Registering problem for Cisco 7942 with Asterisk

I have my Cisco 7942 loaded with my XML configuration file but it couldn't register the line.

When I open the console on the Asterisk, the phone IP doesn't appear in the server and I believe the phone don't go to my Asterisk for registration. Have any idea what's missing in my configuration file. I'm using firmware SIP42.9-0-3.

Great THanks

AXISHKAsked:
Who is Participating?
 
DrDamnitCommented:
I don't understand this question:
"Seem like Asterisk don't disgard the login. Any idea ?"

What do you mean don't disgard? Are you saying that the phone doesn't register?

I don't think this is a complete log. You have a SIP 100 trying, followed by a SIP 401 Unauthorized, which tells the phone "I need you to authenticate", followed by a REGISTER...

There must be more.
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DrDamnitCommented:
Please post your sip.conf and your Cisco phone configs.
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AXISHKAuthor Commented:
Only report of locale error when loading the XML configuration.

The panel displays "Registering" and keep loading for around 10min. Afterwards, the message dispear but the line doesn't register.

Will there be the firmware version ? The firmware is marked as 7942G but actually, my phone model is 7942. However, only 7942G firmware is availabe in the Cisco web.

Any advise is apprecitead.

http://www.cisco.com/cisco/software/release.html?mdfid=281346593&flowid=5266&softwareid=282074288&release=9.1(1)SR1&relind=AVAILABLE&rellifecycle=&reltype=latest


Thanks

SEP108CCF74A9C8.cnf.xml
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DrDamnitCommented:
Still need sip.conf.
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AXISHKAuthor Commented:
Here is the file. tks
sip.conf.txt
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DrDamnitCommented:
You're using FreePBX, so I need screen shots of the sip account for the Cisco account. (sip.conf just shows me includes).
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AXISHKAuthor Commented:
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DrDamnitCommented:
Your sip configs and Cisco configs appear to match.

Here's the next step:

To better assist you, I need to see what is going on. Please use the following procedure to create a log file of the Asterisk CLI:

1. Open the CLI using this:

       asterisk -rvvvvvv | tee /var/log/asterisk.log

This puts the a debug log into the file /var/log/asterisk.log

2. Reproduce the problem.
3. Make calls
4. Exit the CLI after you get the error.
5. Post the log here (as "code" or as an attachment).
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AXISHKAuthor Commented:
I have monitor the asterisk log before and actually, the inital registration has never go to Asterisk. I have also try ot disabled the firewall in Centos but it doesn't help any.

When I ssh to the Cisco phone, the following is observed :

3298: NOT 09:58:53.190892 JVM: SIPCC-SIP_REG_STATE: 1/51, sip_reg_sm_process_event: SIP_REG_STATE_IDLE <- E_SIP_REG_TMR_EXPIRE
3299: ERR 09:58:53.191566 JVM: ccsip_register_send_msg: Error: cc_cfg_table is null.
3300: NOT 09:58:53.192175 JVM: SIPCC-UI_API: 1/0, ui_set_sip_registration_state: 0
3301: NOT 09:59:53.190899 JVM: SIPCC-SIP_REG_STATE: 1/51, sip_reg_sm_process_event: SIP_REG_STATE_IDLE <- E_SIP_REG_TMR_EXPIRE
3302: ERR 09:59:53.191608 JVM: ccsip_register_send_msg: Error: cc_cfg_table is null.

So, it seems that Cisco phone has never inital a registration to Asterisk. I really don't have any idea on that. Any advise is appreciated.

Thanks
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DrDamnitCommented:
Did you set core debug on as well as sip set debug on in the CLI?
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AXISHKAuthor Commented:
Here is the command I run on Asterik

asterisk –vvvcgr
sip set debug

Is that enought ?

Thanks

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DrDamnitCommented:
Should be. And with debug on, there is still not attempt to connect from the phone?
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AXISHKAuthor Commented:
Yes, pretty sure. No registration from phone ...

http://www.fonality.com/trixbox/forums/trixbox-forums/trixbox-endpoints/cisco-9971-and-tftp

I have put the following in my configuration file and reload the sip configuration but it doesn't help.

We are using ATCOM phone and softphone (Eyebeam) and it works fine. Really not idea why it happen like this.....


sip.conf:
tcpenable=yes
tcpbindaddr=0.0.0.0

[<my extension no>]
transport=tcp

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DrDamnitCommented:
My only suggestion at this point is to make sure that you have configured the XML properly, and you need to double check all type cases. The Cisco's are case sensitive.
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AXISHKAuthor Commented:
Is there a way to vertify the cnf.xml, without the Cisco Call Manager ...

Tks
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AXISHKAuthor Commented:
Finally fix the XML and the registeration process could be logged in Asterisk. Seem like Asterisk don't disgard the login. Any idea ?

Thanks

------------->
--- (14 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 192.168.1.193 : 49164 (NAT)

<--- Transmitting (NAT) to 192.168.1.193:49164 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.193:5060;branch=z9hG4bK6fdf070a;received=192.168.1.193
From: <sip:2893@192.168.1.32>;tag=108ccf74a9c80005be8aa1f2-019f374a
To: <sip:2893@192.168.1.32>
Call-ID: 108ccf74-a9c80002-8188eea8-5c56bd88@192.168.1.193
CSeq: 104 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


<------------>

<--- Transmitting (NAT) to 192.168.1.193:49164 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.193:5060;branch=z9hG4bK6fdf070a;received=192.168.1.193
From: <sip:2893@192.168.1.32>;tag=108ccf74a9c80005be8aa1f2-019f374a
To: <sip:2893@192.168.1.32>;tag=as25c37266
Call-ID: 108ccf74-a9c80002-8188eea8-5c56bd88@192.168.1.193
CSeq: 104 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1316e00b"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '108ccf74-a9c80002-8188eea8-5c56bd88@192.168.1.193' in 32000 ms (Method: REGISTER)

<--- SIP read from 192.168.1.193:49164 --->
REGISTER sip:192.168.1.32 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.193:5060;branch=z9hG4bK6fdf070a
From: <sip:2893@192.168.1.32>;tag=108ccf74a9c80005be8aa1f2-019f374a
To: <sip:2893@192.168.1.32>
Call-ID: 108ccf74-a9c80002-8188eea8-5c56bd88@192.168.1.193
Max-Forwards: 70
Date: Fri, 02 Nov 2007 21:56:04 GMT
CSeq: 104 REGISTER
User-Agent: Cisco-CP7942G/8.3.0
Contact: <sip:2893@192.168.1.193:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-0000-0000-108ccf74a9c8>";+u.sip!model.ccm.cisco.com="434"
Supported: (null),X-cisco-xsi-6.0.2
Content-Length: 0
Reason: SIP;cause=200;text="cisco-alarm:25 Name=SEP108CCF74A9C8 Load=SIP42.8-3-3SR2S Last=initialized"
Expires: 3600
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AXISHKAuthor Commented:
Fixed the issue, turn off "nat" on the extension and everythings seem works.

Thanks.
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AXISHKAuthor Commented:
tks
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