nathans

Hi, Tried the installer on for OS X found on https://vodia.com/documentation/install_mac and it gives an unsupported OS version. Choosing continue regardless and the install exits with a fail without installing anything. Please advise. Tanks Nathan

Actually, let me ask this in another way: HQ PBX has the 192.168.30.0/24 network with the snomONE being the 192.168.30.2 and public IP 75.10.10.10 Branch A has the 192.168.33.0/24 and it's phones connect back to HQ's PBX at 192.168.30.2 via the VPN. It this the correct IP Routing List entry? 192.168.30.0/255.255.255/192.168.30.2 192.168.33.0/255.255.255.0/192.168.30.2 0.0.0.0/0.0.0.0/75.10.10.10 THANKS! Nathan

Thanks. Sorry, missed the "1" when I did the copy/paste. I got the parts of the private/public IPs all right. Did not get what you mentioned about the alias? can you explain that part a little more? Thanks!

Hi, Here is our issue. Hopefully someone has some ideas as we are stuck right now: 1) Two main offices each with their own snomONE (4.5) and a bunch of remote offices connected via ipsec VPN and a bunch of floating sip phones connecting via public internet to either of the main PBXs 2) All calls between offices and phones work perfectly and have been like that for the last 5 years or so until we recently had to switch phone providers and had to create a new SIP trunk instead of the old analog trunk we were using. 3) For the SIP trunk to work, we have tried every single setting and configuration we can think of. The only way we manage to make sound travel in both directions (the PBXs are being a Meraki firewall with the correct ports opened and NAT) is to add to the snomONEs a IP Routing List entry like 92.168.0.0/255.255.255.0/192.168.0.7 0.0.0.0/0.0.0.0/75.150.87.9 With this entry, the new SIP trunk work perfectly BUT we loose connectivity of the remote phones. Moreover, the calls between extensions between the main offices only last less than 2 minutes before the call gets dropped. Any ideas? Thanks all for the help Nathan

Hi, We have two geographically distant, offices both running the same setup: 1- same firewall and same configuration 2- A site-to-site VPN with no restrictions in any direction 3- A PBX running at each location on a windows 2008 server We created a trunk at each end as a SIP Gateway, each with the far ends internal IP (via the VPN) as their proxy address and have a dial plan sending all calls to each office according to their extension numbering (7xxx and 8xxx respectively) The issue we have is the following: 1) When office A calls an extension on office B, we get the call through perfectly with good audio and all, but the call is always dropped at around the 2 minute mark. 2) When office B calls an extension of office A, the call goes through and can last hours without any issues. These happens no matter what extension call or what extension is called. Any ideas? Firewall is not an issue (identical equipment and configuration on both ends) VPN is clean and identical both ends (using Meraki MX90 appliances to do a site-to-site VPN) all configuration of the PBX is almos identical, expect the extension numbering and local trunks. Thanks! Nathan

We currently have and administer a bunch of snomONE 4.5 installations. If we can help somehow, please let us know. We would be happy to help/point you in the right direction. Nathan

Hi, Had the same issue on a snomone 4.5 installed on a windows server. NO problems with the time NTP server but somehow the CDRs froze. Did the above and all seems working again.

We have fixed this by taking away the Ip Routing entry we had on one of the sides. On it we had the internal/external IPs for our remote phones. At least we fixed the interoffice trunk dialing but now we broke the outside phone lines

Hi, the Explicitly list addresses for inbound traffic is empty right now. Should these set to the private IP of the far end someone? Thanks Nathan

HI, We have a mystery between two snomone boxes that have been working for almost 2 years. Recently due to security issues we had to switch the inter-office trunk to run across a ipSEC VPN instead of the wild internet. The ipsec has no restrictions whatsoever and traffic flows without issues across it. The trunk is configured as as generic SIP gateway pointing to the remote internal IP of the snomones. Call go throughout with no issue and audio flows perfectly. The only issues we have right now, is that from when Office A dials Office B, the call gets disconnected at exactly 1:30mins. When Office B dials Office A the calls work normally and do not get disconnected until they finish. Any ideas where this limit of 1:30mins or disconnect comes from? We looked everywhere/ Any ideas are most welcomed! Thanks! Nathan

Xinity, It is just like you said. The recording in your case will include the "greeting" and the "instructions" for the IVR - all in the same recording. Then in the IVR section of the AA you will disable the "Play default welcome message:" and then upload or record your greeting + instruction to the enabled "Override for default:" Cheers Nathan

Hi, We have a very strange behavior on one of our analog trunks. This is a new new analog trunk on a Grandstream 8 FXO gateway. Calls in and out are working perfectly except that incoming calls. This is what happens: 1) All calls to any of the FXo numbers are redirected to a AA using the "Send call to extension:" setting on the redirection settings of the trunk 2) The call gets answered on the AA correctly ​3) The problem is that if the user dials an extension, say 8806 the system dials 8888 instead. 4) if there user dials any of the digits (1,2,3..) of the IVR menu, it also sometimes goes to any other extension. This behavior ONLY happens when ringing any of the DIDs from the outside. If from any extension I dial the AA number, it works perfectly and as expected. It is something happening when the trunk hands the call to the AA. Any ideas??? Thanks Nathan this is on a snomone running 4.5.0.1075 Delta Aurigids

HI, We have been asked to integrate an Avaya IP Office with a snomone. The idea is to have the extension of one system call the other's extensions. We thought that just creating a trunk between them should be enough (with the required dial plan logic on it) , but have not been able to make this work at all. Anyone has any experience with this or similar Avaya setup? Thanks for the help!

Hi AG1, We have several snomOne+ with both analog and digital sangoma cards. I agree the setup is far from being a turkey solution, but it is not that difficult once you get into the snom mindset of doing things. A couple of things: 1) did u run the autocalibration for the sangoma card? 2) are the analog lines directly to the sangoma or is there any equipment in between? I can not comment about the patton...tried one to configure one, but I hear their technical support are great. Try them out. Once the snomone+ is working, is pretty good.

Thanks! Nothing special here: the system is racked in an IT closet with a whole bunch of equipment all chilled and cozy in a good environment with stable AC and good UPSs. I'm doing daily backups right now. From some of the logs in the /var/ folder it may sees like a HDD issue. But I thought these boxes had SDD HDs which I thought were a little bit less prone to regular HDD issues. I guess I was wrong...

HI, One of our snomONE+ E1 is getting worse by the day. It started about 2-3 weeks ago when it will suddenly "freeze" not allowing any in/out calls. The web interface would also be not responsive and all phone will loose registration. On the console connected directly to the VGA we would see nothing. The only way to get it back is to power cycle the unit. As time has gone by, this started occurring once every 2 days, and now it has beed happening at least once a day, even on saturdays/sundays with 0 load/calls on it. Nothing was changed in the setup/network/topology. It just seems dies at random times. Any ideas? pointers as to what to look for and fix this?? Thanks! Nathan

perfect. Message sent. Thanks!

Hi, I'm trying to PM but get an error: The member pbx support cannot receive any new messages

305xxxxxx is the external number calling in to the trunk number (361xxxxxx) No idea what "packetrino" is

After a few months, we have yet to make this work. I'm seeing something strange on the logs that maybe a clue to somebody out there. Its seems like the incoming call is detected and recognized and it fails to be tranfer to the extension 70 which is the incoming send all extension. Have no clue why....the calling party juts hears silence for about 10 seconds and then gets a fast busy. Any ideas anyone? Thanks!! [8] 2012/08/08 11:58:02: Trunk: Check if the call to +1361xxxxxx comes from the cell phone +1305xxxxxxx [8] 2012/08/08 11:58:02: To is <sip:+1361xxxxxxx@75.149.181.121:5060;user=phone>, user 0, domain 1 [8] 2012/08/08 11:58:02: Send call to extension ERE returned 70 [5] 2012/08/08 11:58:02: Domain trunk vitelity@snom.beta-brain.com sends call to 70 in domain snom.beta-brain.com [5] 2012/08/08 11:58:04: SIP Tx udp:64.2.142.15:5060: INVITE sip:305xxxxxxx2@64.2.142.15 SIP/2.0 Via: SIP/2.0/UDP 75.149.181.121:5060;branch=z9hG4bK-14b4687fd863a2e50da8ec5032dfbeaa;rport From: <sip:361xxxxxxx@75.149.181.121:5060>;tag=15a98ce6fa To: "+1305xxxxxxx" <sip:305xxxxxxx@64.2.142.15>;tag=as1430a5bc Call-ID: 5bcde1726c80fb7f212923382857158a@64.2.142.15 CSeq: 30655 INVITE Max-Forwards: 70 Contact: <sip:nsandler@75.149.181.121:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snomONE/4.5.0.1090 Epsilon Geminids Content-Type: application/sdp Content-Length: 314 v=0 o=- 1472081629 1472081629 IN IP4 75.149.181.121 s=- c=IN IP4 75.149.181.121 t=0 0 m=audio 60702 RTP/AVP 18 0 8 101 a=rtpmap:18 g729/8000 a=fmtp:18 annexb=no a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtcp-xr:rcvr-rtt=all voip-metrics a=sendrecv [5] 2012/08/08 11:58:04: SIP Rx udp:64.2.142.15:5060: SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 75.149.181.121:5060;branch=z9hG4bK-14b4687fd863a2e50da8ec5032dfbeaa;received=75.149.181.121;rport=5060 From: <sip:361xxxxxxx@75.149.181.121:5060>;tag=15a98ce6fa To: "+1305xxxxxxx" <sip:305xxxxxxx@64.2.142.15>;tag=as1430a5bc Call-ID: 5bcde1726c80fb7f212923382857158a@64.2.142.15 CSeq: 30655 INVITE User-Agent: packetrino Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0

Hi, We are trying to interconnect a Cisco Call Manager with a snomONE system as a way expand capabilities at a customer location. Ideally we would like to connect both systems via a "inter office" trunk so that calls can flow in both directions between systems. Juts like with two snomONEs. Unfurtunatelly we have no experience with Cisco and are at lost as how to start. Any ideas or pointers from anybody? Thanks Nathan

Hi Mark, Even with training installing and configuring a snomone+ is not a trivial process. You should have some definitions regarding extension, call groups and networking in general before you start. Please post a little more info so that we can help you get started. Also, have you reviewed the book? It is a good starting point.

We have the same issue (posted before about this) at one of our locations with all snom phones and a snomONE appliance. Call pickup/listen-in in just stops working on all phones. So far, the only way to get this working again is to reboot. Unfortunately we are having to do this every 2 or 3 weeks. Any insights would be most welcomed.

For us worked changing the DNS for the real IP of gmal's imap server.

We have the same gmail issue not working any more on 2 appliances and it has not clear itself. Any ideas? [3] 2012/06/07 23:44:22: Could not connect to [2001:4860:800a::6d]:465 [1] 2012/06/07 23:44:22: Could not send via TCP: 52 bytes [5] 2012/06/07 23:46:22: SMTP: Timeout [3] 2012/06/07 23:46:27: Could not connect to [2001:4860:800a::6d]:465 [1] 2012/06/07 23:46:27: Could not send via TCP: 52 bytes