Solved siptrunck.a1.net Austria

[axp]

Hi all,

I have installed 3CX at my site. Inbound Calls are working fine, but outbound calls are not working. always get "no connection under this number"

Has anyone installed a Call Profile for this Provider? I think iam just missing some Headers to send to A1.net.

so far,
Ferdinand

 

[leejor]

Look a the 3CX Activity Log to see if the calls are even making it to your provider, then being rejected. It may be an outbound rule issue.

 

[axp]

01/15/2021 9:20:10 PM - L:62.1[Extn:20] got Terminated Recv Req BYE from 127.0.0.1:5080 tid=Pjb26ad482-86dc-4a13-ae72-f13992660174 Call-ID=2469cc7f-09e1-4938-a82e-a6632f1bb1e4: BYE sip:[email protected]:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK-524287-2---Pjb26ad482-86dc-4a13-ae72-f13992660174;rport=5080 Via: SIP/2.0/UDP [::ffff:172.16.0.216]:36116;branch=z9hG4bK-524287-1---tunneltid;rport;tnlid=clnt.0-62b20ee6-eafa-4303-9338-d703b46b232b_116797545 Via: SIP/2.0/TCP 127.0.0.1:50195;rport;branch=z9hG4bKPjb26ad482-86dc-4a13-ae72-f13992660174;alias Max-Forwards: 69 Record-Route: <sip:[email protected]:5080;user=proxy;uri=clnt.0-62b20ee6-eafa-4303-9338-d703b46b232b_116797545> To: <sip:[email protected];tag3cx=3798b773-7905-4746-8cf8-73d1168ee482>;tag=204c2125 From: "plop" <sip:[email protected]>;tag=cac573e7-17cb-4f72-b6ad-e792f6722fd6 Call-ID: 2469cc7f-09e1-4938-a82e-a6632f1bb1e4 CSeq: 13909 BYE User-Agent: 3CXPhone for Android 16.7.0.23 Content-Length: 0

it seems that the call doesn't even makes it to the trunk.

 

[leejor]

I don't see anything saying the it does not match a rule, but that may still be the case as the log may require more detail to be shown.. Looks like the number is in the full, international format (leading +). Have you setup 3CX to use E164 rules, to convert numbers, in that format, to one that is used by your provider? (convert the + to a "standard" prefix?)

 

[axp]

Hi Thanks for your information, but it hink i know now what the problem may be:

16.01.2021 10:05:27 - L:1.1[Extn:20] got Connected.UAS Send 200/INVITE from 0.0.0.0:0 tid=Pj01620b5a58bf407cafd3c5e8c944baea Call-ID=e80e61ac7cae4612b5721fa7acd5b44b: SIP/2.0 200 OK Via: SIP/2.0/TCP 172.16.0.53:56825;rport=56825;branch=z9hG4bKPj01620b5a58bf407cafd3c5e8c944baea;alias Require: timer Contact: <sip:[email protected]:5060;transport=TCP> To: <sip:[email protected];tag3cx=192614a0f7c04eecb9961865593ec141>;tag=4f631728 From: "F Hr"<sip:[email protected]>;tag=bd707eb5b4194aa28b7ad0bd7512c1d6 Call-ID: e80e61ac7cae4612b5721fa7acd5b44b CSeq: 27158 INVITE Session-Expires: 1800;refresher=uac Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.0.451 (451) Content-Length: 369 v=0 o=3cxVCE 2047967134 619417522 IN IP4 172.16.56.198 s=3cxVCE Audio Call c=IN IP4 172.16.56.198 t=0 0 m=audio 12000 RTP/AVP 3 18 0 8 9 96 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 m=video 0 RTP/AVP 31 c=IN IP4 127.0.0.1

It doesn't really make any difference what i change in the outbound rules of the siptrunk, it doesn't change those Parameters, i tried to set different options for the to and from Host part, but nothing is changing.

maybe my Phone doesnt't obey the rules (3cx softphone) or doesn't know whoch trunk/route to go.


so far, Thanks in advance.

 

[YiannisH_3CX]

Hi @axp

I would recommend using a supported provider from our supported providers list: https://www.3cx.com/partners/sip-trunks/

Also i see you are using v18 which is fine for testing purposes but it is not suited for production just yet as it's a very early version. If you plan on using the PBX for production purposes i would recommend installing the latest V16 version.

 

[leejor]

maybe my Phone doesnt't obey the rules (3cx softphone) or doesn't know whoch trunk/route to go.

The phone (originating SIP device), does not have to know how to route a call. That is done by the PBX based on the number sent in the Invite, and your outbound rules (for external calls). If that part is done correctly, then you would see a log of the call being sent to your provider. It is then up to them if they accept the call, or send back a message indicating that they have rejected it for a some reason. If the calls continue to fail, the Activity Log will tell you how it progresses through the PBX, and at what stage it fails. But, as i said, you may need to set the log to provide more details (Verbose mode).

The number does include a leading +, which must be accommodated, usually with the E164 settings..

 

[axp]

Thanks,

i had set the outbound ruls wrong, niw everything seems to work for now.

Thanks everyone


Bottom line: check your outbound rules

 

[YiannisH_3CX]

Glad to see the issue has been resolved