Thanks… I hopefully this help someone
Update this thread
Finally I did register the phones in Asterisk. And I would like to share some import tips I found.
Setting Cisco phone over Asterisk could be really tricky. And here all the information I found from many sources.
This is my environment
- Elastix 2.4.0 and FreePBX
- Cisco phone 7906G
- Tftpd32 (TFTP Server)
- XML Editor file I used editiX XML Editor (useful to find typing mistakes)
- SIP Firmware (I successfully uses SIP11.8-5-4S and SIP11.9-4-2-1S) both work fine. Just with my phone the 9x version runs slow the 8x version is faster but have a small error in Time configuration but still register the hour and date correctly.
I got several error learning about theses phone, so here we go.
- Blank screen or just the cisco logo without the phone start
a. For these I will just assume that you know how to set the TFTP server and DHCP to update the firmware ( Note: if you did the reset and you cannot access the phone setting do not worry working in the same network subnet if you have the TFTP server Up the phone will start an ARC to find a TFTP server)
b. You probably saw in many sites about the reset holding # key during the start and after the blinked light enter the sequence 12345678*0#
i. This code clean the configuration
c. And the other code 3491672850*# delete the firmware
So, in both case you can uses and you can recover the phone if you did use the first one you can update your phone from one firmware to other. But if you did the second one will be a little harder you will need to uses the exact firmware, that’s means you will have to try old the firmware’s until find the right one (Probably SCCP or SIP) so write down the current firmware before do that but after that you can migrate from one firmware to other
- Auth fail during firmware update
a. This error is not completed supported but I found something save my life, if you are trying to update from SCCP to SIP some of the firmware´s has issues with the authentication
i. Try this if you have any firmware SCCP or SIP and get the error just update or downgrade to SCCP firmware SCCP 8.4.4 from this firmware I was able to update to SIP without any authentication error.
ii. Just a little piece of advice. You already know that you need a lot files for update (zip pack you download from cisco page) and some xml files. Try first just using XMLDefault.cnf.xml, it is faster and for the beginning you have more control of the XML file and errors linked to them.
I used this code:
Default>
2002
2001
2000
2427
2428
AsterikIPServer
SIP11.9-3-1SR4-1S
Just need to edit two values
Your Asterisk Server and the firmware you want to update
Note: < loadInformation369 model> this number 369 change with the model but I do not know if there important keep it, just use the write number for your model.
- Registering process in Asterisk.
a. That was the trickiest part, one you have the SEP.cnf.XML, the problem there are several of these model of files on Internet but most of them did not work for me. You can try download any of them but the most important part I will put it here.
i. Nat part
false
ii. Sip Line part (here it is where authentication has place
9
Luis-Manuel
EXTENSION-USER
Luis-Manuel
EXTENSION-USER
<proxy>USECALLMANAGER</proxy>
<port>5060</port>
<autoAnswer>
<autoAnswerEnabled>2</autoAnswerEnabled>
</autoAnswer>
<callWaiting>3</callWaiting>
<authName> EXTENSION-USER </authName>
<authPassword> EXTENSION-PASS </authPassword>
<sharedLine>false</sharedLine>
<messageWaitingLampPolicy>1</messageWaitingLampPolicy>
<messagesNumber>*97</messagesNumber>
<ringSettingIdle>4</ringSettingIdle>
<ringSettingActive>5</ringSettingActive>
<forwardCallInfoDisplay>
<callerName>true</callerName>
<callerNumber>false</callerNumber>
<redirectedNumber>false</redirectedNumber>
<dialedNumber>true</dialedNumber>
</forwardCallInfoDisplay>
</line>
</sipLines>
Note: Just change the part refers to user name and password but DO NOT FORGET THIS LINE
USECALLMANAGER
if you do not put just like this the phone will never register on Asterisk, just put and places exactly like that.
You must complete the SEPxxxxx.cnf.XML file but these were the crucial point for me.
4. In Elastix.
a. Well, for me it was impossible make work the phones over UDP so I enabled the TCP transport, because I am using Elastix these are the files but in other distributions can change
i. sip_general_custom.conf
tcpenable=yes
tcpbindaddr=0.0.0.0
ii. sip_custom_post.conf
EXTENSION
transport=tcp
Note:You must add one entrance for each extension
Lines configuration
NAT value to "never"
Quality to “No”
And That’s it guys.
Good luck