE or update options visit:> http://lists.digium.com/mailman/listinfo/asterisk-users
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