Incoming call issue with Gradwell Trunk

[max1]

Hi,

I appreciate that Gradwell is not on your approved and supported VoIP provider list but unfortunately I am forced to work with this provider for the time being.

I can make outgoing calls via the Gradwell trunk however I am unable to receive calls. I have used Wireshark on the server and see the SIP invite messages however there is no response from 3CX.

The server is v14 Cloud on Windows 2012 R2 with public IP, therefor no NAT is involved. I suspect the issue is to do with the Inbound Parameters but the combination of things I have tried still fails.

Any and all advice would be of great help. Below is the SIP trace from Wireshark.

Thank you

Max

Session Initiation Protocol (INVITE)
Request-Line: INVITE sip:[email protected] SIP/2.0
Method: INVITE
Request-URI: sip:[email protected]
Request-URI User Part: 01234567890
Request-URI Host Part: 123.123.123.12
[Resent Packet: True]
[Suspected resend of frame: 1093]
Message Header
Via: SIP/2.0/UDP 109.224.240.232:5060;branch=z9hG4bK239134b6
Transport: UDP
Sent-by Address: 109.224.240.232
Sent-by port: 5060
Branch: z9hG4bK239134b6
Max-Forwards: 70
From: <sip:[email protected]>;tag=as7d0907fd
SIP from address: sip:[email protected]
SIP from address User Part: 07123456789
SIP from address Host Part: sip.gradwell.net
SIP from tag: as7d0907fd
To: <sip:[email protected]>
SIP to address: sip:[email protected]
SIP to address User Part: 01234567890
SIP to address Host Part: 123.123.123.12
Contact: <sip:[email protected]:5060>
Contact URI: sip:[email protected]:5060
Contact URI User Part: 07123456789
Contact URI Host Part: 109.224.240.232
Contact URI Host Port: 5060
Call-ID: [email protected]:5060
CSeq: 102 INVITE
Sequence Number: 102
Method: INVITE
User-Agent: Asterisk PBX 11.19.0-gw0.60.1
Date: Mon, 23 Nov 2015 16:18:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-ID: 1448295521.793233b375331bf2
sess-id: 1448295521
Content-Type: application/sdp
Content-Length: 322
Message Body
Session Description Protocol
Session Description Protocol Version (v): 0
Owner/Creator, Session Id (o): root 1928609658 1928609658 IN IP4 109.224.240.232
Owner Username: root
Session ID: 1928609658
Session Version: 1928609658
Owner Network Type: IN
Owner Address Type: IP4
Owner Address: 109.224.240.232
Session Name (s): Asterisk PBX 11.19.0-gw0.60.1
Connection Information (c): IN IP4 109.224.240.232
Connection Network Type: IN
Connection Address Type: IP4
Connection Address: 109.224.240.232
Time Description, active time (t): 0 0
Session Start Time: 0
Session Stop Time: 0
Media Description, name and address (m): audio 10334 RTP/AVP 8 0 18 101
Media Type: audio
Media Port: 10334
Media Protocol: RTP/AVP
Media Format: ITU-T G.711 PCMA
Media Format: ITU-T G.711 PCMU
Media Format: ITU-T G.729
Media Format: DynamicRTP-Type-101
Media Attribute (a): rtpmap:8 PCMA/8000
Media Attribute Fieldname: rtpmap
Media Format: 8
MIME Type: PCMA
Sample Rate: 8000
Media Attribute (a): rtpmap:0 PCMU/8000
Media Attribute Fieldname: rtpmap
Media Format: 0
MIME Type: PCMU
Sample Rate: 8000
Media Attribute (a): rtpmap:18 G729/8000
Media Attribute Fieldname: rtpmap
Media Format: 18
MIME Type: G729
Sample Rate: 8000
Media Attribute (a): fmtp:18 annexb=no
Media Attribute Fieldname: fmtp
Media Format: 18 [G729]
Media format specific parameters: annexb=no
Media Attribute (a): rtpmap:101 telephone-event/8000
Media Attribute Fieldname: rtpmap
Media Format: 101
MIME Type: telephone-event
Sample Rate: 8000
Media Attribute (a): fmtp:101 0-16
Media Attribute Fieldname: fmtp
Media Format: 101 [telephone-event]
Media format specific parameters: 0-16
Media Attribute (a): ptime:20
Media Attribute Fieldname: ptime
Media Attribute Value: 20
Media Attribute (a): sendrecv

 

[JonnyM]

Have you got your phone number listed on the next tab along, under "source identification by DID"? I have that set to To: User Part, and the Gradwell number in the box below in the non-+44 format, e.g. 01xxxxxxxxx

 

[jpillow]

Did you add the DID under source id---> source identification by DID? Also do you have the SIP trunk set to require registration for in and outbound calls? Whats your trunk set as?

 

[max1]

Hi,
Yes I have added it under source ID and tried To:User Part and Request Line URI: User Part.
The trunk is set to not require registration as Gradwell do not require it. You must specify the IP of the PBX in their Control Panel.
I'm really stumped!
Thanks

 

[lneblett]

You might take a look at the 3CX server log as well as the Security tab to see if the IP might be blocked. The log may provide more insight into the reason.

 

[max1]

I'm such an idiot :roll:

I forgot that with Cloud server the ports for inbound SIP are incremented from 5060 for instance 1 etc. Turns out all I needed to do was add 13060: to the SIP URL with the provider and boom it works!

FML

Lesson to us all. Read the manual!