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About shigeru

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  1. Hello! Reinstating the banner graphics and restarting the unit worked out. Thanks.
  2. I am using VLC to stream a net radio. Is the following setting not compatible? :sout=#transcode{vcodec=none,acodec=s16l,ab=64,channels=1,samplerate=8000}:rtp{dst=192.168.x.x,port=25000} :sout-keep
  3. My logo banner is green, but does not show the logo banner image at all.
  4. Hello! The MoH with RTP streaming is still not working with 5.2.2.
  5. I get this way with Chrome, Safari and IE, well with Chrome I get WebRTP working. So it is kind of universal. And it is still this way with 5.2.2. The very disturbing problem is that the user can hardly see the Home | Help | Logout link on top right. Hope you have it fixed very soon. Regards,
  6. Hello! I set the logos and custom banners all over, but I get this on WAC. Could you tell me what is going on? Software-Version: 5.2.1 (Vodia mini PBX) Build Date: Mar 4 2014 11:30:31 License Status: Vodia PBX premium 10 (Vodia mini PBX (xxxxxxxxxxxx)) xxx-xxx-xxx-xxx License Duration: Permanent Additional license information: Domains: 1/1, Calls: 0/10, G729A: 5, Extensions: 7/10, Attendants: 1/10, Callingcards: 1/1, Hunt Groups: 4/10, Paging Groups: 1/10, Service Flags: 2/10, IVR Nodes: 0/10, Agent Groups: 1/10, Conference Rooms: 1/10, CO Lines: 0/10, Adhoc Recording, WebRTC support, Fax2Email Working Directory: /pbx MAC Addresses: xxxxxxxxxxxx xxxxxxxxxxxx DNS Servers: IPv4 Addresses:,x.x.x.x,x.x.x.x CDR Records: O=40, T=18, E=40, I=24 Calls: total: 30/10 current: 0/0 SIP packet statistics: Tx: 28236 Rx: 28183 Emails: Successful sent:7 Unsuccessful attempts:83 Available file system space: 94% Uptime: 23:36:57 Memory=53523K Number of HTTP sessions: 6 (2 requests) Domain Statistics: 1 (19 users)
  7. Hello! I have found it, the default "Maximum Number of Calls" was set to 0 and that was the issue. Thanks.
  8. Hello! I think RTP streaming also has a bug. It only plays for the initial moment of transfer to the orbit and rest of time time remains silent. Thanks.
  9. Hello! I have been working on the configuration migration from CentOS to Vodia mini PBX. And finally managed to get all the configuration copied over, I think I did it right. But when I hooked one Snom870 with PnP provisioning, it registers, but all the call out will be rejected with the log entry below. [2] 19:04:51.388 GENE: Port 3: Reject call because of too many calls in domain The phone says "Service Unavailable" though all trunks are registered for calling out and in. What seems to be the problem here? It has only one phone registered to the PBX, and it is licensed to 10 extensions. Thanks.
  10. Hello! I looked into the snom forum and found that this issue is to do with their new firmware. Sorry to have bothered you.
  11. Hello! After updating snomONE to 5.1.3, the PnP provisioned snom870s stopped making a ringtone sound when receiving calls. When I checked Preference settings I find the ringtone provisioned properly. The phone lights and display indicates properly, the phone just does not make any noise when called. Please help. Thanks. Below is the content of snom_3xx_phone.xml; <?xml version="1.0" encoding="utf-8"?> <phone-settings>{loop-start 1} <user_active idx="{lc}" perm="RW">on</user_active> <using_server_managed_dnd idx="{lc}" perm="RW">on</using_server_managed_dnd> <user_realname idx="{lc}" perm="RW">{display-name}</user_realname> <user_idle_text idx="{lc}" perm="RW">{account}: {display-name}</user_idle_text> <user_name idx="{lc}" perm="RW">{account}</user_name> <user_pname idx="{lc}" perm="RW">{account}</user_pname> <user_host idx="{lc}" perm="RW">{domain}</user_host> <user_pass idx="{lc}" perm="RW">{sip-pass}</user_pass> <user_mailbox idx="{lc}" perm="RW">{account}</user_mailbox> <user_srtp idx="{lc}" perm="RW">on</user_srtp> <user_auth_tag idx="{lc}" perm="RW">on</user_auth_tag> <user_symmetrical_rtp idx="{lc}" perm="R">off</user_symmetrical_rtp> <user_auto_connect idx="{lc}" perm="RW">off</user_auto_connect> <user_descr_contact idx="{lc}" perm="RW">off</user_descr_contact> <user_xml_screen_url idx="{lc}" perm="RW">{https-url tftp}/snom870-idle-screen-without-image.xml</user_xml_screen_url> <user_proxy_require idx="{lc}" perm="RW">buttons-{attribute model}</user_proxy_require> <user_outbound idx="{lc}" perm="RW">{outbound-proxy snom_transport}</user_outbound> <user_dp_str idx="{lc}" perm="RW">{dialplan snom}</user_dp_str> <codec_priority_list idx="{lc}" perm="RW">pcmu,pcma,gsm,g722,g726-32,g729,telephone-event</codec_priority_list> <codec_size idx="{lc}" perm="RW">20</codec_size> <stun_server idx="{lc}" perm="R"></stun_server> <user_dtmf_info idx="{lc}" perm="R">off</user_dtmf_info> <user_server_type idx="{lc}" perm="R">pbxnsip</user_server_type> <user_subscription_expiry idx="{lc}" perm="R">3600</user_subscription_expiry> <stun_binding_interval idx="{lc}" perm="R"></stun_binding_interval> <user_dynamic_payload idx="{lc}" perm="R">off</user_dynamic_payload> <dfks idx="{lc}" perm="RW">on</dfks> <record_missed_calls idx="{lc}" perm="RW">on</record_missed_calls>{loop-end} <dkey_snom perm="RW">url {http-url snom}/menu.xml?auth=basic</dkey_snom> <dkey_directory perm="RW">keyevent F_DIRECTORY_SEARCH</dkey_directory> <dkey_menu perm="RW">keyevent F_MENU</dkey_menu> <dkey_record perm="RW">url {http-url snom}/recording.xml?auth=basic</dkey_record> <timezone perm="RW">USA</timezone> <dnd_on_code perm="RW"> </dnd_on_code> <dnd_off_code perm="RW"> </dnd_off_code> <utc_offset perm="RW">{tz gmt-offset}</utc_offset> <dst perm="RW">{tz dst-snom}</dst> <http_user perm="RW">{admin-user}</http_user> <http_pass perm="RW">{admin-pass}</http_pass> {ifn_parm mac} <http_client_user perm="RW">{mac}</http_client_user> <http_client_pass perm="RW">{mac-hash}</http_client_pass> {fin_parm mac} <with_flash perm="RW">off</with_flash> <language perm="RW">{enum extension lang_audio English sp=Espanol de=Deutsch fr=Francais nl=Nederlands ru=Russian dk=Dansk it=Italiano pl=Polski tr=Turkce cn=Chinese}</language> <web_language perm="RW">{enum extension lang_web English sp=Espanol de=Deutsch fr=Francais nl=Nederlands ru=Russian dk=Dansk it=Italiano pl=Polski tr=Turkce cn=Chinese}</web_language> <tone_scheme perm="RW">{enum domain lang_tones USA au=AUS at=AUT ch=CHN dk=DNK fr=FRA de=GER uk=GBR in=IND it=ITA jp=JPN mx=MEX nl=NLD no=NOR nz=NZL sp=ESP sw=SWE ch=SWI}</tone_scheme> <time_24_format perm="RW">{enum extension lang_audio on en=off uk=on}</time_24_format> <date_us_format perm="RW">{enum extension lang_audio off en=on uk=on}</date_us_format> <cw_dialtone perm="RW">{enum extension lang_audio off de=on}</cw_dialtone> <multicast_listen perm="RW">on</multicast_listen> <mc_address idx="1" perm="RW">{multicast-adrport 0}</mc_address> <mc_address idx="2" perm="RW">{multicast-adrport 1}</mc_address> <mc_address idx="3" perm="RW">{multicast-adrport 2}</mc_address> <mc_address idx="4" perm="RW">{multicast-adrport 3}</mc_address> <mc_address idx="5" perm="RW">{multicast-adrport 4}</mc_address> <mc_address idx="6" perm="RW">{multicast-adrport 5}</mc_address> <mc_address idx="7" perm="RW">{multicast-adrport 6}</mc_address> <mc_address idx="8" perm="RW">{multicast-adrport 7}</mc_address> <mc_address idx="9" perm="RW">{multicast-adrport 8}</mc_address> <mc_address idx="10" perm="RW">{multicast-adrport 9}</mc_address> <codec_tos perm="RW">{global tos_rtp}</codec_tos> <register_http_contact>on</register_http_contact> <update_policy perm="RW">auto_update</update_policy> <challenge_response perm="RW">off</challenge_response> <ntp_server perm="RW">{tz ntp-adr}</ntp_server> <block_url_dialing perm="RW">on</block_url_dialing> <transfer_on_hangup perm="RW">off</transfer_on_hangup> <ignore_security_warning perm="RW">on</ignore_security_warning> <answer_after_policy perm="RW">idle</answer_after_policy> <aoc_amount_display perm="RW">charged</aoc_amount_display> <admin_mode_password>{admin-pin}</admin_mode_password> <admin_mode_password_confirm>{admin-pin}</admin_mode_password_confirm> <cancel_desktop>on</cancel_desktop> <rtcp_xr>voip-metrics stat-summary=loss,dup,jitt</rtcp_xr> <auto_connect_indication_tone>off</auto_connect_indication_tone> <ldap_lookup_ringing>off</ldap_lookup_ringing> <ldap_sort_results>on</ldap_sort_results> <ldap_search_filter>(|(sn=%)(gn=%))</ldap_search_filter> <ldap_number_filter>(|(telephoneNumber=%)(mobile=%))</ldap_number_filter> <ldap_name_attributes>cn sn givenName</ldap_name_attributes> <ldap_number_attributes>telephoneNumber mobileTelephoneNumber</ldap_number_attributes> <ldap_display_name>%cn</ldap_display_name> <ldap_predict_text>off</ldap_predict_text> <perform_initial_query_in_ldap_state>on</perform_initial_query_in_ldap_state> <ldap_server>{ip-adr}</ldap_server> <ldap_port>{ldap-port}</ldap_port> <ldap_base>ou=people</ldap_base> <ldap_username>{domain}\{account}</ldap_username> <ldap_password>{web-pass}</ldap_password> <ldap_max_hits>50</ldap_max_hits> <xml_notify>on</xml_notify> <allow_rtp_on_mute>on</allow_rtp_on_mute> <phone_name>{account}@{domain}</phone_name> <admin_mode perm="">off</admin_mode> <attended_transfer_on_ringing>on</attended_transfer_on_ringing> <prioritise_pbx_number_lookup>off</prioritise_pbx_number_lookup> <led_on perm="RW">ON BUSY IN_A_CALL CALLING IN_A_MEETING URGENT_INTERRUPTIONS_ONLY DND UNAVAILABLE ACTIVE INACTIVE BE_RIGHT_BACK AWAY SEIZED CONNECTED ON_HOLD OFFHOOK RINGBACK I-Am-Ready AVAILABLE I-Am-Busy PhoneHasCall PhoneHasMissedCalls CurrentIdentityHasVoiceMessages PhoneHasVoiceMessages Red.on Green.on Orange.on</led_on> <led_blink_slow perm="RW">PARKED HOLDING I-Am-Almost-Ready PhoneHasCallInStateHolding Red.holding Green.holding Orange.holding Red.parked Green.parked Orange.parked</led_blink_slow> <led_blink_medium perm="RW">RECORDING MESSAGE Red.recording Green.recording Orange.recording Red.message Green.message Orange.message</led_blink_medium> <led_blink_fast perm="RW">RINGING PICKUP PhoneHasCallInStateRinging Red.pickup Green.pickup Orange.pickup</led_blink_fast> <led_red perm="RW">BUSY IN_A_CALL CALLING IN_A_MEETING URGENT_INTERRUPTIONS_ONLY HOLDING DND I-Am-Busy Red.on Red.off Red.pickup Red.park Red.message</led_red> <led_green perm="RW">AVAILABLE I-Am-Ready I-Am-Almost-Ready Green.on Green.off Green.pickup Green.park Green.message</led_green> <led_orange perm="RW">Orange.on Orange.off Orange.pickup Orange.park Orange.message</led_orange> <dfks perm="RW">on</dfks> <display_method>display_name_number</display_method> <ignore_security_warning>on</ignore_security_warning> <dim_timer perm="RW">604799</dim_timer> </phone-settings>
  12. Hmm... We have been using the +[number] notation for almost all the electronic phone book just to unify and simplify matters. So some users are used to omit the preceding + for domestic numbers when typing, others place the 0 instead of + for every calls. So these kind of changes without explicitly mentioned in the release note potentially cause big hustle. We have configured the PBX Dial Plan to correspond the user input and select the trunk accordingly. Anyway, the issue has been resolved for now.
  13. It seems that one have to modify the dialplans.xml file, there is no GUI way of addressing this, correct?
  14. Well, what I am looking for is a way to stop snom phones from stripping the 1 in the 11 digit user input. Are you telling me that the new version 5.1.x os snomONE pbx does not have such capability?
  15. Well, then why is this change at this time from 5.0.x to 5.1.x in provisioning? Also, why the phone have to be provisioned to remove the first digit of user input, anyway? I would like to know what I have to do to stop the phone from eliminating the first digit before forwarding the inquiry to snomONE. We have our dial plan that is heavily relying on the phones not cutting out user input. Thanks.
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