chrispopp

Is there a way for us to add the IP manually so that some of these phones register?

Added this to the pnp_grandstream.xml <file name="grandstream-gxp2130.xml" encoding="xml"> <pattern mac="3">cfgc074ad######.xml</pattern> <user-agent>Grandstream.*GXP21(30|35|40|60|70)</user-agent> <prefix>true</prefix> <vendor>Grandstream</vendor> </file> Still does not work. [9] 10:25:25.499 PROV: Provision for User-Agent Grandstream Model HW GXP2140 SW 1.0.9.135 DevId c074adxxxxxx [8] 10:25:25.499 PROV: HTTP: Received request for file cfg.xml from Host: localpbx.com User-Agent: Grandstream Model HW GXP2140 SW 1.0.9.135 DevId c074adxxxxxx Accept: */* [8] 10:25:25.499 PROV: PnP: Extracted MAC for file cfg.xml [7] 10:25:25.499 PROV: HTTP: Error generating or finding grandstream-reject2, Send back 404 Not Found Any ideas?

Grandstream now has the C074AD mac range on their new phone. Just an FYI

Was wondering if you had any intention to secure the system with a Voip blacklist. http://www.voipbl.org/ I think it would be a good feature to defend against unwanted attacks. We use it on some Asterisk and Freeswitch servers and it almost cut our hacking attempts to 0. Just an idea.

Oh I see what you mean. Don't think this will work. So there's no other way?

How can I limit an extension to only be able to dial 2 specific extensions? Say you have a system with 10 extensions, and you program a phone onto one extension. How can I limit that only two of these extensions can be dialled. Dialing any of the other 7 would make the call fail. I tried using a dial plan, by only setting the two extensions and using NOT ALLOWED * for everything else. Did not work. My dial plan is: 99;+;;150;;;false 100;+;;155;;;false 110;-;;*;;;false

Has the phonebridge been integrated to make and receive calls, or only for logging?

Thanks for getting back. I did enable that feature because we need to be able to retrieve the passwords in order to automate a provisioning softphone we created. But I didn't think that the same feature would enable it on the web interface... There should still be a way to disable showing passwords in clear text on the web interface.

Jeez... Just testing more settings. Seems that all passwords are shown. Below is provisioning details.

It's also shown under the extension edit. Right in the password field...

I noticed that in the latest version the SIP/WEB passwords are shown in clear text if someone changes them under the Registration tab. Can this be disabled?

Thank you. Going to try

That puts the First + Last. Isn't there a way to separate them?

When using paramenters to provission, there seems to be {account}: The account name of the current user. But what if you just want the first name or last name? Are there any parameters for those?

I did this. http://pbxip/reg_status.htm?save=save&max_ring_duration=3600 , double checked that the pbx.xml got updated. Then restarted the service. But it still does not go past 120 seconds. Ringing a hunt group. First stage is an extension with 120 seconds, but it disconnects before going to the second stage. If I put 60 seconds per stage. It goes to both stages then disconnects.

Can the ring time be increased past the maximum? The option is currently set to 180 seconds. This is the highest number. Can I somehow increase it to 3600? This is an older version 52 I was thinking to modify it directly in pbx.xml

Used Vernon's recommendation and it worked perfectly.

No that won't work either. If the call is forwarded out, then it considers it released, and will accept new calls.

Tried it and it doesn't work. Say you put 2 in the total calls/total calls on trunk. One call is active, the next one gets disconnected.

How can I call forward to an extension if calls on the trunk are busy. What I'm trying to test is: To only allow one active call per system. For example, someone calls in reaches an auto attendant, the caller presses the key and then the call is forwarded to an external number. The second call that comes in, to go directly to voicemail.

Fair statement. Thanks!

Your sentence seems to say that SOAP is very old... Then why do we use it? The content is always "SOAP" there is no JSON. It is a very old setting, for those who still remember what SOAP is. The information I gathered was from the documentation from here: https://doc.vodia.com/acdreport

The documentation says the following: The PBX can send updates to an external server using HTTP or HTTPS. The setting for this is at the bottom of the ACD page, called "Queue status URL". The setting may take one of the following values (only one URL at a time is possible): json/jsons: When using the json scheme, the PBX sends a JSON object to the address provided behind the scheme. For example, "json://acd.domain.com/acd123" would send the content to "http://acd.domain.com/acd123". When using jsons, the system uses https instead of http. soap/soaps: When using the soap scheme, the PBX sends a XML object to the address provided behind the scheme. For example, "soap://acd.domain.com/acd123" would send the content to "http://acd.domain.com/acd123". When using soaps, the system uses https instead of http. But the validation only accepts http://xyz... json:// also doesn't work

It was always like this... It recently changed. Tested on old versions and it works working before. That's what the "Record calls from hunt group to extension" setting is for. Otherwise it's a useless option, since it doesn't do anything.

"So I created a hunt group, and the phone number goes directly to it. I have also set under general settings that all calls should be recorded. upload pic But incoming calls do not get recorded if they go directly to a hunt group. Turning on the feature "Record incoming calls to hunt group" works. It makes much more sense that the General Settings should override this feature. As it used to...