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Here are the steps I took to get my SPA941 and SPA942 phones auto-provisioned. All phones are running the 6.1.3(a) firmware. My PBX is running on a Windows 2008 64bit server. I first provisioned these phones under pbxnsip version 2.1.10. We are now running version 3.2. 1.) Posted the attached files in the PBX install dir \ html folder. You may want to merge the spa_phone section of the pnp.xml file with your existing pnp.xml file to not overwrite the settings for other vendors you may have already customized in that file. You will probably need to change the dial plan in the <!-- Dial Plan --> section of the spa_phone.xml file to suit your needs. 2.) Added the option 66 in the DHCP pool specifying the IP of the PBX server. 3.) Added the MAC address of the phone sets to the extensions - the Account \ Registration \ Bind to MAC Address field in the PBX admin for the domain. 4.) Make sure other customizations to the phone won't conflict - reset to factory defaults then continue after the phone resets. 5.) Browsed to the web GUI of the phone. Selected Admin Login and advanced mode. 6.) In the web GUI of the phone, on the Provisioning tab, in the Profile Rule field I pasted: http://<pbx server ip>/provisioning/spa_phone_$MA.cfg 7.) After saving the settings the phone resets. After a bit it should pick up it's configs from the PBX then reset again. You should now see the username and extension registered on the phone and in various fields in the phone's web GUI. There is much that can be customized and set via the template config files - ours is pretty default. That should do it. -b spa_phone_files.zip

I just upgraded to v3.2.0.3144 and while it allows the addition of a port to a URL it does not allow additional paths in the URL like the global CDR SOAP field does. The URL to our SOAP service is http://localhost:81/CDRWeb/Default.aspx. I had to edit the XML config file for the agent group directly and reset the pbxnsip service so it would keep the change. I can now not change anything else on the Agent Group Edit page or it will clear those settings. Those fields should allow for more than just the domain.com:port URLs. -b

Anyone?

Would someone mind posting their provisioning and dialplan files for snom phones? None of the installs I have done (Windows installs) have contained anything in the html directory. -b

Can someone post the snom provisioning files that belong in the html folder? Am I missing something about where the provisioning files for various phone types are posted? They don't install with the pbxnsip software install, they aren't posted in the wiki - and I can only find some files attached in various forum topics. It seems to me the default provisioning files - xml, txt files, etc - should be posted along with the info about each phone vendor in the wiki. Sure would make provisioning new phones easier when we buy one from a new manufacturer instead of having to hunt around and ask around piecing the files together. -b

It seems to be more of a reset of the phone - a really quick reboot. They seem to be random and not based on any noticeable events. I even did a factory reset on some to see if it would help - but no change.

UPDATE: It looks now like the MWI is cleared because the phones are resetting. Ever since the update to 3.0 the phones have been having frequent resets. I have not noticed it on my phoneset - although other users see as many as 25 resets a day. I tried the latest firmware on the phones but the resets keep happening. Could there be some change from 2.x to 3.0 that requires a tweak in settings on the SPA-942 phones? -b

All we have are SPA941 and SPA942 phones so I don't have other MFGs phones to compare to. Under 2.x though the MWI was working fine. This started happening only after the switch to 3.0. I will add the IP of the remote syslog server to see if the phone logs anything.

After the upgrade to version 3.0.1.3023 we have been noticing our Linksys SPA942 phones have started having their MWI light randomly cleared somehow on their own without the user first checking the voicemail. With out the indicator on users are not notcing the voice mails they have because they cant see a reminder on their phone. We tried resetting the phones and the PBX server but it has not resolved the issue.

I upgraded to v3.0.1.2023 (win32). We run the interface secure using the self signed SSL certificate. When navigating around the account listing pages there appears to be some element on the listing page - possibly and image - that is referencing non-SSL space. This results in the browser message: "This page contains both secure and nonsecure items. Do you want to display the nonsecure items?" Yes/No. Has anyone run into this same message? I imagine it would be something that pbxnsip has to fix in the code - looks like a reference to an image that has the http:// hardcoded so it will not load secure in SSL.

I am looking to get an actual copy of those files - I can't find them anywhere and they were not installed with the pbx software.

Could someone post the basic Polycom PNP files? Namely I am looking for the files that the pnp.xml file calls for: polycom_master.xml polycom_phone.xml

We had an issue recently with our SIP trunk provider where some "routing" on their end caused us to lose all inbound/outbound call abilities. I don't know if the trunk was connected or not - but our DIDs did not make it to the PBX from the outside world and we could not call out. I need to find a way to monitor for this. Has anyone setup monitoring for various SIP trunk issues like this - trunk connectivity and DID routing? I would rather be notified by our monitoring system or some other kind of script than have users tell me that their calls dropped or they cannot call out or in. In addition our monitoring system could monitor uptime so I have ammo when pursuing SLA payouts.

Yeah, or a way to turn off that prompt altogether would be nice as well. We have a custom recording listing the buttons to push. Then right after than the auto voice comes on telling you to push a button. We really don't need that. Do we have to replace that global wav file with a blank wav file or something?

What about a Gadget for use on the Vista sidebar?

It would be helpful if there was an audio file manager page within the administration tool to manage all of the audio files for a domain. This way all files could be uploaded to a repository then selected on MoH, IVR nodes and Attendants rather than uploading a file each time. In addition it would help if the name of the current file was listed in the field rather than it being blank and waiting for a new file to be uploaded.

If I add the MAC address of a SIP device (Linksys SPA942) to the registration of multiple extensions the phone auto provisions with the multiple extensions correctly. However, how can I specify the order that the extensions are provisioned on the phone. It looks like they get provisioned in numeric order but I need to specify which extension gets registered to Ext 1 on the phone so that it is the default when picking up the hand set and displaying the user name on the phone display.

That did it. The phone is registered and usuable! I had to add the MAC address to the Bind MAC field for the extension I want the phone to register as. IF I put in * then the phone wouldn't register and I couldn't figure out how to associate the phone with the extension. So provisioning additional settings like dial plans, address books, etc... is done through the spa_phone.xml file correct?

1.) I posted the spa_phone file attached to the last post as spa_phone.xml 2.) I added this to pnp.xml and restarted pbxnsip: <file name="spa_phone.xml" encoding="xml"> <pattern>!spa_phone_([0-9a-f]{12})\.cfg!\1!</pattern> <vendor>Linksys SPAxxx</vendor> </file> 3.) In the profile field of the phone I pasted: http://192.168.32.30/spa_phone_$MA.cfg 4.) DHCP serves up option 66 as 192.168.32.30 5.) I tried entering both the MAC and * in the Bind Mac address field for an extension. It is almost as though the phone isn't trying to register with these settings. If I set the proxy address and specify a userid/password it regiaters no problem.

If I just give the phone the IP of the pbx: Profile Rule: 192.168.32.30 Then the phone doesn't register. In fact there are no log entries so it looks like it doesn't even try.

Doesn't look like the SPA942 has tftp options via the interface. Is it just a matter of specifying the tftp port after the IP of the pbx? And by setting passwords - do you mean the extension passwords? There is no place to specify password in Admin/Ports/TFTP

I added the IP address of the pbxnsip server as an alias to the domain and that allowed the SPA942 phone to register. So I guess what I really want to do is have the domain be named something like pbx.domain.com and have a real DNS record pointing to it so that the phones register to that rather than an IP address. Thanks for the idea! It is kind of weird that when a domain (and extensions I noticed) has an alias defined it creates a second entry in the list of domains. Kind of confusing actually.

Has anyone setup pbxnsip to work with Free World Dialup? The wiki has a doc to set it up (http://wiki.pbxnsip.com/index.php/Free_World_Dialup) but it doesn't work using the sample settings. The status always returns "408 Request Timeout (Registration failed, retry after 60 seconds)" in the list of Trunks. The logfile shows: 0] 2008/06/04 09:58:31: SIP Tx udp:65.39.205.114:5082: REGISTER sip: SIP/2.0 Via: SIP/2.0/UDP 192.168.32.30:5060;branch=z9hG4bK-a67737b760b064f951a1305d9940bb11;rport From: <sip:908397@>;tag=29846 To: <sip:908397@> Call-ID: 7s9f3wqr@pbx CSeq: 8386 REGISTER Max-Forwards: 70 Contact: <sip:908397@192.168.32.30:5060;transport=udp;line=c4ca4238>;+sip.instance="<urn:uuid:e1803b84-d405-457c-aa6b-cb39b09b5986>" User-Agent: pbxnsip-PBX/2.1.7.2461 Expires: 3600 Content-Length: 0

Does anyone have the lines necessary to add to the provisioning script to auto provision the Linksys SPA942 phones? This is a must for us before we settle on a PBX.

My Linksys SPA942 phones won't register. The log returns 404 NOT FOUND. SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.32.210:50000;branch=z9hG4bK-927479b5;rport=50000 From: <sip:4014@192.168.32.30>;tag=6b0cf7b4e82b8244o0 To: <sip:4014@192.168.32.30>;tag=1fed52e972 Call-ID: 75fb3539-f570584d@192.168.32.210 CSeq: 28749 REGISTER Content-Length: 0 From this post: http://forum.pbxnsip.com/index.php?showtopic=660, where does the domain get set in the SPA942 phone? What else would 404 NOT Found mean? -b