Asterisk handles incoming in error

[nimeth (Nimeth Vorng)]

Hi there,

I’m having high-level of configuration problem with my FreePBX Distro - CentOS 7 installed on different network of a LAN that supposed to receive incoming call from another asterisk server version 11 on another network.

Below is simple trunk setting using GUI.
type=friend
qualify=yes
port=5060
insecure=port,invite
host=xxx.xxx.xxx.xxx
dtmfmode=rfc2833
disallow=all
context=9292-ivr
canreinvite=no
allow=ulaw
;username=userid
;secret=password
;context=from-trunk

Status of the sip trunk seems ok as below
Name/username Host Dyn Forcerport Comedia ACL Port Status Description
1000 (Unspecified) D No No A 0 UNKNOWN
CamGSM 172.16.100.117 Yes Yes 5060 OK (1 ms)

Inbound route configured as screen shot below.

My custom destination was setup as below.

My dialplan was setup to handle incoming call with DID 9292 as below.

When I called to a number that supposed to be routed by another asterisk server to this asterisk server with ‘9292’ as DID, the call seemed reach this server because I could the caller number xxxxxxxxx but ended immediately and this asterisk server logged error as below.
[2017-12-13 16:22:35] NOTICE[169955]: res_pjsip/pjsip_distributor.c:649 log_failed_request: Request ‘OPTIONS’ from ‘“asterisk” sip:[email protected]’ failed for ‘172.16.100.117:5060’ (callid: [email protected]:5060) - No matching endpoint found
[2017-12-13 16:22:40] NOTICE[169955]: res_pjsip/pjsip_distributor.c:649 log_failed_request: Request ‘INVITE’ from ‘“012551622” sip:[email protected]’ failed for ‘172.16.100.117:5060’ (callid: [email protected]:5060) - No matching endpoint found

Please somebody help me!

Thanks in advance.
Nimeth

[danielf]

Hi @nimeth,

Switch your trunk to a chan_sip trunk and not a chan_pjsip trunk.
It would be easier to integrate it with your other Asterisk server.

Do not forget to keep an eye on the ports of the chan_sip (usually 5061 with pjsip enabled).

Thank you,

Daniel Friedman
Trixton LTD.

[lgaetz (Lorne Gaetz)]

I agree with Daniel, you appear to have mixed up your chan_SIP/PJSIP ports. If you are using chan_sip trunks, make sure you are using to the chan_sip bind port.

[nimeth (Nimeth Vorng)]

Thank you all for your replay and sorry for being quiet for 2 weeks due to my intensive mission to finish another project during this December and just finish end-user training yesterday.

With kind support from a guy at the other asterisk server, the incoming calls could finally reach our IVR app. One of the things he looked at, is edit of pjsip.endpoint.conf as below.

[anonymous]
type=endpoint
context=9292-ivr
allow=ulaw,alaw,gsm
transport=0.0.0.0-udp
;transport=udp,tcp,ws,wss

Thanks again all for your inputs.
Nimeth

[nimeth (Nimeth Vorng)]

Hey,

Because incoming calls are able to reach their destination (IVR App) so, on my Asterisk server, I continued to add Outbound Route usingFreePBX GUI, but when I submit, it threw error below.

PDOException (42S22)
SQLSTATE[42S22]: Column not found: 1054 Unknown column 'time_mode' in 'field list'

What to do so we can add outbound route?

Nimeth