Xlite works, asterisk returns "403 forbidden" ..logs include

Hi all:

Has anyone here tried using indigonetworks’ SIP line with Asterisk? There SIP account worked fine with Xlite. And I was able to register SIP account with Asterisk. However, when I try to make calls through these trunks. All I get was the 403 forbidden message back from the provider. Nothing much informational. And believe me that I have tried every combination of trunk settings there would be trying to dial out with these SIP lines.

Here is a copy of the log:

– Executing [s@macro-dialout-trunk:19] Dial(“SIP/705-0000007f”, “SIP/2420000000-out/12423273792,300,”) in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Audio is at 5060
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x1000 (g722) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 69.4.164.10:5060:
INVITE sip:12423273792@nas-sbc-01.srg.com.bs;user=phone SIP/2.0
Via: SIP/2.0/UDP MY-IP:5060;branch=z9hG4bK231d0451;rport
Max-Forwards: 70
From: “2420000000” sip:2420000000@sia-nas01ca146.srg.com.bs;tag=as6dcfa14f
To: sip:12423273792@nas-sbc-01.srg.com.bs;user=phone
Contact: sip:2420000000@MY-IP:5060
Call-ID: 42a9088e539b4e94664ee67531b07880@sia…146.srg.com.bs
CSeq: 102 INVITE
User-Agent: xlite
Date: Mon, 01 Aug 2011 20:19:15 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 292

v=0
o=root 674195259 674195259 IN IP4 MY-IP
s=xlite
c=IN IP4 MY-IP
t=0 0
m=audio 19460 RTP/AVP 0 8 3 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv


– Called 2420000000-out/12423273792

<— SIP read from UDP:69.4.164.10:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP MY-IP:5060;received=MY-IP;branch=z9hG4bK231d0451;rport=52056
From: “2420000000” sip:2420000000@sia-nas01ca146.srg.com.bs;tag=as6dcfa14f
To: sip:12423273792@nas-sbc-01.srg.com.bs;user=phone
Call-ID: 42a9088e539b4e94664ee67531b07880@sia…146.srg.com.bs
CSeq: 102 INVITE

<------------->
— (6 headers 0 lines) —

<— SIP read from UDP:69.4.164.10:5060 —>
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP MY-IP:5060;received=MY-IP;branch=z9hG4bK231d0451;rport=52056
From: “2420000000” sip:2420000000@sia-nas01ca146.srg.com.bs;tag=as6dcfa14f
To: sip:12423273792@nas-sbc-01.srg.com.bs;user=phone;tag=aprqngfrt-i40hac20000c6
Call-ID: 42a9088e539b4e94664ee67531b07880@sia…146.srg.com.bs
CSeq: 102 INVITE

<------------->
— (6 headers 0 lines) —
Transmitting (NAT) to 69.4.164.10:5060:
ACK sip:12423273792@nas-sbc-01.srg.com.bs;user=phone SIP/2.0
Via: SIP/2.0/UDP MY-IP:5060;branch=z9hG4bK231d0451;rport
Max-Forwards: 70
From: “2420000000” sip:2420000000@sia-nas01ca146.srg.com.bs;tag=as6dcfa14f
To: sip:12423273792@nas-sbc-01.srg.com.bs;user=phone;tag=aprqngfrt-i40hac20000c6
Contact: sip:2420000000@MY-IP:5060
Call-ID: 42a9088e539b4e94664ee67531b07880@sia…146.srg.com.bs
CSeq: 102 ACK
User-Agent: xlite
Content-Length: 0

Can someone please provide a working sample sip.conf for Indigo?

I am also attaching below:

  1. diagnostic log from my x-lite that works with default settings.
  2. x-lite screenshots.

download link: ge.tt/9ZWyfX6?c

I’m using asterisk 1.4.21.2 but had the same issue with 1.8.4.1.

thank you in advance for any help…

any ideas? big mystery :confused: