Hello All<br>I am trying to configure freeswitch so that it sends outgoing calls to the PSTN through voicepulse<br>My configuration is as follows.<br>I created a file $PREFIX/conf/sip_profiles/external/voicepulse.xml<br><br>
&lt;include&gt;<br>&lt;!-- West Coast gateways --&gt;<br>  &lt;gateway name=&quot;voicepulse&quot;&gt;<br>    &lt;param name=&quot;username&quot; value=&quot;my-user&quot;/&gt;<br>    &lt;param name=&quot;realm&quot; value=&quot;<a href="http://sjc-primary.voicepulse.com">sjc-primary.voicepulse.com</a>&quot;/&gt;<br>
    &lt;param name=&quot;password&quot; value=&quot;my-password&quot;/&gt;<br>    &lt;param name=&quot;proxy&quot; value=&quot;<a href="http://sjc-primary.voicepulse.com">sjc-primary.voicepulse.com</a>&quot;/&gt;<br>    &lt;param name=&quot;expire-seconds&quot; value=&quot;600&quot;/&gt;<br>
    &lt;param name=&quot;register&quot; value=&quot;true&quot;/&gt;<br>  &lt;/gateway&gt;<br>  &lt;gateway name=&quot;voicepulse-backup&quot;&gt;<br>    &lt;param name=&quot;username&quot; value=&quot;my-user&quot;/&gt;<br>
    &lt;param name=&quot;realm&quot; value=&quot;<a href="http://sjc-backup.voicepulse.com">sjc-backup.voicepulse.com</a>&quot;/&gt;<br>    &lt;param name=&quot;password&quot; value=&quot;my-password&quot;/&gt;<br>    &lt;param name=&quot;extension&quot; value=&quot;1000&quot;/&gt;<br>
    &lt;param name=&quot;proxy&quot; value=&quot;<a href="http://sjc-backup.voicepulse.com">sjc-backup.voicepulse.com</a>&quot;/&gt;<br>    &lt;param name=&quot;expire-seconds&quot; value=&quot;600&quot;/&gt;<br>    &lt;param name=&quot;register&quot; value=&quot;true&quot;/&gt;<br>
  &lt;/gateway&gt;<br>&lt;/include&gt;<br><br>I also have a dial plan defined as follows<br><br>&lt;extension name=&quot;Long Distance - voicepulse&quot;&gt;<br>    &lt;condition field=&quot;destination_number&quot; expression=&quot;^(\d{10})$&quot;&gt;<br>
      &lt;action application=&quot;set&quot; data=&quot;effective_caller_id_number=12223334444&quot;/&gt;<br>      &lt;!-- If your provider does not provide ringback (180 or 183) you may simulate<br>        ringback by uncommenting the following line. --&gt;<br>
      &lt;!-- action application=&quot;ringback&quot; /--&gt;<br>      &lt;action application=&quot;bridge&quot; data=&quot;sofia/gateway/voicepulse/$1&quot;/&gt;<br>     &lt;/condition&gt;<br>   &lt;/extension&gt;<br><br>
<br>When I dial an external number using extension 1000 I get the following message on the CLI<br><br>]<br>freeswitch@ubuntu&gt; 2009-11-10 00:35:44.365614 [NOTICE] switch_channel.c:602 New Channel sofia/internal/<a href="mailto:1000@74.207.249.79">1000@74.207.249.79</a> [e4301180-cdba-11de-a864-8927fe94a9f0]<br>
2009-11-10 00:35:44.366623 [INFO] mod_dialplan_xml.c:315 Processing Paul-&gt;5555555555 in context default<br>2009-11-10 00:35:44.368645 [NOTICE] switch_channel.c:602 New Channel sofia/external/5555555555 [e43092f4-cdba-11de-a864-8927fe94a9f0]<br>
2009-11-10 00:35:47.59221 [NOTICE] sofia_glue.c:2698 Pre-Answer sofia/external/5555555555!<br>2009-11-10 00:35:47.59221 [INFO] switch_ivr_originate.c:2017 Sending early media<br>2009-11-10 00:35:47.60524 [INFO] mod_sofia.c:1506 Ring SDP:<br>
v=0<br>o=FreeSWITCH 1257800805 1257800806 IN IP4 74.207.249.79<br>s=FreeSWITCH<br>c=IN IP4 74.207.249.79<br>t=0 0<br>m=audio 30542 RTP/AVP 0 101<br>a=rtpmap:0 pcmu/8000<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-16<br>
a=silenceSupp:off - - - -<br>a=ptime:20<br>a=sendrecv<br><br>2009-11-10 00:35:47.60524 [NOTICE] mod_sofia.c:1509 Pre-Answer sofia/internal/<a href="mailto:1000@74.207.249.79">1000@74.207.249.79</a>!<br>2009-11-10 00:35:51.449542 [NOTICE] sofia.c:3849 Hangup sofia/external/5555555555 [CS_EXCHANGE_MEDIA] [NORMAL_TEMPORARY_FAILURE]<br>
2009-11-10 00:35:51.452539 [NOTICE] switch_ivr_bridge.c:419 Hangup sofia/internal/<a href="mailto:1000@74.207.249.79">1000@74.207.249.79</a> [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE]<br>2009-11-10 00:35:51.454125 [NOTICE] switch_core_session.c:1086 Session 1 (sofia/internal/<a href="mailto:1000@74.207.249.79">1000@74.207.249.79</a>) Ended<br>
2009-11-10 00:35:51.454125 [NOTICE] switch_core_session.c:1088 Close Channel sofia/internal/<a href="mailto:1000@74.207.249.79">1000@74.207.249.79</a> [CS_DESTROY]<br>2009-11-10 00:35:51.454125 [NOTICE] switch_core_session.c:1086 Session 2 (sofia/external/5555555555) Ended<br>
2009-11-10 00:35:51.454125 [NOTICE] switch_core_session.c:1088 Close Channel sofia/external/5555555555 [CS_DESTROY]<br><br><br>I am really new to VOIP and having a hard time with this. I am really not sure how to proceed. Any help would be really appreciated.<br>
<br>Thanks<br>Paul<br>