chrispopp

I am also interested in the same setup. Did you figure it out? Trying to use something like this, but it doesn't work: {from-extension}{ext-ani}{from-trunk}{rpi}

Found some issues with the user web interface. 1. Logout button does not work. Console says: Uncaught TypeError: Cannot read property 'signOut' of null at Object.logout (utils.js?v=61.0:1) 2. In the address book. If updating/editing a contact, it creates a new one, instead of updating the old one. 3. Erasing a contact, the confirm window is blank, but the button works ok. 4. Logging in from the admin to the user level, the address book does not populate. But it works fine if you go directly at user level.

They're not modified... Seems it's a front end. Using the inspect and removing the "hide" value, I can turn make it so that it keeps it on the phone, but it still tries to send the email anyways.

when i was testing with 61

Seems I have found a bug with the voicemail to email. If you enable it for a user. Say you send the voicemail to email to mike@email.com. If the user changes his mind, and you set back the value (Send a mailbox message by email) to Do not send email. It still tries to send emails. Even if you remove the email. [8] 9:29:19.373 EMAI: Sending voicemail message [8] 9:29:19.373 EMAI: Preparing to send voicemail message [8] 9:29:19.374 EMAI: No valid destination address for sending email to user

Thanks, tested with the latest one, and still does the same.

My apologies... I now realized it! Thank you.

Sorry I don't seem to understand. The file cannot be uploaded onto the page in version 61 but in version 56 it works... I cannot find any documentation that shows how to deploy this automated message. Where would I be able to upload 2 files? Also, how can I upload a file onto a service flag?

Also tried it on an older version and the upload file under the page works...

Oh Yes, now I see it! I did upload the wav file and it plays fine under Upload file section when I click on the play button. I even uploaded it to an auto-attendant, and it works. But when I upload it via the web interface under the Paging, it gives me the check mark (OK) when I submit. If I exit the page and come back in (or refresh), it automatically reverts back to no file. Then if I refresh: If I ring the paging group, it disconnects me without paging.

What about the Playback method from a page group. If you select a WAV file and press save, it doesn't save it. And if you call it it gives a Forbidden. The extension is authorized. Also Live and Pre-Recorded methods work, just cannot play a wav file... Seems it's broken too This is the log error: [5] 16:30:14.036 APP: Paging: no static file associated with the paging group 165@test.reallycool.com 2018/8/24 16:30:14 Rx: tls:198.251.111.111:36243 (1361 bytes) INVITE sip:165@test.reallycool.com;user=phone SIP/2.0 Via: SIP/2.0/TLS 172.16.16.119:36243;branch=z9hG4bK-8k7vhrbqfhh7;rport From: "Name Test" <sip:299@test.reallycool.com>;tag=a8uqu88njj To: <sip:165@test.reallycool.com;user=phone> Call-ID: 313533353134323631323330353732-y5m9wv6rpvup CSeq: 1 INVITE Max-Forwards: 70 User-Agent: snom760/8.7.5.35 Contact: <sip:299@172.16.16.119:36243;transport=tls;line=3b4da0oi>;reg-id=1 X-Serialnumber: 00041371ZZZZ P-Key-Flags: resolution="31x13", keys="4" Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Session-Expires: 3600 Min-SE: 90 Proxy-Require: buttons Content-Type: application/sdp Content-Length: 504 v=0 o=root 2080425959 2080425959 IN IP4 172.16.16.119 s=call c=IN IP4 172.16.16.119 t=0 0 m=audio 61676 RTP/AVP 0 8 3 9 99 18 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:n8DPLX5e4V/3XGHIr4FQK0A3vre0JLDtTBrkeVEe a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:9 G722/8000 a=rtpmap:99 G726-32/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=rtcp-xr:voip-metrics stat-summary=loss,dup,jitt a=sendrecv 2018/8/24 16:30:14 Tx: tls:198.251.111.111:36243 (352 bytes) SIP/2.0 100 Trying Via: SIP/2.0/TLS 172.16.16.119:36243;branch=z9hG4bK-8k7vhrbqfhh7;rport=36243;received=198.251.111.111 From: "Name Test" <sip:299@test.reallycool.com>;tag=a8uqu88njj To: <sip:165@test.reallycool.com;user=phone>;tag=cbfc9bc1ff Call-ID: 313533353134323631323330353732-y5m9wv6rpvup CSeq: 1 INVITE Content-Length: 0 2018/8/24 16:30:14 Tx: tls:198.251.111.111:36243 (1072 bytes) SIP/2.0 200 Ok Via: SIP/2.0/TLS 172.16.16.119:36243;branch=z9hG4bK-8k7vhrbqfhh7;rport=36243;received=198.251.111.111 From: "Name Test" <sip:299@test.reallycool.com>;tag=a8uqu88njj To: <sip:165@test.reallycool.com;user=phone>;tag=cbfc9bc1ff Call-ID: 313533353134323631323330353732-y5m9wv6rpvup CSeq: 1 INVITE Contact: <sip:299@192.168.1.222:5062;transport=tls> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: Vodia-PBX/61.0 P-Asserted-Identity: "PAGING TEST" <sip:165@test.reallycool.com> Content-Type: application/sdp Content-Length: 394 v=0 o=- 47595 47595 IN IP4 192.168.1.222 s=- c=IN IP4 192.168.1.222 t=0 0 m=audio 26500 RTP/AVP 0 18 9 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:oEJMO6NyIETDa98ZNhunSOeewRkJS6QTxNq1YOS/ a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=rtcp-xr:rcvr-rtt=all voip-metrics a=sendrecv 2018/8/24 16:30:14 Tx: tls:198.251.111.111:36243 (528 bytes) BYE sip:299@172.16.16.119:36243;transport=tls;line=3b4da0oi SIP/2.0 Via: SIP/2.0/TLS 192.168.1.222:5062;branch=z9hG4bK-9cbe27be9ee99c11e467985b5ba0dd91;rport From: <sip:165@test.reallycool.com;user=phone>;tag=cbfc9bc1ff To: "Name Test" <sip:299@test.reallycool.com>;tag=a8uqu88njj Call-ID: 313533353134323631323330353732-y5m9wv6rpvup CSeq: 9218 BYE Max-Forwards: 70 Contact: <sip:299@192.168.1.222:5062;transport=tls> P-Asserted-Identity: "PAGING TEST" <sip:165@test.reallycool.com> Content-Length: 0 2018/8/24 16:30:14 Rx: tls:198.251.111.111:36243 (505 bytes) ACK sip:299@192.168.1.222:5062;transport=tls SIP/2.0 Via: SIP/2.0/TLS 172.16.16.119:36243;branch=z9hG4bK-ahgyn0v9e11k;rport From: "Name Test" <sip:299@test.reallycool.com>;tag=a8uqu88njj To: <sip:165@test.reallycool.com;user=phone>;tag=cbfc9bc1ff Call-ID: 313533353134323631323330353732-y5m9wv6rpvup CSeq: 1 ACK Max-Forwards: 70 User-Agent: snom760/8.7.5.35 Contact: <sip:299@172.16.16.119:36243;transport=tls;line=3b4da0oi>;reg-id=1 Proxy-Require: buttons Content-Length: 0

I see it says this on the vodia.com website (https://vodia.com/en/paging) I don't see any documentation... Scheduled playback. You may schedule paging announcements, for example for school bells or for playing out the good night story in a hospital.

Any news on this. We have several clients that want to be able to create scheduled announcements. For example a school that wants to have the ring bell at 8:45AM, 1:00PM... Even better, to play a recording such as a wavefile.

If I open the two extensions for MAC based provisioning, reboot the phone after, it picks up the correct extensions again. Rebooting the phone after the prov timeout happens, it registers the second extension twice again.

I erased all xml files. Still does it... First provision works fine. After reboot, one extension becomes duplicated. This is what gets generated in the "generated" folder. And this gets generated in the second identity, not the primary folder. <?xml version="1.0" encoding="utf-8"?> <phone-settings e="2"> <user_active idx="1" perm="RW">on</user_active> <using_server_managed_dnd idx="1" perm="RW">on</using_server_managed_dnd> <using_server_managed_fwd_all idx="1" perm="RW">on</using_server_managed_fwd_all> <using_server_managed_fwd_busy idx="1" perm="RW">on</using_server_managed_fwd_busy> <using_server_managed_fwd_time idx="1" perm="RW">on</using_server_managed_fwd_time> <user_realname idx="1" perm="RW">NameDisplay NameDisplay</user_realname> <user_idle_text idx="1" perm="RW">605: NameDisplay NameDisplay</user_idle_text> <user_name idx="1" perm="RW">605</user_name> <user_pname idx="1" perm="RW">00041371XXXX</user_pname> <user_pass idx="1" perm="RW">l04sxr</user_pass> <user_host idx="1" perm="RW">onetel.cooltel.com</user_host> <user_mailbox idx="1" perm="RW">605</user_mailbox> <user_srtp idx="1" perm="RW">on</user_srtp> <user_auth_tag idx="1" perm="RW">on</user_auth_tag> <user_symmetrical_rtp idx="1" perm="R">off</user_symmetrical_rtp> <user_auto_connect idx="1" perm="RW">off</user_auto_connect> <user_descr_contact idx="1" perm="RW">off</user_descr_contact> <user_xml_screen_url idx="1" perm="RW"></user_xml_screen_url> <user_outbound idx="1" perm="RW">sip:onetel.cooltel.com:5062;transport=tls</user_outbound> <user_dp_str idx="1" perm="RW">|sip:\1@\d|d|sip:\1@\d|d |sip:\1@\d|d |sip:\1@\d|d |sip:\1@\d|d \1@\d|d |^(611)$|sip:\1@\d|d |^(411)$|sip:\1@\d|d |^([2-9]{1}[0-9]{9})$</user_dp_str> <codec_priority_list idx="1" perm="RW">pcmu,pcma,gsm,g722,g726-32,g729,telephone-event</codec_priority_list> <codec_size idx="1" perm="RW">20</codec_size> <keyboard_lock_emergency idx="1"></keyboard_lock_emergency> <stun_server idx="1" perm="R"></stun_server> <user_dtmf_info idx="1" perm="R">off</user_dtmf_info> <user_server_type idx="1" perm="R">pbxnsip</user_server_type> <user_subscription_expiry idx="1" perm="R">3600</user_subscription_expiry> <stun_binding_interval idx="1" perm="R"></stun_binding_interval> <user_dynamic_payload idx="1" perm="R">off</user_dynamic_payload> <dfks idx="1" perm="RW">on</dfks> <accept_event_talk_without_sdp idx="1" perm="RW">on</accept_event_talk_without_sdp> <record_missed_calls idx="1" perm="RW">on</record_missed_calls> <timezone perm="RW">USA-5</timezone> <dnd_on_code perm="RW"></dnd_on_code> <dnd_off_code perm="RW"></dnd_off_code> <utc_offset perm="RW">-18000</utc_offset> <dst perm="RW">3600 03.02.07 02:00:00 11.01.07 02:00:00</dst> <http_user perm="RW">admin</http_user> <http_pass perm="RW">coolpass</http_pass> <http_scheme perm="RW">off</http_scheme> <http_client_user perm="RW">00041371XXXX</http_client_user> <http_client_pass perm="RW">l04oisxr</http_client_pass> <with_flash perm="RW">off</with_flash> <language perm="RW">English</language> <web_language perm="RW">English</web_language> <tone_scheme perm="RW">USA</tone_scheme> <time_24_format perm="RW">on</time_24_format> <date_us_format perm="RW">off</date_us_format> <dialnumber_us_format perm="RW">off</dialnumber_us_format> <cw_dialtone perm="RW">off</cw_dialtone> <multicast_listen perm="RW">on</multicast_listen> <mc_address idx="1" perm="RW"></mc_address> <mc_address idx="2" perm="RW"></mc_address> <mc_address idx="3" perm="RW"></mc_address> <mc_address idx="4" perm="RW"></mc_address> <mc_address idx="5" perm="RW"></mc_address> <mc_address idx="6" perm="RW"></mc_address> <mc_address idx="7" perm="RW"></mc_address> <mc_address idx="8" perm="RW"></mc_address> <mc_address idx="9" perm="RW"></mc_address> <mc_address idx="10" perm="RW"></mc_address> <codec_tos perm="RW">ef</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">pool.ntp.org</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>1234</admin_mode_password> <admin_mode_password_confirm>1234</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>onetel.cooltel.com</ldap_server> <ldap_port>389</ldap_port> <ldap_base>ou=people</ldap_base> <ldap_username>onetel.cooltel.com\605</ldap_username> <ldap_password>if11rty3c</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>605@onetel.cooltel.com</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> <dkey_directory perm="RW">keyevent F_DIRECTORY_SEARCH</dkey_directory> <dkey_menu perm="RW">url http://onetel.cooltel.com:80/snom/menu.xml?auth=basic</dkey_menu> <gui_fkey1>keyevent F_DIRECTORY_SEARCH</gui_fkey1> <gui_fkey2>keyevent F_CALL_LIST</gui_fkey2> <gui_fkey3>keyevent F_REDIRECT</gui_fkey3> <gui_fkey4>keyevent F_SETTINGS</gui_fkey4> <dfks perm="RW">on</dfks> <display_method>display_name_number</display_method> <call_screen_fkeys_on_connected perm="RW">F_TRANSFER F_HOLD F_REC F_DUAL_AUDIO F_CONF_ON F_NEXT_CALL_SCREEN</call_screen_fkeys_on_connected> </phone-settings>

Genius! Thank you.

The reason for this, is to have two different servers and use the alias as a fallback DNS in case the main one fails. {domain} is the main one... I need the alias.

Anything?

I'm testing with a Snom 760. I added the MAC address to two extensions. Open both extensions for MAC Based Provisioning. The phone provisions both identities perfectly. BUT after a reboot, the phone creates two of the same identity number. So if I provision extension 201 and 202. It works fine, but after the reboot, the extensions on the phone are 202 and 202. This is on the latest version.

Checked this link: https://vodia.com/documentation/phone_provisioning_variables I want to be able to provision the domain alias. How would I be able to do that? It's not on the list. for example, say my pbx has 2 domains: pbx1.coolesttel.com and the alias pbx2.worstltel.com. I would like to be able to provision the second one.

Changed the API address for Snom to: https://provisioning.snom.com:8083/xmlrpc But it still does not work. [8] 15:29:45.217 PROV: RPS sub 000413XXXXXX [8] 15:29:45.217 PROV: Other request pending

When do you think they will be supported?

How do you provision the Aastra 6739i over WAN? There's a provision URL but don't know which URL to add..

From the logs it seems that the vodia send s the information correctly, it's the SRAPS that is not adding the profile... Opened a ticket with Snom and they don't seem to know where the problem is.

Wouldn't this make the service unusable? What's the point if we cannot redirect it back to our service?