Outgoing calls fail on a new freepbx install

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.

Anyone willing to help??

Incoming calls don’t work either.

There’s not much people here with GV + the new connection experience.

However, try posting a call log https://wiki.freepbx.org/display/SUP/Providing+Great+Debug#ProvidingGreatDebug-AsteriskLogs

I have successfully set up freepbx with gvsip (google voice) for free calling. I followed RonR’s instructions on dslreports website: https://www.dslreports.com/forum/r30661088-PBX-FreePBX-for-the-Raspberry-Pi

Before managing to set up gvsip successfully I had tried Raspbx and raspian with freepbx/asterisk installed. The hardest nut to crack was getting google voice to work (gvsip).

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