gifti

I've set the general setting parameter http://wiki.snomone.com/index.php?title=From_To_Headers to "yes" and changed the account display format to "Lastname, Firstname" but I get the same result on the telephone.

On outgoing internal or external calls to destinations that are listed in the adressbook of the pbx, we only see the called number in the telephone display. But incoming internal or external calls are showing number AND name of the calling party. Internal Call from 21 to 20 in 21 DIsplay: < 20 > 21 My Name Internal Call from 20 to 21 in 21 Display < 20 Other Name > 21 My Name Vodia: 5.1.1 Snom 370: 8.7.3.25.5 regards gifti

It was the domain setting. Many thanks !!!

I bought 5 additional extension licenses for my snomone (mini) twenty "off the menu". After a license activation, 20/25 extensions shown in the general status (before 20/20). Then I've created a new extension. But the extension has a little "B" and can't registrate. Reboot could not solve the problem ... Status General Software-Version: 5.1.1 (snom ONE mini) Build Date: Aug 17 2013 11:37:15 License Status: Vodia PBX twenty (pbx.ggizef.lokal) 5D1-T3Y-CGT-xxx License Duration: Permanent Additional license information: Domains: 1/1, Calls: 0/10, G729A: 10, Extensions: 20/25, Attendants: 1/4, Callingcards: 0/2, Hunt Groups: 4/4, Paging Groups: 0/0, Service Flags: 2/20, IVR Nodes: 3/20, Agent Groups: 0/1, Conference Rooms: 1/2, CO Lines: 0/20, Adhoc Recording, Barge, Listen, Whisper, Trunk Accounting, Prepaid, Fax2Email Working Directory: /usr/local/snomONE MAC Addresses: 000413441xxx DNS Servers: 192.168.0.151 192.168.0.183 192.168.0.154 IPv4 Addresses: 127.0.0.1 192.168.0.200 CDR Records: Duration(30d): trunk = 1000, extension = 337, ivr = 1000 Calls: Total 1/1, Active 0/0 Calls SIP packet statistics: Tx: 3144 Rx: 3152 Emails: Successful sent: 6 Unsuccessful attempts: 0 Available file system space: 46% Uptime: 2014/10/30 16:05:05 (uptime: 0 days 00:37:28) (55673 59568-0) WAV cache: 0 Number of HTTP sessions: Sessions: 2; Threads: SIP=14, HTTP=5 Domain Statistics: Total Domains: 1, Total Accounts: 33 Status Domain On this page you find an overview on the domain in general and about the accounts used in the domain. More information can be found in the call log, the page for the current calls and in the overview on registered phones. Version: 5.1.1 (snom ONE mini) Extensions 20 Attendants 1 Conferences 1 Hunt Groups 4 Agent Group (Queues) 0 Calling Cards 0 Paging Groups 0 Service Flags 2 IVR Nodes 3 CO Lines 0 Registered Extensions 18 Total 32 regards gifti

Bei VOIP über ein ISDN Gateway funktioniert die Weiterleitung der original Nummer ohne Probleme. Man muss allerdings das kostenpflichtige Feature "CLIP no screening" aktiviert haben. Ich habe an den Trunkeinstellungen der snomONE nichts geändert. Diversion: {rfc} >> Standard(RFC). Es kommt hier auf die Einstellungen im Gateway an. Bei unserem Patton Gateway musste ich ein 3 stufige "Complex Function" bei abgehenden Calls hinterlegen. Sie überschreibt die ausgehende Nummer mit der Rufnummer des orginal Anrufers calling-e164 Of > calling-e164 To. Grüße Gifti

Good hint, thank you! IPV6 was disabled on my nic.

Good hint, thank you! Fortunatly, the network was reachable again. IPV6 was disabled on my nic.

I was able to find the ip with a network scanner. Everthing is fine. Phew ...

Hello, unfortunatly, I've deleted the reset-botton.sh script on my snomONE mini (backup system) and now I'm ultimately locked out. . Is there any way to bring the system back to life (send hardware to snom/vodia)? Can I reset an activate the license to another hardware at least? regards gifti

Hello, unfortunatly, I've deleted the reset-botton.sh script on my snomONE mini (backup system) and now I'm ultimately locked out. . Is there any way to bring the system back to life (send hardware to snom/vodia)? Can I save the license to another system (centOS VM) at least? regards gifti

I've tried it again and now it works!? But this "open pickup" is generally an uncertain method. You never know what you get . I'll should think about it. Thanks for supporting me!

- 5.1.1 (snom ONE mini) - phones are connected - picking the first ringing call from a huntgroup is no problem - when a call for the huntgroup is etablished, I can't pickup a concurrent ringing call on this huntgroup Example1 - good: - Huntgroup: 10 - Extensions in stage 1: 11, 12, 13 and 14 - incoming call to 10 - 11 to 14 are ringing simultaneously - I'm able to pickup the call from extension 15 Example2 - wrong: - Huntgroup: 10 - Extensions in stage 1: 11, 12, 13 and 14 - established call for huntgroup 10 connected on extension 11 - another incoming call to 10 at the same time - 12, 13 and 14 are ringing simultaneously - I can't pick the call from extension 15

Hello, I've got a huntgroup with 4 extensions behind (stage 1). When the first call for the hg is ringing, every other extension on the pbx is able to pick it up with *87 (works as desired). But the 2nd, 3rd, etc. call isn't pickable during a established call for this hg. You can answer concurrent calls only direct at the 4 huntgroup extensions. Is it possible to pickup concurrent calls on a huntgroup? Imo, that should work!? regards gifti

After updating firmware from 8.7.3.19: http://provisioning.snom.com:80/download/fw/snom370-8.7.3.19-SIP-f.bin to 8.7.3.25.5: http://provisioning.snom.com:80/download/fw/snom370-8.7.3.25.5-SIP-f.bin CFNA issues are resolved .

snomONE mini 4.5 $499 Upgrade to 5.x.x twenty standard $424.5 (50%) ------------------------------------------------- $923,5 ================================================= I've bought two minis and I'am very happy with it . Reliable, quiet and economical (power over ethernet).

Its is only when you call "remote" over the website and "call forward no answer xx seconds" is enabled. What is the difference to a normal "phone initiated" call? The first INVITE starts from the PBX and shoud not allow refer, doesn't it? INVITE sip:26@192.168.0.210:1752;transport=tcp;line=ya734foz SIP/2.0 Via: SIP/2.0/TCP 192.168.0.200:5060;branch=z9hG4bK-66360e544144f248b39928874c97d11a;rport From: <sip:26@pbx.ggizef.lokal>;tag=1845693600 To: <sip:xxxxxxxxxx@pbx.ggizef.lokal> Call-ID: 1b8dcdd5@pbx CSeq: 14894 INVITE Max-Forwards: 70 Contact: <sip:26@192.168.0.200:5060;transport=tcp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snomONE/5.1.1 Call-Info: <sip:26@pbx.ggizef.lokal>;answer-after=0 Content-Type: application/sdp Content-Length: 324 But I'am not a SIP expert. It isn't a serious error. Workaround -> disable CFNA before call remote. I wait for the new release .

Following issue with snomONE 5.1.1: - Call forward no answer (10sek) enabled for extension xy to number 123 - starting a new call over the website using javascript function "makecall" from extension xy to number 456 - after short ringing the connection to 456 is established for 10 seconds - "SIP/2.0 302 Moved Temporarily" SIP Request to 123 - call to 456 is abandoned - dood dood dood ... snomONE snom370 1 : |T-5060-------INVITE------1752->| 2 : |<-5060-100 Trying/INVITE-1752-T| 3 : |<-5060-180 Ringing/INVIT-1752-T| 4 : |T-5060------PRACK-1752-------->| 5 : |<-52266-STUN_BIND_REQUE-49196-U| 6 : |<-52266--audio/PCMU(0)--49196-U| 7 : |<-5060----200 Ok/PRACK---1752-T| 8 : |<-5060---200 Ok/INVITE---1752-T| 9 : |T-5060----------ACK------1752->| 10: |U-52266--audio/PCMU(0)--49196->| 11: |T-5060---------INVITE----1752->| 12: |<-5060---200 Ok/INVITE---1752-T| 13: |T-5060-------PRACK-------1752->| 14: |<-5060-----SUBSCRIBE-----1752-T| 15: |T-5060--------SIP/2.0----1752->| 16: |<-5060------REGISTER-----1752-T| 17: |T-5060--------SIP/2.0----1752->| 18: |<-5060-302 Moved Tempora-1752-T| 19: |T-5060----------ACK------1752->| 20: |<-5060--------BYE--------1752-T| 21: |T-5060--------SIP/2.0----1752->| 22: |<-5060-----SUBSCRIBE-----1752-T| 23: |T-5060--------SIP/2.0----1752->| When I try the same scenario with "Call forward no answer" disabled then the call is not abandoned. regards gifti

Thank you! But when I use tar to compress the directory (also without recordings and audio), the snomONE mini cpu is overloaded and the system is 5 minutes down . I've found another solution. I've copied the whole /usr/local/snomONE directory on a mounted share (once a day, this is cpu gently). This share is mounted on both systems. If snomONEmini1 fails, snomONEmini2 saves the whole directory back. I've moved the two pbxctrl_license.xml files in a higher directory and linked them symbolic to /usr/local/snomONE. So, they are not affected (cp -P). cd /usr/local/snomONE mv pbxctrl_license.xml ../ ln -s ../pbxctrl_license.xml ./ cp -fruvP /usr/local/snomONE /mnt/myshare regards gifti

Okay, please help me again! I know the root password and I'am logged in at #/usr/local/snomONE. Which files/directories should I select? Is there any script to do this? I don't want to lose/reset the license or something else. I have a second snomONE MINI as backup-system and would like to be able to synchronize the whole configuration between these two systems. Is it possible to mount the same /usr/local/snomONE directory on two systems with different licence keys? Scenario: snomONEmini #1 has a hardware defect snomONEmini #2 is now connected manually to network, mount the same /usr/local/snomONE directory -> configuration, cdr, addressbooks proceeding without gaps

Hello, the snomONE mini CPU performance exceeds 90% when I started to generate a backup file or import some addresses over the webinterface. During import/backup processes, the snomONE mini loses the connection to the phones, webinterface isn't reachable and incoming calls get lost . I've tried to reduce the bandwith of tar and the xml-generator pipe (pv -L 100k), but the scripts are not publicly available, aren't it? Can you fix that in a new release for snomONE mini? Could you post a script, to make the backup manually? Second problem, since 5.1.1 it isn't possible to make backups greater than 10MB. But I need this feature because the address book has more than 7000 records (~14MB tar archiv). Could you expand the dropdown "max_tar" on reg_settings.htm (1, 5, 10, 20MB)? regards gifti

Ahhh .... first call -> connect -> hold -> second call -> connect -> conference -> ready to speak . Is it possible to request the second call and the conference in one step without extra hold, like 'normal' phones do? I mean: first call -> connect -> conference -> second call -> connect -> ready to speak ?

Thank you for your support. But I think the problem were this tree lines in the snom_3xx_phone.xml. <setting_server perm="RW">http://192.168.0.200/prov/snom370.htm</setting_server> <pnp_config perm="RW">on</pnp_config> <settings_refresh_timer perm="RW">3600</settings_refresh_timer> I suppose, that when the snom370 processes these lines, it starts immediately a new provisioning request. The response contains these lines again and the phone starts a new request. And again ... and again ... By the way, we have no problems with 8.7.3.19 on snom370 !

Hello, we use PNP manual setup since 4.x versions, because our phones have static ip adresses. Now i've tried to register a new extension/snom370 with pnp: configured a static IP Adress to the new phone entered the setting url http://192.168.0.200/prov/snom370.htm and activate pnp created the extension and bounded it with the mac adress of the new phone opened the account for 10min provisioning reboot the phone After reboot, the phone repeats the provisioning request over and over again. [8] 2013/09/18 13:28:39: HTTP: file snom370.htm based on template snom_redirect.xml is sent to 192.168.0.202 [8] 2013/09/18 13:28:39: HTTP: Received request for file snom370-xxxxxxxxxxxx.htm from 192.168.0.202 [8] 2013/09/18 13:28:39: Provisioning file snom370-xxxxxxxxxxxx.htm looking for MAC xxxxxxxxxxxx [8] 2013/09/18 13:28:39: PnP: Using the credentials of 21@pbx.ggizef.lokal for file snom_370.xml [8] 2013/09/18 13:28:39: Generated PnP file: <?xml version="1.0" encoding="utf-8"?> <setting-files> <file url="https://192.168.0.200:443/prov/snom_3xx_fw.xml?model=snom370" /> <file url="https://192.168.0.200:443/prov/snom_3xx_phone-xxxxxxxxxxxx.xml?model=snom370" /> ... </setting-files> [8] 2013/09/18 13:28:39: HTTP: file snom370-xxxxxxxxxxxx.htm based on template snom_370.xml is sent to 192.168.0.202 [8] 2013/09/18 13:28:40: HTTP: Received request for file snom_3xx_fw.xml from 192.168.0.202 [8] 2013/09/18 13:28:40: HTTP: file snom_3xx_fw.xml based on template snom_3xx_fw.xml is sent to 192.168.0.202 [8] 2013/09/18 13:28:42: HTTP: Received request for file snom_3xx_phone-xxxxxxxxxxxx.xml from 192.168.0.202 [8] 2013/09/18 13:28:42: Provisioning file snom_3xx_phone-xxxxxxxxxxxx.xml looking for MAC xxxxxxxxxxxx [8] 2013/09/18 13:28:42: PnP: Using the credentials of 21@pbx.ggizef.lokal for file snom_3xx_phone.xml [8] 2013/09/18 13:28:42: Generated PnP file (first 2K): <?xml version="1.0" encoding="utf-8"?> <phone-settings> <user_active idx="1" perm="RW">on</user_active> <using_server_managed_dnd idx="1" perm="RW">on</using_server_managed_dnd> <user_realname idx="1" perm="RW">xxx xxx</user_realname> ... [8] 2013/09/18 13:28:42: HTTP: file snom_3xx_phone-xxxxxxxxxxxx.xml based on template snom_3xx_phone.xml is sent to 192.168.0.202 [8] 2013/09/18 13:28:42: HTTP: Received request for file snom370.htm from 192.168.0.202 [8] 2013/09/18 13:28:42: HTTP: file snom370.htm based on template snom_redirect.xml is sent to 192.168.0.202 [8] 2013/09/18 13:28:42: HTTP: Received request for file snom370-xxxxxxxxxxxx.htm from 192.168.0.202 [8] 2013/09/18 13:28:42: Provisioning file snom370-xxxxxxxxxxxx.htm looking for MAC xxxxxxxxxxxx [8] 2013/09/18 13:28:43: PnP: Using the credentials of 21@pbx.ggizef.lokal for file snom_370.xml [8] 2013/09/18 13:28:43: Generated PnP file: <?xml version="1.0" encoding="utf-8"?> <setting-files> <file url="https://192.168.0.200:443/prov/snom_3xx_fw.xml?model=snom370" /> <file url="https://192.168.0.200:443/prov/snom_3xx_phone-xxxxxxxxxxxx.xml?model=snom370" /> ... [8] 2013/09/18 13:28:43: HTTP: file snom370-xxxxxxxxxxxx.htm based on template snom_370.xml is sent to 192.168.0.202 [8] 2013/09/18 13:28:44: HTTP: Received request for file snom_3xx_fw.xml from 192.168.0.202 ... The snom370 display shows the new extension name but its not active. Repeating Prov.Request 1 ... During the request/repsonse of provisioning messages, i can't reach the webinterface of the phone. 5.0.10a (snom ONE mini) 8.7.3.19 (snom370) What could be the problem? regards gifti

I've tried to make a 3-way conference with the standard conference button on the phone. But nothing happens? Transfering with the transfer button works well ... Any ideas?

Hello snommies, 5.0.10a runs for 9 days and 2 hours without any "codec_list is empty" or "Unsupported Media" errors on our snomONE mini. A total 1200 successful inbound/outbound calls . Congratulations, the error seems to be resolved !!! Greetings gifti