Vodia PBX

Well, the service provider obviously uses software that discards the line parameter in the URI. This is not RFC compliant, and that is because of the problem we are facting here - someone (the PBX) registeres several contacts and when the call comes it it needs to know where exactly it should go. The workaround it to send the call to a tel:alias. But this is not very reliable and the PBX needs to perform a table scan on the trunks to locate the right trunk. If you have a lot of trunks that will take some time, and if you have a lot of calls the CPU will start choking. Maybe you can tell the service provider they need to fix this problem. The RFC was released in 2001 and it is time to support this.

What you could do is copy the message to different mailboxes. Check out http://wiki.pbxnsip.com/index.php/Mailbox, there you can see how to copy a message. Though this is not done automatically, someone must manually do the copy.

What phone? Did you go through the Wiki? If there is something missing on the Wiki we should update it, so that it works right from the beginning.

Maybe we should make this a feature. Now that we have a file-system based spool directory and a thread that can take forever to prepare the email it would be possible.

You mean the DTMF done coming from the phone?

ACK. Another fix for 2.1.11.

Well that is the purpose of the affinity... Worst case is that the Linux distribution is not supporting it, then there is not way to make sure that the core does not shift processes around.

On the PBX, it is not possible right now... Any chance to do that on the phone? What phone are you using?

Whow, you are right! Will be fixed in 2.1.11.

Could be... As long as it works we are happy already. The PBX uses a library to access the dongle, and that library is probably a little bit too pragmatic.

No. Better use G.729A - just 2 kbit more, much better quality and much less CPU intensive. BTW the RTP header overhead is already in the 12-24 kbit/s range, compressing to 5.3 kbit/s is completely pointless IMHO.

Just set the standard dial plan in the domain to something restrictive. Then when callers want to place an outbound call, they need to use the "calling-card" account and use their extension and PIN code before they can place a call.

That sounds like a serious routing problem. The really really best way is to have a fixed IP address and transparently route it to the PBX (no NAT), then you will be able to have a stable operation. Next on the preference list would be DMZ NAT with a SIP replacement (tricky, but possible to have it working in a stable way). Stuff like STUN etc is so crazy instable that it is a waste of time and support money. Check out http://wiki.pbxnsip.com/index.php/Office_w...ic_IP_addresses, if you did not do that yet.

Well, the rules are defined in http://wiki.pbxnsip.com/index.php/Outbound_Calls_on_Trunk. In the next version we'll have a new setting called "ANI" for every account, where you can explicity put the caller-ID that you want to this specific account. I think we need to write something up that explains how this whole Caller-ID presentation works in theory and in real ITSP-operator life.

Did you turn password provisioning on (admin/ports/tftp)? By default it is off - because tftp has no security. I am not very happy with the way the provisioning files are "moderated" at the moment by us (pbxnsip). Maybe we should have a public CVS server or a Wiki where everybody can edit those files, so that we can share the best configuration.

Well, at the moment that is a feature. The user-initiated recording is supposed to be for recording only short parts of a conversation, like someone saying a telephone number of address or something that you don't want or can't to scribble down while talking. For these purposes it seems reasonable to put that into the (private) mailbox of the user, and possibly forward it by email. What is the use case for recording to the directory?

That is a feature! Imagine you have 1000000000 matches.... The first 9 just fill the display, then you need to enter more digits (1-9) to refine the search.

A hunt group is definitively a "stress test" to the phone. It receives a large number of messages in a short period of time. Maybe what you can do it log only SIP packets to/from the IP address of the phone and write the log to a file. Then we can find out if the messages are going out as they should.

Anyway... Just edit the template above and then you should be able to provision the 300 as you like.

We addressed that in the next version 3.0. Short-term workaround is to put your own snom_300_fkeys.xml in the html directory: <?xml version="1.0" encoding="utf-8"?> <functionKeys>{if_button dnd none} <fkey idx="dnd" context="active" perm="RW">{enum_button dnd button+dnd private=line}</fkey>{fi_button dnd none}{if_button 1 none} <fkey idx="0" context="active" perm="RW">{enum_button 1 button+1 private=line}</fkey>{fi_button 1 none}{if_button 2 none} <fkey idx="1" context="active" perm="RW">{enum_button 2 button+2 private=line}</fkey>{fi_button 2 none} <fkey idx="2" context="active" perm="RW">{parameter key3}</fkey> <fkey idx="3" context="active" perm="RW">{parameter key4}</fkey> <fkey idx="4" context="active" perm="RW">{parameter key5}</fkey> <fkey idx="5" context="active" perm="RW">{parameter key6}</fkey> </functionKeys> Do you see the parameters key3..6 in the PnP parameters in the web interface (admin/pnp settings)?

For what phone type?

The PBX displays the first 9 entries that match. Because there could be thousands, there is a limit. The point is here that you need to "drill down" by entering more digits. For example, if you search "Valerio", you would enter <Adressbook>, 8 (TUV), 2 (ABC), 5 (JKL) than then usually the list should be short enough that you can use the arrow keys to select the right entry.

We were thinking about an option so that after the PBX starts up it changes the account. That means after settings the priority and everything, the PBX would change the account to an user account, reducing the risk of malicious things and then start operation. Do you think that would help?

If they belong into different areas, it might be better to post them seperately. Makes it later easier to find the answers. Well, probably the easiest way is to check out the netstat command and find out what ports the PBX opened. The only dynamic ports (apart from DNS, which you don't have to take care of) is the RTP ports. A good read is http://wiki.pbxnsip.com/index.php/Office_w...ic_IP_addresses. Sorry I don't get the question here... You want to trunk into an existing (analog) PBX? Sipura sold their stuff a couple of years ago to Linksys, maybe http://wiki.pbxnsip.com/index.php/Linksys can help here.

Well, that does not help because the file has no reference to the MAC. Better do the following: Put the following file into the html directory and name it "snom_3xx.xml": <?xml version="1.0" encoding="utf-8"?> <setting-files> <file url="{https-url}/snom_3xx_phone-{mac}.xml?model={attribute model}" /> <file url="{https-url}/snom_3xx_fkeys-{mac}.xml" /> <file url="{http-url}/snom_general.xml" /> <file url="{http-url snom}/web_lang.xml" /> <file url="{http-url snom}/gui_lang.xml" /> </setting-files> Then you can put another file into the html directory and call it "snom_general.xml", where you can put whatever you want to put in all of the 320/360/370 phones.