Update on 7970 registration to Asterisk with 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) : 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 :
forums.digium.com/viewtopic.php? … light=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@172.19.125.13;tag=001e4a5f12700002ab51cff4-e26d9841
To: sip:2001@172.19.125.13
Call-ID: 001e4a5f-12700002-e1c0d642-bb021e13@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@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@172.19.125.13;tag=001e4a5f12700002ab51cff4-e26d9841
To: sip:2001@172.19.125.13;tag=as3f746d9f
Call-ID: 001e4a5f-12700002-e1c0d642-bb021e13@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@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@10.16.121.170’ in 32000 ms (Method: REGISTER)
Retransmitting #1 (NAT) to 10.16.121.170:49309:
OPTIONS sip:2001@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@172.19.125.13;tag=as76e8e4a2
To: sip:2001@10.16.121.170:5060;transport=udp
Contact: sip:Unknown@172.19.125.13
Call-ID: 0ceb39367da8e62636859beb017f91e5@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@172.19.125.13;tag=001e4a5f12700002ab51cff4-e26d9841
To: sip:2001@172.19.125.13
Call-ID: 001e4a5f-12700002-e1c0d642-bb021e13@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@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@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 file for this hard phone is at cid-ff3ef0764138e401.skydrive.li … 70.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

I think your problem is in the header line

Supported: (null),X-cisco-xsi-6.0.2

() is not allowed here, but Cisco does not seem to care. This breaks the register METHOD.

If anyone who knows if there is a way through the config files to remove the (null) component of the supported field, I for one would love to hear it.

Regards

Mark Dutton

Hi,

I suggest you SCCP, it works almost out of the box, you only need the SEPMAC.cnf.xml hosted on a tftp server.

It works fine for me, I used the chan_sccp not chan_skinny (never tried it).

bye

PS: if you need I can post sample files.

Yep I know SCCP works and Cisco would rather I used it, but I want to use SIP as this is the standard. Fixing this problem also fixes it for any other SIP PBX around.

Thanks for the offer to post samples, but what I (and others with this problem I suspect) would like to see is a way to get Cisco handsets to send properly formatted SIP headers.

Bump… Has anyone succeeded with the IP Communicator using SIP?