Are you using the default dialplan? I think you might just need to ignore early media on your bridge app. If you are using the default.xml file then locate "Local_Extension" and the bridge line:<br><br> <action application="bridge" data="user/${dialed_extension}@${domain_name}"/><br>
<br>Change it to this, then try again:<br><br> <action application="bridge" data="{ignore_early_media=true}user/${dialed_extension}@${domain_name}"/><br><br>If I understand correctly, the "symptom" you are experiencing is the normal operation of the bridge app (and it's cousin, the originate API command). When the b-leg sends back media, including ringing, the bridge (or the originate) will be considered "successful," and in the case of bridge, the b-leg's audio (the early media) will be connected to the a-leg. If you set ignore_early_media=true then you are explicitly telling the bridge app that you only want to connect the b-leg to the a-leg if the b-leg actually answers.<br>
<br>I hope that made sense...<br><br>-MC<br><br><br><div class="gmail_quote">On Thu, Jun 23, 2011 at 9:32 PM, Max Alex <span dir="ltr"><<a href="mailto:max.asterisk@gmail.com">max.asterisk@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">Hi,<br>Thanks for reply.<br>Current scenario is below.<br><br>PSTN call -> sangoma device -> freeswitch incoming context -> routed to 1001 -> ringing (Answered on cellphone)<br>
Here when it is routed to 1001 the call it is started ringing, but on phone that call is answered and hearding sound of ringing.<br>
<br>Required flow:<br>PSTN call -> sangoma device -> freeswitch incoming context -> routed to 1001 -> ringing (Ringing on cellphone)<br><br>I hope it is understable, the call should not be answered until 1001 answer it, right not when it is started ring it is answered on cell phone.<br>
that should not be happen as it is not answered yet.<br><br>Waiting for your reply.<div class="im"><br><br clear="all">Thanks,<br>Max Alex<br>Voip Developer<br><br>
<br><br></div><div><div></div><div class="h5"><div class="gmail_quote">On Fri, Jun 24, 2011 at 5:48 AM, Michael Collins <span dir="ltr"><<a href="mailto:msc@freeswitch.org" target="_blank">msc@freeswitch.org</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0pt 0pt 0pt 0.8ex;border-left:1px solid rgb(204, 204, 204);padding-left:1ex">
I'm not sure I understand the problem. What is happening vs. what you believe should be happening?<br><font color="#888888">-MC</font><div><div></div><div><br><br><div class="gmail_quote">On Thu, Jun 23, 2011 at 3:31 AM, Max Alex <span dir="ltr"><<a href="mailto:max.asterisk@gmail.com" target="_blank">max.asterisk@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0pt 0pt 0pt 0.8ex;border-left:1px solid rgb(204, 204, 204);padding-left:1ex">Hi,<br>Thanks for your reply.<br>Here is my configuration and log <br><a href="http://pastebin.freeswitch.org/16571" target="_blank">http://pastebin.freeswitch.org/16571</a><br>
<br>I am using A200 analog sangoma device with freeswitch, it is working fine but when it is routing call to 1001 then it is answered.<br>
Please provider your suggestions.<br><br clear="all">Thanks,<br><font color="#888888">Max Alex<br>Voip Developer</font><div><div></div><div><br><br>
<br><br><div class="gmail_quote">On Wed, Jun 22, 2011 at 8:51 PM, Michael Collins <span dir="ltr"><<a href="mailto:msc@freeswitch.org" target="_blank">msc@freeswitch.org</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0pt 0pt 0pt 0.8ex;border-left:1px solid rgb(204, 204, 204);padding-left:1ex">
I thought the A200 was an analog card? Maybe I have my numbers mixed up...<div><br></div><div>Go ahead and collect a debug log of this call. It might help to have your configs posted as well. Use <a href="http://pastebin.freeswitch.org" target="_blank">pastebin.freeswitch.org</a>. See this wiki article for tips on how to collect information:</div>
<div><a href="http://wiki.freeswitch.org/wiki/Reporting_Bugs" target="_blank">http://wiki.freeswitch.org/wiki/Reporting_Bugs</a></div><div><br></div><div>-MC<br><br><div class="gmail_quote"><div><div></div><div>
On Wed, Jun 22, 2011 at 3:23 AM, Max Alex <span dir="ltr"><<a href="mailto:max.asterisk@gmail.com" target="_blank">max.asterisk@gmail.com</a>></span> wrote:<br>
</div></div><blockquote class="gmail_quote" style="margin:0pt 0pt 0pt 0.8ex;border-left:1px solid rgb(204, 204, 204);padding-left:1ex"><div><div></div><div>Hi,<br>I have installed freeswitch latest version with sangoma card A200 as well,<br>
I have installed and configured freetdm module with wanpipe drivers for freeswitch,<br>
We are properly receiving the inbound calls in public context and then we are routing that call to 1001 extension,<br>
it is properly routing to 1001 as well, but we have one issue while routing on 1001.<br><br>Here is the issue description.<br>I am calling from my cell phone to that DID number of pri line, and then it will start ringing on 1001 extension,<br>
When 1001 extension start ringing the call is answered on my cell phone,<br>it is something like freeswitch preanswer or autoanswer the call, how can i stop this answer call when it is ringing on 1001 extension,<br>Waiting for good reply.<br>
<br clear="all">Thanks,<br><font color="#888888">Max Alex<br><br><br>
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