<div dir="ltr">Hi Abdul,<div><br></div><div>Kindly share the whole FS console logs (enable sip debug inside the logs too) , can you modify the bridge statement as this:</div><div><br></div><div><span style="color:rgb(0,0,0);font-family:Calibri,Arial,Helvetica,sans-serif;font-size:16px"><action application="bridge" data="sofia/<b>external</b>/$1@</span><span style="color:rgb(0,0,0);font-family:Calibri,Arial,Helvetica,sans-serif;font-size:16px">AbdulkamailioSIP.com"/></span><br></div><div><span style="color:rgb(0,0,0);font-family:Calibri,Arial,Helvetica,sans-serif;font-size:16px"><br></span></div><div><span style="color:rgb(0,0,0);font-family:Calibri,Arial,Helvetica,sans-serif;font-size:16px">If you have saved your kamailio as a gateway then you can alternatively dial it as following:</span></div><div><span style="color:rgb(0,0,0);font-family:Calibri,Arial,Helvetica,sans-serif;font-size:16px"><br></span></div><div><span style="color:rgb(0,0,0);font-family:Calibri,Arial,Helvetica,sans-serif;font-size:16px"><action application="bridge" data="sofia/<b>gateway</b>/</span><b>GOOD_GATEWAY</b><span style="color:rgb(0,0,0);font-family:Calibri,Arial,Helvetica,sans-serif;font-size:16px">/$1</span><span style="color:rgb(0,0,0);font-family:Calibri,Arial,Helvetica,sans-serif;font-size:16px">"/></span><span style="color:rgb(0,0,0);font-family:Calibri,Arial,Helvetica,sans-serif;font-size:16px"><br></span></div><div><span style="color:rgb(0,0,0);font-family:Calibri,Arial,Helvetica,sans-serif;font-size:16px"><br></span></div>Where <b>GOOD_GATEWAY</b> is the gateway name from an xml file. Here is how.<div><font face="monospace, monospace"><br></font></div><div><font face="monospace, monospace">FreeSWITCH:~# cd /usr/local/freeswitch/conf/sip_profiles/external/<br><br>FreeSWITCH-A:~# vim kamailio.xml<br><br>Insert these Lines in this file:<br><br><include><br> <gateway name="<b>GOOD_GATEWAY</b>"><br> <param name="username" value="nothing"/><br> <param name="password" value="doesn't_matter"/><br> <param name="proxy" value="192.168.30.3"/> <!--SET IP OF KAMAILIO HERE --><br> <param name="register" value="false"/><br> <param name="retry-seconds" value="10"/><br> <param name="caller-id-in-from" value="true"/><br> <param name="extension-in-contact" value="true"/><br> <param name="ping" value="25"/><br> <param name="inbound-late-negotiation" value="true"/><br> <param name="context" value="default"/><br> </gateway><br></include></font><div><span style="color:rgb(0,0,0);font-family:Calibri,Arial,Helvetica,sans-serif;font-size:16px"><br></span></div>Also, if you don't use gateway approach can you make sure that from your FS the domain name 'AbdulKamailioSIP.com' resolves to IP of Kamailio Server. <br><br>I've a feeling that this email should be in Freeswitch mailing list, not in Kamailio's/</div><div><br></div><div>Regards,</div><div>Sammy</div><div><br></div><div><div><span style="color:rgb(0,0,0);font-family:Calibri,Arial,Helvetica,sans-serif;font-size:16px"><br></span></div></div></div><div class="gmail_extra"><br><div class="gmail_quote">On Wed, Feb 10, 2016 at 5:00 PM, malik sherif <span dir="ltr"><<a href="mailto:asherif74@hotmail.com" target="_blank">asherif74@hotmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div dir="ltr">
<div style="font-size:12pt;color:#000000;background-color:#ffffff;font-family:Calibri,Arial,Helvetica,sans-serif">
<p>Hello,</p>
<p>I am using Kamailio and freeswitch to setup SBC but the I attempted to make a call it just goes to the voice mail.
<br>
</p>
<p>Here is what freeswitch is displaying. <br>
</p>
<p>Thanks for your help in advance <br>
</p>
<p>Abdul<br>
</p>
<p><br>
</p>
<p><br>
</p>
<p>freeswitch@linux-ix64> 2016-02-10 10:54:16.663387 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/102@AbdulKamailioSIP.com [12f87c10-f3be-43ee-b038-f6647e5af373]<br>
2016-02-10 10:54:16.683337 [INFO] mod_dialplan_xml.c:635 Processing 102 <102>->kb-102 in context public<br>
2016-02-10 10:54:16.683337 [NOTICE] switch_ivr.c:1861 Transfer sofia/internal/102@AbdulKamailioSIP.com to XML[kb-102@default]<br>
2016-02-10 10:54:16.683337 [INFO] mod_dialplan_xml.c:635 Processing 102 <102>->kb-102 in context default<br>
2016-02-10 10:54:16.683337 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/102@AbdulkamailioSIP.com [0c6c8dda-34fc-45a0-a6a2-8e82ff3a9be3]<br>
2016-02-10 10:54:18.183346 [NOTICE] sofia.c:7539 Hangup sofia/internal/102@AbdulkamailioSIP.com [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE]<br>
2016-02-10 10:54:18.183346 [NOTICE] switch_core_session.c:1641 Session 2 (sofia/internal/102@AbdulkamailioSIP.com) Ended<br>
2016-02-10 10:54:18.183346 [NOTICE] switch_core_session.c:1645 Close Channel sofia/internal/102@AbdulkamailioSIP.com [CS_DESTROY]<br>
2016-02-10 10:54:18.183346 [INFO] mod_dptools.c:3244 Originate Failed. Cause: NORMAL_TEMPORARY_FAILURE<br>
2016-02-10 10:54:18.183346 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/102@AbdulKamailioSIP.com!<br>
2016-02-10 10:54:18.183346 [NOTICE] mod_dptools.c:1268 Channel [sofia/internal/102@AbdulKamailioSIP.com] has been answered<br>
2016-02-10 10:54:32.043345 [NOTICE] sofia.c:952 Hangup sofia/internal/102@AbdulKamailioSIP.com [CS_EXECUTE] [NORMAL_CLEARING]<br>
2016-02-10 10:54:32.063338 [NOTICE] switch_core_session.c:1641 Session 1 (sofia/internal/102@AbdulKamailioSIP.com) Ended<br>
2016-02-10 10:54:32.063338 [NOTICE] switch_core_session.c:1645 Close Channel sofia/internal/102@AbdulKamailioSIP.com [CS_DESTROY]<br>
<br>
</p>
<p><br>
</p>
<p>Any idea as to how to implement this command on freeswitch dial plan, I am not sure what to use for gw1</p>
<p></p>
<pre><action application="bridge" data="{sip_invite_domain=${sip_from_host}}sofia/gateway/gw1/$<a href="mailto:1@domain.org" target="_blank">1@domain.org</a>"/></pre>
<br>
<p></p>
<p><br>
</p>
<p><br>
</p>
<p><br>
</p>
<p>From Freeswitch dial plan<br>
</p>
<p><br>
</p>
<p><extension name="kbridge"><br>
<condition field="destination_number" expression="^kb-(.+)$"><br>
<action application="set" data="proxy_media=true"/><br>
<action application="set" data="call_timeout=50"/><br>
<action application="set" data="continue_on_fail=true"/><br>
<action application="set" data="hangup_after_bridge=true"/><br>
<action application="set" data="sip_invite_domain=AbdulkamailioSIP.com"/><br>
<action application="export" data="sip_contact_user=ufs"/><br>
<action application="bridge" data="sofia/$${domain}/$1@AbdulkamailioSIP.com"/><br>
<action application="answer"/><br>
<action application="voicemail" data="default ${domain_name} $1"/><br>
</condition><br>
</extension><br>
<br>
</p>
<p><br>
</p>
<p><br>
</p>
<p><br>
</p>
<br>
<br>
<div style="color:rgb(0,0,0)">
<hr style="display:inline-block;width:98%">
<div dir="ltr"><font style="font-size:11pt" color="#000000" face="Calibri, sans-serif"><span class=""><b>From:</b> sr-users <<a href="mailto:sr-users-bounces@lists.sip-router.org" target="_blank">sr-users-bounces@lists.sip-router.org</a>> on behalf of SamyGo <<a href="mailto:govoiper@gmail.com" target="_blank">govoiper@gmail.com</a>><br>
</span><b>Sent:</b> Friday, January 29, 2016 5:02 PM<div><div class="h5"><br>
<b>To:</b> Kamailio (SER) - Users Mailing List<br>
<b>Subject:</b> Re: [SR-Users] Fw: Kamailio and freeswitch integration for SBC</div></div></font>
<div> </div>
</div><div><div class="h5">
<div>
<div dir="ltr">
<div><span style="white-space:pre-wrap">Sorry for last email:
<div style="white-space:normal"><font face="monospace, monospace"><span style="white-space:pre-wrap"></span>if (!lookup("location")) {</font></div>
<div style="white-space:normal"><font face="monospace, monospace"><span style="white-space:pre-wrap"></span>$var(rc) = $rc;</font></div>
<div style="white-space:normal"><font face="monospace, monospace"><span style="white-space:pre-wrap"></span>route(TOVOICEMAIL);</font></div>
<div style="white-space:normal"><font face="monospace, monospace"><span style="white-space:pre-wrap"></span>t_newtran();</font></div>
<div style="white-space:normal"><font face="monospace, monospace"><span style="white-space:pre-wrap"></span>switch ($var(rc)) {</font></div>
<div style="white-space:normal"><font face="monospace, monospace"><span style="white-space:pre-wrap"></span>case -1:</font></div>
<div style="white-space:normal"><font face="monospace, monospace"><span style="white-space:pre-wrap"></span>case -3:</font></div>
<div style="white-space:normal"><font face="monospace, monospace"><span style="white-space:pre-wrap"></span>send_reply("404", "Not Found");</font></div>
<div style="white-space:normal"><font face="monospace, monospace"><span style="white-space:pre-wrap"></span>exit;</font></div>
<div style="white-space:normal"><font face="monospace, monospace"><span style="white-space:pre-wrap"></span>case -2:</font></div>
<div style="white-space:normal"><font face="monospace, monospace"><span style="white-space:pre-wrap"></span>send_reply("405", "Method Not Allowed");</font></div>
<div style="white-space:normal"><font face="monospace, monospace"><span style="white-space:pre-wrap"></span>exit;</font></div>
<div style="white-space:normal"><font face="monospace, monospace"><span style="white-space:pre-wrap"></span>}</font></div>
<div style="white-space:normal"><font face="monospace, monospace"><span style="white-space:pre-wrap"></span>}</font></div>
That is where you get 404 Not Found. What I see is that you're registering users with domain as AbdulKamailioSIP.com but when your FreeSwitch sends call to Kamailio the RURI becomes:
<b>INVITE <a href="mailto:sip%3A7632689993@10.22.52.2" target="_blank">sip:7632689993@10.22.52.2</a> SIP/2.0</b> Which is definitely not matching any User like: INVITE sip:7632689993@<b>AbdulKamailioSIP.com</b> SIP/2.0 So, you need to go in your FS dialplan and make sure you
set the proper Domains before sending call out, there are couple of ways to do this.
<b>1 - </b>Using FreeSWITCH to set FROM domain: <a href="https://wiki.freeswitch.org/wiki/Variable_sip_invite_domain" target="_blank">
https://wiki.freeswitch.org/wiki/Variable_sip_invite_domain</a> <b>2 - </b>Use custom SIP header from FS to contain a domain name, and in Kamailio set headers as you require; something like this: Attach a SIP Header in FS dialplan before sending call out to
Kamailio, say <font face="monospace, monospace">X-USER-DOMAIN: AbdulKamailioSIP.com
</font>Next when I receive call in Kamailio.cfg I detect this header <font face="monospace, monospace">
if(is_present_hf("X-USER-DOMAIN")) { $ru = "sip:" + $rU + "@" + $hdr(X-USER-DOMAIN); $td = $hdr(X-USER-DOMAIN); }
</font>In option 2 you must do it before executing record_route() functions, so possibly need to do this inside your FSINBOUND route. I prefer option 1. PS: Wireshark highlights any custom SIP headers in sky blue, that doesn't mean there is any error in there.</span></div>
<div><span style="white-space:pre-wrap"><br>
</span></div>
<div><span style="white-space:pre-wrap">Regards,</span></div>
<div><span style="white-space:pre-wrap">Sammy</span></div>
<div><span style="white-space:pre-wrap"><br>
</span></div>
</div>
<div class="gmail_extra"><br>
<div class="gmail_quote">On Fri, Jan 29, 2016 at 11:47 AM, SamyGo <span dir="ltr">
<<a href="mailto:govoiper@gmail.com" target="_blank">govoiper@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div dir="ltr">Hi Abdul,
<div><br>
This is where you are getting your 404 NOT Found from Kamailio:</div>
<div><br>
</div>
<div><br>
</div>
</div>
<div>
<div>
<div class="gmail_extra"><br>
<div class="gmail_quote">On Thu, Jan 28, 2016 at 4:30 PM, malik sherif <span dir="ltr">
<<a href="mailto:asherif74@hotmail.com" target="_blank">asherif74@hotmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div dir="ltr">
<div style="font-size:12pt;color:#000000;background-color:#ffffff;font-family:Calibri,Arial,Helvetica,sans-serif">
<p>I will also run the commands that suggested.<br>
</p>
<br>
<br>
<div style="color:rgb(0,0,0)">
<hr style="display:inline-block;width:98%">
<div dir="ltr"><font style="font-size:11pt" color="#000000" face="Calibri, sans-serif"><span><b>From:</b> sr-users <<a href="mailto:sr-users-bounces@lists.sip-router.org" target="_blank">sr-users-bounces@lists.sip-router.org</a>> on behalf of SamyGo <<a href="mailto:govoiper@gmail.com" target="_blank">govoiper@gmail.com</a>><br>
</span><b>Sent:</b> Thursday, January 28, 2016 6:08 PM<span><br>
<b>To:</b> Kamailio (SER) - Users Mailing List<br>
</span><b>Subject:</b> Re: [SR-Users] Fw: Kamailio and freeswitch integration for SBC</font>
<div> </div>
</div>
<div>
<div>
<div>
<div dir="ltr">I believe Daniel is busy with FOSDEM , <br>
<br>
<br>
Abdul can you confirm that you're still getting this output in FS console: <br>
<font face="times new roman, serif"><br>
2016-01-13 05:37:29.572184 [INFO] mod_dialplan_xml.c:635 Processing 7632689991 <7632689991>->kb-7632689993 in context default<br>
2016-01-13 05:37:29.572184 [CRIT] mod_dptools.c:1638 WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING
<br>
2016-01-13 05:37:29.572184 [CRIT] mod_dptools.c:1638 Open /usr/local/freeswitch/conf/vars.xml and change the default_password.<br>
2016-01-13 05:37:29.572184 [CRIT] mod_dptools.c:1638 Once changed type 'reloadxml' at the console.<br>
2016-01-13 05:37:29.572184 [CRIT] mod_dptools.c:1638 WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING
<br>
<font color="#ff0000">2016-01-13 05:37:39.632245 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/<a href="mailto:7632689993@10.22.52.2" target="_blank">7632689993@10.22.52.2</a> [d52b6ef9-c4f6-4edf-aff9-8a8da3761788]</font><br>
<font color="#ff0000">2016-01-13 05:37:39.632245 [NOTICE] sofia.c:7539 Hangup sofia/internal/<a href="mailto:7632689993@10.22.52.2" target="_blank">7632689993@10.22.52.2</a> [CS_ROUTING] [UNALLOCATED_NUMBER]</font><br>
<br>
</font>Please paste your complete dialplan here as well, though this clearly states that the number it tried to dial is not registered or unable to dial to.<font face="times new roman, serif"><font color="#000000"><br>
</font></font>please paste out the content of the following command just before dialing:<font face="times new roman, serif"><b style="color:rgb(255,0,0)"><br>
fs_cli> show registrations<br>
</b><br>
</font>Also, it will help you find out useful info about why it shows you UNALLOCATED NUMBER if you enable the sofia sip debug by using the following command.<font face="times new roman, serif"><br>
<b style="color:rgb(255,0,0)">fs_cli> sofia global siptrace on<br>
</b><br>
</font>Once you execute the above command make a call to destination and see what FreeeSWITCH is trying to do.
<br>
<br>
Thanks,<br>
Sammy.</div>
<div class="gmail_extra"><br>
<div class="gmail_quote">On Thu, Jan 28, 2016 at 11:23 AM, malik sherif <span dir="ltr">
<<a href="mailto:asherif74@hotmail.com" target="_blank">asherif74@hotmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div dir="ltr">
<div style="font-size:12pt;color:#000000;background-color:#ffffff;font-family:Calibri,Arial,Helvetica,sans-serif">
<p><br>
</p>
<p>Any hint?<br>
</p>
<br>
<div style="color:rgb(0,0,0)">
<hr style="display:inline-block;width:98%">
<div dir="ltr"><font style="font-size:11pt" color="#000000" face="Calibri, sans-serif"><span><b>From:</b> sr-users <<a href="mailto:sr-users-bounces@lists.sip-router.org" target="_blank">sr-users-bounces@lists.sip-router.org</a>> on behalf of malik sherif <<a href="mailto:asherif74@hotmail.com" target="_blank">asherif74@hotmail.com</a>><br>
</span><b>Sent:</b> Tuesday, January 26, 2016 11:35 PM<br>
<b>To:</b> Kamailio (SER) - Users Mailing List; <a href="mailto:miconda@gmail.com" target="_blank">
miconda@gmail.com</a>
<div>
<div><br>
<b>Subject:</b> Re: [SR-Users] Kamailio and freeswitch integration for SBC</div>
</div>
</font>
<div> </div>
</div>
<div>
<div>
<div>
<div style="font-size:12pt;color:#000000;background-color:#ffffff;font-family:Calibri,Arial,Helvetica,sans-serif">
<p>Thanks again and here is the pcap file.</p>
<p>Thanks</p>
<p>Abdul<br>
</p>
<br>
<br>
<div style="color:rgb(0,0,0)">
<hr style="display:inline-block;width:98%">
<div dir="ltr"><font style="font-size:11pt" color="#000000" face="Calibri, sans-serif"><b>From:</b> Daniel-Constantin Mierla <<a href="mailto:miconda@gmail.com" target="_blank">miconda@gmail.com</a>><br>
<b>Sent:</b> Friday, January 22, 2016 8:46 AM<br>
<b>To:</b> malik sherif; Kamailio (SER) - Users Mailing List<br>
<b>Subject:</b> Re: [SR-Users] Kamailio and freeswitch integration for SBC</font>
<div> </div>
</div>
<div>Can you attach the pcap file - copy&paste inline makes it imposible to read and digest it with a traffic analyzer (e.g., wireshark).<br>
<br>
Cheers,<br>
Daniel<br>
<br>
<div>On 21/01/16 18:31, malik sherif wrote:<br>
</div>
<blockquote type="cite">
<div style="font-size:12pt;color:#000000;background-color:#ffffff;font-family:Calibri,Arial,Helvetica,sans-serif">
<p><br>
</p>
<br>
<br>
<div style="color:rgb(0,0,0)">
<hr style="display:inline-block;width:98%">
<div dir="ltr"><font style="font-size:11pt" color="#000000" face="Calibri, sans-serif"><b>From:</b> sr-users
<a href="mailto:sr-users-bounces@lists.sip-router.org" target="_blank"><sr-users-bounces@lists.sip-router.org></a> on behalf of malik sherif
<a href="mailto:asherif74@hotmail.com" target="_blank"><asherif74@hotmail.com></a><br>
<b>Sent:</b> Wednesday, January 20, 2016 9:55 PM<br>
<b>To:</b> Kamailio (SER) - Users Mailing List<br>
<b>Subject:</b> Re: [SR-Users] Kamailio and freeswitch integration for SBC</font>
<div> </div>
</div>
<div>
<div style="font-size:12pt;color:#000000;background-color:#ffffff;font-family:Calibri,Arial,Helvetica,sans-serif">
<p>Copy and paste part of tcmdump and highlighted the 404. 10.22.52.2 is the server IP address</p>
<p>Thanks again</p>
<p>Abdul<br>
</p>
</div>
</div>
</div>
</div>
</blockquote>
<br>
<div style="font-size:12pt;color:#000000;background-color:#ffffff;font-family:Calibri,Arial,Helvetica,sans-serif">
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<div style="font-size:12pt;color:#000000;background-color:#ffffff;font-family:Calibri,Arial,Helvetica,sans-serif">
<div style="color:rgb(0,0,0)">
<div>
<blockquote type="cite">
<div style="font-size:12pt;color:#000000;background-color:#ffffff;font-family:Calibri,Arial,Helvetica,sans-serif">
<div style="color:rgb(0,0,0)">
<div>
<div style="font-size:12pt;color:#000000;background-color:#ffffff;font-family:Calibri,Arial,Helvetica,sans-serif">
<div style="color:rgb(0,0,0)">
<div><a href="http://kb.asipto.com/freeswitch:kamailio-3.3.x-freeswitch-1.2.x-sbc" style="display:table-cell;text-align:center" target="_blank"><img style="display:inline-block;max-width:250px;max-height:250px;min-height:49px;width:180px;border-width:0px;vertical-align:bottom" height="49" width="180" src="http://kb.asipto.com/_media/wiki:logo.png"></a>
</div>
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</blockquote>
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<br>
<pre cols="72">--
Daniel-Constantin Mierla
<a href="http://twitter.com/#!/miconda" target="_blank">http://twitter.com/#!/miconda</a> - <a href="http://www.linkedin.com/in/miconda" target="_blank">http://www.linkedin.com/in/miconda</a>
Book: SIP Routing With Kamailio - <a href="http://www.asipto.com" target="_blank">http://www.asipto.com</a>
<a href="http://miconda.eu" target="_blank">http://miconda.eu</a></pre>
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</div>
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</div>
<br>
_______________________________________________<br>
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list<br>
<a href="mailto:sr-users@lists.sip-router.org" target="_blank">sr-users@lists.sip-router.org</a><br>
<a href="http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users" rel="noreferrer" target="_blank">http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users</a><br>
<br>
</blockquote>
</div>
<br>
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</div>
</div>
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</blockquote>
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<br>_______________________________________________<br>
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list<br>
<a href="mailto:sr-users@lists.sip-router.org">sr-users@lists.sip-router.org</a><br>
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