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lleo19

Occasional Visitor
Hi All,
Long time reader, first post!
It was Michael's review of the Snom m3 that enticed me to update my home/office telecommunications into the 21st century.
I got myself a Snom m3 with 4 handsets and installed Trixbox on an old P3 system to act as PBX.
As I am not yet ready to cut the POTS lines, I also got a Rhino R8FXX-EC analog card with a dual FXO module to interface with the PSTN.
On the VOIP side I signed up for service from an ITS, for now without a DID number.

I understand that the snom m3 does not need a PBX and could register directly with the ITS provider, but that is very limiting. My plans call for an IVR, voicemail to email, fax to email, and later possibly expand with GSM trunks and other geeky features.

I have created four extensions in Trixbox and directed each one to ring on one of the handsets. Everything seems to work fine. Calls between the handsets work just fine, both through the PBX or the Snom gateway directly as intercom. Calls from the outside PSTN also work fine and placing calls to the outside are directed correctly correctly based on my dial plan.

For now, I have set the PBX that when I get a call on the POTS line, the Trixbox PBX routes the call to a call group and rings all handsets. If there is no answer, than it goes to a voicemail associated with a 5th extension.

My problem is that after I take the call on one of the handsets, there will be a mysterious message displayed about a "missed call". Acknowledging the message will show you the call-log that I actually took the call. Very annoying!

Now having 4 handsets on 4 extensions normally will have 4 different voicemails associated. Regardless how many I set up and how I tried to set it up, I cannot make the snom handsets to display messages about voicemails received, not to mention anything about number of voicemails pending etc.

Michael or anyone else using snom m3 out there with Asterisk based PBX, can you please share your set up, as these limitations are much more severe in my mind than just not being able to centrally provision phone-book entries.

As far as details: all handsets and gateway are updated to firmware 1.16 and I am centrally provisioning the snom configuration through trixbox based TFTP server.
Thanks in advance, Lorant
 
Hmmm, that's an odd one. I've done a lot of work with asterisk so I'll take a shot...

So it's the handset itself that displays a missed call? I'm assuming so, in which case I dont so much think that's an issue with Asterisk, as much as the handset. Asterisk doesn't normally pass on information about missed calls, that's usually a function of the phone/device itself. So I'm guessing that there's something wierd going on with the handset that thinks it missed the call. Not sure there's much in Asterisk's config you'll be able to do about that.

Just in case some sort of funky configuration is getting pushed down to the device, perhaps try manually provisioning the device by manually adding an SIP device.

Asterisk should be pushing voicemail info to the phone though. Perhaps log into the Trixbox forums and see what they have to say. There's a lot of really good help on there.
 
Thanks for your reply
I think it is the handset, or better yet the gateway that generates this message.
Not sure if you are familiar with Snom m3, but it has a gateway to which the DECT handsets are registered and the gateway interfaces with the PBX or can register directly with upto 8 SIP registrars over the internet.

I tried the trixbox forums, but the overall feedback about snom m3 is not at all positive. Can you direct me where to find or guide me how to troubleshoot or monitor the asterisk voicemail info push?
Thanks, Lorant
 
Thanks for your reply
I think it is the handset, or better yet the gateway that generates this message.
Not sure if you are familiar with Snom m3, but it has a gateway to which the DECT handsets are registered and the gateway interfaces with the PBX or can register directly with upto 8 SIP registrars over the internet.

I tried the trixbox forums, but the overall feedback about snom m3 is not at all positive. Can you direct me where to find or guide me how to troubleshoot or monitor the asterisk voicemail info push?
Thanks, Lorant

Your best bet would be getting some logs from the server and posting them online on the TB forums. I never dealt with snom myself though. Not entirely sure where the current logs are stored with asterisk, but that's easy enough to find. My guess is it's something funky with the snom gateway, because I rarely ever hear of VM notifications not working properly.
 
My use of the phone with Asterisk is substantially different than what you describe. I have never established multiple registrations for the phones on my Astlinux server. That was never required.

Each registration in the m3 base was to a different server or ITSP. My in-house server is only one of these.

Here's the SIP.CONF entry for the snom:

[4002]
; SNOM
type=friend
host=dynamic
username=something
secret=something
qualify=2000
dtmfmode=rfc2833
context=from-inhouse
canreinvite=no

I don't have it subscribed to any VM as my desk phone displays MWI.

I also see that if an account is set to ring multiple handsets in the snom's internal routing the handsets that were not answered display a "missed call" message. This has not bothered me at all. I makes a certain amount of sense.

I'll have a play with the VM/MWI settings and see what I come up with.

Michael
 
I thought I would tag onto this thread as you might be able to help me. I have 4 M3's newly installed. They are all configured with Asterisk using a very simple Trixbox install. I have two POTs lines going into the server, identically configured. When I call line 2 all phones ring just fine. When I call line 1 all phones ring once then lose the connection to the gateway. They do this every single time. I was running the latest firmware so I thought I would step back to 1.15 to see if that made any difference, it did not. What can I post here to aid people in helping me to diagnose this issue? It seems to be an issue with the Snom gateway not the asterisk server.
 
Solved

My use of the phone with Asterisk is substantially different than what you describe. I have never established multiple registrations for the phones on my Astlinux server. That was never required.

Each registration in the m3 base was to a different server or ITSP. My in-house server is only one of these.

Here's the SIP.CONF entry for the snom:

...snip...

Michael,
Thanks for your post and the config information. Actually it forced me to rethink my dial plan. The fact that Snom can have 8 different identities and 8 handsets associated limited my thinking, but now with a single registration all handsets ring on incoming calls and when the call is taken, I do not get the "missed call" message. This also helped me to properly set up the voicemail and now everything works as it should.
I can't wait to try the new firmware and be able to load the same directory on all handsets.
Again, thanks and all the best
 
Last edited:
I thought I would tag onto this thread as you might be able to help me. I have 4 M3's newly installed. They are all configured with Asterisk using a very simple Trixbox install. I have two POTs lines going into the server, identically configured. When I call line 2 all phones ring just fine. When I call line 1 all phones ring once then lose the connection to the gateway. They do this every single time. I was running the latest firmware so I thought I would step back to 1.15 to see if that made any difference, it did not. What can I post here to aid people in helping me to diagnose this issue? It seems to be an issue with the Snom gateway not the asterisk server.

Well, me no expert by any means, but I also happen to use Trixbox with my Snoms and kept the POTS lines as well. You are not mentioning if you have monitored the incoming call in Asterisk CLI (ssh to your box and enter "asterisk -rvvvvvvvvvv" command)
Also not sure what do you mean loosing connection to gateway, it is that the handset displays something about "looking for the gateway"?
If you are logged in to the Snom website, do you get anything in the Snom log? Is your config file parsed without an error at gateway's start?
Are incoming calls routed through the same registration/extension to Snom for both lines or they ring separate extensions/registrations? Did you try to swap the lines at the FXO if it makes any difference?
OK, these are just a few of my questions, not sure if it helps at all, but the fact that your setup works with at least one line is encouraging. Me thinks that it may be a setup issue.
Cheers, Lleo
 
Thanks for responding. The phones are displaying "Searching for gateway" - which I think means they have lost wireless connection with the Snom base station. They reconnect within about 30 seconds. I am assuming that something about this particular POTS line is causing the SNOM base station to reset or something else.

I am not that Asterisk proficient - I set up everything through the web interface although I am able to connect to the server and get to the linux prompt and manipulate files. I have been looking at the log and so far have not been able to decipher it to the point that I see something that could explain this issue. I have posted the log (and what I did for this portion of the log) and the configuration files below, maybe that is helpful?

I did switch the lines at the server and the problem still occurred on the same line (in other words it followed the physical POTS line) - so it is obviously something to do with the incoming line?

Both lines are from Verizion (Fios) and configured the same except one has caller id (the problem line) and the other does not. I have actually ordered caller id on the second line so they will soon be the same.

I am bringing home a wired phone from the office today to try it to see if it does the same thing.

I uploaded the config and log files along with this message.
 
Here is the file attachment with the config and log files. Thanks.
 

Attachments

  • config and log.zip
    7.5 KB · Views: 255
I also have a question about the configuration you used to avoid the missed call message on all of the phones. Currently I have 4 phones and I have set them all up as separate identities. I set up a ring group that rings all phones. However, when one phone is answered, all others show the Missed Call screen. I think you solved this in your configuration, how did you actually do it?
 
New development - I turned on Caller Id on the second line and it is doing exactly the same thing as the first line. Apparently it is the Verizon Caller id that is causing the problem. Any thoughts on how I might sort this out? Thanks.
 
I also have POTS lines from Verizon, although not on FIOS :rolleyes:
and I have CID service as well.
Regarding the "missed call" message: I made this mistake myself initially to set up as many identities as the number of handsets, but Michael's message helped. You do not need them. You may need two if you have two lines and your dial plan should look something like:
-call on line1 comes and rings a ringgroup that has your two identies on a hunting strategy. It will ring the first identity all handsets and you can take the call on any of them.
-while you have the call on line1, if a call comes in on line2 which will ring on the same ringgroup, but will skip to the second registered identity as the first will report back busy, and will ring all handsets, including the one you are on the first call and you can take line2 from any of the handsets.

For the above to work, you will have to place checkmarks for both identities and all handsets on the Snom website, under "telephony settings"/"destination for incoming call"

Now of course you can get more complicated than the above. I have my 4 handsets setup as follows to serve my home and office.
- 1 extension rings on only 1 handset only at night that is in my bedroom.
- during office hours two handsets ring in my office.
- during non-office hours, but not night all handsets ring in the house
- during lunch time none of the phones ring, but goes directly to voicemail.
- I have only 1 voicemail box that is also associated with all handsets, so the MWI will display a waiting voicemail on all handsets, and I can check messages from each of the handsets.

The possibilities are "almost" unlimited. ;)
 
Upon re-reading my reply to your question, I realized that my answer may not sound simple or clear. Bottom line regarding the "missed call" message (also per Michael's response) if there are more than one extensions/identities on which an incoming call does ring you will get a missed call message as the Snom treats those as separate calls. On the contrary, you can have a call reaching the Snom gateway through a single registration and control through the snom website all the handsets that will ring.
Hope his is more clear
 
So, I should setup the SNOM gateway as one extension in Asterisk? Then all will ring when that extension is rung but the Missed Call message will not display if one of them is answered. If I am correct, this also means that calling between handsets can only be done with the SNOM intercom feature vs. dialing extensions.

Now, if I do this, what happens if a call comes in and is answered, and then someone picks up another handset. Will it break in on the existing call - like a party line?
 
No, you can have more than one extension registered, but you have to make sure that a call rings only one extension. The number of handsets to ring this extension you can control on the snom website.
You could do something like:

Handset 1 2 3 4
201 x - - -
202 - x - -
203 - - x -
204 - - - x
205 x x x x

This way you have extensions 201-204 to ring only on one of the handsets which will also allow to call between them through PBX.
Incoming calls should ring on extension 205 which will ring on all handsets and can be taken on any of them. You can try if the next call will also go through extension 205, otherwise you may need to set up another extension like 205.

The "missed call" message will appear when you send an incoming call to more than one extension like 201 and 202. Even if you take the call through 201, from 202 perspective looks as a missed call.
 

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