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<body class='hmmessage'>Hi there!<BR>
<BR>
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 : <A href="http://forums.digium.com/viewtopic.php?t=19160">http://forums.digium.com/viewtopic.php?t=19160</A> <BR>
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. <BR>
I went through another post here on this subject at : <BR><A href="http://forums.digium.com/viewtopic.php?t=15212&highlight=7970">http://forums.digium.com/viewtopic.php?t=15212&highlight=7970</A> <BR>
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=" <BR>
However, the phone is still stuck in Registering, and I see these new messages on the asterisk CLI : <BR>
<BR><-------------> <BR>--- (13 headers 0 lines) --- <BR>Using latest REGISTER request as basis request <BR>Sending to 10.16.121.170 : 49309 (NAT) <BR>
<--- Transmitting (NAT) to 10.16.121.170:49309 ---> <BR>SIP/2.0 100 Trying <BR>Via: SIP/2.0/UDP 10.16.121.170:5060;branch=z9hG4bK12e80d0f;received=10.16.121.170 <BR>From: <sip:2001@172.19.125.13>;tag=001e4a5f12700002ab51cff4-e26d9841 <BR>To: <sip:2001@172.19.125.13> <BR>Call-ID: <A href="mailto:001e4a5f-12700002-e1c0d642-bb021e13@10.16.121.170">001e4a5f-12700002-e1c0d642-bb021e13@10.16.121.170</A> <BR>CSeq: 101 REGISTER <BR>User-Agent: Asterisk PBX <BR>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY <BR>Supported: replaces <BR>Contact: <sip:2001@172.19.125.13> <BR>Content-Length: 0 <BR>
<BR><------------> <BR>d2armyFreePBX*CLI> <BR><--- Transmitting (NAT) to 10.16.121.170:49309 ---> <BR>SIP/2.0 200 OK <BR>Via: SIP/2.0/UDP 10.16.121.170:5060;branch=z9hG4bK12e80d0f;received=10.16.121.170 <BR>From: <sip:2001@172.19.125.13>;tag=001e4a5f12700002ab51cff4-e26d9841 <BR>To: <sip:2001@172.19.125.13>;tag=as3f746d9f <BR>Call-ID: <A href="mailto:001e4a5f-12700002-e1c0d642-bb021e13@10.16.121.170">001e4a5f-12700002-e1c0d642-bb021e13@10.16.121.170</A> <BR>CSeq: 101 REGISTER <BR>User-Agent: Asterisk PBX <BR>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY <BR>Supported: replaces <BR>Expires: 3600 <BR>Contact: <sip:2001@10.16.121.170:5060;transport=udp>;expires=3600 <BR>Date: Thu, 29 Nov 2007 14:00:55 GMT <BR>Content-Length: 0 <BR>
<BR><------------> <BR>Scheduling destruction of SIP dialog <A href="mailto:'001e4a5f-12700002-e1c0d642-bb021e13@10.16.121.170'">'001e4a5f-12700002-e1c0d642-bb021e13@10.16.121.170'</A> in 32000 ms (Method: REGISTER) <BR>Retransmitting #1 (NAT) to 10.16.121.170:49309: <BR>OPTIONS sip:2001@10.16.121.170:5060;transport=udp SIP/2.0 <BR>Via: SIP/2.0/UDP 172.19.125.13:5060;branch=z9hG4bK1d9fde6c;rport <BR>From: "Unknown" <sip:Unknown@172.19.125.13>;tag=as76e8e4a2 <BR>To: <sip:2001@10.16.121.170:5060;transport=udp> <BR>Contact: <sip:Unknown@172.19.125.13> <BR>Call-ID: <A href="mailto:0ceb39367da8e62636859beb017f91e5@172.19.125.13">0ceb39367da8e62636859beb017f91e5@172.19.125.13</A> <BR>CSeq: 102 OPTIONS <BR>User-Agent: Asterisk PBX <BR>Max-Forwards: 70 <BR>Date: Thu, 29 Nov 2007 14:00:54 GMT <BR>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY <BR>Supported: replaces <BR>Content-Length: 0 <BR>
<BR>--- <BR>d2armyFreePBX*CLI> <BR><--- SIP read from 10.16.121.170:49309 ---> <BR>REGISTER sip:172.19.125.13 SIP/2.0 <BR>Via: SIP/2.0/UDP 10.16.121.170:5060;branch=z9hG4bK12e80d0f <BR>From: <sip:2001@172.19.125.13>;tag=001e4a5f12700002ab51cff4-e26d9841 <BR>To: <sip:2001@172.19.125.13> <BR>Call-ID: <A href="mailto:001e4a5f-12700002-e1c0d642-bb021e13@10.16.121.170">001e4a5f-12700002-e1c0d642-bb021e13@10.16.121.170</A> <BR>Max-Forwards: 70 <BR>Date: Fri, 02 Nov 2007 23:25:54 GMT <BR>CSeq: 101 REGISTER <BR>User-Agent: Cisco-CP7970G/8.3.0 <BR>Contact: <sip:2001@10.16.121.170:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-0000-0000-001e4a5f1270>";+u.sip!model.ccm.cisco.com="30006" <BR>Supported: (null),X-cisco-xsi-6.0.2 <BR>Content-Length: 0 <BR>Expires: 3600 <BR>
<BR>
These messages repeat again and again. It does not look like the "SIP/2.0 200 OK" message is any better than 401 before. <BR>
My config in sip_additional.conf is : <BR>
[2001] <BR>type=friend <BR>password=2001 <BR>record_out=Adhoc <BR>record_in=Adhoc <BR>qualify=yes <BR>port=5060 <BR>pickupgroup= <BR>nat=yes <BR><A href="mailto:mailbox=2001@device">mailbox=2001@device</A> <BR>host=dynamic <BR>dtmfmode=rfc2833 <BR>disallow= <BR>dial=SIP/2001 <BR>context=from-sip <BR>canreinvite=no <BR>callgroup= <BR>callerid=device <2001> <BR>allow= <BR>accountcode= <BR>call-limit=50 <BR>
<BR>
My updated SEP<MAC> file for this hard phone is at <A href="http://cid-ff3ef0764138e401.skydrive.live.com/self.aspx/Public/SEP001E4A5F1270.cnf.xml">http://cid-ff3ef0764138e401.skydrive.live.com/self.aspx/Public/SEP001E4A5F1270.cnf.xml</A> <BR>
On the phone side when I ssh in, "show register" shows : <BR>
LINE REGISTRATION TABLE <BR>Proxy Registration: ENABLED, state: IDLE <BR>line APR state timer expires proxy:port <BR>---- --- ------------- ---------- ---------- ---------------------------- <BR>1 .1x REGISTERING 0 0 172.19.125.13:5060 <BR>2 ... NONE 0 0 undefined:0 <BR>3 ... NONE 0 0 undefined:0 <BR>4 ... NONE 0 0 undefined:0 <BR>5 ... NONE 0 0 undefined:0 <BR>6 ... NONE 0 0 undefined:0 <BR>7 ... NONE 0 0 undefined:0 <BR>8 ... NONE 0 0 undefined:0 <BR>1-BU .1x REGISTERING 3600 17 172.19.125.13:5060 <BR>
Note: APR is Authenticated, Provisioned, Registered <BR>
<BR>Please help, thanks <BR>John<BR><BR><br /><hr />You keep typing, we keep giving. Download Messenger and join the i’m Initiative now. <a href='http://im.live.com/messenger/im/home/?source=CRM_WL_joinnow' target='_new'>Join in!</a></body>
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