check if the authentication password is set correct and the same on both phone and cisco side or reset it on both phone and Cisco sides.

- Are both phones shown at Brekeke SIP Sever/Registered Clients page?
- capture packets at Brekeke server and check what error response is sent for the second phone's register request

- is the caller registered at pbx/sip server? - can the caller phone call to other extensions without issue? - is there ARS used for the conference call? - capture packets at pbx server machine and check error response for conference call and who sent error response - is there any error message ...

check the following setting at Cisco side:
- if brekeke server IP is set in Cisco SIP trunk
- if 172.16.1.10 is the primary handler IP address at cisco side for receiving calls

http://www.brekeke-sip.com/bbs/viewtopic.php?t=7468

what happen if set b2bua off?

- what dial plan rules are created at brekeke server? paste them in order shown from Brekeke SIP Server - What transport is used at Cisco side? if it is tcp, need to add following line in the dial plan rule "Deploy Patterns" for calls sent to Cisco side Deploy patterns $transport = tcp - also ...

how about Radius server?
http://wiki.brekeke.com/wiki/Radius-Server

you may need session plug-in to modify the requests in the middle of a session
http://www.brekeke.com/support/sip_document.php
"Developer's Guide: Accounting Plug-in"

Advanced edition of Brekeke SIP Server supports TLS and convert to other transport with $transport setting in dial plan rule "Deploy Patterns" as wiki page below http://wiki.brekeke.com/wiki/how-to-convert-tls-to-another-transport-protocol if the call is to UA registered at Brekeke SIP Server ...

Rule: Accept calls from the other server ------------------------------- [Matching Patterns] $request = ^INVITE $addr = other_server_IP_address To = sip:(.+)@ [Deploy Patterns] $auth = false To = sip:%1@ ------------------------------- Rule: Send calls to the other server ...

for the client registered through VPN, set dial plan rule in wiki page
http://wiki.brekeke.com/wiki/Brekeke-settings-for-VPN

Capture packets at Brekeke server and check - Is there SDP shown in the INVITE packets sent from Cisco? - if there is any session timer, Alert-Info, P-Location headers used at cisco invite, disable it from cisco side or add the dial plan rule "from cisco" as in doc also add "deploy patterns" the ...

http://wiki.brekeke.com/wiki/Brekeke-PBX-v3-Quickstart check setup as wiki page above - is there any sip users shown from sip server side "Registered Clients" side? - did you assign these registered users in pbx user extensions / phones page? - capture packets at brekeke server and check what error ...

http://www.brekeke.com/doc/sip/rauland_ ... _bssv3.pdf

check if there is SDP in the invite sent from cisco to brekeke as in doc above section 2 - 2.1 Enable the MTP (Media Termination Point Required)

did you set 171.214.248.32 in sip server "Interface Address" field and restart sip server?
capture packets and check if phone register request sent to sip server and what response from sip server

compare packets of register between xlite and mobile device for difference
maybe there is route to other address in mobile device's register request?

if the same issue happen with other phone registration? like register with xlite
is there upper registration set at brekeke server configuration/SIP page?
capture packets about registration and check if From and To header are the same IP in register request sent from phone

check if you set any upper registration or dial plan to route the register to other IP

there is sample at
http://www.brekeke.com/downloads/pbx-developer-tool.php

did you set port forwarding at router at Brekeke side and external IP at Brekeke SIP Server?
http://wiki.brekeke.com/wiki/Set-port-forwarding

create dial plan rule to catch the call from/to this user and set $rtp=on in rule [Deploy Patterns] but not sure if above setting will solve your issue. In v3.3.x there are following settings at "RTP" page maybe it will solve the issue Port mapping ->source port Send UA's remote address -> yes