Hi there, I want to test inbound and outbound call follow on ASTPP .
Inbound: ISP → ASTPP → Sip device in ASTPP or Call center
Example: Call to 02xxxxxx → DIDs → Sip device of Customer (111)
Outbound: Sip device in ASTPP → ISP
Example: 111 dial 02xxxxx or 842xxxxx
But any way I trying to call in or call out, I will get error No route to destination.
Please help me fix it :(((
How about posting the freeswitch logs also?
The answer will be in your freeswitch log. You can also watch it in real time from the fs_cli console. It probably cannot find a matching a rate to the number you are dialing.
Hi Tai36, I notice that the number you dialed is 11 digits long and the display number is 10 digits. Is this correct? Are you is Aus.?
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 210.211.108.3;branch=z9hG4bKfVX6Saqyu~fHa;rport=5060
From: “0964348489” sip:0964348489@125.235.38.182;tag=8HQQ7gHa6ecKg
To: sip:02862856569@210.211.108.3:5060
Call-ID: 94df6ac3-c6ff-123c-b6b8-941882812ba8
CSeq: 72429792 INVITE
User-Agent: ASTPP
Content-Length: 0
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 210.211.108.3;branch=z9hG4bKfVX6Saqyu~fHa;rport=5060
Max-Forwards: 69
From: “0964348489” sip:0964348489@125.235.38.182;tag=8HQQ7gHa6ecKg
To: sip:02862856569@210.211.108.3:5060;tag=30tS83ZmUvZKK
Call-ID: 94df6ac3-c6ff-123c-b6b8-941882812ba8
CSeq: 72429792 INVITE
User-Agent: ASTPP
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: path, replaces
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Reason: Q.850;cause=3;text=“NO_ROUTE_DESTINATION”
Content-Length: 0
ACK sip:02862856569@210.211.108.10:5060 SIP/2.0
Via: SIP/2.0/UDP 210.211.108.3;branch=z9hG4bKfVX6Saqyu~fHa;rport
From: “0964348489” sip:0964348489@125.235.38.182;tag=8HQQ7gHa6ecKg
To: sip:02862856569@210.211.108.3:5060;tag=30tS83ZmUvZKK
Call-ID: 94df6ac3-c6ff-123c-b6b8-941882812ba8
CSeq: 72429792 ACK
Content-Length: 0
The log of freeswitch has been updated below
Hi neilsheppard, both numbers are syntactically correct, 10 digits is the phone number and 11 digits is the landline phone number and I come from Vietnam. Can you help me check this issue?
@tai36 , it seems you don’t have ASTPP DB profile loaded, share output of ‘sofia status’ from fs_cli, other thing is check the response of commaond ‘mod_exists mod_lua’ in fs_cli
You still have not posted freeswitch log data. The info you posted is SIP debug info. That will not show the problem so you should disable that. You want fs_cli data which shows ASTPP db access where it tries to match rates to phone numbers as you are making calls. 99% of the time that is the problem and you will see a message in there saying it can’t find a match. This is assuming you have the basics set up correctly such as your SIP trunk.
Oh, How can I display this information you need to check? Please give me step by step to help me show it.
Hi Knerd, I get same issue with you after I fix error : “cannot open /usr/share/freeswitch/scripts/astpp/astpp.lua: No such file or directory” by cp file to /usr/share/freeswitch/scripts/astpp. Besides reinstalling from scratch, is there any way to fix this error directly so I can continue using it? My ASTPP is version 6.
It seems ODBC is not installed or not configured properly. Look at /etc/odbc.ini.
I tested according to your instructions and this is the result I got
Looks like maybe you have not installed the ODBC connector package.
apt install mysql-connector-odbc.
Test odbc driver
odbcinst -s -q
Test odbc connection
isql -v astpp astppuser ${USER_PASSWORD}
quit
Same issue here. I tried to follow all the suggestions and seems that I have my odbc package running and connecting Astpp just fine. Still “Context default not found, No Route, Aborting” error.
SIP gateway and Trunk provider connects perfectly. I just can’t place a call.
2023-10-hhhhh 23:20:46.326412 94.40% [INFO] switch_cpp.cpp:1465 [ASTPP] Product ID : 1
2023-10-hhhhh 23:20:46.326412 94.40% [INFO] switch_cpp.cpp:1465 [ASTPP] Package Type : 1 Call Direction : outbound [0:inbound,1:outbound,2:both]
2023-10-hhhhh 23:20:46.326412 94.40% [DEBUG] switch_cpp.cpp:1465 [ASTPP] [GET_COUNTER_INFO] Query :SELECT used_seconds FROM counters WHERE accountid = 2 AND package_id = 1 AND status=1 LIMIT 1
2023-10-hhhhh 23:20:46.326412 94.40% [INFO] switch_cpp.cpp:1465 [ASTPP] Remaining minutes : 1500
2023-10-hhhhh 23:20:46.326412 94.40% [ERR] mod_lua.cpp:202 /usr/share/freeswitch/scripts/astpp/lib/astpp.functions.lua:510: attempt to compare nil with number
stack traceback:
/usr/share/freeswitch/scripts/astpp/lib/astpp.functions.lua:510: in function 'package_calculation'
...hare/freeswitch/scripts/astpp/scripts/astpp.dialplan.lua:193: in main chunk
[C]: in function 'dofile'
/usr/share/freeswitch/scripts/astpp/astpp.lua:85: in main chunk
2023-10-hhhhh 23:20:46.326412 94.40% [ERR] mod_lua.cpp:269 LUA script parse/execute error!
2023-10-hhhhh 23:20:46.326412 94.40% [WARNING] mod_dialplan_xml.c:669 **Context default not found**
2023-10-hhhhh 23:20:46.326412 94.40% [INFO] switch_core_state_machine.c:306 **No Route, Aborting**
2023-10-hhhhh 23:20:46.326412 94.40% [NOTICE] switch_core_state_machine.c:307 Hangup sofia/default/100123@myserver1 [CS_ROUTING] **[NO_ROUTE_DESTINATION**]
2023-10-hhhhh 23:20:46.326412 94.40% [DEBUG] switch_core_state_machine.c:640 (sofia/default/100123@myserver1) State ROUTING going to sleep
I just fixed by removing the astpp database and re-imported it (I followed this individual steps from:
under the part “Configure MySQL” only)