<html><body><div style="color:#000; background-color:#fff; font-family:times new roman, new york, times, serif;font-size:12pt"><div><span><div>Hi,</div><div><br></div><div>As you said create new extensions for flex client and h323 endpoint(Ekiga softphone).</div><div>I created two new extensions 1103.xml for flex client and 1104.xml for ekiga soft phone.</div><div>I login using 1103 on flex client and make call to 1104.</div><div>Now when I call from flex client(1103) to Ekiga softphone(1104),ekiga phone rings and when i answer the </div><div>call ,call terminates immediately by showing local user rejected call.</div><div><br></div><div>In another case where I call from Ekiga phone(1104) to flex client(1103),it says person at 1104 is </div><div>not available and says record your voice mail.In this case flex client did not get any incoming call.</div><div><br></div><div><br></div><div>Here is my default.xml</div><div><br></div><div><extension
name="Local_Extension" continue="true"></div><div> <condition field="destination_number" expression="^(1104)$"></div><div> <action application="export" data="dialed_extension=$1"/></div><div> <action application="set" data="effective_caller_id_number=${dialed_extension}"/></div><div> <!--action application="set" data="call_timeout=30"/--></div><div> <!--action application="set" data="hangup_after_bridge=true"/--></div><div> <action application="set" data="bypass_media=true"/></div><div> <action application="bridge" data="user/${dialed_extension}@${domain_name}"/></div><div> </condition> </div><div><br></div><div> <condition field="destination_number" expression="^(1103)$"></div><div>
<action application="export" data="dialed_extension=$1"/></div><div> <action application="set" data="effective_caller_id_number=${dialed_extension}"/></div><div> <!--action application="set" data="call_timeout=30"/--></div><div> <!--action application="set" data="hangup_after_bridge=true"/--></div><div> <action application="set" data="bypass_media=true"/></div><div> <action application="bridge" data="user/${dialed_extension}@${domain_name}"/></div><div> </condition> </div><div></extension></div><div><br></div><div>Correct me if my dialstrings are wrong.</div><div>Tell me if any other changes to made in other files also.</div><div>How can I make both way calling ?Please help me its emergency , I have demo next
week. </div><div><br></div><div>-Navnath.</div><div><br></div><div><br></div><div><br></div><div style="font-family: 'times new roman', 'new york', times, serif; font-size: 12pt;"><br></div></span></div><div style="font-family: 'times new roman', 'new york', times, serif; font-size: 12pt;"><br></div> <div style="font-family: 'times new roman', 'new york', times, serif; font-size: 12pt;"> <div style="font-family: 'times new roman', 'new york', times, serif; font-size: 12pt;"> <div dir="ltr"> <hr size="1"> <font size="2" face="Arial"> <b><span style="font-weight:bold;">From:</span></b> "freeswitch-users-request@lists.freeswitch.org" <freeswitch-users-request@lists.freeswitch.org><br> <b><span style="font-weight: bold;">To:</span></b> freeswitch-users@lists.freeswitch.org <br> <b><span style="font-weight: bold;">Sent:</span></b> Thursday, 2 May 2013 8:59 PM<br> <b><span style="font-weight: bold;">Subject:</span></b> FreeSWITCH-users Digest,
Vol 83, Issue 10<br> </font> </div> <div class="y_msg_container"><br>----- Forwarded Message -----<br><br>Send FreeSWITCH-users mailing list submissions to<br> <a ymailto="mailto:freeswitch-users@lists.freeswitch.org" href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a><br><br>To subscribe or unsubscribe via the World Wide Web, visit<br> <a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br>or, via email, send a message with subject or body 'help' to<br> <a ymailto="mailto:freeswitch-users-request@lists.freeswitch.org" href="mailto:freeswitch-users-request@lists.freeswitch.org">freeswitch-users-request@lists.freeswitch.org</a><br><br>You can reach the person managing the list at<br> <a
ymailto="mailto:freeswitch-users-owner@lists.freeswitch.org" href="mailto:freeswitch-users-owner@lists.freeswitch.org">freeswitch-users-owner@lists.freeswitch.org</a><br><br>When replying, please edit your Subject line so it is more specific<br>than "Re: Contents of FreeSWITCH-users digest..."<br><br>Today's Topics:<br><br> 1. make ekiga to flex call (Navnath Sonavne)<br> 2. Re: make ekiga to flex call (Brian Foster)<br> 3. Re: Check if UA is still there? (mehroz)<br> 4. Re: Check if UA is still there? (Ken Rice)<br> 5. Re: Check if UA is still there? (mehroz)<br> 6. Re: Check if UA is still there? (Michael Collins)<br><div id="yiv6395193712"><div><div style="color: rgb(0, 0, 0); background-color: rgb(255, 255, 255); font-family: 'times new roman', 'new york', times, serif; font-size: 12pt;"><div><div><font face="times new roman, new york, times, serif">Hi All,</font></div><div><font face="times new roman, new
york, times, serif"><br></font></div><div><font face="times new roman, new york, times, serif">I am using flex client(at 192.168.9.165) given in freeswitch source </font></div><div><font face="times new roman, new york, times, serif">to make call to Ekiga(h323 protocol) softphone(at 192.168.8.27) via freeswitch server(at 192.168.8.41).</font></div><div><font face="times new roman, new york, times, serif">I have two users registered on freeswitch with extension as 1100 and 1101 in default context.</font></div><div><font face="times new roman, new york, times, serif">I loged in using one of extension in flex client,then I dail another extension. </font></div><div><font face="times new roman, new york,
times, serif">After I dail,call goes to Ekiga softphone at 192.168.8.27 because I have made changes </font></div><div><font face="times new roman, new york, times, serif">in default.xml accordingly to forward calls to this ip.</font></div><div><font face="times new roman, new york, times, serif">Here is My default.xml part :</font></div><div><font face="times new roman, new york, times, serif"><br></font></div><div><font face="times new roman, new york, times, serif"><extension name="Local_Extension"></font></div><div><font face="times new roman, new york, times, serif"> <condition field="destination_number" expression="^(\d{4})$"></font></div><div><font face="times new roman, new york, times, serif"> <action application="export" data="dialed_extension=$1"/></font></div><div><font face="times new roman, new york, times, serif"> <action application="set"
data="effective_caller_id_number=${dialed_extension}"/></font></div><div><font face="times new roman, new york, times, serif"> <action application="set" data="PEER_IP=192.168.8.27"/></font></div><div><font face="times new roman, new york, times, serif"> <action application="set" data="call_timeout=30"/></font></div><div><font face="times new roman, new york, times, serif"> <action application="set" data="hangup_after_bridge=true"/></font></div><div><font face="times new roman, new york, times, serif"> <action application="set" data="proxy_media=true"/></font></div><div><font face="times new roman, new york, times, serif"> <action application="bridge" data="h323/$1@${PEER_IP}"/></font></div><div><font face="times new roman, new york, times, serif"> </condition>
</font></div><div><font face="times new roman, new york, times, serif"></extension></font></div><div><font face="times new roman, new york, times, serif"><br></font></div><div><font face="times new roman, new york, times, serif">Ekiga phone can answer the call and there is audio transmission from both end successfully.</font></div><div><font face="times new roman, new york, times, serif"><br></font></div><div><font face="times new roman, new york, times, serif">But this is one way calling i.e. from flex to ekiga.</font></div><div><font face="times new roman, new york, times, serif">Now I want to call from Ekiga softphone to flex client.</font></div><div style="color: rgb(0, 0, 0); font-size: 16px; font-family: 'times new roman', 'new york', times, serif; background-color: transparent; font-style: normal;"><font face="times new roman, new york, times, serif">How to call flex client from ekiga
softphone?</font></div><div><font face="times new roman, new york, times, serif"><br></font></div><div><font face="times new roman, new york, times, serif">Anybody please me on this issue.</font></div><div><font face="times new roman, new york, times, serif"><br></font></div><div><font face="times new roman, new york, times, serif">-Navnath.</font></div><div><font face="times new roman, new york, times, serif"><br></font></div><div><font face="times new roman, new york, times, serif"><br></font></div><div><font face="times new roman, new york, times, serif"><br></font></div><div style="font-family: 'times new roman', 'new york', times, serif; font-size: 12pt;"><br></div></div></div></div></div><div id="yiv6395193712"><div dir="ltr">According to your dialplan you're forcing all calls that hit Local_Extension (any call with a destination number of 4 digits) to call your ekiga softphone. Your other problem lies in the fact that you should have
different dialplans for different methods of calling unlike endpoints one for h232 one for flex.</div>
<div dir="ltr">See below:</div>
<div dir="ltr">-BDF<br>
On May 2, 2013 9:05 AM, "Navnath Sonavne" <<a rel="nofollow" ymailto="mailto:navnath.sonavne@yahoo.com" target="_blank" href="mailto:navnath.sonavne@yahoo.com">navnath.sonavne@yahoo.com</a>> wrote:<br>
><br>
> Hi All,<br>
><br>
> I am using flex client(at 192.168.9.165) given in freeswitch source <br>
> to make call to Ekiga(h323 protocol) softphone(at 192.168.8.27) via freeswitch server(at 192.168.8.41).<br>
> I have two users registered on freeswitch with extension as 1100 and 1101 in default context.<br>
> I loged in using one of extension in flex client,then I dail another extension. <br>
> After I dail,call goes to Ekiga softphone at 192.168.8.27 because I have made changes <br>
> in default.xml accordingly to forward calls to this ip.<br>
> Here is My default.xml part :<br>
><br>
> <extension name="Local_Extension"><br>
> <condition field="destination_number" expression="^(\d{4})$"><br>
> <action application="export" data="dialed_extension=$1"/><br>
> <action application="set" data="effective_caller_id_number=${dialed_extension}"/><br>
> <action application="set" data="PEER_IP=192.168.8.27"/> <<Bad: you should never hardcode a specific endpoint IP if this dialplan should be dialing others as well<br>
> <action application="set" data="call_timeout=30"/><br>
> <action application="set" data="hangup_after_bridge=true"/><br>
> <action application="set" data="proxy_media=true"/><br>
> <action application="bridge" data="h323/$1@${PEER_IP}"/> <<Bad: you set your dialplan to bridge the call to your softphone<br>
> </condition> <br>
> </extension><br>
><br>
> Ekiga phone can answer the call and there is audio transmission from both end successfully.<br>
><br>
> But this is one way calling i.e. from flex to ekiga.<br>
> Now I want to call from Ekiga softphone to flex client.<br>
> How to call flex client from ekiga softphone?<br>
><br>
> Anybody please me on this issue.<br>
><br>
> -Navnath.<br>
><br>
><br>
><br>
><br>
><br>
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</div></div>What if we want to even hungup the channels associated with those users? does<br>freeswitch helps ?<br><br>Like, i have a scenario in which , an active call disrupts , once BOTH users<br>gets disconnected from network.<br>Here, both users gets unregistered after option messages but the call<br>sustains. Is there any thing i can handle this scenario?<br><br><br><br><br>--<br>View this message in context: <a href="http://freeswitch-users.2379917.n2.nabble.com/Check-if-UA-is-still-there-tp6451091p7590307.html" target="_blank">http://freeswitch-users.2379917.n2.nabble.com/Check-if-UA-is-still-there-tp6451091p7590307.html</a><br>Sent from the freeswitch-users mailing list archive at Nabble.com.<br><br><br><br>options pings are really just for registered users. a user does not have to be registered to place a call. more useful in your scenario is rtp timers and session timers<br><br>Ken<br>Sent from my iPad<br><br>On May 2, 2013, at 8:49, mehroz
<<a ymailto="mailto:mehroz.ashraf85@gmail.com" href="mailto:mehroz.ashraf85@gmail.com">mehroz.ashraf85@gmail.com</a>> wrote:<br><br>> What if we want to even hungup the channels associated with those users? does<br>> freeswitch helps ?<br>> <br>> Like, i have a scenario in which , an active call disrupts , once BOTH users<br>> gets disconnected from network.<br>> Here, both users gets unregistered after option messages but the call<br>> sustains. Is there any thing i can handle this scenario?<br>> <br>> <br>> <br>> <br>> --<br>> View this message in context: <a href="http://freeswitch-users.2379917.n2.nabble.com/Check-if-UA-is-still-there-tp6451091p7590307.html" target="_blank">http://freeswitch-users.2379917.n2.nabble.com/Check-if-UA-is-still-there-tp6451091p7590307.html</a><br>> Sent from the freeswitch-users mailing list archive at Nabble.com.<br>> <br>>
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href="mailto:FreeSWITCH-users@lists.freeswitch.org">FreeSWITCH-users@lists.freeswitch.org</a><br>> <a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br>> UNSUBSCRIBE:<a href="http://lists.freeswitch.org/mailman/options/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/options/freeswitch-users</a><br>> <a href="http://www.freeswitch.org/" target="_blank">http://www.freeswitch.org</a><br><br><br>i cannot use session-timers, disabled due to an issue with video calling....<br><br>and rtp timers just does not seems to be working!<br><br>i have <br><br>But once network disconnects from both clients (works good, with a single<br>client network outage), FS does not hangs up the call ..... <br><br><br><br>--<br>View this message in context: <a
href="http://freeswitch-users.2379917.n2.nabble.com/Check-if-UA-is-still-there-tp6451091p7590309.html" target="_blank">http://freeswitch-users.2379917.n2.nabble.com/Check-if-UA-is-still-there-tp6451091p7590309.html</a><br>Sent from the freeswitch-users mailing list archive at Nabble.com.<br><br><br><br><div id="yiv6395193712"><div dir="ltr"><br><div class="yiv6395193712gmail_extra"><br><br><div class="yiv6395193712gmail_quote">On Thu, May 2, 2013 at 8:13 AM, mehroz <span dir="ltr"><<a rel="nofollow" ymailto="mailto:mehroz.ashraf85@gmail.com" target="_blank" href="mailto:mehroz.ashraf85@gmail.com">mehroz.ashraf85@gmail.com</a>></span> wrote:<br>
<blockquote class="yiv6395193712gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">i cannot use session-timers, disabled due to an issue with video calling....<br>
<br>
and rtp timers just does not seems to be working!<br></blockquote><div>Can you elaborate on this? Do you have a full pcap and FS debug log of a call with RTP in both directions and then the incoming RTP stream stopping and yet the call not being disconnected?<br>
<br></div><blockquote class="yiv6395193712gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
<br>
i have<br>
<br>
But once network disconnects from both clients (works good, with a single<br>
client network outage), FS does not hangs up the call .....<br></blockquote><div>Correct. No SIP signaling and no session timers means there's nothing left to tell FS that the call should be torn down, except for the RTP timers mentioned above. That's why you'll need to supply a complete debug w/ pcap in order to find out why RTP timers are not working.<br>
<br></div><div>-MC<br></div></div><br clear="all"><br>-- <br>Michael S Collins<br>Twitter: @mercutioviz<br><a rel="nofollow" target="_blank" href="http://www.freeswitch.org/">http://www.FreeSWITCH.org</a><br><a rel="nofollow" target="_blank" href="http://www.cluecon.com/">http://www.ClueCon.com</a><br>
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