407 Proxy Authentication Required when trying to make call to du.ae

Hi all,

I am trying to connect my FreePBX to du.ae sip trunk.

Trunk details:

type=peer
host=5.32.4.225
outboundproxy=10.59.108.25
qualify=yes
fromdomain=5.32.4.225
fromuser=97145627500p
realm=5.32.4.225 
authuser=97145627500p 
secret=XXXX  
context=from-pstn-toheader
insecure=port,invite  
dtmfmode=rfc2833
directmedia=no
disallow=all
allow=ulaw
bindaddr=10.15.47.142

Trunk is reachable and well connected, but when I am trying to call via the trunk I am getting this error:

<--- SIP read from UDP:10.59.108.25:5060 --->
SIP/2.0 407 Proxy Authentication Required
Call-ID: [email protected]:5060
Via: SIP/2.0/UDP 151.253.188.7:5060;received=10.15.47.142;branch=z9hG4bK186f3623;rport=5060
To: <sip:5.32.4.225>;tag=64dbdea1-6656fa1e614d7c
From: "Unknown" <sip:[email protected]>;tag=as64de688c
CSeq: 102 OPTIONS
Date: Wed, 29 May 2024 09:49:18 GMT
Warning: 399 sbc.du.com "IP association no match, user not registered"
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS
Reliably Transmitting (NAT) to 10.59.108.25:5060:
OPTIONS sip:10.59.108.25 SIP/2.0
Via: SIP/2.0/UDP 151.253.188.7:5060;branch=z9hG4bK622fd97a;rport
Max-Forwards: 70
From: "Unknown" <sip:[email protected]>;tag=as25bec40e
To: <sip:10.59.108.25>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-16.0.40.7(18.16.0)
Date: Wed, 29 May 2024 08:14:45 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---


I am going out with the usable IP recieved from the trunk - 10.15.47.142.

Details in the trunk are correct and no issue with trunk connectivity, only when call is made via this one.

Any ideas? thanks in advance.

The log doesn’t show any call attempt (INVITE transaction) and 407 is not an error. Also you are using an obsolete and unsupported channel driver.

This is the actual error.

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You are sending from 10.15.47.142, but the Via (sent-by) header says 151.253.188.7, which I assume is your public IP. Asterisk uses that value when it is behind a NAT and believes the destination is on the public internet. I suspect that the 10.59.108.25 actual destination is reachable without NAT, but in Asterisk SIP Settings, Local Networks is not correctly set. Try including
10.0.0.0/8
After Submit and Apply Config, you must restart Asterisk.

If that’s not your issue, is your trunk registered? If not, what gets logged (if anything) when it attempts to register? If it is registered, do incoming calls work? If not, what gets logged on an attempted incoming call? If incoming calls are ok, please post an incoming INVITE, as well as an outgoing INVITE and the associated replies.

Also, please explain why you are trying to do this with the chan_sip driver, instead of pjsip.

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