Jump to content


  • Posts

  • Joined

  • Last visited

Everything posted by voipguy

  1. You didn't follow my instructions. I said "Place the "snom_D765.xml" file in the PBX \ html directory. If the html directory doesn't exist then create it." Both ver 4 and ver 5 have the /webpages directory, I never said place it in the /webpages directory. For the firmware files place them in the tftp directory and you can also place the firmware files in the /html directory and test to see if the phones will pull the firmware from the PBX tftp or html directories after you tell the PBX what path to get the firmware files from.
  2. Place the "snom_D765.xml" file in the PBX \ html directory. If the html directory doesn't exist then create it. That's how it was done in the ver 4.5 days, let me know if it works. I think you can upload the snom firmware files to the PBX \ tftp directory, if it doesn't exist then create it.
  3. Was anyone in the OS changing/updating the OS date/time? Or maybe a Linux system process is auto updating the system time? I've done this myself, I noticed my PBX time was off by like 3 minutes so I went into the PBX Linux and ran command to update the system time and then noticed the live calls got dropped and the call time length was like 10 hours for the call.....the call got dropped because my global max call time is set to 4 hours. Now I know to only update the OS/Linux time when no active/live calls are running.
  4. Hi Vodia PBX, So can you add this feature to the PBX? The limit your talking about doesn't help us at all because it's only if the customer makes a outbound call, we need a Domain Maximum call duration to control inbound & outbound. Thanks
  5. Hmmmm.....I didn't see this in the 5.2.2 release notes, If I hadn't read this post I wouldn't have know this new feature was added.
  6. If you don't want to wait for Vodia to add in the 10 hour setting I'm pretty sure you can do it via editing the web page and adding in the 10 option. I remember from back in the day when I needed longer time options for an ACD and Pradeep said you could just edit the web page and the software will use that new time setting.
  7. We have a hosted license. We have a mix of residential and business customers. All of our packages are "unlimited" talk time. Some of our residential customers like to call their spouse and stay on the phone after they have both fallen asleep - yes this happens a lot. We have the global setting "Maximum call duration:" set to 3 hours so it kills those types of calls but then our business customers in other Domains need longer call times. Using your method I don't think would even work - wouldn't the global setting "Maximum call duration:" set at 3 hours kill a call at the 3 hour mark even though you might have added outbound credits for a certain account? I think it would. Plus we don't use calling card accounts. Anyways that's not a solution for us or most other people that use this software, were not going to start developing some method of calling card accounts and distribute credits. We have been using the software from ver2 onwards and more and more of the "Global" settings have been moved into the "Domain" settings which makes the software more powerful. Please add my feature request to the wish list. Thanks
  8. Hi, Feature request - "Maximum call duration:" per Domain, not Global like it is now. Thanks
  9. Be careful with that feature - I believe it does more then reset their voicemail.
  10. The Record feature is a paid option - check your license to make sure you have it.
  11. So the pbx wont be using xml for its database anymore? What kind of database will it be using?
  12. Since ver 4 and up I think they use 2 cores, one core runs all the sip/calls and the second core handles other stuff like the web interface etc.
  13. voipguy

    DND grr

    Not sure what that "{ifn_button dnd none}" means.....I didn't modify any of that....I created a snom_7xx_custom.xml file and put in my 2 custom DND settings so when the snom 7XX phones do their PnP it won't add the *78/*79 to the Snom 7XX phones. I'm using ver 4.5 and I think your using ver 5.X.....not sure how much ver 5.X has changed or where you would setup a PNP setting to send a custom xml file to the 7XX phones.
  14. voipguy

    DND grr

    ...just for the 7XX series snom phones.....you don't want to globally set that because then all your snom 3XX and 8XX DND will stop working. ...it will still sync with the PBX but it wont try to make an audio call....meaning it won't try to dial *78 or *79 audio call which hangs on the snom 7XX phones. The DND will then work and sync....even if you turn on DND in the PBX web for a snom 7xx phone it will work and not hang the call.
  15. voipguy

    DND grr

    Is this only happening on the SNOM 7XX series phones? If yes, then I think my "fix" will solve the problem. See my thread: http://forum.snomone.com/index.php/topic/6670-snom-720-dnd-setting/ Do a quick test....in your SNOM 7XX phone web GUI under the Preference tab then find DND On: and the DND Off: remove the *78 and *79....save the settings....then press your SNOM 7X DND button and see if that fixes your problem. If it does then create a SNOM_7XX_Custom.xml file to send to your snom phones when they do PnP with custom DND settings like this...please note you need the 1 space between the > < : <dnd_on_code perm="RW"> </dnd_on_code> <dnd_off_code perm="RW"> </dnd_off_code>
  16. ver 5.1 release notes: "When creating an extension, there is no more field for passwords. Instead, the PBX is generated good passwords by default. Also, the PBX automatically opens the new extension for provisioning (unless disabled by a new settings)." Does this mean we can't choose our own passwords when creating an extension? This would be a major set back if it's true.
  17. Hi Snom ONE, Thanks for replying. I can't do the work around, I need to find out why it's happening - it's just a setting somewhere among the 200 plus settings in the TG784n. Can you tell me why snomone sends a notify msg to the TG784n ? It seems for all the failed calls it's when snomone sends the notify msg. Thanks,
  18. Hi, Hoping someone can help me with this or give me some ideas of what is causing this. Using snomone ver 4.5 the voip device is a Thomson TG784n firmware ver 8.43 This is a random problem. When you call the TG784N voip number from your cell or landline the TG784N will ring normally....but sometimes when I call the TG784n it goes straight to voicemail? The TG784n doesn't even ring - the person that has the TG784n doesn't even know somebody just tried to call them. I can see the TG784n is registered to my someone PBX. I know the TG784n is getting the call because I can see in the TG784n logs it is receiving the sip invite from my someone PBX. I did a wireshark trace on my PBX and I can see the TG784n did send back a 486 busy - but why? The TG784n was not in a call. This is 100% not a NAT issue. Here is the log from my TG784n for a failed call - goes straight to voicemail - because TG784n sends 486 to pbx: Info Jul 22 12:43:02 VOIP: [784n] [4165551234] [FXS1] INVITE - SIP message received Info Jul 22 12:43:02 VOIP: [784n] [4165551234] [FXS1] 401 Unauthorized - SIP message sent Info Jul 22 12:43:03 VOIP: [784n] [4165551234] [FXS1] ACK - SIP message received Info Jul 22 12:43:03 VOIP: [784n] [4165551234] [FXS1] INVITE - SIP message received Info Jul 22 12:43:04 FIREWALL event (1 of 3): created rules Info Jul 22 12:43:05 VOIP: [784n] [4165551234] [FXS1] 180 Ringing - SIP message sent Info Jul 22 12:43:07 VOIP: [784n] [4165551234] [FXS1] NOTIFY - SIP message received Info Jul 22 12:43:07 VOIP: [784n] [4165551234] [FXS1] 401 Unauthorized - SIP message sent Info Jul 22 12:43:07 VOIP: [784n] [4165551234] [FXS1] NOTIFY - SIP message received Info Jul 22 12:43:07 VOIP: [784n] [4165551234] [FXS1] 200 OK - SIP message sent Info Jul 22 12:43:27 VOIP: [784n] [4165551234] [FXS1] REGISTER - SIP message sent Info Jul 22 12:43:27 VOIP: [784n] [4165551234] [FXS1] 200 Ok - SIP message received Here is log from snomone - same call: [5] 2013/07/22 12:44:53: Identify trunk (IP address and DID match) 98 [9] 2013/07/22 12:44:53: Resolve 2473446: aaaa udp VoipProxyIP 5060 [9] 2013/07/22 12:44:53: Resolve 2473446: a udp VoipProxyIP 5060 [9] 2013/07/22 12:44:53: Resolve 2473446: udp VoipProxyIP 5060 [6] 2013/07/22 12:44:53: Call-leg 241: Sending RTP for 17566fdc17@VoipProxyIP to (ip removed):6210, codec not set yet [8] 2013/07/22 12:44:53: Incoming call: Request URI sip:4165551234@snomoneipaddress:5060, To is <sip:4165551234@snomoneipaddress:5060> [5] 2013/07/22 12:44:53: Domain trunk 507265@snomonedomainname sends call to 4165551234 in domain snomonedomainname [8] 2013/07/22 12:44:53: Set the To domain based on To user 4165551234@snomonedomainname [9] 2013/07/22 12:44:53: Resolve 2473447: url sip:4165551234@ipaddress:5060 [9] 2013/07/22 12:44:53: Resolve 2473447: udp ipaddress 5060 [8] 2013/07/22 12:44:54: Answer challenge with username 4165551234 [9] 2013/07/22 12:44:54: Resolve 2473448: udp ipaddress 5060 udp:1 [9] 2013/07/22 12:44:54: Resolve 2473449: udp ipaddress 5060 udp:1 [9] 2013/07/22 12:44:54: Message repetition, packet dropped [7] 2013/07/22 12:44:54: Call f15f0326@pbx: Clear last INVITE [9] 2013/07/22 12:44:54: Resolve 2473450: url sip:4165551234@ipaddress:5060 [9] 2013/07/22 12:44:54: Resolve 2473450: udp ipaddress 5060 [5] 2013/07/22 12:44:54: INVITE Response 486 Busy Here: Terminate f15f0326@pbx [6] 2013/07/22 12:44:54: Call-leg 241: Codec g729/8000 is chosen for call id 17566fdc17@VoipProxyIP [9] 2013/07/22 12:44:54: Resolve 2473451: aaaa udp VoipProxyIP 5060 [9] 2013/07/22 12:44:54: Resolve 2473451: a udp VoipProxyIP 5060 [9] 2013/07/22 12:44:54: Resolve 2473451: udp VoipProxyIP 5060 [8] 2013/07/22 12:44:54: Hangup: Call 242 not found [5] 2013/07/22 12:44:54: set codec: codec g729/8000 is set to call-leg 241 When the TG784n sends back " 401 Unauthorized - SIP message sent" my snomone pbx logs this "Answer challenge with username 4165551234" There must be a setting inside the TG784n that I can switch off that snomone doesn't like or maybe a setting in snomone that has to be changed? Things/patterns I've noticed: 1. Why does the TG784n send a 401 when snomone sends invite to it for a call? Yes snomone answer the nonce security question with the right answer and the call continues but why does the TG784n send the 401 in the first place? Even on a successful call where the TG784n phone does ring this initial 401 happens. On my Speedtouch 780wl firmware ver 7.4 which is the modem model before the 784N modem it doesn't send a sip 401 message back to the pbx and I don't see NOTIFY messages coming from PBX to TG784n. 2. When the call is successful on the TG784n I still see the sip 401 but I don't see the "NOTIFY - SIP message received" - the call works and the TG784n rings? It's almost like the TG784n is saying to snomone I'm busy 486 so I can't deal with your NOTIFY message that you just sent me, were in the middle of trying to setup a call here. 3. If I turn the voicemail box off in snomone and I dial the TG784n and the call fails I hear a busy signal because the TG784n sent a 486, when mailbox is enabled I don't hear busy signal and it goes straight to voicemail - I know this is how it's suppose to work when snomone gets a 486 back but why are we getting a 486 back from the TG784n? Any help would be appreciated. I've been working on this problem for over a week, I must have made a 100 test changes. Thanks
  19. http://snomone.com/alacarte Automatic Recording License for automatic recording of inbound and outbound calls USD 995.00
  20. You have to buy that feature in ver 5. Your license probably doesn't have it - that's why you can't see the settings.
  21. snomONE, Don't worry about the images going to the provisioned phones - that's more difficult and not important right now...the web interface for ver 5 branding is more important and easier to get working.
  22. I don't think you have to hit enter/checkmark......once you hit the Directory button it lists all the names....you then start entering the letters and you will see in the display it start narrowing down the list to the matches of the letters you entered.
  23. On ver this does not work. It won't record call forwarded external numbers.
  • Create New...