<span class="re6"><span class="nu0">Hi,<br>Thanks for your reply.<br>I have enabled logger as per your help.<br>I have given completed log on following link of pastebin.<br><a href="http://pastebin.freeswitch.org/16616">http://pastebin.freeswitch.org/16616</a><br>
<br>You can see this line as it is showing answered and this call is answered on my cell phone but on softphone it is ringing and i have rejected from there.<br>2011</span><span class="nu0">-06</span><span class="nu0">-28</span> <span class="nu0">17</span>:<span class="nu0">47</span>:<span class="nu0">52.909657</span> <span class="br0">[</span>NOTICE<span class="br0">]</span> mod_freetdm.c:<span class="nu0">1953</span> Channel <span class="br0">[</span>FreeTDM/<span class="nu0">2</span>:<span class="nu0">1</span>/<span class="nu0">01234567890</span><span class="br0">]</span> has been answered <br>
<br>It is pre answering the cell phone when it is ringing on phone<br>Waiting for your help.<br><br clear="all"></span>Thanks,<br>Max Alex<br>Voip Developer<br><br>
<br><br><div class="gmail_quote">On Mon, Jun 27, 2011 at 9:15 PM, Michael Collins <span dir="ltr"><<a href="mailto:msc@freeswitch.org">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;">
Get a complete, unedited, unabridged console debug log w/ siptrace and put it on pastebin w/ "FreeSWITCH Log" for the syntax highlighting. Use "sofia global siptrace on" to make sure you can all SIP traffic.<br>
<font color="#888888">
<br>-MC</font><div><div></div><div class="h5"><br><br><div class="gmail_quote">On Mon, Jun 27, 2011 at 6:05 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 reply,<br>I have tried the same way and reloaded freeswitch, but still it is answered on first ring of the call.<br>When it is ringing the call on 1001 and the same time it is answered on cell phone, so something is done when it is ringing on 1001.<br>
<br>Here is the dialplan for the same<br><br> <extension name="Local_Extension"><br> <condition field="destination_number" expression="^(10[01][0-9])$"><br> <action application="set" data="dialed_extension=$1"/><br>
<action application="export" data="dialed_extension=$1"/><br> <action application="bind_meta_app" data="2 b s record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/><br>
<action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/>--><br> <action application="set" data="ringback=${us-ring}"/><br>
<action application="set" data="transfer_ringback=$${hold_music}"/><br> <action application="set" data="call_timeout=30"/><br> <action application="set" data="hangup_after_bridge=true"/><br>
<action application="set" data="continue_on_fail=true"/><br> <action application="hash" data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/><br>
<action application="hash" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/><br> <action application="set" data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}"/><br>
<action application="hash" data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/><div><br> <action application="bridge" data="{ignore_early_media=true}user/${dialed_extension}@${domain_name}"/><br>
</div>
<action application="sleep" data="1000"/><br> <action application="voicemail" data="default ${domain_name} ${dialed_extension}"/><br> </condition><br>
</extension><br><br><br>Please help me for this issue.<div><br><br clear="all">Thanks,<br>Max Alex<br>Voip Developer<br><br>
<br><br></div><div><div></div><div><div class="gmail_quote">On Fri, Jun 24, 2011 at 11:38 AM, Michael Collins <span dir="ltr"><<a href="mailto:msc@freeswitch.org" target="_blank">msc@freeswitch.org</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
Are you using the default dialplan? I think you might just need to ignore early media on your bridge app. If you are using the default.xml file then locate "Local_Extension" and the bridge line:<br><br> <action application="bridge" data="user/${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><font color="#888888"><br>-MC</font><div><div></div><div><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" 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 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><br><br clear="all">Thanks,<br>Max Alex<br>Voip Developer<br><br>
<br><br></div><div><div></div><div><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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