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