Parks

pbxnsip can do this with the snom phones but they need to be running firmware version 7.x first before any auto provisioning works. You also have to put the right bin file in the directory so when the snom phone looks for it, it sees that their is a newer version and downloads, installs and reboots if needed. I'm running pbxnsip version 3.1.2.3120 so if you have a newer release it might be different.

Before setting the dialplans up I tested setting the trunk ANI and my personal caller ID still comes up.

We are running a multi tenant deployment and want to see how we can force the source caller id to be the main office line rather the extensions caller id. We're trying to reduce the amount of e911 registrations required for customers with DIDs. Thanks of the input.

Just trying to see how G729a is working on other companies multi tenant deployments. We would like to put some customers trunks to use it primarily. Any feedback would be really helpful, thank.

Yes adding +1 worked and will be fine for using but would they still 11 digits and be able to get their voicemails sense we registered with the +?

We are starting to test for residential service and setup some test accounts. I couldn't get any of the ATAs to register so I thought I'd try to register my x-lite to confirm and couldn't. Here is the log file and everything should be fine. We're using 10 digit btn as their account/extension under 1 domain called residential.vonvox.net. Current system running 3.1.2.3120 [7] 2009/06/24 19:33:02: SIP Rx udp:75.149.48.107:8800: REGISTER sip:residential.vonvox.net SIP/2.0 Via: SIP/2.0/UDP 192.168.5.230:8800;branch=z9hG4bK-d8754z-4a3a8e79161c7f1e-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:9254185059@192.168.5.230:8800;rinstance=a2889ba708bc4675> To: "Test User2"<sip:9254185059@residential.vonvox.net> From: "Test User2"<sip:9254185059@residential.vonvox.net>;tag=2219607b Call-ID: Yzk0YmFjZGJiOWQxODZhOWUxZDMyMmEwODdkMmZlMGE. CSeq: 1 REGISTER Expires: 300 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: X-Lite release 1100l stamp 47546 Content-Length: 0 [7] 2009/06/24 19:33:02: SIP Tx udp:75.149.48.107:8800: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.5.230:8800;branch=z9hG4bK-d8754z-4a3a8e79161c7f1e-1---d8754z-;rport=8800;received=75.149.48.107 From: "Test User2" <sip:9254185059@residential.vonvox.net>;tag=2219607b To: "Test User2" <sip:9254185059@residential.vonvox.net>;tag=07d93e67c8 Call-ID: Yzk0YmFjZGJiOWQxODZhOWUxZDMyMmEwODdkMmZlMGE. CSeq: 1 REGISTER Content-Length: 0 Thanks for the help!

When will this change? Also we haven't made upgrades sense December because when ever there is one fixing something there is something else that gets messed up. Secondly pbxnsip changes so much on how they process info it makes it difficult to make a change and very nerve racking. We normally would have to do all the quality assurance because it doesn't get done. I guess we can setup another lab to test with.

This is a general voicemail extension therefore there aren't any registrations on it. We have the receptionists two phones and to allow access and send MWI to their extensions. Hope this is more clear.

We have a client that is finding old message days after they have been left. Just today she went to lunch and came back to find message waiting via the MWI and checked them. They were from Friday but MWI didn't let her know until today randomly. This is also happening to one other client as well.

I'll upload to our web server and email the link shortly. Thanks.

YES

We've rebuilt the server already. If we waited for assistance we would have lost way to many customers.

We are using server 2003 and we rebuilt the server last night. Nothing else was using the port either as we checked that before rebuild.

Doesn't work again after reboot. Here is what in the pbx.xml <log_level>7</log_level> <log_filename /> <log_length>500</log_length> <log_keep>3</log_keep> <log_sip_register>true</log_sip_register> <log_sip_subnot>false</log_sip_subnot> <log_sip_options>true</log_sip_options> <log_sip_dialog>true</log_sip_dialog> <log_sip_watchlist /> <log_sip_level>7</log_sip_level> <log_sip_routing>false</log_sip_routing> <log_event_general>true</log_event_general> <log_event_sip>true</log_event_sip> <log_event_media>true</log_event_media> <log_event_app>true</log_event_app> <log_event_email>false</log_event_email> <log_event_web>true</log_event_web> <log_event_register>true</log_event_register> <log_event_snmp>true</log_event_snmp> <log_event_trunk>true</log_event_trunk> <log_event_soap>true</log_event_soap> <log_event_tftp>true</log_event_tftp> <log_event_pstn>false</log_event_pstn> <log_event_sql>true</log_event_sql>

Finally got it working. I'll now be able to login and check the log details and will open the pbx.xml and find that line and confirm that we have a $. However we don't have any logs under the PBX directory. c:\program files\pbx

Don't know where you want me to look for this? I read the wiki link you sent me but cannot find where that log file would be located in the directory. Can I provide you a download link in a private message for the zipped up directory?

Can the service be started using "Log On As" the system admin rather than Local System. I don't get the time out issue but the website and phones still don't work.

Found out this am that servie wasn't running so I restarted. This has happen a couple of times where it just stops running for no apparent reason. After restarting the service this am I'm still unable to get to the web page via localhost on machine or from www. Please help with what ever you can.

I have been asking for this for months. Don't think it's possible nor something they're focusing on. But I'm curious to see what they say.

What about someone who we can pay as we don't have any experience on doing this type of conversion. Thanks

We have purchased from royaltyfreemusic.com but now need it converted to be compatible in PBXnSIP. Can someone point me in the right direction for getting this done. I have no problem paying a little for getting it done quickly either. Thanks for the help.

So you don't have any ideas on what else I can possibly try? Thanks for looking at the pcap.

You can download pcap here www.trivosoitsolutions.com/Downloads/TrivosoIT.pcap I can let you know the good and bad calls in a private message.

We're a small hosting company and some clients when making calls don't hear their calling party but the called party can hear them. If the called party calls our clients back nothing is wrong. This is ONLY with Comcast subscribers which makes this difficult to resolve if we even can. The worse part is that our clients never know if or when it happens because they never know when someone is going to have the service.

I know for my scenario I don't have any redirection involved.