jag

This sounds like you have reached the max number of redirects allowed. This is a feature to stop run-away calls from looping and looping. To fix this.... Stop the service Go to the Working Directory of the PBX Edit pbx.xml Look for the XML setting max_loop Change it to the required value Start the service. That should do it.

Would a license assigned to a dongle work on MAC OS?

A few fax numbers, when dialed, will occasional play a recording, “please enter your destination number, terminate the number with a # sign”. The recording is definitely coming from his pbx. I verified this as there were no active trunk calls while this was happening. It seems as though the call is being redirected to a calling card account, yet none exist.

We are having trouble with our fax, so we switched trace on and we get 1000's of these messages in the trace. At first we thought it was the M3 phone, so we stopped that and these message still appeared. What do they mean? [9] 2008/08/18 16:40:55: DTMF: Power: 0 0 0 0 0 0 0 0 0 [9] 2008/08/18 16:40:55: DTMF: Power: 0 0 0 2 0 2 0 0 0 [9] 2008/08/18 16:40:55: DTMF: Power: 0 0 0 2 0 0 0 0 0 [9] 2008/08/18 16:40:55: Last message repeated 2 times [9] 2008/08/18 16:40:55: DTMF: Power: 0 0 0 0 0 0 0 1 1 [9] 2008/08/18 16:40:55: DTMF: Power: 0 0 0 0 0 0 1 0 0 [9] 2008/08/18 16:40:55: DTMF: Power: 0 0 0 0 0 0 0 0 0 [9] 2008/08/18 16:40:55: Last message repeated 3 times [9] 2008/08/18 16:40:55: DTMF: Power: 2 0 0 0 2 2 0 1 1 [9] 2008/08/18 16:40:55: DTMF: Power: 0 0 0 0 2 0 0 0 0 [9] 2008/08/18 16:40:55: DTMF: Power: 2 0 0 2 0 0 0 1 0 [9] 2008/08/18 16:40:55: DTMF: Power: 0 0 0 0 0 0 0 0 1 [9] 2008/08/18 16:40:55: DTMF: Power: 0 0 2 0 0 0 0 0 0 [9] 2008/08/18 16:40:55: DTMF: Power: 0 0 0 0 0 0 0 0 0 [9] 2008/08/18 16:40:55: DTMF: Power: 0 0 2 0 0 2 0 0 0 [9] 2008/08/18 16:40:55: DTMF: Power: 0 0 0 0 0 0 0 0 1 [9] 2008/08/18 16:40:55: DTMF: Power: 0 0 0 0 0 0 1 1 0 [9] 2008/08/18 16:40:55: DTMF: Power: 0 0 0 0 0 0 1 0 1 [9] 2008/08/18 16:40:55: DTMF: Power: 2 0 0 0 0 0 0 1 0 [9] 2008/08/18 16:40:55: DTMF: Power: 0 0 0 0 0 0 1 0 0 [9] 2008/08/18 16:40:55: DTMF: Power: 0 2 0 0 0 0 0 0 0 [9] 2008/08/18 16:40:55: DTMF: Power: 2 0 0 0 0 0 1 0 0 [9] 2008/08/18 16:40:55: DTMF: Power: 0 2 0 2 0 0 0 0 1

Do the Message Summary lines under the registration tab of an extension show as Registrations for SNMP Reporting? message-summary 901 sip:901@192.168.0.2:2051;line=1wvo15xt 59

Normally the User = the extension number of the account which was auto provisioned and the password is the SIP password from the account. If you don't hand out the SIP password then they will never enter the phone (unless they crack it)

I installed IPTABLES and set it up to allow only ports 22(ssh) and 5060/5061 on both tcp/udp, 8085/8086 on tcp and the entire RTP range in udp. Do you see any problems with this and pbxnsip? Has anyone else used IPTABLES and pbxnsip V2?

We tried that and someone made their network 'Static'... when the phone booted up it, it failed to get DHCP and gave them access to the phone. It allowed them to configure the account, registrar, password and outbound proxy. Is there anyway we can block the phone from being changed except from pbxnsip? OR Is there any secret way of only allowing the user to get to the IP Address of the configuration e.g. DCHP or Static IP Address.

We are trying to block out the user from changing any of the settings in any Snom phone with the exception of the DHCP or Static IP address. Basically, when a phone is shipped to a user, it needs to be completly locked down and the user is not allow access to the phone. Can this be done via Auto Provisioning?

How about http://uk3.php.net/curl

Auto Provisioning Snom 320/360 phone allows me to view the Address book for the domain, we scroll down and view x records and then page through them. Can the same be done for a Snom 300 phone considering it only has 1 line to display? If yes, is that automatically done through Auto Provisioning?

You need to order G729 licenses with your reseller, they are $12 each per line and need enough to cover all calls on the pbx.

What does this mean? 5] 2008/08/05 16:55:56: Tuning to new SSRC

The additions/changes will be done in a few weeks when the talent comes back from holiday. Many thanks to Ramond from Adinets in the Netherlands for translating the text

6. Don't waste you money on a multiple core CPU. pbxnsip will only use 1 core at a time. 7. Deploy in a good datacentre, bad connectivity or up time will kill your business in a day

You should be able to manually edit the pbx.xml file in the working directory and change the ports from 80 and 443. That should work.

The messages used for the AA are global messages. The PBX automatically askes for the Extension after the user doesn't input something (normally a few seconds).

For your first request of different ring tones and possibly an LCD message you need to use the Hunt group functionality. Create 2 hunt groups Name each one Set the ring tone alert Set the Display From Header to be Group with CLI Put each extension in the hunt group. Now set the AA to take them to the required hunt group So if the user presses 1 for Sales the LCD should display "Sales (1234567890)" Recording Voice Prompts There are two ways of recording a personalized AA message. Dial *98xxx where xxx is your AA number OR Record a wave file and up load it in the AA screen through the upload wave file field and press save, now dial the AA and check the message. Jonathan Greenwood pbxnsip Europe

Try setting a Domain alias of the IP address of the trunk.. INVITE sip:3474244022@64.113.246.222:5060 SIP/2.0 Create an domain ALIAS of 64.113.246.222 for the domain, that may solve the problem.

I'm trying to understand the difference between these following fields; NAT Refresh time Minimum Registration Time: Is either connected? What would happen if you had 30 (NAT) and 120 (MRT)? I need to have an understanding of this. The wiki states that the MRT is the time that the phone waits before it try's and reconnects after a disconnect. Does it keep it separate to the NATf refresh?

If I set the lines field on an extension to 2 what will the PBX do with regards to calls? Will it use this field to restrict the total number of calls = 2? If the third call took place or came in from a gatewy, what would happen to that call? Thanks jag

I see this in our System Status page: Emails: Successful sent: 71 Unsuccessful attempts: 9 (Warning: Last email could not be sent!) Any chance how we can find out which email it is or more information about the failed email. I looked in the Spool Directory under the main PBX directory and this is empty. Thanks

You need to declare each extension as a tel:alias. So you would create in the alias column of extension 200 the following "tel:200". This is used as a Global Scope alias and therefore allows you to cross-domains

When you have a the option to ask for a participants name, can this be done without a PIN Code?

[5] 2008/01/03 16:23:09: load: Index domain not set What does this mean?