Using a 8861 phone with an asterisk server or other third party platform

[joelblack1]

We want to use some 8861 phones with our Asterisk server. However, we can't figure out how to configure them with account information. When using an TFTP server to configure, we keep getting an error that it couldn't find the sl-be-sip.jar file.

1. Are we correct that using a TFTP server is the way to configure these phones?

2. What is a sl-be-sip.jar and where can we find it?

[Manish Gogna]

Hi,

Please check the following post

https://supportforums.cisco.com/discussion/12211536/need-help-register-cisco-cp-6941-cp-8941-asterisk-server

Manish

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[Leo Laohoo]

1.  What error messages are you getting?  Settings > Admin > Status > Status Messages? 

2.  What firmware is the phone running? 

3.  DHCP Option 150 present?

4.  Post your SEPmacaddress.cnf.xml file.

5.  With your Asterisk Server, what is under Extension > extension number > Transport?

[joelblack1]

1.  What error messages are you getting?  Settings > Admin > Status > Status Messages? 

“Error updating locale”  (we are also seeing “VPN not configured”, but we do not know if this is actually an error message, or just informational.)

On the TFTP server, the log shows that the phone is attempting to download sl-be-sip.jar, but is unable to do so because it doesn’t exist.  This seems to coincide with the “Error updating locale” message.

2.  What firmware is the phone running? 

Active load: sip88xx.10-3-1-20

3.  DHCP Option 150 present?

Yes

4.  Post your SEPmacaddress.cnf.xml file.

See attached.

5.  With your Asterisk Server, what is under Extension > extension number > Transport?

Still looking for this setting.  Since we are running Thirdlane PBX on top of Asterisk, those directions do not seem to apply.  However, I would point out that it doesn’t appear that the phone is getting far enough in the process that this is an issue as yet

Thanks!

[Leo Laohoo]

 “Error updating locale” 

Don't worry about this error message.  What else?  What error messages do you see?

<transportLayerProtocol>4</transportLayerProtocol> 

Change the value to "2".  

2. What is a sl-be-sip.jar and where can we find it?

Go HERE.

[joelblack1]

Thanks for the information.  The “Error updating locale” was indeed related to the sl-be-sip.jar file, and just yesterday we were able to find the file in the CME locale pack from Cisco.  We had previously tried the process in that link you posted without success.

We also just yesterday found the change for “transportLayerProtocol” on http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+79x1+xml+configuration+files+for+SIP, and this did solve the final connection issue.  We are now connecting and registering to third-party SIP servers!  Thanks for your help.

One remaining issue is that in this process, it has disabled the web GUI access.  Any idea how to re-enable this?

[Leo Laohoo]

One remaining issue is that in this process, it has disabled the web GUI access.  Any idea how to re-enable this?

<webAccess>0</webAccess>
<sshAccess>0</sshAccess>
<sshPort>22</sshPort>

Once you get the configuration "perfected", can you please attach it to this thread (minus username & passwords, of course) so other people will be able to compare their configs with yours?

Please don't forget to rate our useful posts. 

[Mohammed al Baqari]

Hi,

Adding to what Leo mentioned, I suggest to download all phone files from cisco website and upload them to TFTP server to make sure that the phone gets them during registration process

[denisnone]

We connected successfully 8861 to our Asterisk server, it registers, calls and even sounds. But the problem I can't solve is that 8861 doesn't show call history, missed calls or recents. In "Application" menu there is no "Recents" section under #1. The first icon is "Settings". This looks strange. Do I missed something in cnf.xml or it uses recents pulled somehow from the server? And actually no Directory application, my contacts are unavailable. Even local contacts from bluetooth mobile phone. Do anybody know how to handle this?

[Leo Laohoo]

 But the problem I can't solve is that 8861 doesn't show call history, missed calls or recents.

Post the SEPmacaddress.cnf.xml file file and we'll have a look.

[denisnone]

I started all over again with cnf.xml I took from this Jira issue: https://issues.asterisk.org/jira/browse/ASTERISK-13145 Entered my credentials and servers, now "Recents" works. Maybe it was some broken xml tag structure, I preferred not to dig into it. Now whats interesting is that "Directory" still doesn't work. I know it's server-based and to make it work I need to create a webserver dynamic page which renders xml on the fly to feed to the phone. This is more complex task. But still I guess there will be no opportunity to add records using the phone, only to read them. And finally I wanted to run somehow the visual voicemail. Seems like it works like that Directory. I read in Cisco's datasheet that they will make a firmware for Third-Party Call Control support. But who knows when will it be ready? Or did they decide to drop it silently?