Android app fails with "Not found", web client works fine

[turkeyman]

I'm trying to work out why my android app can't make calls.

It appears to provision correctly, and shows "Ready for calls" in the top right.

Dialing a number (local extension, or external number) fails with "Not found".



Comparing the activity logs, I can see that for some reason, calls from the app fail to find an extension, or fail to match any outbound rule. Calls from the web client work fine.



Here is the log for the Android client which FAILS to call a local extension:
-------------------------------------------------------------------------------------------------
05/06/2021 1:46:34 PM - [CM503014]: Call(C:16): No known route from Extn:232 to target: <sip:001@FQDN:0/UDP>
05/06/2021 1:46:34 PM - Call(C:16): from Extn:232 to 001 doesn't match any known extension number.
05/06/2021 1:46:34 PM - [CM503010]: Call(C:16): Making route(s) from Extn:232 to <sip:001@FQDN:0/UDP>
05/06/2021 1:46:34 PM - [CM505001]: Endpoint Extn:232: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CXPhone for Android 16.6.2.89] PBX contact: [sip:[email protected]:5060]
05/06/2021 1:46:34 PM - [CM500002]: Call(C:16): Info on incoming INVITE from Extn:232:
Invite-IN Recv Req INVITE from 127.0.0.1:5080 tid=Pjba3a13b3-4a87-460f-83e4-dd84aaaeafc8 Call-ID=8f417d3d-1017-4f23-a136-2c0a41fc0c33:
INVITE sip:[email protected]:5060;transport=UDP;tag3cx=4dd7c54c-c577-4cb5-8687-2a8bbcb1dc42 SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK-524287-2---Pjba3a13b3-4a87-460f-83e4-dd84aaaeafc8;rport=5080
Via: SIP/2.0/UDP [::ffff:IP_ADDR]:38254;branch=z9hG4bK-524287-1---tunneltid;rport;tnlid=clnt.0-b56c23f8-20ac-44a8-965c-9df4bdb3af6d_19517955
Via: SIP/2.0/TCP 127.0.0.1:50195;rport;branch=z9hG4bKPjba3a13b3-4a87-460f-83e4-dd84aaaeafc8;alias
Max-Forwards: 69
Record-Route: <sip:[email protected]:5080;user=proxy;uri=clnt.0-b56c23f8-20ac-44a8-965c-9df4bdb3af6d_19517955>
Contact: "Manu Evans" <sip:[email protected]:5060;rinstance=0-b56c23f8-20ac-44a8-965c-9df4bdb3af6d>
To: <sip:001@FQDN;tag3cx=4dd7c54c-c577-4cb5-8687-2a8bbcb1dc42>
From: "Manu Evans"<sip:[email protected]>;tag=56ab998c-6cc3-4734-a535-da3546caa959
Call-ID: 8f417d3d-1017-4f23-a136-2c0a41fc0c33
CSeq: 9822 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="Ztq73gig0L",realm="3CXPhoneSystem",nonce="414d53596093669911:1e26652a7069a0db07032170be67c0ed",uri="sip:001@FQDN;tag3cx=4dd7c54c-c577-4cb5-8687-2a8bbcb1dc42",response="2e4c906d6276fa2e5f94443e4c87486f",algorithm=MD5
Supported: replaces, 100rel, timer, norefersub
User-Agent: 3CXPhone for Android 16.6.2.89
Content-Length: 464

v=0
o=- 3829261596 3829261596 IN IP4 192.168.0.46
s=pjmedia
c=IN IP4 127.0.0.1
b=AS:117
t=0 0
a=X-nat:0
m=audio 14100 RTP/AVP 120 9 8 0 3 18 96
c=IN IP4 127.0.0.1
b=TIAS:96000
b=AS:117
a=sendrecv
a=rtpmap:120 opus/48000/2
a=fmtp:120 useinbandfec=1
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
m=video 0 RTP/AVP 31
c=IN IP4 127.0.0.1


And here is the log for comparison from the web client which SUCCEEDS to make the call:
------------------------------------------------------------------------------------------------------------------------
05/06/2021 1:47:41 PM - [CM503002]: Call(C:17): Alerting Extn:001 by contact <sip:[email protected]:33959/tcp>
05/06/2021 1:47:41 PM - [CM503025]: Call(C:17): Calling T:Extn:001@[Dev:sip:[email protected]:33959;transport=tcp;avaya-sc-enabled] for L:17.1[Extn:232]
05/06/2021 1:47:41 PM - [CM503027]: Call(C:17): From: Extn:232 ("Evans, Manu" <sip:[email protected]:0>) to T:Extn:001@[Dev:sip:[email protected]:33959;transport=tcp;avaya-sc-enabled]
05/06/2021 1:47:41 PM - [CM503004]: Call(C:17): Route 1: from L:17.1[Extn:232] to T:Extn:001@[Dev:sip:[email protected]:33959;transport=tcp;avaya-sc-enabled]
05/06/2021 1:47:41 PM - [Flow] No office hours set, office hours assumed
05/06/2021 1:47:41 PM - [Flow] Call(C:17): has built target endpoint: Extn:001 for call from L:17.1[Extn:232]
05/06/2021 1:47:41 PM - [Flow] Target endpoint for 001 is Extn:001
05/06/2021 1:47:41 PM - [CM503010]: Call(C:17): Making route(s) from Extn:232 to <sip:[email protected]:5060/UDP>
05/06/2021 1:47:41 PM - [CM505001]: Endpoint Extn:232: 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]
05/06/2021 1:47:41 PM - [CM500002]: Call(C:17): Info on incoming INVITE from Extn:232:
Invite-IN Recv Req INVITE from 127.0.0.1:5063 tid=4ab4ea69cbcbe979 Call-ID=Qn5Fuad-rgVAyBKZp1eTvw..:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5063;branch=z9hG4bK-524287-1---4ab4ea69cbcbe979;rport=5063
Max-Forwards: 70
Contact: <sip:[email protected]:5063;rinstance=9fe555e5bde8e631>
To: <sip:[email protected]:5060>
From: "Evans, Manu"<sip:[email protected]>;tag=ad8fa54e
Call-ID: Qn5Fuad-rgVAyBKZp1eTvw..
CSeq: 2 INVITE
Subject:
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Content-Type: application/sdp
Proxy-Authorization: Digest username="Ztq73gig0L",realm="3CXPhoneSystem",nonce="414d5359609366dc51:9d5d5a798b60cb23453a949356d1a511",uri="sip:[email protected]:5060",response="db9cf4409049bd1bf997dbc6ff712dcc",algorithm=MD5
Supported: replaces
User-Agent: 3CX WebRTC proxy
Content-Length: 595

v=0
o=3cxVCE 18446744071954281983 1238651491 IN IP4 127.0.0.1
s=3cxVCE Audio Call
c=IN IP4 127.0.0.1
t=0 0
m=audio 8514 RTP/AVP 111 103 104 9 0 8 106 105 13 110 112 113 126
a=rtpmap:111 opus/48000/2
a=fmtp:111 minptime=10;useinbandfec=1
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:110 telephone-event/48000
a=rtpmap:112 telephone-event/32000
a=rtpmap:113 telephone-event/16000
a=rtpmap:126 telephone-event/8000
a=ptime:20
a=sendrecv


There's a few things that look odd to me in both logs.

The hostnames look odd for a start...

In the failing log it says "From: "Manu Evans"<sip:[email protected]>" which shows the local LAN IP for the PBX, not accessible from the internet.

In the succeeding log it says: "From: "Evans, Manu"<sip:[email protected]>", I'm not sure how localhost is useful in an invite header? ...but it seems to work.



A bit further on, the failing log says: "Making route(s) from Extn:232 to <sip:001@FQDN:0/UDP>" it shows the public-facing hostname which seems right, but port 0 seems odd?

The succeeding log says: "Making route(s) from Extn:232 to <sip:[email protected]:5060/UDP>" still the odd use of localhost, but the port is correct... and this works.



All that said, it doesn't seem to be a problem with firewall or access to the server, because the PBX receives the INVITE from the Android client (I am able to paste these logs), so the problem seems to happen a step later...



Please help?

 

[turkeyman]

I wonder if there's a problem with NAT?
The FQDN resolves to the correct ip, which is then translated to 10.6.0.2 by destination NAT for incoming connections. Perhaps the incoming INVITE contained the hostname, and the PBX resolved the IP post translation before writing to the log? I still don't really see how this is relevant though, because 3CX attempts to place the call regardless, and seems to fail when looking up the destination extension or outbound rule.
I'm still suspicious of that weird '0' port number though..

 

[ChrisC_3CX]

First off, please provide us with the information mentioned her: https://www.3cx.com/community/threads/every-new-post-must-contain.20492/

Also, please let us know if this is affecting one or more android clients. Does creating a new test extension from scratch and provisioning it to an android device yield the same results?

 

[turkeyman]

The missing information:
Phone model: OnePlus 7 Pro
Android version: 11
App version: 16.7.5.103
PBX version: 16.0.9
Network connection: WiFi
Same network: No, phone is behind NAT on public internet

It seems to affect all Android clients.
Creating a new test extension and provisioning to the device shows the same problem.

Since I created this post, the Android client has received an update; the text "Not found" is not displayed anymore as I described, instead it now enters the call UI, plays a mid-pitch tone four times fairly fast, and then immediately returns to the dialer UI with no user input at any point.
I guess the issue hasn't changed, but the way the app notifies of this issue has changed, although it's equally unhelpful! It would be nice if it said what the problem is...

 

[turkeyman]

I noticed the call log looks a little different now than the one I showed in the OP; here is a log from a failed call attempt from the Android client:

05/30/2021 10:41:07 PM - Leg L:8.1[Unknown:] is terminated: Cause: BYE from PBX

05/30/2021 10:41:07 PM - [CM502001]: Source info: From: "Test Testson"<sip:[email protected]>;tag=81f7ecff-7604-43eb-92c4-02601a380551; To: <sip:[email protected];tag3cx=4057ce94-1738-476a-872a-a3e6085e03b4>

05/30/2021 10:41:07 PM - Leg L:8.1[Unknown:] is terminated: Cause: BYE from PBX 05/30/2021 10:41:07 PM - [CM502001]: Source info: From: "Test Testson"<sip:[email protected]>;tag=81f7ecff-7604-43eb-92c4-02601a380551; To: <sip:[email protected];tag3cx=4057ce94-1738-476a-872a-a3e6085e03b4> 05/30/2021 10:41:07 PM - [CM503013]: Call(C:8): Incoming call rejected, caller is unknown; msg=Invite-IN Recv Req INVITE from 127.0.0.1:5080 tid=Pja25f985c-546d-406b-894d-7a804c659dcb Call-ID=ae271051-067d-4e96-b11d-6c624186eaba: INVITE sip:[email protected]:5060;transport=UDP;tag3cx=4057ce94-1738-476a-872a-a3e6085e03b4 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK-524287-2---Pja25f985c-546d-406b-894d-7a804c659dcb;rport=5080 Via: SIP/2.0/UDP [::ffff:124.170.191.180]:37540;branch=z9hG4bK-524287-1---tunneltid;rport;tnlid=clnt.0-900d008e-8752-4e7b-961f-39e248e91bd2_64858763 Via: SIP/2.0/TCP 127.0.0.1:50195;rport;branch=z9hG4bKPja25f985c-546d-406b-894d-7a804c659dcb;alias Max-Forwards: 69 Record-Route: <sip:[email protected]:5080;user=proxy;uri=clnt.0-900d008e-8752-4e7b-961f-39e248e91bd2_64858763> Contact: "Test Testson" <sip:[email protected]:5060;rinstance=0-900d008e-8752-4e7b-961f-39e248e91bd2> To: <sip:[email protected];tag3cx=4057ce94-1738-476a-872a-a3e6085e03b4> From: "Test Testson"<sip:[email protected]>;tag=81f7ecff-7604-43eb-92c4-02601a380551 Call-ID: ae271051-067d-4e96-b11d-6c624186eaba CSeq: 32113 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="MVTKaY7saC",realm="3CXPhoneSystem",nonce="414d535960b387e363:e730d21d5448f1bab8d816aeebcc76ef",uri="sip:[email protected];tag3cx=4057ce94-1738-476a-872a-a3e6085e03b4",response="2f3905a2d217b5d0409d5417e8ef4353",algorithm=MD5 Supported: replaces, 100rel, timer, norefersub User-Agent: 3CXPhone for Android 16.7.5.103 Content-Length: 381 v=0 o=- 3831367266 3831367266 IN IP4 192.168.0.46 s=pjmedia c=IN IP4 127.0.0.1 b=AS:84 t=0 0 a=X-nat:0 m=audio 14052 RTP/AVP 9 8 0 18 96 c=IN IP4 127.0.0.1 b=TIAS:64000 b=AS:84 a=sendrecv a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 m=video 0 RTP/AVP 31 c=IN IP4 127.0.0.1

05/30/2021 10:41:07 PM - Timezone offset: 600 min.

It's interesting that the activity log now says "Incoming call rejected, caller is unknown", but To and From both look fine.
I can send text messages between these 2 extensions using the same Android client...

 

[turkeyman]

One further data point; when the Android client RECEIVES a call from a working client (using the web client from outside the network, it works), the Android client promptly receives the invite and the phone rings, but if I answer the call, it does the same as above; enters the call UI, immediately plays 4 mid-tones, then hangs up. The dialing client keeps ringing and doesn't seem to notice or receive notification that the Android client failed to begin the call.

 

[Vali_3CX]

If you call the voicemail (999) or echo service (*777) does it work properly?

 

[turkeyman]

No, the same problem.

 

[NickD_3CX]

In the Management Console, go to Settings --> Network and select the "FQDN" tab.

In the "Local SIP Domain" field, do you see your correct 3CX FQDN?
If not, correct it, press OK, then restart all 3CX Services from the "Dashboard".

 

[turkeyman]

That worked, thanks!
That said, can you explain why that changed anything? The option "Allow calls from/to external SIP URIs" is un-checked, it looks like that field should be ignored if the option is disabled...

This is a tangent question, but my FQDN appears to be mycompany.3cx.com.au, which was apparently issued by 3CX. I'd like to use my own domain; sip.mycompany.com.au, but the fiend to enter the FQDN is locked and I can't seem to be able to change it. How can I correct this, and it also mentioned that I need to configure my domain correctly?

 

[NickD_3CX]

Great!

You can't change your FQDN from there, and generally speaking you should never change the Local SIP Domain either.

If you want to change your FQDN, you need to backup (w/o FQND information), release your FQDN from your license key from your customer portal, uninstall, re-install while restoring backup and set the new FQDN during the installations.