I recently set up freepbx for google voice on a raspberry pi 3B. I followed xekon’s wiki (and another similar freepbx wiki) and was able to install and set up everything. However, I have tried to call out but I receive an error message stating that the call cannot be made because of a 503 error. When I use ekiga to call my extension in asterisk, ekiga says that the call was successfully made.
Here are asterisk cli messages and call logs:
Asterisk CLI: (Asterisk version: 13.20.0)
<— Received SIP request (673 bytes) from UDP:192.168.1.78:5060 —>
SUBSCRIBE sip:[email protected]:5160 SIP/2.0
Call-ID: [email protected]
Content-Length: 0
CSeq: 8002 SUBSCRIBE
From: <sip:[email protected]>;tag=SP12207e046…
Max-Forwards: 70
To: <sip:[email protected]>
Via: SIP/2.0/UDP 192.168.1.78:5060;branch=z9hG4bK-f397d70;rport
Authorization: DIGEST algorithm=MD5,nc=00000001,qop=auth,cnonce="45ccbc715…",nonce="1534002611/1d97f7789d01c65361e75b…",opaque="762c996e6c…",realm="asterisk",response="270ac84d3384fdd7ca8762cad. . .",uri="sip:[email protected]:5160",username="101"
User-Agent: OBIHAI/OBi100-1.3.0.2872
Event: message-summary
Contact: <sip:[email protected]:5060>
Expires: 3600
<— Transmitting SIP response (328 bytes) to UDP:192.168.1.78:5060 —>
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.1.78:5060;rport=5060;received=192.168.1.78;branch=z9hG4bK-f397d70
Call-ID: [email protected]
From: <sip:[email protected]>;tag=SP12207e0464…
To: <sip:[email protected]>;tag=z9hG4bK-f…
CSeq: 8002 SUBSCRIBE
Server: FPBX-14.0.3.12(13.20.0)
Content-Length: 0
– Transmitting SIP response (409 bytes) to UDP:192.168.1.68:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.68:5060;rport=5060;received=192.168.1.68;branch=z9hG4bK-59103130
Call-ID: [email protected]
From: <sip:[email protected]>;tag=SP12207e0464b19e13b
To: <sip:[email protected]>;tag=z9hG4bK-59103130
CSeq: 53223 REGISTER
Date: Sat, 11 Aug 2018 16:09:55 GMT
Contact: <sip:[email protected]:5060>;expires=59
Server: FPBX-14.0.3.12(13.20.0)
Content-Length: 0
Asterisk Report (call logs):
Fri, Aug 10 2018 10:46 AM 00:00:47 Unknown <101> *43
Fri, Aug 10 2018 10:47 AM 00:00:01 Unknown <101> hung up
- Fri, Aug 10 2018 1:25 PM 00:00:10 Unknown <101> 8189434612
Fri, Aug 10 2018 1:25 PM 00:00:01 Unknown <101> hung up
- Fri, Aug 10 2018 1:38 PM 00:00:08 Unknown <101> 8189434612
Fri, Aug 10 2018 1:38 PM 00:00:00 Unknown <101> hung up
- Fri, Aug 10 2018 2:29 PM 00:00:08 Unknown <101> 8189434612
Fri, Aug 10 2018 2:30 PM 00:00:00 Unknown <101> hung up
- Fri, Aug 10 2018 2:48 PM 00:00:09 Unknown <101> 8189434612
Fri, Aug 10 2018 2:48 PM 00:00:00 Unknown <101> hung up
- Fri, Aug 10 2018 2:53 PM 00:00:09 Unknown <101> 8189434612*43
Fri, Aug 10 2018 2:53 PM 00:00:00 Unknown <101> hung up
- Fri, Aug 10 2018 3:36 PM 00:00:01 Unknown <101> 8189434612
Fri, Aug 10 2018 3:36 PM 00:00:01 Unknown <101> hung up
- Fri, Aug 10 2018 4:50 PM 00:00:00 Unknown <101> 8189434612
Fri, Aug 10 2018 4:50 PM 00:00:00 Unknown <101> hung up
- Fri, Aug 10 2018 5:12 PM 00:00:08 Unknown <101> 8189434612
Fri, Aug 10 2018 5:12 PM 00:00:00 Unknown <101> hung up
- Fri, Aug 10 2018 5:12 PM 00:00:00 Unknown <101> 18189434612
Fri, Aug 10 2018 5:12 PM 00:00:00 Unknown <101> hung up
- Fri, Aug 10 2018 5:29 PM 00:00:08 Unknown <101> 8189434612
Fri, Aug 10 2018 5:29 PM 00:00:00 Unknown <101> hung up
- Fri, Aug 10 2018 8:56 PM 00:00:01 Unknown <101> 8189434612
Fri, Aug 10 2018 8:56 PM 00:00:00 Unknown <101> hung up
- Fri, Aug 10 2018 8:58 PM 00:00:01 Unknown <101> 8189434612
Fri, Aug 10 2018 8:58 PM 00:00:01 Unknown <101> hung up
- Fri, Aug 10 2018 8:58 PM 00:00:00 Unknown <101> 18189434612
Fri, Aug 10 2018 8:58 PM 00:00:00 Unknown <101> hung up
- Fri, Aug 10 2018 9:38 PM 00:00:00 Unknown <101> 8189434612
Fri, Aug 10 2018 9:38 PM 00:00:00 Unknown <101> hung up
- Fri, Aug 10 2018 9:39 PM 00:00:02 Unknown <101> 18189434612
Fri, Aug 10 2018 9:39 PM 00:00:01 Unknown <101> hung up
- Sat, Aug 11 2018 9:00 AM 00:00:01 Unknown <101> 8189434612
Sat, Aug 11 2018 9:00 AM 00:00:00 Unknown <101> hung up
I am puzzled as to why caller 101 is unknown since that is the name of my single extension. Also I am curious why asterisk notes that the sip call is okay, but also notes that the same sip is unrecognised. Does this mean that I need to whitelist the port numbers?
Any help
or advise would be greatly appreciated.
Don M.