08-17-2015 10:11 AM - edited 03-17-2019 04:01 AM
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?
08-17-2015 10:07 PM
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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08-18-2015 01:11 AM
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?
08-18-2015 10:39 AM
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!
08-21-2015 01:06 AM
“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.
08-21-2015 08:39 AM
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?
08-21-2015 05:05 PM
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?
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08-21-2015 07:35 AM
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
10-10-2015 12:52 PM
10-10-2015 04:37 PM
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.
10-17-2015 02:00 PM
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