<div dir="ltr">Hi all,<br><br>So how to open JIRA ticket bcoz I don't have any idea of that <br><br><div class="gmail_quote">On Mon, Dec 5, 2011 at 7:43 PM, Danny Nicholas <span dir="ltr"><<a href="mailto:danny@debsinc.com">danny@debsinc.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">Not my idea - just what I came across on google - probably should open a<br>
JIRA issue so it gets "really resolved" instead of hit-and-miss patching.<br>
<div><div></div><div class="h5"><br>
-----Original Message-----<br>
From: <a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a><br>
[mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a>] On Behalf Of Paul Belanger<br>
Sent: Saturday, December 03, 2011 2:38 PM<br>
To: <a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a><br>
Subject: Re: [asterisk-users] video calls not working<br>
<br>
On 11-11-21 10:07 AM, Danny Nicholas wrote:<br>
> Two items<br>
><br>
> #1 you only need 1 disallow=all in your sip.conf definition<br>
><br>
> #2 you need to patch rtp.c to define 126 as FORMAT_H263 - this is an<br>
> xlite response to Asterisk starting music-on-hold during the connect<br>
> pause. The r on the dial command attempts to do a "faux ring" which<br>
> xlite interprets as a MOH request, so if you don't want to<br>
> patch/recompile, just take the r off of Dial.<br>
><br>
Why are you manually patching asterisk? Have you created an issue in JIRA<br>
about this?<br>
<br>
--<br>
Paul Belanger<br>
Digium, Inc. | Software Developer<br>
twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at:<br>
<a href="http://digium.com" target="_blank">http://digium.com</a> & <a href="http://asterisk.org" target="_blank">http://asterisk.org</a><br>
<br>
--<br>
_____________________________________________________________________<br>
-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> -- New to<br>
Asterisk? Join us for a live introductory webinar every Thurs:<br>
<a href="http://www.asterisk.org/hello" target="_blank">http://www.asterisk.org/hello</a><br>
<br>
asterisk-users mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
<a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>
<br>
<br>
--<br>
_____________________________________________________________________<br>
-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>
New to Asterisk? Join us for a live introductory webinar every Thurs:<br>
<a href="http://www.asterisk.org/hello" target="_blank">http://www.asterisk.org/hello</a><br>
<br>
asterisk-users mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
<a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>
</div></div></blockquote></div><br><br clear="all"><br>-- <br><div dir="ltr"><br>Thanks and regards<br><br> Virendra Bhati<br>+91-8885268942<br>Software Engineer<br></div><br>
</div>