John Constalgie
2007-Nov-30 22:56 UTC
[asterisk-users] Problem registering Cisco 7970 phone with Asterisk 1.4 running FreePBX
Hi there! I am having problems registering my 7970 hardphone with Asterisk 1.4(with FreePBX interface). I had an earlier post about trying to get it to work first with a 7970 emulator (Cisco IP Communicator) on the Asterisk Forum : http://forums.digium.com/viewtopic.php?t=19160 Instead I decided to try the real phone instead, and was able to advance further. The firmware was able to install smoothly but I am stuck at the registration part. I went through another post here on this subject at : http://forums.digium.com/viewtopic.php?t=15212&highlight=7970 This helped me get past the SIP 401 Unauthorized error when I went into the sip_additional.conf file and changed the "secret=" line to "password=" However, the phone is still stuck in Registering, and I see these new messages on the asterisk CLI : <-------------> --- (13 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 10.16.121.170 : 49309 (NAT) <--- Transmitting (NAT) to 10.16.121.170:49309 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.16.121.170:5060;branch=z9hG4bK12e80d0f;received=10.16.121.170 From: <sip:2001 at 172.19.125.13>;tag=001e4a5f12700002ab51cff4-e26d9841 To: <sip:2001 at 172.19.125.13> Call-ID: 001e4a5f-12700002-e1c0d642-bb021e13 at 10.16.121.170 CSeq: 101 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:2001 at 172.19.125.13> Content-Length: 0 <------------> d2armyFreePBX*CLI> <--- Transmitting (NAT) to 10.16.121.170:49309 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.16.121.170:5060;branch=z9hG4bK12e80d0f;received=10.16.121.170 From: <sip:2001 at 172.19.125.13>;tag=001e4a5f12700002ab51cff4-e26d9841 To: <sip:2001 at 172.19.125.13>;tag=as3f746d9f Call-ID: 001e4a5f-12700002-e1c0d642-bb021e13 at 10.16.121.170 CSeq: 101 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 3600 Contact: <sip:2001 at 10.16.121.170:5060;transport=udp>;expires=3600 Date: Thu, 29 Nov 2007 14:00:55 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog '001e4a5f-12700002-e1c0d642-bb021e13 at 10.16.121.170' in 32000 ms (Method: REGISTER) Retransmitting #1 (NAT) to 10.16.121.170:49309: OPTIONS sip:2001 at 10.16.121.170:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 172.19.125.13:5060;branch=z9hG4bK1d9fde6c;rport From: "Unknown" <sip:Unknown at 172.19.125.13>;tag=as76e8e4a2 To: <sip:2001 at 10.16.121.170:5060;transport=udp> Contact: <sip:Unknown at 172.19.125.13> Call-ID: 0ceb39367da8e62636859beb017f91e5 at 172.19.125.13 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 29 Nov 2007 14:00:54 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- d2armyFreePBX*CLI> <--- SIP read from 10.16.121.170:49309 ---> REGISTER sip:172.19.125.13 SIP/2.0 Via: SIP/2.0/UDP 10.16.121.170:5060;branch=z9hG4bK12e80d0f From: <sip:2001 at 172.19.125.13>;tag=001e4a5f12700002ab51cff4-e26d9841 To: <sip:2001 at 172.19.125.13> Call-ID: 001e4a5f-12700002-e1c0d642-bb021e13 at 10.16.121.170 Max-Forwards: 70 Date: Fri, 02 Nov 2007 23:25:54 GMT CSeq: 101 REGISTER User-Agent: Cisco-CP7970G/8.3.0 Contact: <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-6.0.2 Content-Length: 0 Expires: 3600 These messages repeat again and again. It does not look like the "SIP/2.0 200 OK" message is any better than 401 before. My config in sip_additional.conf is : [2001] type=friend password=2001 record_out=Adhoc record_in=Adhoc qualify=yes port=5060 pickupgroup= nat=yes mailbox=2001 at device host=dynamic dtmfmode=rfc2833 disallow= dial=SIP/2001 context=from-sip canreinvite=no callgroup= callerid=device <2001> allow= accountcode= call-limit=50 My updated SEP<MAC> file for this hard phone is at http://cid-ff3ef0764138e401.skydrive.live.com/self.aspx/Public/SEP001E4A5F1270.cnf.xml On the phone side when I ssh in, "show register" shows : LINE REGISTRATION TABLE Proxy Registration: ENABLED, state: IDLE line APR state timer expires proxy:port ---- --- ------------- ---------- ---------- ---------------------------- 1 .1x REGISTERING 0 0 172.19.125.13:5060 2 ... NONE 0 0 undefined:0 3 ... NONE 0 0 undefined:0 4 ... NONE 0 0 undefined:0 5 ... NONE 0 0 undefined:0 6 ... NONE 0 0 undefined:0 7 ... NONE 0 0 undefined:0 8 ... NONE 0 0 undefined:0 1-BU .1x REGISTERING 3600 17 172.19.125.13:5060 Note: APR is Authenticated, Provisioned, Registered Please help, thanks John _________________________________________________________________ You keep typing, we keep giving. Download Messenger and join the i?m Initiative now. http://im.live.com/messenger/im/home/?source=TAGLM -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071130/6b1e16dd/attachment.htm
John Constalgie
2007-Dec-02 20:58 UTC
[asterisk-users] Problem registering Cisco 7970 phone with Asterisk 1.4 running FreePBX
Hi there! I am having problems registering my 7970 hardphone with Asterisk 1.4(with FreePBX interface). I had an earlier post about trying to get it to work first with a 7970 emulator (Cisco IP Communicator) on the Asterisk Forum : http://forums.digium.com/viewtopic.php?t=19160 Instead I decided to try the real phone instead, and was able to advance further. The firmware was able to install smoothly but I am stuck at the registration part. I went through another post here on this subject at : http://forums.digium.com/viewtopic.php?t=15212&highlight=7970 This helped me get past the SIP 401 Unauthorized error when I went into the sip_additional.conf file and changed the "secret=" line to "password=" However, the phone is still stuck in Registering, and I see these new messages on the asterisk CLI : <-------------> --- (13 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 10.16.121.170 : 49309 (NAT) <--- Transmitting (NAT) to 10.16.121.170:49309 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.16.121.170:5060;branch=z9hG4bK12e80d0f;received=10.16.121.170 From: <sip:2001 at 172.19.125.13>;tag=001e4a5f12700002ab51cff4-e26d9841 To: <sip:2001 at 172.19.125.13> Call-ID: 001e4a5f-12700002-e1c0d642-bb021e13 at 10.16.121.170 CSeq: 101 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:2001 at 172.19.125.13> Content-Length: 0 <------------> d2armyFreePBX*CLI> <--- Transmitting (NAT) to 10.16.121.170:49309 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.16.121.170:5060;branch=z9hG4bK12e80d0f;received=10.16.121.170 From: <sip:2001 at 172.19.125.13>;tag=001e4a5f12700002ab51cff4-e26d9841 To: <sip:2001 at 172.19.125.13>;tag=as3f746d9f Call-ID: 001e4a5f-12700002-e1c0d642-bb021e13 at 10.16.121.170 CSeq: 101 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 3600 Contact: <sip:2001 at 10.16.121.170:5060;transport=udp>;expires=3600 Date: Thu, 29 Nov 2007 14:00:55 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog '001e4a5f-12700002-e1c0d642-bb021e13 at 10.16.121.170' in 32000 ms (Method: REGISTER) Retransmitting #1 (NAT) to 10.16.121.170:49309: OPTIONS sip:2001 at 10.16.121.170:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 172.19.125.13:5060;branch=z9hG4bK1d9fde6c;rport From: "Unknown" <sip:Unknown at 172.19.125.13>;tag=as76e8e4a2 To: <sip:2001 at 10.16.121.170:5060;transport=udp> Contact: <sip:Unknown at 172.19.125.13> Call-ID: 0ceb39367da8e62636859beb017f91e5 at 172.19.125.13 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 29 Nov 2007 14:00:54 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- d2armyFreePBX*CLI> <--- SIP read from 10.16.121.170:49309 ---> REGISTER sip:172.19.125.13 SIP/2.0 Via: SIP/2.0/UDP 10.16.121.170:5060;branch=z9hG4bK12e80d0f From: <sip:2001 at 172.19.125.13>;tag=001e4a5f12700002ab51cff4-e26d9841 To: <sip:2001 at 172.19.125.13> Call-ID: 001e4a5f-12700002-e1c0d642-bb021e13 at 10.16.121.170 Max-Forwards: 70 Date: Fri, 02 Nov 2007 23:25:54 GMT CSeq: 101 REGISTER User-Agent: Cisco-CP7970G/8.3.0 Contact: <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-6.0.2 Content-Length: 0 Expires: 3600 These messages repeat again and again, until I do "sip set debug off". It does not look like the "SIP/2.0 200 OK" message is any better than 401 before. My config in sip_additional.conf is : [2001] type=friend password=2001 record_out=Adhoc record_in=Adhoc qualify=yes port=5060 pickupgroup= nat=yes mailbox=2001 at device host=dynamic dtmfmode=rfc2833 disallow= dial=SIP/2001 context=from-sip canreinvite=no callgroup= callerid=device <2001> allow= accountcode= call-limit=50 When I do "sip show peers" , I see : Name/username Host Dyn Nat ACL Port Status 2001/2001 (Unspecified) D N 0 UNKNOWN 1 sip peers [Monitored: 0 online, 1 offline Unmonitored: 0 online, 0 offline] My updated SEP<MAC> file for this hard phone is at http://cid-ff3ef0764138e401.skydrive.live.com/self.aspx/Public/SEP001E4A5F1270.cnf.xml On the phone side when I ssh in, "show register" shows : LINE REGISTRATION TABLE Proxy Registration: ENABLED, state: IDLE line APR state timer expires proxy:port ---- --- ------------- ---------- ---------- ---------------------------- 1 .1x REGISTERING 0 0 172.19.125.13:5060 2 ... NONE 0 0 undefined:0 3 ... NONE 0 0 undefined:0 4 ... NONE 0 0 undefined:0 5 ... NONE 0 0 undefined:0 6 ... NONE 0 0 undefined:0 7 ... NONE 0 0 undefined:0 8 ... NONE 0 0 undefined:0 1-BU .1x REGISTERING 3600 17 172.19.125.13:5060 Note: APR is Authenticated, Provisioned, Registered Please help, thanks John _________________________________________________________________ Put your friends on the big screen with Windows Vista? + Windows Live?. http://www.microsoft.com/windows/shop/specialoffers.mspx?ocid=TXT_TAGLM_CPC_MediaCtr_bigscreen_102007 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071202/252f51ad/attachment.htm
Edwin Lam
2007-Dec-03 19:34 UTC
[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.xmltry 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