AXISHK
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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
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
Please post your sip.conf and your Cisco phone configs.
ASKER
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
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
Still need sip.conf.
ASKER
Here is the file. tks
sip.conf.txt
sip.conf.txt
You're using FreePBX, so I need screen shots of the sip account for the Cisco account. (sip.conf just shows me includes).
ASKER
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).
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).
ASKER
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_st ate: 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
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_st
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
Did you set core debug on as well as sip set debug on in the CLI?
ASKER
Here is the command I run on Asterik
asterisk –vvvcgr
sip set debug
Is that enought ?
Thanks
asterisk –vvvcgr
sip set debug
Is that enought ?
Thanks
Should be. And with debug on, there is still not attempt to connect from the phone?
ASKER
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
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
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.
ASKER
Is there a way to vertify the cnf.xml, without the Cisco Call Manager ...
Tks
Tks
ASKER
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= z9hG4bK6fd f070a;rece ived=192.1 68.1.193
From: <sip:2893@192.168.1.32>;ta g=108ccf74 a9c80005be 8aa1f2-019 f374a
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= z9hG4bK6fd f070a;rece ived=192.1 68.1.193
From: <sip:2893@192.168.1.32>;ta g=108ccf74 a9c80005be 8aa1f2-019 f374a
To: <sip:2893@192.168.1.32>;ta g=as25c372 66
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-a9c80002-8188eea 8-5c56bd88 @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= z9hG4bK6fd f070a
From: <sip:2893@192.168.1.32>;ta g=108ccf74 a9c80005be 8aa1f2-019 f374a
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@192.168.1.193:50 60;transpo rt=udp>;+s ip.instanc e="<urn:uu id:0000000 0-0000-000 0-0000-108 ccf74a9c8> ";+u.sip!m odel.ccm.c isco.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
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=
From: <sip:2893@192.168.1.32>;ta
To: <sip:2893@192.168.1.32>
Call-ID: 108ccf74-a9c80002-8188eea8
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=
From: <sip:2893@192.168.1.32>;ta
To: <sip:2893@192.168.1.32>;ta
Call-ID: 108ccf74-a9c80002-8188eea8
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-a9c80002-8188eea
<--- 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=
From: <sip:2893@192.168.1.32>;ta
To: <sip:2893@192.168.1.32>
Call-ID: 108ccf74-a9c80002-8188eea8
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@192.168.1.193:50
Supported: (null),X-cisco-xsi-6.0.2
Content-Length: 0
Reason: SIP;cause=200;text="cisco-
Expires: 3600
ASKER CERTIFIED SOLUTION
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ASKER
Fixed the issue, turn off "nat" on the extension and everythings seem works.
Thanks.
Thanks.
ASKER
tks