Tom Waterman

Ok in my office I have about 62 Snom M3 phones. We just went live today. I have received about 6 or 7 reports of calls dropping in midsentance. The calls are not on mute and I am running a permanent key. These calls go from the PBX to Audiocodes FXO gateways. It is very sporadic. Theres is also no NAT that goes on in this situation. My cpu usage is very low and I am working on getting a wiresharp capture the next time it is reported. I am running version 3.1.2.3120. Does anyone have any ideas? Thank you for your help in advance. Tom

I am on PBXnSIP version 3.1.2.3120

I am automatically provisioning the phones with an xml file. I have the TFTP password set to ALWAYS. When I loose the settings I am also unable to log into the web and it fails to register however, I can ping it and see it in wireshark requestin aI have noticed that after I provision the phone you log into the web interface with the phone username and password which is awesome and my directory works as well. I am using the Snom 370.

Ok FINALLY! I was able to provision the phone and get the address book to load. But if I rebbot the phone everything fails. If I look in the wireshark capture I can see this: Subscription-State: terminated;reason=noresource I am not sure if this has anything to do with this or not. If I reset the phone to defaultsd it works fine. Any ideas? Tom

I am ruynning version 3.1.2.3120. I just tried creating the trunks for the first time. I may try to edit the xml manually.

Hmm what does this mean from the log file. I can see it deleting some of my co lines. [0] 2009/02/05 08:17:44: Database sanity check: Deleting coline 15 [0] 2009/02/05 08:17:44: Database sanity check: Deleting coline 16 [0] 2009/02/05 08:17:44: Database sanity check: Deleting coline 17 [1] 2009/02/05 08:17:44: Starting up version 3.1.2.3120 [1] 2009/02/05 08:17:45: Working Directory is C:\Program Files\pbxnsip\PBX [5] 2009/02/05 08:17:45: Starting threads [5] 2009/02/05 08:17:45: Set processor affinity to 1 [0] 2009/02/05 08:17:45: UDP: bind() to port 69 failed [0] 2009/02/05 08:17:45: Could not open UDP port 69 for TFTP [5] 2009/02/05 08:18:58: Identify trunk (domain name match) 2 [2] 2009/02/05 08:19:27: HTTP address is empty

That is the problem that I have. I don't know why the PnP configuration does not work. I have followed the Wiki. When I boot the phone and log into it I can get pbx to assign the mac address to the users extension but the dir (phonebook) does not work. I have 2 deployments next week and another in a month or so. So I need to get this ironed out. The firmware issue is solved. After downloading the original firmware the phone is working again! Now if I can get that phone book issue taken care of.

How many co lines can I have a single trunk? I have a mediant 1000 that has a T1 with 23 channels. Do I need to create more than 1 trunk group. I see in my single trun group that I am missing a handful of the lines I know I put in there. Thank you Tom

I have 2 questions about the snom 370. First, I have a snom 370 that was running firmware 7.1.30 and I upgraded it to 7.3.14 Now I have a nice BRICK. When I boot it the phone hangs or it will boot and ask you to confirm the language although there are no choices to choose from. You can exit out of it but the phone fails to work. I have tried downgrading and loading new and old firmware via the tftp option all with the same results. Anyone know how to fix this or where I can get a copy of 7.1.30? Thank for the help. Second, I want to push down the domain directory to all phones on the network. Most are 370's. I have followed the wiki with negative results. I would appreciate any help as I am desperate. Thank you Tom

Yeah I tried this as well and it did not work. It keep setting my time zone to -4:30 and I need -5:00 and it messes with Daylight savings time even when I tur it all off. I am at the end of my rope.

No dice. I have found the section edited it and I am still an hour off. My time is 1:02pm and the phones say 2:02pm. I have daylight savings time enabled on the pbx but something is still wrong.

I have found it and edited it. I'll report back.

Is this global configurration file on the phone or in the PBX? I have been changeing the file on the phone under the advanced tab is a time server section. If it is on the PBX how do I find it?

A couple of days ago I deployed 65 phones in my company. All but 2 are snome M3. All of the lines were set and I set a time server in each of the bases as well as upgraded the firmware. Yesterday I came in and tried to work on auto provisioning (big mistake) and all of a sudden as soon as I rebooted a phone it would wipe out the first extension and provision it with a new one from the pbx. Don't even ask me how that happened. I believfe everything is good now but In the base of the M3 I still have the wrong time server. I have cleared it out and saved it a hundred times but each time I reboot the phone this pool.ntp.org is in there and all of the times are incorrect. I have checked our own time server on the network and it is correct. Any help would be great. We need to roll these out tomorrow. Tom

I think I must be missing something that would fix this issue. I want the ability to record calls on any extension I choose on a per call basis. I want to start recording before the call begins and end it when the call ends. I do not want to record all calls to or from an extension. This seems very simple so I think I am missing something in the setup. Right now I can get it to record all calls so I know the recording feature works. Tom

I am deployiing 60+ Snom M3 phones in our office. I have a paging group setup and I want the phones to auto answer on the page with the speaker phone enabled. Can this be done? I am having a hard time finding any info on this. Thank you in advance.

I do not have an electronic version of the file to place in the lic window. I thought the USB dongle made it so you did not need that. I didn't think you needed both. I'll email Kevin right now.

I have the blue usb dongle that Kevin sent to me.

I have our license on a USB drive attached to the PBXnSIP server. Yet my product still says it is unlicensed. Am I doing something wrong?

Could you send me that as well. I get the same error.

I found out what it was. I had my domain name in the setting set as the FQD name but in the sip header I was sending the ip address. so I added the alias of localhost to the domain and it works like a charm.

Yeah that's what I thought. This is running on Server 2008 and I have disable windows firewall all together. I even stoppped the service. No luck.

I have my PBX in the DMZ( I read the wiki article on this) and I can see my registration request hitting it and I can see the replies on my Snom M3 gateway. But my error is 404 not found and on the phone it says unknow sip account. Here is the sip trace from my M3 phone. I have 60 of these to deploy by weeks end! Thank you. Sent to udp:172.31.3.10:5060 at 26/01/2009 12:10:26 (458 bytes) REGISTER sip:172.31.3.10 SIP/2.0 Via: SIP/2.0/UDP 172.16.50.1;branch=z9hG4bKxyf.fzoxcxpecm9oyzah302ktx Max-Forwards: 70 From: <sip:817@172.31.3.10>;tag=sfe862h1 To: <sip:817@172.31.3.10> Call-ID: 8uh1h72ojm CSeq: 160257 REGISTER Contact: <sip:817@172.16.50.1> Allow: INVITE, CANCEL, BYE, ACK, REGISTER, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE, INFO, PRACK Expires: 60 User-Agent: snom-m3-SIP/01.22 (MAC=0004132A2F3D; HW=1) Content-Length: 0 Received from udp:172.31.3.10:5060 at 26/01/2009 12:10:26 (243 bytes) SIP/2.0 404 Not Found Via: SIP/2.0/UDP 172.16.50.1;branch=z9hG4bKxyf.fzoxcxpecm9oyzah302ktx From: <sip:817@172.31.3.10>;tag=sfe862h1 To: <sip:817@172.31.3.10>;tag=82a4ffb026 Call-ID: 8uh1h72ojm CSeq: 160257 REGISTER Content-Length: 0

The secondary server will have all of the same information the primary does. Should I use the same IP address? That way if server 1 fails I would power on server 2. But the phone would all need to register again, correct? But a feature in a new reales to make this easier would be great, especially as we try to roll this out to mission critical places.

How can I setup a redundant/failover installation of PBXnSIP. I have 2 servers in 2 different locations that are connected via a run of point to point fiber with just media converters at each end. I want 1 server to be the main server but should it ever die I need to be able to use the backup. Is this possible and if so is there a document that provides instruction. Thank you in advance! Tom