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ASTPP Fix errors

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.

Skills: Asterisk PBX, Linux, MySQL, PHP, VoIP

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About the Employer:
( 12 reviews ) Juiz de Fora, Brazil

Project ID: #16067213

11 freelancers are bidding on average $142 for this job

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voipmanvn

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

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iKcon

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.

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tsebro

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gilbertarias

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

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Newtechprogramer

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vikasroy63

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 dis More

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