Solved OBiTALK 212 and outbound GV calls

[Steve123]

I've managed to get an Obitalk 212 working with inbound calls following the instructions in an earlier thread. Analog works fine in both directions.

One thing I learned, mentioned here in case someone else has the same problem, is that by default this model takes its configuration from the Obitalk servers. The expert mode on the website is exactly the same as the one you can reach directly on the device, so I wasn't using it. As it turns out, none of my changes were taking effect. To configure the device directly, you have to disable auto-provisioning. See this thread: https://www.obitalk.com/forum/index.php?topic=2347.0 for a discussion.

The problem: Outbound calls don't show any activity in the 3CX activity log, but they do show up in the call log.

Outbound rules are configured so that a 10 digit number prepends "1" and then sends it to the Google Voice trunk. Another rule takes 8 digit numbers and send them to a local FXO on a SPA-3012 that I'm migrating from. On the Obitalk, Google Voice is on SP1/ITSP A, and SP2/ITSP B is for 3CX. I haven't attached the config screen shots from the Obitalk since there's no sign of 3CX initiating the call, neither in it's own log nor that of the Obitalk.

Anyone have any ideas?

 

[Steve123]

OK, it seems the debug level needs to be raised to see outbound calls in the activity log.

Here's what it shows

11/29/2021 12:03:49 PM - Terminated from "Steve"<sip:[email protected]>;tag=6c052179 to <sip:[email protected]:5060>;tag=0640eb5a; reason: Rejected

11/29/2021 12:03:49 PM - L:73.1[Extn:101] Sending: OnSendResp Send 480/INVITE from 0.0.0.0:0 tid=c66d8838c819a176 Call-ID=9adoYhZOfrzoye3RRLa--Q..: SIP/2.0 480 Temporarily Unavailable Via: SIP/2.0/UDP 127.0.0.1:5063;branch=z9hG4bK-524287-1---c66d8838c819a176;rport=5063 To: <sip:[email protected]:5060>;tag=0640eb5a From: "Steve"<sip:[email protected]>;tag=6c052179 Call-ID: 9adoYhZOfrzoye3RRLa--Q.. CSeq: 2 INVITE Content-Length: 0

11/29/2021 12:03:49 PM - SendMsg from <sip:[email protected]:5060>;tag=0640eb5a to "Steve"<sip:[email protected]>;tag=6c052179

11/29/2021 12:03:49 PM - ~Target=Unknown:14076209696@(Ln.10001@Google Voice)

11/29/2021 12:03:49 PM - CAHangup: on leg L:73.1[Extn:101] dn=101

11/29/2021 12:03:49 PM - C:73<-1evice is supported by direct call control procedures. Call will be terminated from Call Manager side

11/29/2021 12:03:49 PM - Hangup request:C:73 legId = 1

11/29/2021 12:03:43 PM - Default best-route IP is determined as 10.65.0.202

11/29/2021 12:03:41 PM - [CM503025]: Call(C:73): Calling T:Line:10001>>14076209696@[Dev:sip:[email protected]:5061] for L:73.1[Extn:101]

11/29/2021 12:03:41 PM - Route to L:73.2[Line:10001>>14076209696] sends Invite-OUT Send Req INVITE from 0.0.0.0:0 tid=76766f3813112415 Call-ID=9DZ9Qlto_mlgtGqSr_ub6Q..: INVITE sip:[email protected]:5060 SIP/2.0 Via: SIP/2.0/ ;branch=z9hG4bK-524287-1---76766f3813112415;rport Max-Forwards: 70 Contact: <sip:10.65.0.202:5060> To: <sip:[email protected]:5060> From: "Steve"<sip:10.65.0.202:5060>;tag=11bf5715 Call-ID: 9DZ9Qlto_mlgtGqSr_ub6Q.. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer Content-Length: 286 v=0 o=3cxPS 19395876921278464 34707248802430977 IN IP4 10.65.0.202 s=3cxPS Audio call c=IN IP4 10.65.0.202 t=0 0 m=audio 7272 RTP/AVP 0 8 9 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv

11/29/2021 12:03:41 PM - Outbound CID reformat rule for DN:10001 is disabled (<Rules />)

11/29/2021 12:03:41 PM - SLA is globally disabled

11/29/2021 12:03:41 PM - Added leg L:73.2[Line:10001>>14076209696] <-> L:73.1[Extn:101]

11/29/2021 12:03:41 PM - [Flow] Call(C:73): making call from L:73.1[Extn:101] to T:Line:10001>>14076209696@[Dev:sip:[email protected]:5061]

11/29/2021 12:03:41 PM - [CM503027]: Call(C:73): From: Extn:101 ("Steve" <sip:[email protected]:0>) to T:Line:10001>>14076209696@[Dev:sip:[email protected]:5061]

11/29/2021 12:03:41 PM - [CM503004]: Call(C:73): Route 1: from L:73.1[Extn:101] to T:Line:10001>>14076209696@[Dev:sip:[email protected]:5061]

11/29/2021 12:03:41 PM - Outbound CID reformat rule for DN:10001 is disabled (<Rules />)

11/29/2021 12:03:41 PM - Line limit check: Current # of calls for line Lc:10001(@Google Voice[<sip:[email protected]:5061/UDP>]) is 0; limit is 1

11/29/2021 12:03:41 PM - Call(C:73): Call from Extn:101 to 4076209696 matches outbound rule 'USA Local'

11/29/2021 12:03:41 PM - [Flow] Call(C:73): has built target endpoint: Out#:>>Rule{USA Local}>>4076209696 for call from L:73.1[Extn:101]

11/29/2021 12:03:41 PM - [Flow] Target endpoint for 4076209696 is Out#:>>Rule{USA Local}>>4076209696

11/29/2021 12:03:41 PM - Looking for outbound rule: dialed = [4076209696], processed: [4076209696]; from-ext:

11/29/2021 12:03:41 PM - [Flow] Building target endpoint to 4076209696 from "Steve" <sip:[email protected]:0>

11/29/2021 12:03:41 PM - [CM503010]: Call(C:73): Making route(s) from Extn:101 to <sip:[email protected]:5060/UDP>

11/29/2021 12:03:41 PM - Remote SDP is set for leg L:73.1[Extn:101]

11/29/2021 12:03:41 PM - [CM505001]: Endpoint Extn:101: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CX WebRTC proxy] PBX contact: [sip:[email protected]:5060]

 

[leejor]

Have a read through this, for the Obi setting that I used.

https://www.3cx.com/community/threads/3cx-with-google-voice-via-obi200-settings.69896/

I will look over my 3CX trunk settings and see what is different on yours. I did notice that you have 10001 and 10002 used in your first attachment, where you GV trunk is 10002.

You may have to set the Activity Logs to Verbose to see all of the call details.

 

[Steve123]

I just checked those and with a fresh retry get the following. I see that it's trying to reach Line 10001, where as GV/Obi should be line 10002.

It's worth mentioning that I have an SPA-1302 as 10001, and this new Obi on 10002

11/29/2021 12:43:32 PM - Currently active calls [none]

11/29/2021 12:43:32 PM - Leg L:78.1[Extn:101] is terminated: Cause: BYE from PBX

11/29/2021 12:43:32 PM - [CM503020]: Call(C:78): Normal call termination. Call originator: Extn:101. Reason: Busy

11/29/2021 12:43:32 PM - L:78.1[Extn:101] failed to reach Line:10001>>14076209696, reason Busy

11/29/2021 12:43:32 PM - Leg L:78.2[Line:10001>>14076209696] is terminated: Cause: 486 Busy Here/INVITE from 10.65.0.204:5061

11/29/2021 12:43:32 PM - Call to T:Line:10001>>14076209696@[Dev:sip:[email protected]:5061] from L:78.1[Extn:101] failed, cause: Cause: 486 Busy Here/INVITE from 10.65.0.204:5061

11/29/2021 12:43:32 PM - [CM503003]: Call(C:78): Call to <sip:[email protected]:5061> has failed; Cause: 486 Busy Here/INVITE from 10.65.0.204:5061

11/29/2021 12:43:32 PM - Timezone offset: 480 min.

11/29/2021 12:43:32 PM - [CM503025]: Call(C:78): Calling T:Line:10001>>14076209696@[Dev:sip:[email protected]:5061] for L:78.1[Extn:101]

11/29/2021 12:43:32 PM - [CM503027]: Call(C:78): From: Extn:101 ("Steve" <sip:[email protected]:0>) to T:Line:10001>>14076209696@[Dev:sip:[email protected]:5061]

11/29/2021 12:43:32 PM - [CM503004]: Call(C:78): Route 1: from L:78.1[Extn:101] to T:Line:10001>>14076209696@[Dev:sip:[email protected]:5061]

11/29/2021 12:43:32 PM - Line limit check: Current # of calls for line Lc:10001(@Google Voice[<sip:[email protected]:5061/UDP>]) is 0; limit is 1

11/29/2021 12:43:32 PM - Call(C:78): Call from Extn:101 to 4076209696 matches outbound rule 'USA Local'

11/29/2021 12:43:32 PM - [Flow] Call(C:78): has built target endpoint: Out#:>>Rule{USA Local}>>4076209696 for call from L:78.1[Extn:101]

11/29/2021 12:43:32 PM - [Flow] Target endpoint for 4076209696 is Out#:>>Rule{USA Local}>>4076209696

11/29/2021 12:43:32 PM - [CM503010]: Call(C:78): Making route(s) from Extn:101 to <sip:[email protected]:5060/UDP>

11/29/2021 12:43:32 PM - [CM505001]: Endpoint Extn:101: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CX WebRTC proxy] PBX contact: [sip:[email protected]:5060]

11/29/2021 12:43:32 PM - [CM500002]: Call(C:78): Info on incoming INVITE from Extn:101: Invite-IN Recv Req INVITE from 127.0.0.1:5063 tid=3a7e5a44fda95714 Call-ID=P1IhSSOt59dVbtzhEzZw..: INVITE sip:[email protected]:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5063;branch=z9hG4bK-524287-1---3a7e5a44fda95714;rport=5063 Max-Forwards: 70 Contact: <sip:[email protected]:5063;rinstance=d0b93f3ed46b7844> To: <sip:[email protected]:5060> From: "Steve"<sip:[email protected]>;tag=21f1a442 Call-ID: P1IhSSOt59dVbtzhEzZw.. CSeq: 2 INVITE Subject: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE Content-Type: application/sdp Proxy-Authorization: Digest username="Bb4sejnZ5b",realm="3CXPhoneSystem",nonce="414d535961a45a7460:d1983cf88728858ee1e404ea7d43cc7e",uri="sip:[email protected]:5060",response="2d687ece73166b556bd96793b66b8352",algorithm=MD5 Supported: replaces User-Agent: 3CX WebRTC proxy Content-Length: 376 v=0 o=3cxVCE 18446744071681058392 173882362 IN IP4 127.0.0.1 s=3cxVCE Audio Call c=IN IP4 127.0.0.1 t=0 0 m=audio 8642 RTP/AVP 0 8 9 109 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000/1 a=rtpmap:109 opus/48000/2 a=fmtp:109 maxplaybackrate=48000;stereo=1;useinbandfec=1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv

 

[Steve123]

Further investigation seems to suggest this is a problem on the Obi side (at least this bronze certified partner says so). But, I can't see why.

 

[Steve123]

Problem solved. In the thread that @leejor referenced above, there is a note about using {>(<1:>xx.):SP1} as the X_InboundCallRoute for SP2 Service. I have no idea why that is required. Perhaps somethings changed on the Obi's recently. In any case, with that route, all works.

Now on to configuring the FXO port...

 

[leejor]

I have only used the Obi200, so no experience making use of the FXO port on a 212. I use an SPA3102/3000 for that sort of thing.

 

[Steve123]

Well, I spoke too soon on the GV and Obi 212. I was able to ring other phones, and still can, but I can't get an answer because of a server error:

11/29/2021 5:22:17 PM - Leg L:106.1[Extn:100] is terminated: Cause: BYE from PBX

11/29/2021 5:22:17 PM - [CM503020]: Call(C:106): Normal call termination. Call originator: Extn:100. Reason: Server Failure

11/29/2021 5:22:17 PM - L:106.1[Extn:100] failed to reach Line:10001>>18042221111, reason Server Failure

11/29/2021 5:22:17 PM - Leg L:106.2[Line:10001>>18042221111] is terminated: Cause: 503 Service Unavailable/INVITE from 10.65.0.204:5061

11/29/2021 5:22:17 PM - Call to T:Line:10001>>18042221111@[Dev:sip:[email protected]:5061] from L:106.1[Extn:100] failed, cause: Cause: 503 Service Unavailable/INVITE from 10.65.0.204:5061

11/29/2021 5:22:17 PM - [CM503003]: Call(C:106): Call to <sip:[email protected]:5061> has failed; Cause: 503 Service Unavailable/INVITE from 10.65.0.204:5061

11/29/2021 5:22:15 PM - [CM505002]: Gateway:[Google Voice] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [OBIHAI/OBi212-3.2.2.8680] PBX contact: [sip:[email protected]:5060]

11/29/2021 5:22:15 PM - [CM503002]: Call(C:106): Alerting Line:10001>>18042221111 by contact <sip:[email protected]:5061/UDP>

11/29/2021 5:22:14 PM - [CM503025]: Call(C:106): Calling T:Line:10001>>18042221111@[Dev:sip:[email protected]:5061] for L:106.1[Extn:100]

11/29/2021 5:22:14 PM - [CM503027]: Call(C:106): From: Extn:100 ("100-09535382" <sip:[email protected]:0>) to T:Line:10001>>18042221111@[Dev:sip:[email protected]:5061]

11/29/2021 5:22:14 PM - [CM503004]: Call(C:106): Route 1: from L:106.1[Extn:100] to T:Line:10001>>18042221111@[Dev:sip:[email protected]:5061]

11/29/2021 5:22:14 PM - Line limit check: Current # of calls for line Lc:10001(@Google Voice[<sip:[email protected]:5061/UDP>]) is 0; limit is 1

11/29/2021 5:22:14 PM - Call(C:106): Call from Extn:100 to 8042221111 matches outbound rule 'USA Local'

11/29/2021 5:22:14 PM - [Flow] Call(C:106): has built target endpoint: Out#:>>Rule{USA Local}>>8042221111 for call from L:106.1[Extn:100]

11/29/2021 5:22:14 PM - [Flow] Target endpoint for 8042221111 is Out#:>>Rule{USA Local}>>8042221111

11/29/2021 5:22:14 PM - [CM503010]: Call(C:106): Making route(s) from Extn:100 to <sip:[email protected]:0/UDP>

11/29/2021 5:22:14 PM - [CM505001]: Endpoint Extn:100: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Cisco/SPA504G-7.6.1] PBX contact: [sip:[email protected]:5060]

11/29/2021 5:22:14 PM - [CM500002]: Call(C:106): Info on incoming INVITE from Extn:100: Invite-IN Recv Req INVITE from 10.65.0.201:5060 tid=-82e40bad Call-ID=[email protected]: INVITE sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP 10.65.0.201:5060;branch=z9hG4bK-82e40bad Max-Forwards: 70 Contact: "100-09535382" <sip:[email protected]:5060> To: <sip:[email protected]> From: "100-09535382"<sip:[email protected]>;tag=84fb15b99d88d2b9o0 Call-ID: [email protected] CSeq: 102 INVITE Expires: 240 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE Content-Type: application/sdp Proxy-Authorization: Digest username="94RVCvC1M8",realm="3CXPhoneSystem",nonce="414d535961a49bc631:d88b24732b47a8e16d81b37b4ad37f58",uri="sip:[email protected]",algorithm=MD5,response="1b61ea38698d2d60b8a7c778b44f2a03" Supported: replaces User-Agent: Cisco/SPA504G-7.6.1 Content-Length: 399 v=0 o=- 139125407 139125407 IN IP4 10.65.0.201 s=- c=IN IP4 10.65.0.201 t=0 0 m=audio 16460 RTP/AVP 0 2 8 9 18 96 97 98 101 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv

11/29/2021 5:22:14 PM - [CM503001]: Call(C:106): Incoming call from Extn:100 to <sip:[email protected]:0>

11/29/2021 5:22:03 PM - Timezone offset: 480 min.

 

[Steve123]

Another clue, the non-connection to the phone happens in inbound and outbound. If I dial in and pick up the handset, it shows "connected", but the mobile I'm dialing in on keeps ringing.

 

[leejor]

11/29/2021 5:22:17 PM - [CM503003]: Call(C:106): Call to <sip:[email protected]:5061> has failed; Cause: 503 Service Unavailable/INVITE from 10.65.0.204:5061

So...if you plug a set into he FXS port of the Obi, can you place calls?

You did confirm that the Obi is set to use port 5061 of the 3CX trunk?

If you look back at the third attachment in your earlier post...SIP Trunks. Why does the Google Voice trunk show Main Trunk Number as 10002 and not 10001?

 

[Steve123]

So...if you plug a set into he FXS port of the Obi, can you place calls?

Yes, and receive.

You did confirm that the Obi is set to use port 5061 of the 3CX trunk?

I'm pretty sure. The Obi config is attached, and the status page.

If you look back at the third attachment in your earlier post...SIP Trunks. Why does the Google Voice trunk show Main Trunk Number as 10002 and not 10001?

The SPA-3102 currently providing PSTN is on 10001.

 

[Steve123]

... Why does the Google Voice trunk show Main Trunk Number as 10002 and not 10001?

This prompted me to change it to 10001 and try again. I don't see any errors, but neither do I get any voice. Just silence. The log output doesn't seem to provide any clues. The call is originating from a Cisco SPA504G (ext 100), and I get the same thing from the 3CX soft phone.

11/30/2021 5:25:47 PM - Leg L:121.2[Line:10001>>19172756975] is terminated: Cause: BYE from local

11/30/2021 5:25:47 PM - [CM503008]: Call(C:121): Call is terminated

11/30/2021 5:25:47 PM - Leg L:121.1[Extn:100] is terminated: Cause: BYE from 10.65.0.201:5060

11/30/2021 5:25:39 PM - Currently active calls - 1: [121]

11/30/2021 5:25:31 PM - [CM503007]: Call(C:121): Extn:100 has joined, contact <sip:[email protected]:5060/UDP>

11/30/2021 5:25:31 PM - [CM503007]: Call(C:121): Line:10001>>19172756975 has joined, contact <sip:[email protected]:5061/UDP>

11/30/2021 5:25:31 PM - L:121.2[Line:10001>>19172756975] has joined to L:121.1[Extn:100]

11/30/2021 5:25:29 PM - [CM505002]: Gateway:[Google Voice] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [OBIHAI/OBi212-3.2.2.8680] PBX contact: [sip:[email protected]:5060]

11/30/2021 5:25:29 PM - [CM503002]: Call(C:121): Alerting Line:10001>>19172756975 by contact <sip:[email protected]:5061/UDP>

11/30/2021 5:25:28 PM - [CM503025]: Call(C:121): Calling T:Line:10001>>19172756975@[Dev:sip:[email protected]:5061] for L:121.1[Extn:100]

11/30/2021 5:25:28 PM - [CM503027]: Call(C:121): From: Extn:100 ("100-09535382" <sip:[email protected]:0>) to T:Line:10001>>19172756975@[Dev:sip:[email protected]:5061]

11/30/2021 5:25:28 PM - [CM503004]: Call(C:121): Route 1: from L:121.1[Extn:100] to T:Line:10001>>19172756975@[Dev:sip:[email protected]:5061]

11/30/2021 5:25:28 PM - Line limit check: Current # of calls for line Lc:10001(@Google Voice[<sip:[email protected]:5061/UDP>]) is 0; limit is 1

11/30/2021 5:25:28 PM - Call(C:121): Call from Extn:100 to 9172756975 matches outbound rule 'USA Local'

11/30/2021 5:25:28 PM - [Flow] Call(C:121): has built target endpoint: Out#:>>Rule{USA Local}>>9172756975 for call from L:121.1[Extn:100]

11/30/2021 5:25:28 PM - [Flow] Target endpoint for 9172756975 is Out#:>>Rule{USA Local}>>9172756975

11/30/2021 5:25:28 PM - [CM503010]: Call(C:121): Making route(s) from Extn:100 to <sip:[email protected]:0/UDP>

11/30/2021 5:25:28 PM - [CM505001]: Endpoint Extn:100: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Cisco/SPA504G-7.6.1] PBX contact: [sip:[email protected]:5060]

11/30/2021 5:25:28 PM - [CM500002]: Call(C:121): Info on incoming INVITE from Extn:100: Invite-IN Recv Req INVITE from 10.65.0.201:5060 tid=-d5e4efe0 Call-ID=[email protected]: INVITE sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP 10.65.0.201:5060;branch=z9hG4bK-d5e4efe0 Max-Forwards: 70 Contact: "100-09535382" <sip:[email protected]:5060> To: <sip:[email protected]> From: "100-09535382"<sip:[email protected]>;tag=eeb6a1e3b8cdc550o0 Call-ID: [email protected] CSeq: 102 INVITE Expires: 240 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE Content-Type: application/sdp Proxy-Authorization: Digest username="94RVCvC1M8",realm="3CXPhoneSystem",nonce="414d535961a5ee0897:f8690cbc526a10d8ddd7cbb1c120ad7b",uri="sip:[email protected]",algorithm=MD5,response="9e3b4666fca5fbfc4eb6c364dff0f400" Supported: replaces User-Agent: Cisco/SPA504G-7.6.1 Content-Length: 399 v=0 o=- 147784536 147784536 IN IP4 10.65.0.201 s=- c=IN IP4 10.65.0.201 t=0 0 m=audio 16486 RTP/AVP 0 2 8 9 18 96 97 98 101 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv

11/30/2021 5:25:28 PM - [CM503001]: Call(C:121): Incoming call from Extn:100 to <sip:[email protected]:0>

11/30/2021 5:25:09 PM - Timezone offset: 480 min.

 

[leejor]

"The SPA-3102 currently providing PSTN is on 10001".

It can't be if trunk 10001 is used for the Obi, it should be using trunk 10002, if that is what you assigned.

The call looks to be going though to 10.65.0.204 correctly, that better be the Obi.

In the attachment, it shows SP3 as being registered to 3CX as well as SP2...why is that? What is that?

 

[Steve123]

"The SPA-3102 currently providing PSTN is on 10001".

It can't be if trunk 10001 is used for the Obi, it should be using trunk 10002, if that is what you assigned.

The SPA-3102 was using 100001 and that was working for FXO calls. My intention is to set up the Obi 'beside' the LinkSys so I could fall back to it for local calls if I stuffed something up. So I had the LinkSys on 10001 and the Obi on 10002. The IPs addresses are:

.201 Cisco IP phone
.202 3CX
.203 SPA-3102
.204 Obi-212

In the attachment, it shows SP3 as being registered to 3CX as well as SP2...why is that? What is that?

SP3 is the start of getting the FXO port working.

I may be lacking some fundmental understanding of how this all hangs together, so if you seen an obvious or stupid mistake, please do point it out.

 

[Steve123]

I've just tested sound between the softphone and handset; this works. So, the problem appears to be sound between the Obi 212 and 3CX. Looking into this now.

 

[leejor]

After setting up the Obi110 a long time ago, then moving the settings to the obi200 when it became obsolete for GV, I gave up on changing anything more. The routing design is as if they purposefully tried to make it as confusing as possible. I'll avoid making any changes as long as it continues to work so i can only offer the settings I've got on mine.. The SPA3102 make complete sense to me, not so the Obi devices.

SP3 is the start of getting the FXO port working.

I assume that it is set to use a port other than 5061, on the Obi.

The trunk numbers, which is using which, makes no mater as long as the devices and trunk setting are completely separate.

My intention is to set up the Obi 'beside' the LinkSys so I could fall back to it for local calls if I stuffed something up.

Do this in the outbound rules, as an alternative route, not by mixing around the trunk numbers.

 

[Steve123]

I assume that it is set to use a port other than 5061, on the Obi.

Correct. Obi/SP3 is on 5062

The trunk numbers, which is using which, makes no mater as long as the devices and trunk setting are completely separate.

Okay. To avoid confusion, I've just changed the Obi to be the GV DID. It can get confusing the way 3CX automatically allocates trunk numbers and authentication ID's.

Do this in the outbound rules, as an alternative route, not by mixing around the trunk numbers.

Exactly. I flip the outbound route for 8 digit numbers to/from the OBI and SPA trunks.

 

[Steve123]

For anyone finding this, the problem with no audio turned out to be caused by a change in Google port numbers and had nothing to do with 3CX or the Obitalk.. GV did work on the POTS phone when I started this process. Apparently the changes are rolling, or perhaps my Chrome softphone held onto the ports. I don't know.

See:
http://www.obitalk.com/forum/index.php?topic=14184.0
https://support.google.com/a/answer/9206518?hl=en#zippy=,outbound-ports-need-to-allow-voice-traffic

for information on what outbound ports need to be opened.

 

[leejor]

So...you had these ports blocked on your firewall? Or, you found you had to do some port forwarding, in your router?

 

[Steve123]

All ports are blocked except those in the whitelist. Once I added some rules to allow the new GV ports, everything works.