Solved No getting Inbound Calls

[Rommel Otero]

Hi everyone

Im pretty new with 3cx, normaly worked with Asterisk

Im having an issue that I have this trunk created and outgoing calls works fine but no inbound

I haved tested the information given by the provider in a Sotfphone and Im able to make call and receive calls, however with the same data in 3cx Im just able to receive calls

Plz can anyone point me in the right direction?

sorry for mistakes, english is not my first language

 

[leejor]

Does the trunk show as Registered in 3CX? Is there an incoming call attempt showing in the 3CX Activity Log?

Have you spoken to your provider about what they see when sending a call to you?

 

[Rommel Otero]

Yes It doesnshow registed the trunk

nop the 3cx logs does no show any activity of and incoming call (I think), I mean

I talked to the provider and what the said was that I should test in a sothphone and if it works your are missing something on your pbx

 

[leejor]

Have you run the Firewall Checker to be sure that messages sent to port 5060 are being passed to the PBX? If the trunks is showing as registered, and you have it set to require that (it will show as registered if you specify not to register), then port 5060 should be getting SIP messages. you may want to see if your provider can tell you, when the trunk is registered, what IP they are sending calls. If that matches your public IP, then I would suspect a router misconfiguration/option issue
If incoming calls are reaching 3Cx, even if they are rejected for some reason, there will be a log.

When you did a test with a softphone, was it on your network?

 

[YiannisH_3CX]

Hello @Rommel Otero

If you are familiar with wireshark then you could try a capture on the 3CX machine and see if Invite messages are reaching the PBX. This could be an issue with Invites not reaching the PBX or the calls being rejected because of source identification.

 

[Rommel Otero]

Solved. Problem was that thr provider delivers the call without cero (0) of the international code, example for UK in isted of 0044xxxxx, it delivers 44xxxxx so inbound rule was not working.

Now I have another issue. I get the inbound fine and if I direct the call to a Queu fine but to the digital receptionist gets the call but the call gets out as soon as I dial an option.

Internal calls to the IVR works fine and the options works fine, the problems is when the call is from outside, here the log

11/15/2018 1:30:14 PM - L:77.1[Line:10000<<50640100188] got Terminated Send Req BYE from 0.0.0.0:0 tid=3f2b292378a3f139 Call-ID=f3a04a86-2e55-4fb1-9936-af0dc3a402d8:
BYE sip:[email protected]:5070 SIP/2.0
Via: SIP/2.0/ ;branch=z9hG4bK-524287-1---3f2b292378a3f139;rport
Max-Forwards: 70
Route: <sip:190.14.152.11;lr;ep;pinhole=UDP:35.232.100.25:5060>
Contact: <sip:[email protected]:5060>
To: 40100188 <sip:[email protected]>;tag=jHb9PiatyjW8OOaA.o
From: <sip:[email protected]>;tag=cd4e952a
Call-ID: f3a04a86-2e55-4fb1-9936-af0dc3a402d8
CSeq: 3 BYE
User-Agent: 3CXPhoneSystem 15.5.15502.6 (15502)
Reason: SIP;text=Application Rejected Sdp(usually no common codec)
Content-Length: 0


11/15/2018 1:30:14 PM - L:77.1[Line:10000<<50640100188] Sending: OnSendReq Send Req BYE from 0.0.0.0:0 tid=f1f9e8301932217b Call-ID=f3a04a86-2e55-4fb1-9936-af0dc3a402d8:
BYE sip:[email protected]:5070 SIP/2.0
Via: SIP/2.0/ ;branch=z9hG4bK-524287-1---f1f9e8301932217b;rport
Max-Forwards: 70
Route: <sip:190.14.152.11;lr;ep;pinhole=UDP:35.232.100.25:5060>
Contact: <sip:[email protected]:5060>
To: 40100188 <sip:[email protected]>;tag=jHb9PiatyjW8OOaA.o
From: <sip:[email protected]>;tag=cd4e952a
Call-ID: f3a04a86-2e55-4fb1-9936-af0dc3a402d8
CSeq: 3 BYE
Reason: SIP;text=Application Rejected Sdp(usually no common codec)
Content-Length: 0

11/15/2018 1:30:14 PM - SendMsg from <sip:[email protected]>;tag=cd4e952a to 40100188 <sip:[email protected]>;tag=jHb9PiatyjW8OOaA.o

11/15/2018 1:30:14 PM - Offer is rejected on Leg L:77.1[Line:10000<<50640100188];
OnOfferRejected Recv 488/INVITE from 190.14.152.11:5060 tid=da9a3207d3cbde09 Call-ID=f3a04a86-2e55-4fb1-9936-af0dc3a402d8:
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP 10.128.0.228:5060;branch=z9hG4bK-524287-1---da9a3207d3cbde09;rport=5060;received=35.232.100.25
To: 40100188 <sip:[email protected]>;tag=jHb9PiatyjW8OOaA.o
From: <sip:[email protected]>;tag=cd4e952a
Call-ID: f3a04a86-2e55-4fb1-9936-af0dc3a402d8
CSeq: 2 INVITE
Allow: INVITE, ACK, BYE, CANCEL, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS, UPDATE
Server: PortaSIP
Content-Length: 0


11/15/2018 1:30:14 PM - L:77.1[Line:10000<<50640100188] Sending: OnSendReq Send Req INVITE from 0.0.0.0:0 tid=77a32b3aa30fbf77 Call-ID=f3a04a86-2e55-4fb1-9936-af0dc3a402d8:
INVITE sip:[email protected]:5070 SIP/2.0
Via: SIP/2.0/ ;branch=z9hG4bK-524287-1---77a32b3aa30fbf77;rport
Max-Forwards: 70
Route: <sip:190.14.152.11;lr;ep;pinhole=UDP:35.232.100.25:5060>
Contact: <sip:[email protected]:5060>
To: 40100188 <sip:[email protected]>;tag=jHb9PiatyjW8OOaA.o
From: <sip:[email protected]>;tag=cd4e952a
Call-ID: f3a04a86-2e55-4fb1-9936-af0dc3a402d8
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE
Content-Type: application/sdp
Supported: replaces, timer
Content-Length: 584

v=0
o=3cxPS 4723175938064384 27072344145002498 IN IP4 35.232.100.25
s=3cxPS Audio call
c=IN IP4 35.232.100.25
t=0 0
m=audio 9114 RTP/SAVP 8 18 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:ma6dhWAEgWThRV3Q2qa92OYpgSVK/1kan6Ek3qXW
a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:QwuGfRfoZIetqIQYWfIcBhWCQXzUCjXlBfZ6kYp+
a=sendonly
m=audio 9114 RTP/AVP 8 18 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=sendonly

1/15/2018 1:30:14 PM - SendMsg from <sip:[email protected]>;tag=cd4e952a to 40100188 <sip:[email protected]>;tag=jHb9PiatyjW8OOaA.o
11/15/2018 1:30:14 PM - Providing SDP for L:77.1[Line:10000<<50640100188]
11/15/2018 1:30:14 PM - Reneg. from L:77.2[Ivr:800], offer
11/15/2018 1:30:14 PM - Remote SDP is set for leg L:77.2[Ivr:800]


11/15/2018 1:30:14 PM - Offer SDP arrived on Leg L:77.2[Ivr:800];
OnOffer Recv Req INVITE from 127.0.0.1:5483 tid=556e883e2418f75c Call-ID=0Xf2naKMe826dSiLfp5YUw..:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5483;branch=z9hG4bK-524287-1---556e883e2418f75c;rport=5483
Max-Forwards: 70
Contact: <sip:[email protected]:5483;rinstance=0d92bea19aed6dea>
To: "50640100188:Entrada"<sip:[email protected]:5060;nf=e>;tag=da90e15a
From: <sip:[email protected]>;tag=0d1d905f
Call-ID: 0Xf2naKMe826dSiLfp5YUw..
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Content-Type: application/sdp
Supported: replaces
User-Agent: 3CX IVR
Content-Length: 578

v=0
o=3cxVCE 367296861 18446744073208379696 IN IP4 127.0.0.1
s=3cxVCE Audio Call
c=IN IP4 127.0.0.1
t=0 0
m=audio 12076 RTP/AVP 0 8 3 112 18 13 97 9 111 110 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:112 opus/48000/2
a=fmtp:112 maxplaybackrate=16000;sprop-maxcapturerate=16000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:13 CN/8000
a=rtpmap:97 SPEEX/8000
a=rtpmap:9 G722/8000
a=rtpmap:111 iLBC/8000
a=fmtp:111 mode=30
a=rtpmap:110 iLBC/8000
a=fmtp:110 mode=20
a=rtpmap:101 telephone-event/8000
a=ptime:20
a=sendonly

 

[leejor]

Reason: SIP;text=Application Rejected Sdp(usually no common codec)

This is probably the error message to concentrate on. Do calls direct to an extension do the same thing?

 

[Rommel Otero]

inbound call goes to IVR (800) with 4 option

 

[leejor]

Try sending an incoming call direct to one extension, as a test, see if that works.

 

[Rommel Otero]

yes it works fine, if I call internal to IVR works fine, problems is just witch external calls

 

[cobaltit]

Have you tried turning off SRTP?

 

[Rommel Otero]

YES!!!! That was, but why? I mean I read somewhere, cant remenber where, best practice suggest that option to be enable

 

[YiannisH_3CX]

YES!!!! That was, but why? I mean I read somewhere, cant remenber where, best practice suggest that option to be enable

Not all providers support SRTP so if your provider does not support this option the call will be rejected. Incoming calls however will continue to work as the call will be offered without SRTP and the 3CX phone system will reply accordingly.

 

[YiannisH_3CX]

Glad to see the issue has been resolved