Tim

I was going to ask if Apache HTTPD and the pbxctrl are bound to different ports, but it seems that, at least in the build that I am running, it doesn't the prevent pbxctrl process from starting. It does throw a "FATAL" error, so I would have expected it to exit, not finish starting. Do you have logging to a file enabled and does it show any errors? I am seeing the errors below when I start pbxctrl after running the following commands. We are currently running build pbxctrl-centos5-3.0.1.3014. # nc -l 80 & # nc -l 443 & # /etc/init.d/pbxctrl start From my pbx.log: [0] 20080915060138: Could not bind socket to port 80 on IP 0.0.0.0 [0] 20080915060138: FATAL: Could not open TCP port 80 for HTTP/HTTPS [0] 20080915060138: Could not bind socket to port 80 on IP [::] [0] 20080915060138: FATAL: Could not open TCP port 80 for HTTP/HTTPS [0] 20080915060138: Could not bind socket to port 443 on IP 0.0.0.0 [0] 20080915060138: FATAL: Could not open TCP port 443 for HTTP/HTTPS Tim

Adding '!E!156!' to the DTMF Match List seems to have fixed the problem. Thanks for the help.

If I add '!^E$!156!' would that match and loop back to extension 156 or do I need to have the DTMF input in there to make it trigger?

Not convenient, but if one would happen to have said "specially designed cable" it would be an option for a recovery in an emergency.

What about the DTMF not being recognized after the message loops? Am I missing something in my configuration? Tim

Is there anything that can be done for the White boxes? Is there a way to boot into Single user mode with the serial console, to reset the password, perhaps?

I have setup an IVR node, which now that I have figured out how they work, I think is a great feature. It is the IVR node that is giving me problems with the DTMF press not being recognized after the message completes and starts replaying. It would also be cool if the IVR node had an option to wait n seconds before repeating the message, but we can just add a few seconds of silence before we end the recording for now.

We have the admin password for the web interface, it is just the root password that seems to have been lost. This works for both the older white boxes as well as the new black ones? Tim

Is it possible to recover the root password on the CS410? Most devices like this I have worked with have always had some way for tech support to recover after a lost password, have you guys at pbxnsip built something into the CS410? Tim Donahue

UB == Universal Binary That would allow the same binary to run on both the Intel and PPC architectures. I would say that might not be a good idea. You would likely run into some performance issues seeing as this is a realtime application and we have seen issues with video/voice processing in universal binaries in the past. If you want to support the older PPC based systems for something as sensitive as voice handling, a native build is the only way to go. Tim

Normally I would point you to the mailbox access instructions, but they seem to be a little out of date. Would it be possible for someone at pbxnsip to get the instructions on the Wiki updated, it was rather confusing to hear one thing and see the other on the wiki. To copy or move a message to another voicemail, press 6, then you can move, copy, or annotate and forward the message. Tim

We have a client that has several informational type of recordings in their AA. Originally we had these messages setup as voicemail boxes with a max of 0 messages. This worked fine until the client decided that they wanted a zero out option in them, and they didn't like the canned "Press zero for more assistance" recording at the end of their message. All of them have a simple zero out option to go back up to the previous branch of the AA. I have the routing working correctly, but the problem I am having is the IVR node stops responding to input after the first loop of the message. If the caller listens to the entire message, and the message starts repeating, the system accepts the DTMF input as shown by the log message but the call never routes to its destination. [6] 2008/09/09 18:39:53: Received DTMF 0 We are currently running the pbxctrl-centos5-3.0.1.3014 build. Thanks in advance for your help. Tim

I have the outbound proxy setup in the trunk (in fact, the trunk type is outbound proxy). I also just setup DNS SRV records for a testing domain on that system and I am still unable to get calls to route inbound without the trunk being set to "Global". We use the domain name in the URI of the SIP INVITE coming from our SBC. Anything else I can try? -- Tim

We are having a problem with inbound call routing. I setup individual proxies on our SBC for each of the domains on the PBX, all on different IP addresses, but that calls are still not being associated with the correct domain. All calls are giving me a message "Received incoming call without trunk information and user has not been found". The only thing that helped was putting all the trunks that have inbound calls coming to them into "Global" mode. We are currently running pbxctrl-centos5-3.0.0.2991. Do you know if there is a new build that fixes this trunk association problem?

Seems to be working now. Thanks for your help. Tim

Where does this go? It is loaded in the "Reload Configuration Files" section of the Configuration page, correct? Tim

I am trying to get a 7960 to be provisioned by the PnP functionality of the pbxnsip. I keep getting this error message over and over, "TFTP: File SIP000ED71098C1.cnf not found." That MAC address is in the Bind To field for the extension. Any clues as to what is happening? Here is the full log. [6] 2008/05/30 16:29:41: TFTP: File CTLSEP000ED71098C1.tlv not found [6] 2008/05/30 16:29:41: TFTP: File SEP000ED71098C1.cnf.xml not found [6] 2008/05/30 16:29:41: TFTP: File SIP000ED71098C1.cnf not found [6] 2008/05/30 16:29:41: TFTP: File MGC000ED71098C1.cnf not found [6] 2008/05/30 16:29:41: TFTP: File XMLDefault.cnf.xml not found [6] 2008/05/30 16:29:41: TFTP: Request SIPDefault.cnf [7] 2008/05/30 16:29:41: UDP: Opening socket [7] 2008/05/30 16:29:41: Open TFTP port 2052 [8] 2008/05/30 16:29:41: TFTP: Transfer finished successfully [7] 2008/05/30 16:29:41: UDP: Opening socket [7] 2008/05/30 16:29:41: Open TFTP port 2052 [6] 2008/05/30 16:29:41: TFTP: Request P0S3-08-6-00.loads [8] 2008/05/30 16:29:41: TFTP: Transfer finished successfully [6] 2008/05/30 16:29:58: TFTP: Request SIPDefault.cnf [7] 2008/05/30 16:29:58: UDP: Opening socket [7] 2008/05/30 16:29:58: Open TFTP port 2052 [8] 2008/05/30 16:29:58: TFTP: Transfer finished successfully [6] 2008/05/30 16:29:58: TFTP: File SIP000ED71098C1.cnf not found [6] 2008/05/30 16:30:44: TFTP: File CTLSEP000ED71098C1.tlv not found [6] 2008/05/30 16:30:44: TFTP: File SEP000ED71098C1.cnf.xml not found [6] 2008/05/30 16:30:44: TFTP: File SIP000ED71098C1.cnf not found [6] 2008/05/30 16:30:44: TFTP: File MGC000ED71098C1.cnf not found [6] 2008/05/30 16:30:44: TFTP: File XMLDefault.cnf.xml not found [6] 2008/05/30 16:30:44: TFTP: Request SIPDefault.cnf [7] 2008/05/30 16:30:44: UDP: Opening socket [7] 2008/05/30 16:30:44: Open TFTP port 2052 [8] 2008/05/30 16:30:44: TFTP: Transfer finished successfully [7] 2008/05/30 16:30:44: UDP: Opening socket [7] 2008/05/30 16:30:44: Open TFTP port 2052 [6] 2008/05/30 16:30:44: TFTP: Request P0S3-08-6-00.loads [8] 2008/05/30 16:30:44: TFTP: Transfer finished successfully [6] 2008/05/30 16:31:01: TFTP: Request SIPDefault.cnf [7] 2008/05/30 16:31:01: UDP: Opening socket [7] 2008/05/30 16:31:01: Open TFTP port 2052 [8] 2008/05/30 16:31:01: TFTP: Transfer finished successfully [6] 2008/05/30 16:31:01: TFTP: File SIP000ED71098C1.cnf not found Thanks for the help. Tim

We are using the PnP provisioning where possible for our clients. We have run into one problem so far with this. How can you configure parameters such as auto_dial through the PnP provisioning system? Normally I would use a Perl script to accomplish this through the web interface of the phones, but with the interfaces password protected, it is difficult to do this in any automated way. Tim

This is for one of the (many) test cases for the InterOp testing. I will contact the engineer and find out if the session timers are necessary to successfully complete this test case. Thanks for your help. Tim

Is it possible for the moderator to add a participant to a conference call, either in Ad-hoc or in Scheduled mode? Tim Donahue

Does pbxnsip have the ability to suport SIP Session Timers? We are currently doing an InterOp test with Level3 and one of the tests they are requiring is to have SIP Session Timers enabled for calls longer than 15 minutes. Tim Donahue

I noticed that the CS410 update to version 3.0.0.2899 still points to the 2.1.7 release notes. What are the differences between the 2.1.7 software release and the major version number bump to 3.x for the CS410? We have several CS410s deployed at client sites and we want to make sure that all our bases are covered before we push the update to them. -- Tim Donahue