Level 3 uses port 5070 for their sip server but requires port 5060 to be used on our side; I made changes to the vars.xml config file to support this. I was just able to get the server to answer a call by changing the register value to false and adding an entry in the ACL config for the Level 3 server however now it seems that the audio isn't working correctly. I'm using a simple dial plan to echo the audio back but all I get is dead air. <extension name="level3"> <condition field="destination_number" expression="^13127567000$"> <action application="answer"/> <action application="echo" /> </condition> </extension> On Sep 11, 2011, at 7:47 PM, Nandy Dagondon wrote: in addition to peter's advise, take a look at the SIP port 5070. FS is using port 5080 for the external SIP profile. modify the port number at "external.xml" then delete the port numbers in your proxy settings. On Sun, Sep 11, 2011 at 6:36 PM, Peter Olsson <peter.olsson@visionutveckling.se<mailto:peter.olsson@visionutveckling.se>> wrote: You're not giving us much information here. Please post exactly what doesn't work, and also pastebin the actual logs from FreeSWITCH. /Peter ________________________________________ Från: freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org> [freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org>] för Chad Vogel [cvogel@lyonl.com<mailto:cvogel@lyonl.com>] Skickat: den 11 september 2011 03:28 Till: freeswitch-users@lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org> Ämne: [Freeswitch-users] external sip profile hello, I'm trying to switch from asterisk to freeswitch; however i'm wondering how can I create a sip profile because the sip profile i created doesn't seem to function with level 3. here is the sip profile i created that isn't working: <include> <gateway name="Level3"> <param name="username" value="USERNAME"/> <param name="password" value="PASSWORD"/> <param name="proxy" value="4.55.35.60:5070<http://4.55.35.60:5070/>"/> <param name="expire-seconds" value="3600"/> <param name="register-transport" value="udp"/> <param name="register" value="true"/> </gateway> </include> here is my asterisk profile (it works): [level3_out] type=peer nat=no host=4.55.35.60 username=***Username*** secret=***Password*** dtmfmode=rfc2833 port=5070 [level3_in] nat=no insecure=very dtmfmode=rfc2833 disallow=all context=from-trunk canreinvite=no allow=ulaw&alaw host=4.55.35.60 type=peer port=5070 How can I create a sip profile that will function the same in freeswitch? !DSPAM:4e6c82af32761635315745! FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org<http://www.freeswitch.org/> FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org