Direct RTP connection

stavros · **Posted:** Tue Apr 14, 2009 4:09 am

zane9362 wrote:

So what is it that this string does in the Class.RateEngine.php file.

"* 1000" does exactly what it says; it multiplies by 1,000 (thereby converting seconds into milliseconds).

Quote:

Will this affect the billing?

Yes, as "timeout" intimates, this is how we tell Asterisk the maximum duration of the call.

Quote:

If I take the "* 1000" out of the string then the call will connect for 2 sec and terminate. WHY?

The changes you have made are incorrect in some way. If the call only lasts 1/1,000th as long as funds should permit, it's very likely you're still using L() rather than S().

Quote:

$dialparams = str_replace("%timeout%", min($timeout , $max_long), $A2B->agiconfig['dialcommand_param']);
$dialparams = str_replace("%timeout%", min($timeout * 1000, $max_long), $A2B->agiconfig['dialcommand_param']);

I'm grasping at straws here, but you must replace the existing code rather than copy & paste it then change the 2nd copy.

Quote:

I have tried the following and have no idea if it is correct.

As stated several times already in this thread, the correct setting would be:

Code:

dialcommand_param = "|60|gS(%timeout%)"

I hope borisat2billing doesn't mind me editing his post to correct his typo; it seems to be leading folks astray.

dgeorge2009 · **Joined:** Thu Apr 09, 2009 4:38 am **Posts:** 9

Hi Guys,

I have been reading this thread several times to make sure I do this correctly. I have two questions

1) Will A2b be able to bill the call correctly if RTP is not going through asterisk server?
2) Will it be able to terminate the call when the funds run out?
3) Can this work is my SIP ATA is usign NAT?

Thanks,
Dave

jimmy888 · **Joined:** Wed Apr 01, 2009 6:09 am **Posts:** 5

I have tried both
dialcommand_param = "|60|gS(%timeout%)"
with *1000 change in Class.RateEngine.php,

and
dialcommand_param = "|60|gL(%timeout%)"
only.

when I run "tcpdump udp" to check traffic, I found that the voice rtp traffic from provider came to my a2b, and then from my a2b to my softphone. so it looks as if the re-invite is not working at all, my asterisk is still in the media path.

I read another thready saying that
"You have to set the reinvite=yes AND under general settings don't set the "tr" etc. option". I did that too. and it doesn't help

I then tried to make sure the asterisk is not doing transcoding, so I forced it to use ulaw codec by setting :

disallow=all
allow=ulaw
nat=no

the tcpdump still shows that the traffic is going through my asterisk.

in my a2billing.conf and sip.conf , I have reinvite=yes all explicitly set.

what else can be changed to get this reinvite working so that the media goes directly between the 2 phones ?

I used software xlite behind a NAT as one phone, and my cell phone as another phone to do the test.

dcitelecom · **Posted:** Tue Apr 23, 2013 2:50 am

Is this still valid for the current 2.0 version?
Set dialcommand_param = "|60|S(%timeout%)"
and remove *1000 from line 1271 in Class.RateEngine.php

provided by Star2Billing S.L.

Direct RTP connection

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