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How to configure a Cisco 7911G phone for SIP with Elastix PBX
Hi all,
We bought some Cisco 7911G phones for some new people in the office and we have to configure them to work with our PBX - Elastix. Looked around everywhere including EE and haven't found a complete guide on how to upgrade the phones, a great guide would be something along the lines of what TFTP server to use and config guidelines. We have Cisco support and can download the files necessary but there are so many versions.
Also, something else which is is worthy of noting, we have tried some of the solutions found and one phone won't boot up and the other phone keeps trying to upgrade (it does so in an infinite loop)
Any ideas on how to fix that?
Your support as always is greatly appreciated.
D.
We bought some Cisco 7911G phones for some new people in the office and we have to configure them to work with our PBX - Elastix. Looked around everywhere including EE and haven't found a complete guide on how to upgrade the phones, a great guide would be something along the lines of what TFTP server to use and config guidelines. We have Cisco support and can download the files necessary but there are so many versions.
Also, something else which is is worthy of noting, we have tried some of the solutions found and one phone won't boot up and the other phone keeps trying to upgrade (it does so in an infinite loop)
Any ideas on how to fix that?
Your support as always is greatly appreciated.
D.
If you have problems translating a specific step to spanish let me know of it.
ASKER
Hi, thanx for the info. I managed to get the phone to upload the images and change the network config. Through the instructions - which I had to google translate - I came to the part where I have to create the SEPXXXXXX.cnf file (where X is the mac address) and from there on I am lost - there are a lot of references to things that need to be replaced - where is t says:
The file in which we focus is SEPX.cnf.xml, which introduce the data needed to get the phone as "unprovision" and have them register on our Asterisk server. Looks like this:
<device>
<deviceProtocol> SIP </ deviceProtocol>
<sshUserId> root </ sshUserId>
after that there are references (in red) to all the points needing changed - however I am not sure what everything is referring to. Is there a way to put things in a "for example" format?
Also, the 2 phones previously tampered with are not displaying anything now, they will accept the reset instruction but will not do anything afterwards :(
Any help greatly appreciated.
D.
The file in which we focus is SEPX.cnf.xml, which introduce the data needed to get the phone as "unprovision" and have them register on our Asterisk server. Looks like this:
<device>
<deviceProtocol> SIP </ deviceProtocol>
<sshUserId> root </ sshUserId>
after that there are references (in red) to all the points needing changed - however I am not sure what everything is referring to. Is there a way to put things in a "for example" format?
Also, the 2 phones previously tampered with are not displaying anything now, they will accept the reset instruction but will not do anything afterwards :(
Any help greatly appreciated.
D.
The paragraph just before the xml text states that the file we are focused on is the SEPX.cnf.xml, in order to get the phone out of the "unprovisioned" state. The fields in red that need change are (will have to insert them in their order):
<name>NTP server Asterisk IP</name>
<processNodeName>Asterisk IP </processNodeName>
<phoneLabel>String to be shown on top right corner (no spaces or weird characters)</phoneLabel>
<proxy>Asterisk IP</proxy>
<name>Asterisk's SIP account user name</name>
<displayName>Name that will show up on phones calling this phone</displayName>
<authName>Asterisk authentication user name</authName>
<authPassword>password!</a
<messagesNumber>voicemail number</messagesNumber>
<contact>Asterisk's SIP account user name</contact>
<dialTemplate>dialplan.xml
<pcPort>0 to enable, 1 disable (I think its backwards)</pcPort>
<webAccess>0 to enable, 1 disable</webAccess>
Now, for the bricked phones, please start getting a packet capture from the switchport, you may use cloudshark.org for us to analyze them.
<name>NTP server Asterisk IP</name>
<processNodeName>Asterisk IP </processNodeName>
<phoneLabel>String to be shown on top right corner (no spaces or weird characters)</phoneLabel>
<proxy>Asterisk IP</proxy>
<name>Asterisk's SIP account user name</name>
<displayName>Name that will show up on phones calling this phone</displayName>
<authName>Asterisk authentication user name</authName>
<authPassword>password!</a
<messagesNumber>voicemail number</messagesNumber>
<contact>Asterisk's SIP account user name</contact>
<dialTemplate>dialplan.xml
<pcPort>0 to enable, 1 disable (I think its backwards)</pcPort>
<webAccess>0 to enable, 1 disable</webAccess>
Now, for the bricked phones, please start getting a packet capture from the switchport, you may use cloudshark.org for us to analyze them.
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ASKER
Had to get the phones in the system very urgently so got a local Asterisk company to help.
For the hanging phone, please collect a packet capture from the switchport (this model when used with callmanager has the Span to PC port option disabled by default, don't know the status of the phones config at this point). The capture will tell us what to do for the phone.