Test Call from New providor SBC

This ones a little different.

We are working with a new to Sip providor.

They are sending us a test invite but i cant even see it hit our side this is what they have sent us but im reaching out for wisdom please.

Is there invite correct for us to pickup the call. Ive **** ome of the details for security.

Thanks in advance.

INVITE tel:+44*****770771 SIP/2.0

Via: SIP/2.0/UDP 10.***.***.58:5060;branch=z9hG4bKexebkexadlibcwdwcxcvxklxe;Role=3;Hpt=8e78_16;TRC=ffffffff-ffffffff

Record-Route: <sip:10.***.***.58:5060;transport=udp;lr;Hpt=8e78_16;CxtId=4;TRC=ffffffff-ffffffff;X-HwB2bUaCookie=19023>

Call-ID: isbcyu4a9u9u8y563t5bftx8585yt92utf1x@B.5.103.****.**re.com

From: "Anonymous"<sip:anonymous@anonymous.invalid>;tag=73esxews

To: <sip:+44*****770771@34.****.****.58;transport=udp;user=phone>



Contact: <sip:10.***.***.58;transport=udp;Dpt=f29a-200;Hpt=8e78_16;CxtId=4;TRC=ffffffff-ffffffff>

Max-Forwards: 64

Supported: timer,100rel,histinfo,precondition

Session-Expires: 1800

Min-SE: 600

Privacy: id

P-Charging-Vector: icid-value=AC57D2E3EC1F8201810513223;orig-ioi=10.***.***.46;term-ioi=***.****.com

P-Early-Media: supported

Content-Length: 621

Content-Type: application/sdp


o=- 33568 33568 IN IP4 10.***.***.56

s=SBC call

c=IN IP4 10.***.***.56

t=0 0

m=audio 21066 RTP/AVP 8 102 116 100 107 105 3 106

a=rtpmap:8 PCMA/8000

a=rtpmap:102 AMR/8000

a=rtpmap:116 telephone-event/8000

a=rtpmap:100 AMR/8000

a=fmtp:100 mode-set=0,2,4,7;mode-change-neighbor=1;mode-change-period=2

a=rtpmap:107 AMR/8000

a=fmtp:107 mode-set=0,2,4;mode-change-neighbor=1;mode-change-period=2

a=rtpmap:105 GSM-EFR/8000

a=rtpmap:3 GSM/8000

a=rtpmap:106 GSM-HR/8000


a=curr:qos local none

a=curr:qos remote none

a=des:qos optional local sendrecv

a=des:qos optional remote sendrecv

Ive sent them the trace for what we can see as to be fair it looks fine on first look. Mean while they also have specific requirements for making a call and im struggling getting the FROM format correct. It needs to be -

From: "6666" <sip:FreeSWITCH@46.***.***.106>;tag=pZ6yXv8BaS8Zc

**** Should be sip:+44*****770771@ims.s***.com ******

They also asked if i could add this tag in the URI header - user=phone

Any Ideas how i configure this in fusion. Ive tried altering @ the gateway without success.


Staff member
Your PBX is behind NAT???

In your sip profiles try setting ext-sip-ip and ext-rtp-ip to autonat:X.X.X.X where X.X.X.X is your public IP.

flush memcache and restart freeswitch.
Hi Digi

Thanks for the steer yesterday. Im on outbound now and managed to adjust the invite to the providors requirements except a couple of things

1. They dont want to see the Ext at the beginning of the FROM
2. They would like to see user=phone in the URI header.

Any advice - Matts also helping out.

Kind Regards


Hi Digi

Just been test with the providor as although we are now sending the right invite its not even hitting them with this in the bridge statement. As soon as i remove the call can be sent again but ofcoarse it fails as they want to see the correct invite.

This is the only clue i get in SNGREP
Any Ideas?

Kind Regards