Hi. I am new on ASTPP.
I use the link([login to view URL]) to manual install the ASTPP.
I configure the trunks, rates, sip, gateways, etc. All are configured.
When I try to make a call, show me error on fs_cli and the call hangup:
2018-01-13 13:00:25.568487 [ERR] switch_odbc.c:368 STATE: IM002 CODE 0 ERROR: [unixODBC][Driver Manager]Data source name not found, and no default driver specified
2018-01-13 13:00:25.568487 [CRIT] switch_core_sqldb.c:508 Failure to connect to ODBC ASTPP!
2018-01-13 13:00:25.568487 [ERR] [login to view URL] Connection failed. DBH NOT Connected.
2018-01-13 13:00:25.568487 [DEBUG] [login to view URL] [ASTPP] Database connection fail...!!!
2018-01-13 13:00:25.568487 [DEBUG] [login to view URL] [ASTPP] [LOAD_CONF] Query :SELECT name,value FROM system WHERE group_title IN ('global','opensips','callingcard')
2018-01-13 13:00:25.568487 [ERR] [login to view URL] DBH NOT Connected.
2018-01-13 13:00:25.568487 [ERR] [login to view URL] /usr/local/freeswitch/scripts/astpp/lib/[login to view URL]: assertion failed!
stack traceback:
[C]: in function 'assert'
/usr/local/freeswitch/scripts/astpp/lib/[login to view URL]: in function 'load_conf'
/usr/local/freeswitch/scripts/astpp/[login to view URL]: in main chunk
2018-01-13 13:00:25.568487 [ERR] [login to view URL] LUA script parse/execute error!
The rest of the log are: ( I hidden the server ip with ).
2018-01-13 13:00:25.568487 [WARNING] mod_dialplan_xml.c:667 Context default not found
2018-01-13 13:00:25.568487 [INFO] switch_core_state_machine.c:311 No Route, Aborting
2018-01-13 13:00:25.568487 [NOTICE] switch_core_state_machine.c:312 Hangup sofia/default/3206821246@ [CS_ROUTING] [NO_ROUTE_DESTINATION]
2018-01-13 13:00:25.568487 [DEBUG] switch_core_state_machine.c:643 (sofia/default/3206821246@) State ROUTING going to sleep
2018-01-13 13:00:25.568487 [DEBUG] switch_core_state_machine.c:584 (sofia/default/3206821246@) Running State Change CS_HANGUP (Cur 1 Tot 3)
2018-01-13 13:00:25.568487 [DEBUG] switch_core_state_machine.c:850 (sofia/default/3206821246@) Callstate Change RINGING -> HANGUP
2018-01-13 13:00:25.568487 [DEBUG] switch_core_state_machine.c:852 (sofia/default/3206821246@) State HANGUP
2018-01-13 13:00:25.568487 [DEBUG] mod_sofia.c:438 Channel sofia/default/3206821246@ hanging up, cause: NO_ROUTE_DESTINATION
2018-01-13 13:00:25.568487 [DEBUG] mod_sofia.c:577 Responding to INVITE with: 404
2018-01-13 13:00:25.568487 [DEBUG] switch_core_state_machine.c:60 sofia/default/3206821246@ Standard HANGUP, cause: NO_ROUTE_DESTINATION
2018-01-13 13:00:25.568487 [DEBUG] switch_core_state_machine.c:852 (sofia/default/3206821246@) State HANGUP going to sleep
2018-01-13 13:00:25.568487 [DEBUG] switch_core_state_machine.c:619 (sofia/default/3206821246@) State Change CS_HANGUP -> CS_REPORTING
2018-01-13 13:00:25.568487 [DEBUG] switch_core_state_machine.c:584 (sofia/default/3206821246@) Running State Change CS_REPORTING (Cur 1 Tot 3)
2018-01-13 13:00:25.568487 [DEBUG] switch_core_state_machine.c:938 (sofia/default/3206821246@) State REPORTING
2018-01-13 13:00:25.568487 [INFO] mod_json_cdr.c:271 Process [[login to view URL]]
2018-01-13 13:00:25.568487 [DEBUG] switch_core_state_machine.c:174 sofia/default/3206821246@ Standard REPORTING, cause: NO_ROUTE_DESTINATION
2018-01-13 13:00:25.568487 [DEBUG] switch_core_state_machine.c:938 (sofia/default/3206821246@) State REPORTING going to sleep
In inbound calls, shows the same errors too, but the call stay ringing, but don't redirect to Customers Account.
On the Calls Report dont shows anything.
I need help for fix this errors and make the outbound and inbound calls work.
Hello,
seem you did not success to configure backend in server. i can have a check in all the configure and correct it,
It seem very easy to me. I can help you soon
thanks,
ThanhTruong
Hello. We are iKcon Infotech.
We have been dealing with ASTPP, A2Billing and many other OpenSource System from past several years.
We have reviewed your requirements and will resolve the issue.
Greetings! I will fix errors with ASTPP. We established and use our phone system based on AsteriskPBX. We have technical support team of 12 qualified System Administrators and 5 (five) of them are Senior System Administrators i.e. System Administrators with Level3 experience.
Contact me if interested.
Hello, Im VoIP Engineer with more than 8 years of expertise, im working with ASTPP since its Start, i can help you get it 100%, hit me on chat or skype: montana3601
My 7+ years of experience in VoIP industry and my expertise on VoIP Server Applications (Asterisk, ASTPP, FreePBX, FreeSwitch, A2Billing, GoAutoDial) makes me the best fit for this job. Please let me know if we can discuss further.