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  1. collinsit


    I am not looking to do major recording on it, just the occasional call. Can you give some more information how I could setup basic recording through the system? I am just trying to see if there is any way I can get occasional recording to work or I guess I will have to look into a new system if I can't do it through the CS410 at all. Thanks
  2. collinsit


    Is there any way to record calls with the CS410. I have the white model and currently have it linked to an Exchange server. I also have two Snom phones. I know the CS410 doesn't have much memory to record locally and I don't see any option to record within the web interface anywhere. Is it possible to record calls somewhere else like to the Exchange server either through the pbxnsip phone interface or from the record button on the Snom phones? Thanks
  3. Hi there, I am trying to find a way to dial out from the call logs on the Snom phones. I have a Snom 360 and a Snom 320 and they both have this issue. If a call comes in, the missed call or incoming call history shows the number of the caller. If I try to dial from the history by pressing the check mark I get a message saying that the number i've dialed is not in service. In order to dial out from my system I have to enter 9 before the number and it isn't doing this when I dial from the call logs. Is there any way to prepend a digit to the number when I try to call from the phone logs. Thanks for any help or suggestions.
  4. What change did you make to get past it in your test environment? There really isn't any setting in the firewall as I am forwarding a complete IP through to the PBX.
  5. No, I am running a permanent key. Under status it says: License Status: Appliance Key License Duration: Permanent The calls don't seem to be disconnecting when I am talking on them, just when they are muted which doesn't sound like a license limitation.
  6. I don't think I am using the demo key. I should have a licensed CS410 and did before. Upgrading the software wouldn't have affected my license would it? How would I know if my license is limited to 3 minutes?
  7. I tried the new firmware version on the PBX and I am still having the problem. Like clockwork, after the call is connected and on mute for 3 minutes it disconnects. I will try to get a capture of the traffic tomorrow to see if there is anything that will help there.
  8. It's actaully on a CS410 with the newest firmware release. If you have a patch you want me to try I can give it a shot. Thanks.
  9. That would be great. I really need to get this fixed since I do a lot of calls this way. The way I have been testing this after the call I had yesterday is that i'll dial a land line and answer the call. I then mute the call on the pbx and within a few minutes the call will disconnect. If you can give me any suggestions as soon as possible I would really appreciate it. Thanks
  10. I checked the version on the phones and it looks like they were running 7.1.3. I updated them to 7.1.33 and the problem still kept happening. I then updated one of them to the newest version which is 7.3.7 and I am still dropping the calls. I couldn't find anything in any of their release notes about this problem. You said that I could edit the pbx.xml to change a setting for the one way audio setting. It doesn't sound like the ideal option but if I wanted to test making this change, how do I edit this file? If you can think of any other options that would be appreciated too. I don't think it is a Snom firmware problem since I have now tried it with three different versions.
  11. Hi there, I have a CS410 unit and a couple Snom phones, a 320 and a 360. I have been having an issue with calls dropping if I am not talking on the call. You may wonder why I wouldn't be talking but I often do conference calls or phone seminars so the phone will be on speakerphone and in most cases my microphone will be muted. It seems to happen after about 5 minutes. This has been going on for a while with the 360 but normally the 320 worked fine. I had that problem today with the 360 again so I dialed into the call with the 320 and it was doing it too. After it did it a few times at about 5 minute intervals I tested and just before the call got to about 5 minutes I picked up the handset, unmuted the line and made a noise. I then put it back on speaker and the call went fine. If I did this before about the 5 minute mark the call would go fine but if I forgot it would disconnect again after 5 minutes. I have the newest software release on the CS410 and the newest firmware on the Snom phones so I am not sure what the problem is. Any help would be greatly appreciated.
  12. Thanks for all your help. I have tried everything to get this to work and for some reason couldn't get it going. I was looking through the forums and came across a solution in another post. The user mentioned that he had the same problem and ended up changing the pbxnsip domain back to localhost from the fqdn domain name he was using. Since I had tried everything else, I gave this a shot too and voila it seemed to work. The reason I was looking through the other forum posts is because during my testing a found another error with my setup. When I dial into the pbx from an external number and pickup the call on one of the phones, I can hear the audio from the remote phone but they can't hear anything from me. If I dial out from the PBX this works fine. I know that normally this is considered a NAT problem but this used to work fine and nothing with my NAT setup has changed. I have an access-list on my firewall allowing everything through to the PBX and because I have several public IP addresses I have forwarded an entire IP through to the box so there should be nothing blocking it at all. The only thing that has changed is that I have upgraded my CS410 to the newest firmware software release. The logs see the calls being established but don't give any indication as to why they don't go through. Any ideas about this?
  13. I see the option to set who the call comes from. You're right, I didn't have anything set in this option. I set this option to one of the PBX extensions but I am still getting the error. I do have redirect enabled on the trunk. Does it matter which extension I use for this billing option and do I have to restart the pbx after I make this change? It seems like a pretty simple option but I am still getting the error that I don't have permission to transfer to the extensions.
  14. Ok, I am fine with hitting the checkmark button on the phone when I am making calls for now. All my phones are Snom phones so I will see what I can figure out about this on their site. The only other issue I am having with this Exchange setup is related to the Exchange Auto Attendant. I want to use Exchange for the AA because it supports dial by name and I have the mailboxes there. I setup the VoIP line to automatically go to the auto attendant of the Exchange server. When I dial into the line the Auto Attendant answers fine. The problem is, when I try to enter any extension to connect to a phone I get the error "We are sorry but you are not allowed to place this call". That happens regardless of which extension I dial through the system. The Exchange AA also gives you the option to press # after you dial an extension to go directly to the users voicemail. If I do this I transfer successfully to the mailbox so it seems like the routing within Exchange is working but I am not routing back out to the PBX. I have gone through the forum and the wiki looking for a solution to this but haven't been able to find any configuration information for working with the Exchange Auto Attendant. I used to use the PBXNSIP AA and it worked fine but would like to use the Exchange one now. Any help with this would be appreciated. Thanks
  15. Hi Again, I have been playing with this more and I think it seems to be working now. That last error seemed to be an Exchange problem but I am able to dial into the Exchange server now. I do have one question that isn't totally related to this but does apply somewhat. I now have the PBX setup to require the user to press the checkmark to dial any extension and that is ok for the most part but I have tried to get dialplans to work so that they dial the moment the number is typed. For example; if I dial 7100 I would like it to dial the moment the last digit is entered. The same thing goes if I enter a users extension, I have to press the checkmark to make the call go through. It seems that by changing the default scheme setting the phone never automatically dials and I have to press the checkmark to make the call go through. I have gone through the dialplan documentation but can't still seem to figure this out. Thanks for all your help.
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