<table cellspacing="0" cellpadding="0" border="0" ><tr><td valign="top" style="font: inherit;">Forgive me,<br>I don't know how to turn on the SIP debug mode. This is what it say from FS command line:<br><br>2009-01-13 16:26:46 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing 1000->18187188288in context default<br>2009-01-13 16:26:47 [ERR] mod_sofia.c:2341 sofia_outgoing_channel() Invalid Gateway<br>2009-01-13 16:26:47 [NOTICE] mod_sofia.c:2540 sofia_outgoing_channel() Close Channel N/A [CS_NEW]<br>2009-01-13 16:26:47 [ERR] switch_ivr_originate.c:1110 switch_ivr_originate() Cannot create outgoing channel of type [sofia] cause: [INVALID_NUMBER_FORMAT]<br>2009-01-13 16:26:47 [INFO] mod_dptools.c:1891 audio_bridge_function() Originate Failed. Cause: INVALID_NUMBER_FORMAT<br>2009-01-13 16:26:47 [NOTICE] mod_dptools.c:1918 audio_bridge_function() Hangup sofia/internal/1000@192.168.2.104 [CS_EXECUTE] [INVALID_NUMBER_FORMAT]<br>2009-01-13
16:26:47 [NOTICE] switch_core_session.c:956 switch_core_session_thread() Session 163 (sofia/internal/1000@192.168.2.104) Ended<br>2009-01-13 16:26:47 [NOTICE] switch_core_session.c:958 switch_core_session_thread() Close Channel sofia/internal/1000@192.168.2.104 [CS_HANGUP]<br><br><br>--- On <b>Mon, 1/12/09, Kristian Kielhofner <i><kristian.kielhofner@gmail.com></i></b> wrote:<br><blockquote style="border-left: 2px solid rgb(16, 16, 255); margin-left: 5px; padding-left: 5px;">From: Kristian Kielhofner <kristian.kielhofner@gmail.com><br>Subject: Re: [Freeswitch-users] outbound call, new comer<br>To: freeswitch-users@lists.freeswitch.org<br>Date: Monday, January 12, 2009, 1:07 PM<br><br><pre>In West Philadelphia born and raised...<br><br>Voicepulse seems to be picky about number format. Trying doing full<br>E.164 (+1). Also, make sure your realm is correct. What does a SIP<br>debug look like?<br><br>On 1/12/09, Will Smith
<willbelair@yahoo.com> wrote:<br>><br>><br>> Hi,<br>> I am first time FS user, so it is a bit confused with all the setup. For<br>inbound calls, I tried to add a voicepulse.xml in the sip_profiles/external with<br>the following codes:<br>><br>> <include><br>> <gateway name="voicepulse"><br>> <param name="username" value="3334445555"/><br>> <param name="realm"<br>value="my_sip_provider.com"/><br>> <param name="password" value="3334445555"/><br>> <param name="proxy"<br>value="my_sip_provider.com"/><br>> <param name="expire-seconds" value="600"/><br>> <param name="register" value="true"/><br>> </gateway><br>> <gateway name="voicepulse-backup"><br>> <param name="username" value="3334445555"/><br>> <param name="realm"<br>value="my_sip_provider.com"/><br>> <param name="password"
value="3334445555"/><br>> <param name="extension" value="3334445555"/><br>> <param name="proxy" <br>value="my_sip_provider.com"/><br>> <param name="expire-seconds" value="600"/><br>> <param name="register" value="true"/><br>> </gateway><br>> </include><br>><br>> ------------<br>> and in the conf/dialplan/default.xml file I added:<br>><br>> <!-- Dial any 10 digit number (2223334444) or 1+10 number (12223334444)<br>here --><br>> <extension name="Long Distance - voicepulse"><br>> <condition field="destination_number"<br>expression="^(1{0,1}\d{10})$"><br>> <action application="set"<br>data="effective_caller_id_number=12223334444"/><br>> <!-- If your provider does not provide ringback (180 or 183) you<br>may simulate<br>> ringback by uncommenting the following line. --><br>> <!-- action
application="ringback" /--><br>> <action application="bridge"<br>data="sofia/gateway/voicepulse/$1"/><br>> </condition><br>> </extension><br>><br>><br>> ------------------<br>><br>> For inbound, I added<br>><br>> <extension name="Voicepulse"> <!-- your provider or<br>any name you'd like to call it --><br>> <condition field="destination_number"<br>expression="3334445555"> <!-- your DID for this gateway--><br>> <action application="transfer" data="1001 XML<br>default"/><br>> </condition><br>> </extension><br>><br>> ----------<br>><br>> And, if I dial 3334445555 from a softphone registered with<br>my_sip_provider, I got the message to the voice mail of 1001 - the 1001<br>extension does not ring.<br>> And if from 1001, I dial some real number like 18188892345, I got the<br>error: Invalid Gateway ...
Cannot create outgoing channel of type [fosia] cause:<br>[Invalid_number_format] ...<br>><br>><br>> Would someone please give me some help to set this up. I am a bit confused<br>with these.<br>><br>> Thank you<br>><br>> _______________________________________________<br>> Freeswitch-users mailing list<br>> Freeswitch-users@lists.freeswitch.org<br>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users<br>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users<br>> http://www.freeswitch.org<br>><br>><br><br><br><br>-- <br>Kristian Kielhofner<br>http://blog.krisk.org<br>http://www.submityoursip.com<br>http://www.astlinux.org<br>http://www.star2star.com<br><br>_______________________________________________<br>Freeswitch-users mailing
list<br>Freeswitch-users@lists.freeswitch.org<br>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users<br>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users<br>http://www.freeswitch.org<br></pre></blockquote></td></tr></table><br>