Parks

I'm not following understanding. Currently all snom phones have an ip address in their outbound proxy setting. If we wanted to move them to another cluster and their provisioning address is that of the old cluster we would have to do lots of manual work updating their provisioning addresses to their new server, correct?

No so in the outbound proxy setting we place sip:voip.vonvox.net:5061;transport=tls which gets resolved to sip:216.218.236.2:5061;transport=tls after the snom phone reboots. So now if we wanted to move this customer to another cluster we would have to manually update this field in all their phones. Hope this is more clear.

Is there a way to stop this. If we ever wanted to move the pbxnsip we would have to manually go into each snom endpoint and make the change. This isn't going to be possible without having users down for some time which is unacceptable.

For several months now this hasn't been working but I not need to pnp new phones on this system. Here is what the snom logfile says: [5]24/12/2001 00:00:09: Using GUI language English from: /mnt/snomlang/gui_lang_EN.xml [5]24/12/2001 00:00:10: Using WEB language English from: /mnt/snomlang/web_lang_EN.xml [5]24/12/2001 00:00:10: read_xml_settings: found dial-plan XML header [5]24/12/2001 00:00:10: read_xml_settings: found one byte encoding: 0 [5]24/12/2001 00:00:18: DHCP: Received IP address 192.168.5.201 [0]24/12/2001 00:00:18: tcp::set_port: Default SO_RCVBUF=43689 [0]24/12/2001 00:00:18: tcp::set_port: Default SO_SNDBUF=16384 [5]24/12/2001 00:00:18: Opening TCP socket on port 843 [5]24/12/2001 00:00:23: Setting server was already set: http://voip.vonvox.net/provisioning/snom360-{mac}.htm [5]24/12/2001 00:00:23: Fetching URL: http://voip.vonvox.net/provisioning/snom360-000413291522.htm [5]24/12/2001 00:00:30: sip::process_challenge:No nonce found on response [2]23/12/2001 16:00:33: start_dst(984276000) end_dst(1004839200) offset_dst(3600) offset_utc(-28800) [2]23/12/2001 16:00:33: start DST: 03/11/2001 02:00:00 (984276000) [2]23/12/2001 16:00:33: end DST: 11/04/2001 02:00:00 (1004839200) [5]23/12/2001 16:00:34: read_xml_settings: found phone-book XML header [5]23/12/2001 16:00:34: read_xml_settings: found one byte encoding: 0 [5]23/12/2001 16:00:34: Using GUI language English from: /mnt/snomlang/gui_lang_EN.xml [5]23/12/2001 16:00:35: Using WEB language English from: /mnt/snomlang/web_lang_EN.xml [0]23/12/2001 16:00:36: tcp::set_port: Default SO_RCVBUF=43689 [0]23/12/2001 16:00:36: tcp::set_port: Default SO_SNDBUF=16384 [5]23/12/2001 16:00:36: Opening TCP socket on port 80 [0]23/12/2001 16:00:36: tcp::set_port: Default SO_RCVBUF=43689 [0]23/12/2001 16:00:36: tcp::set_port: Default SO_SNDBUF=16384 [5]23/12/2001 16:00:36: Opening TCP socket on port 443 [2]28/3/2010 15:44:35: start_dst(1268532000) end_dst(1289095200) offset_dst(3600) offset_utc(-28800) [2]28/3/2010 15:44:35: start DST: 03/14/2010 02:00:00 (1268532000) [2]28/3/2010 15:44:35: end DST: 11/07/2010 02:00:00 (1289095200) Please let me know your thoughts, thanks.

I rebuilt the server again last night. So far so good. Keeping my fingers crossed. We didn't update windows through September but rather through April. Hopefully everything is fixed.

We are using pnp. Where is the directory for this that we can look at the settings?

I don't think you read my last post because WE DON'T use windows firewall at all. It's also not on every calls so leaving wireshark on might be hugh but guess I can always try it.

I understand that the default is 160 per snoms wiki but it doesn't go into detail on changing this. I've even when to the gui in the phones and changed the qos diffser in the advanced tab to 25 46 but calls are still be tagged 160. Please let me know if you have dealt with this and how to correct it.

We don't have windows firewall but do use juniper and that's been fine as we have the tcp and udp ports open for the sip and rtp traffic. We have the cdrs duration set to 90 days which should be fine. I can always change sense we don't use the pbxnsip cdrs. What would I be looking for in the pcap?

We're experiencing this and believe it's because of a recent windows update. We reinstalled the os and it still happens. Can or does anyone have any other ideas? Basically the pbxnsip service doesn't auto start on reboot and when starting manually can take a few tries. It goes half way and hangs and sometimes starts and sometimes gives an error message that M$ doesn't have any help with.

Ok. My linux and network engineer is testing with Debian because of how it handles packet management. We're going to deploy with 2 servers to start and use NFS for syncing. Thanks for the input as well.

We want to move away from MS and into something more reliable but would like to know which version is more suggested for clusters. Any suggestions would be great.

That is what I needed to do. Thanks all for your help.

We are testing the newest stable release from our 3.1.x.xxxx version. We're not able to have the switch send the 1 as it always removes it before sending to our class 4 switch for routing. We have the following in the dialplans: xxxxxxx -> 1925* xxxxxxxxxx -> 1* xxxxxxxxxxx -> blank 011* -> 011* Why is it removing the leading 1 every time?

This scenario only applies if the 911 operator needs to call back the caller but the whole point of registering each location is to have it auto route to the nearest 911 call center. enable911 charges you if you go through the national operator an extra $100 per attempt. If we are using 1 trunk to propagate this ANI it's always going to be the same. Am I missing understanding this? Anyway I can call to speak about it?

That would be great, thanks. Were would you be posting this so I can review it?

What version is EPID from. We're running 3.1.2.3120. Plus I thought that the extension ANI will override the domain or trunk ANI??? Also what happens if the company has a few remote workers this will not work.

Each domain has one trunk to our nextone switch which will route the call accordingly.

NO. We need to be able to override extension ANI when calling to 911 with a certain ANI that's registered in 911 dbase. I cannot believe this is so difficult to understand. I'm been trying to get this now for month and no one seems to understand the laws regarding this and PBXnSIP from what I can see doesn't make it very economical for service providers making us register all DIDs with the 911 dbase.

This still doesn't answer my question. All we want is to have everyone from a particular domain to use a specific ANI when calling 911. This is a feature that should be added. This is what I am envisioning: In the domain setting right under emergency number there would be a field for emergency ANI. And let's say that there is 1 home worker that cannot use that ANI of course in the domain right under ANI their would be emergency ANI that would override the domain one. This seems to be the simplest why to cover all needs regarding multi tenant deployments scenarios. I'm just very surprised this really hasn't been an issue before. Registering ever number is a waste of money for both the customer and us and probably lost some customers because of it. Would be very helpful if someone from PBXnSIP would review this and give feedback. Thanks all.

the invite was correct however the ANI still shows the extension/account number. Many carriers will read the use it for the CID as that's what an ANI is. Some will not even complete the call if left blank, they want Not Available in the ANI to represent Private Number. I'll be happy to email someone a cdr showing this.

I have tried that as well on the account level and domain settings. What is pu in the ANI when it actually works?

I don't believe the call back feature actually calls the originating caller back but rather sends an email to the callee letting them know the callers number and they want a call back. We also never got it working so we turned off for our customers. Don't know about your other issue as we don't use Polycom phones.

Some residential customer don't want their number be published when placing calls. If left blank pbxnsip in puts the account number which is the customers 10 digit tn in our case. We want to pass Not Available to the PSTN as some carriers require this in the ANI and cannot be blank. Any suggestions would be great, thanks.

I understand that the audio is compressed and therefore the quality is less. However our customers on our SIP trunking service love it and cannot tell the difference. Most cannot tell the difference in quality especially because so many people use cell phones these days. If every time it gets transcoded it reduces the quality then it really is only get transcoded twice once on the carrier side then on the customers side. Am I looking at that right?