bruce

I guess I'll close this thread with the following thought. I just got off the phone with snom/vodia and must say I'm completely put off by their business tactics. I have a server about to go down and they cannot offer any way to transfer my license to a different mac address UNLESS we pay to upgrade. That is some really bad customer service right there. I'm now left trying to justify the cost of upgrading just to deal with this. I can't even move the NIC since it's legacy PCI. Totally dissatisfied customer.

Well im still on 4.5, with a snom blue license. So I should have no issue once I actually install the correct version now?

*insert cursing here* Cannot believe all this was due to a license issue. And I only noticed it because I happened to look at one of the text files buried in the snomone dir. THanks for all the help. Bruce

So more information. I was poking around and as it turns out when you restore your license doesn't come over. So I put in our license key. All extensions are now showing up as not licensed and still cannot register.

tcp 0 0 0.0.0.0:443 0.0.0.0:* LISTEN 1564/pbxctrl tcp 0 0 0.0.0.0:5060 0.0.0.0:* LISTEN 1564/pbxctrl tcp 0 0 0.0.0.0:5061 0.0.0.0:* LISTEN 1564/pbxctrl tcp 0 0 0.0.0.0:389 0.0.0.0:* LISTEN 1564/pbxctrl tcp6 0 0 :::443 :::* LISTEN 1564/pbxctrl tcp6 0 0 :::5060 :::* LISTEN 1564/pbxctrl tcp6 0 0 :::5061 :::* LISTEN 1564/pbxctrl tcp6 0 0 :::389 :::* LISTEN 1564/pbxctrl udp 0 0 0.0.0.0:57260 0.0.0.0:* 1564/pbxctrl udp 0 0 0.0.0.0:50368 0.0.0.0:* 1564/pbxctrl udp 0 0 0.0.0.0:5060 0.0.0.0:* 1564/pbxctrl udp 0 0 0.0.0.0:69 0.0.0.0:* 1564/pbxctrl udp 0 0 0.0.0.0:39894 0.0.0.0:* 1564/pbxctrl udp 0 0 0.0.0.0:161 0.0.0.0:* 1564/pbxctrl udp6 0 0 :::5060 :::* 1564/pbxctrl udp6 0 0 :::69 :::* 1564/pbxctrl udp6 0 0 :::51935 :::* 1564/pbxctrl udp6 0 0 :::34826 :::* 1564/pbxctrl udp6 0 0 :::161 :::* 1564/pbxctrl unix 2 [ ] DGRAM 6602 1564/pbxctrl Ip of phone is explicitly allowed so its not being blacklisted.. 192.168.1.228.2048 > 192.168.1.188.5060: [udp sum ok] SIP, length: 393 CANCEL sip:100@192.168.1.188;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.1.228:2048;branch=z9hG4bK-lkiuetlkl4em;rport From: "Bruce Markey" <sip:100@192.168.1.188>;tag=hgg0cykxhj To: <sip:100@192.168.1.188;user=phone> Call-ID: 3c2670553252-3omuupi3eb9o CSeq: 1 CANCEL Max-Forwards: 70 Reason: SIP;cause=487;text="Request terminated by user" Proxy-Require: buttons Content-Length: 0 Thats what a -vv tcpdump shows me. Still no registration.

IFconfig is exactly like the old. I can verify that working. 1 external interface with gateway, 1 internal with no gateway. That works. route is correct. Default route goes out where it should. Verified all internet connectivity. I actually totally removed iptables just to get that out of the way. I did a tcpdump on port 5060 testing with 1 phone to see if it registers. 09:35:19.933261 IP 192.168.1.228.2048 > 192.168.1.188.sip: SIP, length: 657 09:35:23.943406 IP 192.168.1.228.2048 > 192.168.1.188.sip: SIP, length: 657 I see the phone ( 228 ) communicating with the snom server ( 188 ) but it will not register.

I thought the same thing. Iptables is accept on all. Resolv.conf is set to same dns servers as old server. I can get everywhere on the internet. What I did find out is. If I do a tcpdump on the external interface and click register its nothing. Its almost like snomone cannot see the interface. I have completely removed the trunk and re set it up. In addition I thought it might be an issue with using Debian 7 since it just came out. So I reverted back to debian 6. Same thing. That's where I'm at now. Thanks Bruce

We've been running Snomone one on a physical server now for about 2 years with no issues. It's running under ubuntu 10.10. I've needed to migrate this to a new server and to upgrade the version of snomone to the newest. So. I've run up a Debian 7 under VMware. Installed the newest version of snomone as per instructions on the wiki. I backed up my config on the old server, imported to the new. Everything came right up. We run dual nics, one internal one external just fyi. Tried to migrate over and our voip trunk will not come up. Constantly gets a 408 error. Upon some more testing found out that running tcpdump shows that when I click register in the admin interface I get 0 traffic on my external nic, hence its not even trying to register and im getting a 408. Am I barking up the wrong tree here trying to get this to run in a vm? Is there something with snomone that I have to set different to make it work. I've verified all network level stuff and it all works. Same settings throughout. The only difference is a new version of snom and the fact that its in a vm. Any help here would be appreciated. Thanks Bruce

Matt Is that available or going to be? I've been fighting with this issue for a bit now. To be able to have a small app that shows who's on what line etc. Thanks Bruce Markey

Lights could work but I'd like to able to see who is on with who. Would make things easier for us dealing with calls. PAC? Thanks Bruce

Not sure if there is an easy way to do this so I thought I'd ask. We have maybe 6 guys on the phone all day. None of can see one another so its hard to know when one of them is on the phone. Now you can log into the web interface and see the Currently Active Calls. This way we can see what extension is on the line. I'd prefer that not everyone was logging into the web interface. Also it's a bit unwieldy to always have to flip over to a web page to view calls. Is there a widget of any kind that anyone has come across that can be kept open and will poll for current calls? Looking for suggestions. Thanks Bruce Markey

I just slapped myself in the forehead. Sheesh. Thanks Bruce

Currently even when we are on the phone and a new call comes in it rings in our ear. Is there a way to disable this? We are all part of a hunt group. What I'd like is for it to ignore an extension if that person is currently on a call. Thanks Bruce

I searched through the forums and didn't see anything specifically concerning this so I figured I'd ask. I'm hoping it's something simple. We have 4 pstn lines through a FXO box and a 4 line SIP trunk. Whenever we get voicemail it sounds great when checked from one of the SNOM phones in the office. But, if you check it from any other phone, or you try to listen to the file sent to your email you cannot hear it. Even with volume all the way up its barely audible. The only thing I have not tested is whether or not there is a difference whether the call comes in on our SIP trunk. All incoming come in over the pstn, and we try to keep the SIP for outgoing only. All calls sound normal. I did see that there is no "setting" for this persay, but maybe someone could point me towards the next step with this problem. Most of our guys check their VM from outside, mostly in email, and this is becoming a problem. Thanks Bruce

After all that, yes I went that route, no pun intended. dual nics 1 live. Everything works. So much easier this way. Home extensions work, softphone on cell works (although not over 3g). Thanks again for all the help. Bruce

Hi all. Here is our current setup, it's very simple and works fine. Internal network (192.168.1.x) - snom server (192.168.1.47) -- Second nic (10.1.10.250) -- connection to comcast business (10.1.10.1) - internet We use 4 pstn lines with an fxo gateway, and also a second trunk to a sip provider for outgoing calls. The reason for the second nice is due to having to connect to the Comcast router directly. It sits on a 1 to 1 nat. (DMZ is already taken by another server) Internally all phones work wonderfully. Obviously this creates problem with any outside extension. IE direct extensions at home, softphones running on smart phones. They register fine but 1 way audio only. Right, RTP and nat not playing nice. Home extension works great with IPSEC vpn. So what we are doing is removing the whole 10 network out of the picture. We have a block of addresses and my original plan was to put the pbx's second nic right on a live address but after thinking about that it might not be the best idea. So here comes introduction of a sip proxy. That way this can sit live on the net and keep our production pbx at least 1 level protected. Not sure if anyone here is using milkfish but that is the plan it so run milkfish on a dd-wrt router. My question is twofold. Is there anything I should be aware of when implementing a sip proxy. From what i've read this eliminates the issues we've had with remote extensions etc. Second, has anyone actually implemented milkfish? Hope this is the right place for this question, I know its not specifically snom but it plays into the whole system. Thanks much Bruce

Ok that makes sense. What makes it stop letting me dial after 2 numbers then. IE lets say I want to dial 268-xxxx. Immediatly after the 26 it stops accepting input. I can't seem to make this go away, I assumed it has to do with extensions. Am I correct in this or am I barking up the wrong tree. Thanks Bruce

I'm guessing this probably has a simple answer but I can't seem to make it work. We have extensions numbered 10000 - 10020. Where/how exactly to I set it up so it recognizes when I'm dialing an extension and when I'm dialing a real number. If i try to dial a local number starting with say a 2 for us, it thinks its an extension. So we end up having to dial the full 1+area code to make the number work. If i'm missing a best practices for extension setup sorry for that. This seems to be the last little hurdle we have to make work before our rollout. Thanks Bruce

Thanks for the info matt, appreciate it.

Is it possible to send voicemail notification to multiple recipients? As in every member of a hunt group. So if we get a tech call and none of our techs can pick it up can it be delivered to all their email? Or do I have to make a catch all for that and then have them all check that? Thanks Bruce