[SR-Users] Routing PSTN calls from Asterisk through Kamailio to UAC & loose_route

[Vitaliy Aleksandrov]

According to your description BYE was sent using the information from 
R-URI which had no 5080 port.
Asterisk should have added port 5080 to the outgoing Invite contact so 
that it could be used for in-dialog routing.

Can you show a full trace with sip traffic between kamailio and 
asterisk. To catch sip traffic on all interfaces use "-i any" option for 
tcpdump or "-d any" for ngrep.

> I've been working on integration of Asterisk and Kamailio, currently on the
> same host with different ports, and have come across a problem with calls that
> originate from the Asterisk side (PSTN/DAHDI) and route through Kamailio to a
> SIP UAC.  In short, when the SIP UAC (10.1.1.9) sends the BYE, loose_route()
> is returning -1 and the BYE is routed back to Kamailio (10.1.1.1:5060) instead
> of Asterisk (10.1.1.1:5080).  I am using the stock WITHINDLG route
> configuration.
>
> RR module settings are as follows:
> modparam("rr", "enable_full_lr", 1)
> modparam("rr", "append_fromtag", 1)
>
> The BYE from the SIP UAC contains the following Route header which only
> contains the contents of Kamailio's Record-Route header.  I have attached the
> full sip trace for review as well.
>
> Route: <sip:10.1.1.1;lr=on;ftag=2400d939-de0b-4456-9e01-f9a3302f3e25;nat=yes>.
>
> What would be the best method to resolve this issue in either Asterisk or
> Kamailio?  Should I manually add a Record-Route header for the Asterisk
> host:port to Kamailio config? Is there something to be done in Asterisk?
>
> Thanks.  -A
>
>
>
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