[Asterisk-Users] app_prepaid NAT issue

[Brian Rathman]

Because whenever I place call from the phone without it going through the
prepaid application, I don't have any audio issues and the route between the
gateway and the phone is built correctly with NAT taken into account. When
the call goes through app_prepaid, a new dial command is issued via
app_prepaid:

line 567:
	res = prepaid_pbx_dial(chan, dialstr);

which then calls this:

line 369:
    app = pbx_findapp("Dial");
    if (app) {
        ret = pbx_exec(chan, app, data, 1);
    } else {
        ast_log(LOG_WARNING, "Could not find application (Dial)\n");
        ret = -2;
    }

I am assuming that this is where the issue is, but I am not famaliar enough
with the rest of the Asterisk code to know where to easily look next for
pbx_exec or what is required for NAT proxy to function correctly. I was
hoping that someone may have experienced something similar or could direct
me in the right direction.

Thanks,
Brian


-----Original Message-----
From: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com]On Behalf Of Brian K. West
Sent: Friday, June 18, 2004 8:15 PM
To: asterisk-users at lists.digium.com
Subject: Re: [Asterisk-Users] app_prepaid NAT issue


Its not an apps place to take nat int account.  WHERE did you get the idea
that it was?

bkw

----- Original Message -----
From: "Brian Rathman" <brian at ilk.com>
To: <asterisk-users at lists.digium.com>
Sent: Friday, June 18, 2004 5:03 PM
Subject: [Asterisk-Users] app_prepaid NAT issue


> I was able to get app_prepaid working, but unfortunately I am getting one
> way audio on the phone that I was placing the call from. It is behind NAT.
> It appears that the app_prepaid is not taking this into consideration
since
> I see:
>
> Jun 18 17:46:25 DEBUG[1133742896]: chan_sip.c:4130 build_route:
build_route:
> Contact hop: <sip:7708183799 at 192.168.1.101:5060;line=jet7pbic>
> Jun 18 17:46:25 DEBUG[1192491824]: rtp.c:1406 ast_rtp_bridge: Oooh,
> 'SIP/7708183799-8d6d' changed end address to 192.168.1.101:10094 (format
6)
> Jun 18 17:46:25 DEBUG[1192491824]: rtp.c:1408 ast_rtp_bridge: Oooh,
> 'SIP/7708183799-8d6d' was 65.202.115.115:10094/(format 6)
>
> Any help would be greatly appreciated.
>
> Thanks,
> Brian
>
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