Patton SN4114 FXO Gateway Setup Problems

[angyen]

Dear members,

I have recently purchased a Patton SN4114.
My settings are as below:-
Self hosted - 3CX PBX (10.10.10.18) - Passed all firewall check in 3cx, using 3cx STUN, extensions with 3CX android app
Patton SN4114 (10.30.10.9) - Connected via VPN, Successfully Provisioned with 3cx config file, Outbound Rules configured

Problems:-
1. Able to receive call via 3CX android app extension. However, only One-way audio. 3CX android app extension can hear from the caller but the caller cannot hear from the extension.
2. Unable to make outbound call - Error message was "Temporarily Unavailable"

I have browse through all the searches and tries to modify the setting but no avails. I would be much appreciated if anyone could assist me to get this up and running.

Thank you in advance.

 

[ChrisC_3CX]

First I recommend trying a few things to narrow this down, because, as you said we may be talking about two entirely different problems.

1. Able to receive call via 3CX android app extension. However, only One-way audio. 3CX android app extension can hear from the caller but the caller cannot hear from the extension.

If you perform an echo test from the android mobile app do you receive audio? Dial *777 from the android device to perform an echo test.

2. Unable to make outbound call - Error message was "Temporarily Unavailable"

• What do you mean by "using 3CX STUN", are you referring to how IP Phones are configured?
• Where are you calling from, is it from the android device? Try calling from a webclient, do you get the same message?
• Did you provision the gateway as per our guide here: https://www.3cx.com/voip-gateways/patton-smartnode-sn4112/ . Also is it running the mentioned firmware version?
• Does the error occur for any number you dial? Have you tried different formats such as in E164 or not and including country code/area code or not? Ideally use a format that would work on the analogue line too.

 

[angyen]

If you perform an echo test from the android mobile app do you receive audio? Dial *777 from the android device to perform an echo test.

Yes. The echo test working fine. In fact I have several extensions and they can call each other and the audio are great without any issue for months since I setup it. Sometimes both extension are in the same LAN, sometimes both are over mobile data, they all works. So it shouldn't be the problem with Android mobile app side.

• What do you mean by "using 3CX STUN", are you referring to how IP Phones are configured?

I am referring to the "External IP Configuration" in 3CX Network setting where the
Primary STUN server - stun-au.3cx.com
Secondary STUN server - stun2.3cx.com
Third STUN server - stun3.3cx.com
Sorry I do not know if this is relevant because I read something about STUN when I search for solution of one way audio

• Where are you calling from, is it from the android device? Try calling from a webclient, do you get the same message?

Android device and webclient both doesn't work. But I can see the error message from webclient as it disappear too fast

• Did you provision the gateway as per our guide here: https://www.3cx.com/voip-gateways/patton-smartnode-sn4112/ . Also is it running the mentioned firmware version?

Yes.I followed the exact same steps as per the guide and, the firmware is R6.9 2017-03-13 H323 SIP FXS FXO.

• Does the error occur for any number you dial? Have you tried different formats such as in E164 or not and including country code/area code or not? Ideally use a format that would work on the analogue line too.

Yes. I tried with area code, without area code it's all the same. can't get through.

 

[ChrisC_3CX]

Make absolutely sure you selected the correct country for the tone set:



Also, try increasing the "Dial Delay" to 5 if you haven't already:

This would of course mean you'll have to wait for something like 5-6 seconds when you make a call, but this is just temporary so that we can eliminate this being the cause.

In the meantime, also go to "Dashboad >> Activity Log >> Settings" and enable "Verbose Logging". When the issue occurs, check the activity log or even event log for errors and maybe provide us with a screenshot. (do remember to conceal any sensitive information)

 

[angyen]

11/17/2021 11:30:09 PM - Default best-route IP is determined as 10.10.10.18

11/17/2021 11:29:59 PM - Updating device Dev(1285845732):[sip:[email protected]:5483;rinstance=fe5f980e452fee09 / 70] by message: DevUpd Recv Req REGISTER from 127.0.0.1:5483 tid=d6e49a0677b19235 Call-ID=n7fEZbQBFAu6uskiK2_nNA..: REGISTER sip:127.0.0.1:15060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5483;branch=z9hG4bK-524287-1---d6e49a0677b19235;rport=5483 Max-Forwards: 70 Contact: <sip:[email protected]:5483;rinstance=fe5f980e452fee09> To: "Conference Room70:"<sip:[email protected]:15060> From: "Conference Room70:"<sip:[email protected]:15060>;tag=7075d15c Call-ID: n7fEZbQBFAu6uskiK2_nNA.. CSeq: 1826 REGISTER Expires: 60 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE Proxy-Authorization: Digest username="70",realm="3CXPhoneSystem",nonce="414d535961951ff668:cc559fd66581eec7de16ee12b9110975",uri="sip:127.0.0.1:15060",response="864001f8509216c92fee775c3e2d4176",algorithm=MD5 Supported: replaces User-Agent: 3CX Conference Room Content-Length: 0

11/17/2021 11:29:58 PM - Default best-route IP is determined as 10.10.10.18

11/17/2021 11:29:53 PM - Updating device Dev(1868174392):[sip:[email protected]:5063;rinstance=f0f49ba47d0ddcff / 90000] by message: DevUpd Recv Req REGISTER from 127.0.0.1:5063 tid=d077ce6c1d4e2970 Call-ID=yL3-NQk9cSZvMSstsil-oA..: REGISTER sip:127.0.0.1:15060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5063;branch=z9hG4bK-524287-1---d077ce6c1d4e2970;rport=5063 Max-Forwards: 70 Contact: <sip:[email protected]:5063;rinstance=f0f49ba47d0ddcff> To: <sip:[email protected]:15060> From: <sip:[email protected]:15060>;tag=71ab4701 Call-ID: yL3-NQk9cSZvMSstsil-oA.. CSeq: 14882 REGISTER Expires: 60 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE Proxy-Authorization: Digest username="90000",realm="3CXPhoneSystem",nonce="414d535961951ff184:147500dfb2153a02ced5cc2dfece042f",uri="sip:127.0.0.1:15060",response="b066e3fe9f2b766dd87aa90a80976e3d",algorithm=MD5 Supported: replaces User-Agent: 3CX WebRTC proxy Content-Length: 0

11/17/2021 11:29:53 PM - Line 90000 got Device: Dev(1868174392):[sip:[email protected]:5063;rinstance=f0f49ba47d0ddcff / 90000]

11/17/2021 11:29:53 PM - Source is identified as trunk Lc:90000(@WebMeeting bridge[<sip:[email protected]:5063/UDP>])

11/17/2021 11:29:53 PM - Source is identified as trunk Lc:90000(@WebMeeting bridge[<sip:[email protected]:5063/UDP>])

11/17/2021 11:29:53 PM - ~Target=Unknown:*PHONE NO. REDACTED*@(Ln.10000@Patton Analog FXO)

11/17/2021 11:29:53 PM - Leg L:20.1[Extn:10] is terminated: Cause: BYE from PBX

11/17/2021 11:29:53 PM - Terminated from "Ang Yen"<sip:[email protected]:15060>;tag=0cd39376-fb3f-4736-b144-f8fecd896c92 to <sip:*PHONE NO. REDACTED*@siow.3cx.jp:15060;tag3cx=f72d8467-246d-4a6a-af67-3bcf5c104c93>;tag=239b5045; reason: Rejected

11/17/2021 11:29:53 PM - L:20.1[Extn:10] Sending: OnSendResp Send 500/INVITE from 0.0.0.0:0 tid=Pj5eb69414-1805-44c4-8170-f749415f4d67 Call-ID=1965a40d-ba76-4bc3-a95d-d013fcabbb58: SIP/2.0 500 Server Internal Error Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK-524287-2---Pj5eb69414-1805-44c4-8170-f749415f4d67;rport=5080 Via: SIP/2.0/UDP 61.6.135.235:45278;branch=z9hG4bK-524287-1---tunneltid;rport;tnlid=clnt.0-30e93ed8-3f0f-4605-abb8-3c379083a0d2_39278200 Via: SIP/2.0/TCP 127.0.0.1:50195;rport;branch=z9hG4bKPj5eb69414-1805-44c4-8170-f749415f4d67;alias To: <sip:*PHONE NO. REDACTED*@siow.3cx.jp:15060;tag3cx=f72d8467-246d-4a6a-af67-3bcf5c104c93>;tag=239b5045 From: "Ang Yen"<sip:[email protected]:15060>;tag=0cd39376-fb3f-4736-b144-f8fecd896c92 Call-ID: 1965a40d-ba76-4bc3-a95d-d013fcabbb58 CSeq: 29476 INVITE Warning: 499 3CX.siow.prima "Server Failure" Content-Length: 0

11/17/2021 11:29:53 PM - SendMsg from <sip:*PHONE NO. REDACTED*@siow.3cx.jp:15060;tag3cx=f72d8467-246d-4a6a-af67-3bcf5c104c93>;tag=239b5045 to "Ang Yen"<sip:[email protected]:15060>;tag=0cd39376-fb3f-4736-b144-f8fecd896c92

11/17/2021 11:29:53 PM - Call(C:20) is terminated

11/17/2021 11:29:53 PM - [CM503020]: Call(C:20): Normal call termination. Call originator: Extn:10. Reason: Server Failure

11/17/2021 11:29:53 PM - [CM503016]: Call(C:20): Attempt to reach <sip:*PHONE NO. REDACTED*@siow.3cx.jp:15060> from Extn:10 has failed. Reason: Server Failure

11/17/2021 11:29:53 PM - [Flow] Current call diversion path:[]

11/17/2021 11:29:53 PM - L:20.1[Extn:10] failed to reach Line:10000>>*PHONE NO. REDACTED*, reason Server Failure

11/17/2021 11:29:53 PM - RerouteReq

11/17/2021 11:29:53 PM - Leg L:20.2[Line:10000>>*PHONE NO. REDACTED*] is terminated: Cause: 503 Service Unavailable/INVITE from 10.30.10.9:5060

11/17/2021 11:29:53 PM - L:20.2[Line:10000>>*PHONE NO. REDACTED*] got Terminated Recv 503/INVITE from 10.30.10.9:5060 tid=c59dae328e3ec456 Call-ID=1Q1g46EfJ-Ds6ndm_aNfGA..: SIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP 10.10.10.18:15060;branch=z9hG4bK-524287-1---c59dae328e3ec456;rport=15060;received=10.200.30.2 To: <sip:*PHONE NO. REDACTED*@10.30.10.9:5060>;tag=3352394669 From: "Ang Yen"<sip:+6*2ND PHONE NO. REDACTED*@10.10.10.18:15060>;tag=77e25656 Call-ID: 1Q1g46EfJ-Ds6ndm_aNfGA.. CSeq: 1 INVITE Server: Patton SN4114 JO EUI 00A0BA0DEC81 R6.9 2017-03-13 H323 SIP FXS FXO M5T SIP Stack/4.2.14.18 Content-Length: 0

11/17/2021 11:29:53 PM - Terminated from <sip:*PHONE NO. REDACTED*@10.30.10.9:5060>;tag=3352394669 to "Ang Yen"<sip:+6*2ND PHONE NO. REDACTED*@10.10.10.18:15060>;tag=77e25656; reason: Error

11/17/2021 11:29:53 PM - Call to T:Line:10000>>*PHONE NO. REDACTED*@[Dev:sip:[email protected]:5060] from L:20.1[Extn:10] failed, cause: Cause: 503 Service Unavailable/INVITE from 10.30.10.9:5060

11/17/2021 11:29:53 PM - L:20.2[Line:10000>>*PHONE NO. REDACTED*] got Failure: Failure Recv 503/INVITE from 10.30.10.9:5060 tid=c59dae328e3ec456 Call-ID=1Q1g46EfJ-Ds6ndm_aNfGA..: SIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP 10.10.10.18:15060;branch=z9hG4bK-524287-1---c59dae328e3ec456;rport=15060;received=10.200.30.2 To: <sip:*PHONE NO. REDACTED*@10.30.10.9:5060>;tag=3352394669 From: "Ang Yen"<sip:+6*2ND PHONE NO. REDACTED*@10.10.10.18:15060>;tag=77e25656 Call-ID: 1Q1g46EfJ-Ds6ndm_aNfGA.. CSeq: 1 INVITE Server: Patton SN4114 JO EUI 00A0BA0DEC81 R6.9 2017-03-13 H323 SIP FXS FXO M5T SIP Stack/4.2.14.18 Content-Length: 0

11/17/2021 11:29:53 PM - [CM503003]: Call(C:20): Call to <sip:*PHONE NO. REDACTED*@10.30.10.9:5060> has failed; Cause: 503 Service Unavailable/INVITE from 10.30.10.9:5060

11/17/2021 11:29:53 PM - Session 16327 has failed in leg L:20.2[Line:10000>>*PHONE NO. REDACTED*] ; Cause: 503 Service Unavailable/INVITE from 10.30.10.9:5060

11/17/2021 11:29:53 PM - Failure from <sip:*PHONE NO. REDACTED*@10.30.10.9:5060>;tag=3352394669 to "Ang Yen"<sip:+6*2ND PHONE NO. REDACTED*@10.10.10.18:15060>;tag=77e25656

11/17/2021 11:29:53 PM - [CM503025]: Call(C:20): Calling T:Line:10000>>*PHONE NO. REDACTED*@[Dev:sip:[email protected]:5060] for L:20.1[Extn:10]

11/17/2021 11:29:53 PM - Route to L:20.2[Line:10000>>*PHONE NO. REDACTED*] sends Invite-OUT Send Req INVITE from 0.0.0.0:0 tid=c59dae328e3ec456 Call-ID=1Q1g46EfJ-Ds6ndm_aNfGA..: INVITE sip:*PHONE NO. REDACTED*@10.30.10.9:5060 SIP/2.0 Via: SIP/2.0/ ;branch=z9hG4bK-524287-1---c59dae328e3ec456;rport Max-Forwards: 70 Contact: <sip:[email protected]:15060> To: <sip:*PHONE NO. REDACTED*@10.30.10.9:5060> From: "Ang Yen"<sip:+6*2ND PHONE NO. REDACTED*@10.10.10.18:15060>;tag=77e25656 Call-ID: 1Q1g46EfJ-Ds6ndm_aNfGA.. 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: 483 v=0 o=3cxPS 17294149707366400 35491612104589313 IN IP4 10.10.10.18 s=3cxPS Audio call c=IN IP4 10.10.10.18 b=AS:117 t=0 0 a=X-nat:0 m=audio 7034 RTP/AVP 8 9 18 120 0 96 c=IN IP4 10.10.10.18 b=TIAS:96000 b=AS:117 a=sendrecv a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:18 G729/8000 a=rtpmap:120 opus/48000/2 a=fmtp:120 useinbandfec=1 a=rtpmap:0 PCMU/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=rtcp:7035 m=video 0 RTP/AVP 31 c=IN IP4 127.0.0.1

11/17/2021 11:29:53 PM - Outbound CID reformat rule for DN:10000 is disabled (<Rules />)

11/17/2021 11:29:53 PM - SLA is globally disabled

 

[angyen]

11/17/2021 11:29:53 PM - Added leg L:20.2[Line:10000>>*PHONE NO. REDACTED*] <-> L:20.1[Extn:10]

11/17/2021 11:29:53 PM - [Flow] Call(C:20): making call from L:20.1[Extn:10] to T:Line:10000>>*PHONE NO. REDACTED*@[Dev:sip:[email protected]:5060]

11/17/2021 11:29:53 PM - [CM503027]: Call(C:20): From: Extn:10 ("Ang Yen" <sip:[email protected]:15060>) to T:Line:10000>>*PHONE NO. REDACTED*@[Dev:sip:[email protected]:5060]

11/17/2021 11:29:53 PM - [CM503004]: Call(C:20): Route 1: from L:20.1[Extn:10] to T:Line:10000>>*PHONE NO. REDACTED*@[Dev:sip:[email protected]:5060]

11/17/2021 11:29:53 PM - Outbound CID reformat rule for DN:10000 is disabled (<Rules />)

11/17/2021 11:29:53 PM - Line limit check: Current # of calls for line Lc:10000(@patton Analog FXO[<sip:[email protected]:5060/UDP>]) is 0; limit is 1

11/17/2021 11:29:53 PM - Call(C:20): Call from Extn:10 to *PHONE NO. REDACTED* matches outbound rule 'Malaysia'

11/17/2021 11:29:53 PM - [Flow] Call(C:20): has built target endpoint: Out#:>>Rule{Malaysia}>>*PHONE NO. REDACTED* for call from L:20.1[Extn:10]

11/17/2021 11:29:53 PM - [Flow] Target endpoint for *PHONE NO. REDACTED* is Out#:>>Rule{Malaysia}>>*PHONE NO. REDACTED*

11/17/2021 11:29:53 PM - Selected prefix: 01

11/17/2021 11:29:53 PM - Looking for outbound rule: dialed = [*PHONE NO. REDACTED*], processed: [*PHONE NO. REDACTED*]; from-ext:

11/17/2021 11:29:53 PM - [Flow] Building target endpoint to *PHONE NO. REDACTED* from "Ang Yen" <sip:[email protected]:15060>

11/17/2021 11:29:53 PM - [CM503010]: Call(C:20): Making route(s) from Extn:10 to <sip:*PHONE NO. REDACTED*@siow.3cx.jp:15060/UDP>

11/17/2021 11:29:53 PM - Remote SDP is set for leg L:20.1[Extn:10]

11/17/2021 11:29:53 PM - [CM505001]: Endpoint Extn:10: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CXPhone for Android 18.0.3.17] PBX contact: [sip:[email protected]:15060]

11/17/2021 11:29:53 PM - Inbound DID: ''; Phonebook Name: ''

11/17/2021 11:29:53 PM - [CM500002]: Call(C:20): Info on incoming INVITE from Extn:10: Invite-IN Recv Req INVITE from 127.0.0.1:5080 tid=Pj5eb69414-1805-44c4-8170-f749415f4d67 Call-ID=1965a40d-ba76-4bc3-a95d-d013fcabbb58: INVITE sip:*PHONE NO. REDACTED*@127.0.0.1:15060;transport=UDP;tag3cx=f72d8467-246d-4a6a-af67-3bcf5c104c93 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK-524287-2---Pj5eb69414-1805-44c4-8170-f749415f4d67;rport=5080 Via: SIP/2.0/UDP 61.6.135.235:45278;branch=z9hG4bK-524287-1---tunneltid;rport;tnlid=clnt.0-30e93ed8-3f0f-4605-abb8-3c379083a0d2_39278200 Via: SIP/2.0/TCP 127.0.0.1:50195;rport;branch=z9hG4bKPj5eb69414-1805-44c4-8170-f749415f4d67;alias Max-Forwards: 69 Record-Route: <sip:[email protected]:5080;user=proxy;uri=clnt.0-30e93ed8-3f0f-4605-abb8-3c379083a0d2_39278200> Contact: "Ang Yen" <sip:[email protected]:5060;rinstance=0-30e93ed8-3f0f-4605-abb8-3c379083a0d2> To: <sip:*PHONE NO. REDACTED*@siow.3cx.jp:15060;tag3cx=f72d8467-246d-4a6a-af67-3bcf5c104c93> From: "Ang Yen"<sip:[email protected]:15060>;tag=0cd39376-fb3f-4736-b144-f8fecd896c92 Call-ID: 1965a40d-ba76-4bc3-a95d-d013fcabbb58 CSeq: 29476 INVITE Session-Expires: 1800 Min-SE: 90 Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Content-Type: application/sdp Proxy-Authorization: Digest username="5dlByUOOTN",realm="3CXPhoneSystem",nonce="414d535961951ff127:1626db03cbb740f8bffa22b997ba1b20",uri="sip:*PHONE NO. REDACTED*@siow.3cx.jp:15060;tag3cx=f72d8467-246d-4a6a-af67-3bcf5c104c93",response="3d300e3630c3805cd942300596027823",algorithm=MD5 Supported: replaces, 100rel, timer, norefersub User-Agent: 3CXPhone for Android 18.0.3.17 Content-Length: 438 v=0 o=- 3846151791 3846151791 IN IP4 10.20.10.112 s=pjmedia b=AS:117 t=0 0 a=X-nat:0 m=audio 10000 RTP/AVP 8 9 18 120 0 96 c=IN IP4 10.20.10.112 b=TIAS:96000 b=AS:117 a=rtcp:10001 a=sendrecv a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:18 G729/8000 a=rtpmap:120 opus/48000/2 a=fmtp:120 useinbandfec=1 a=rtpmap:0 PCMU/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 m=video 0 RTP/AVP 31 c=IN IP4 127.0.0.1

11/17/2021 11:29:53 PM - [CM503001]: Call(C:20): Incoming call from Extn:10 to <sip:*PHONE NO. REDACTED*@siow.3cx.jp:15060>

11/17/2021 11:29:53 PM - Dev(1146703024):[sip:[email protected]:5060;rinstance=0-30e93ed8-3f0f-4605-abb8-3c379083a0d2 / 10] has updated source address: 127.0.0.1:5080/UDP fk=15 tgt=

11/17/2021 11:29:53 PM - Updating device Dev(1146703024):[sip:[email protected]:5060;rinstance=0-30e93ed8-3f0f-4605-abb8-3c379083a0d2 / 10] by message: DevUpd Recv Req INVITE from 127.0.0.1:5080 tid=Pj5eb69414-1805-44c4-8170-f749415f4d67 Call-ID=1965a40d-ba76-4bc3-a95d-d013fcabbb58: INVITE sip:*PHONE NO. REDACTED*@127.0.0.1:15060;transport=UDP;tag3cx=f72d8467-246d-4a6a-af67-3bcf5c104c93 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK-524287-2---Pj5eb69414-1805-44c4-8170-f749415f4d67;rport=5080 Via: SIP/2.0/UDP 61.6.135.235:45278;branch=z9hG4bK-524287-1---tunneltid;rport;tnlid=clnt.0-30e93ed8-3f0f-4605-abb8-3c379083a0d2_39278200 Via: SIP/2.0/TCP 127.0.0.1:50195;rport;branch=z9hG4bKPj5eb69414-1805-44c4-8170-f749415f4d67;alias Max-Forwards: 69 Record-Route: <sip:[email protected]:5080;user=proxy;uri=clnt.0-30e93ed8-3f0f-4605-abb8-3c379083a0d2_39278200> Contact: "Ang Yen" <sip:[email protected]:5060;rinstance=0-30e93ed8-3f0f-4605-abb8-3c379083a0d2> To: <sip:*PHONE NO. REDACTED*@siow.3cx.jp:15060;tag3cx=f72d8467-246d-4a6a-af67-3bcf5c104c93> From: "Ang Yen"<sip:[email protected]:15060>;tag=0cd39376-fb3f-4736-b144-f8fecd896c92 Call-ID: 1965a40d-ba76-4bc3-a95d-d013fcabbb58 CSeq: 29476 INVITE Session-Expires: 1800 Min-SE: 90 Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Content-Type: application/sdp Proxy-Authorization: Digest username="5dlByUOOTN",realm="3CXPhoneSystem",nonce="414d535961951ff127:1626db03cbb740f8bffa22b997ba1b20",uri="sip:*PHONE NO. REDACTED*@siow.3cx.jp:15060;tag3cx=f72d8467-246d-4a6a-af67-3bcf5c104c93",response="3d300e3630c3805cd942300596027823",algorithm=MD5 Supported: replaces, 100rel, timer, norefersub User-Agent: 3CXPhone for Android 18.0.3.17 Content-Length: 438 v=0 o=- 3846151791 3846151791 IN IP4 10.20.10.112 s=pjmedia b=AS:117 t=0 0 a=X-nat:0 m=audio 10000 RTP/AVP 8 9 18 120 0 96 c=IN IP4 10.20.10.112 b=TIAS:96000 b=AS:117 a=rtcp:10001 a=sendrecv a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:18 G729/8000 a=rtpmap:120 opus/48000/2 a=fmtp:120 useinbandfec=1 a=rtpmap:0 PCMU/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 m=video 0 RTP/AVP 31 c=IN IP4 127.0.0.1

11/17/2021 11:29:53 PM - IncomingCall: C:20 from <sip:[email protected]:15060/UDP> to <sip:*PHONE NO. REDACTED*@127.0.0.1:15060/UDP>

11/17/2021 11:29:53 PM - Added leg L:C:20.1[No endpoint yet]

11/17/2021 11:29:53 PM - Call from "Ang Yen"<sip:[email protected]:15060>;tag=0cd39376-fb3f-4736-b144-f8fecd896c92 to <sip:*PHONE NO. REDACTED*@siow.3cx.jp:15060;tag3cx=f72d8467-246d-4a6a-af67-3bcf5c104c93>;tag=239b5045

11/17/2021 11:29:50 PM - Updating device Dev(3686830):[sip:[email protected]:5060 / 10000] by message: DevUpd Recv Req REGISTER from 10.10.10.3:5060 tid=0f0d97f8ccdd589c8 Call-ID=db58c1f3d05796f8: REGISTER sip:10.10.10.18:15060 SIP/2.0 Via: SIP/2.0/UDP 10.30.10.9:5060;branch=z9hG4bK0f0d97f8ccdd589c8;received=10.10.10.3 Max-Forwards: 70 Contact: <sip:[email protected]:5060> To: <sip:[email protected]> From: <sip:[email protected]>;tag=04ca589752 Call-ID: db58c1f3d05796f8 CSeq: 32114 REGISTER Expires: 300 Proxy-Authorization: Digest username="10000",realm="3CXPhoneSystem",nonce="414d535961951fee37:4f66a12646bfbe08926734afd26b208f",uri="sip:10.10.10.18:15060",response="35ca1ee7b361545912fa625baa4b543a",algorithm=MD5 User-Agent: Patton SN4114 JO EUI 00A0BA0DEC81 R6.9 2017-03-13 H323 SIP FXS FXO M5T SIP Stack/4.2.14.18 Content-Length: 0

11/17/2021 11:29:50 PM - Line 10000 got Device: Dev(3686830):[sip:[email protected]:5060 / 10000]

11/17/2021 11:29:50 PM - Source is identified as trunk Lc:10000(@patton Analog FXO[<sip:[email protected]:5060/UDP>])

11/17/2021 11:24:05 PM - There's another STUN server that resolves to the same IP: 54.39.182.217:3478/UNKNOWN_TRANSPORT fk=0 tgt=; ignored

11/17/2021 11:21:24 PM - Set log verbosity to MaxLevel = 2, Severity mask = 7

 

[angyen]

The above is the log for one failed outgoing call.
Toneset is correct and Dail Delay has been set to 5 seconds.
The error message was Service Unavailable

 

[angyen]

This is the incoming call log with just one way audio.
3CX android extension can hear from the incoming caller but the incoming caller cannot hear from the 3CX android extension.

 

[ChrisC_3CX]

The Gateway is indeed replying with "503 Service Unavailable". This error could really mean anything and I can't be 100% sure as this is not a 3CX error. That said, it may be related to the gateway configuration or the PSTN Line itself. To eliminate this being related to the PSTN Line, a simple test would be to plug an analogue phone directly to the line and see if that works.

Regarding the audio issue could you make this test, enable call recording for the extension and make the call to reproduce the issue. Then listen to the recording and check whether you can hear the user audio. This will let us know if the audio from the user is reaching the PBX or not.

 

[angyen]

The Gateway is indeed replying with "503 Service Unavailable". This error could really mean anything and I can't be 100% sure as this is not a 3CX error. That said, it may be related to the gateway configuration or the PSTN Line itself. To eliminate this being related to the PSTN Line, a simple test would be to plug an analogue phone directly to the line and see if that works.

Regarding the audio issue could you make this test, enable call recording for the extension and make the call to reproduce the issue. Then listen to the recording and check whether you can hear the user audio. This will let us know if the audio from the user is reaching the PBX or not.

I have tested the PSTN Line and it's working fine when I use an analogue phone to make a call and receive a call. So it's not the Line problem.

For audio issue, as I am a free plan user, I couldn't enable Call Recording. However, I did test and two way audio could reach the PBX by an internal call from 2 different user with android app with the "PBX Delivers Audio" option ticked in both user.

 

[ChrisC_3CX]

I have tested the PSTN Line and it's working fine when I use an analogue phone to make a call and receive a call. So it's not the Line problem.

Since you've configured the gateway as per our guides and have also confirmed the PSTN line to be working, that leaves the gateway. That said, I'd highly recommend you reach out to Patton so that they can hopefully clarify why the gateway is replying with that SIP message.

However, I did test and two way audio could reach the PBX by an internal call from 2 different user with android app with the "PBX Delivers Audio" option ticked in both user.

So internal calls work but external have the audio issue? That is another way to confirm that two way audio from and to the PBX from an android app works. It would seem this is related to the issue with the outbound calls. I'd recommend sorting out the first problem and only spend time on the second after that has been resolved as they may very be related so solving one may get rid of the other.

Do let me know how it goes so that we can look into it further if needed.