In West Philadelphia born and raised... Voicepulse seems to be picky about number format. Trying doing full E.164 (+1). Also, make sure your realm is correct. What does a SIP debug look like?
On 1/12/09, Will Smith <[email protected]> wrote: > > > Hi, > I am first time FS user, so it is a bit confused with all the setup. For > inbound calls, I tried to add a voicepulse.xml in the sip_profiles/external > with the following codes: > > <include> > <gateway name="voicepulse"> > <param name="username" value="3334445555"/> > <param name="realm" value="my_sip_provider.com"/> > <param name="password" value="3334445555"/> > <param name="proxy" value="my_sip_provider.com"/> > <param name="expire-seconds" value="600"/> > <param name="register" value="true"/> > </gateway> > <gateway name="voicepulse-backup"> > <param name="username" value="3334445555"/> > <param name="realm" value="my_sip_provider.com"/> > <param name="password" value="3334445555"/> > <param name="extension" value="3334445555"/> > <param name="proxy" value="my_sip_provider.com"/> > <param name="expire-seconds" value="600"/> > <param name="register" value="true"/> > </gateway> > </include> > > ------------ > and in the conf/dialplan/default.xml file I added: > > <!-- Dial any 10 digit number (2223334444) or 1+10 number (12223334444) here > --> > <extension name="Long Distance - voicepulse"> > <condition field="destination_number" expression="^(1{0,1}\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> > > > ------------------ > > For inbound, I added > > <extension name="Voicepulse"> <!-- your provider or any name you'd like to > call it --> > <condition field="destination_number" expression="3334445555"> <!-- your > DID for this gateway--> > <action application="transfer" data="1001 XML default"/> > </condition> > </extension> > > ---------- > > And, if I dial 3334445555 from a softphone registered with my_sip_provider, I > got the message to the voice mail of 1001 - the 1001 extension does not ring. > And if from 1001, I dial some real number like 18188892345, I got the error: > Invalid Gateway ... Cannot create outgoing channel of type [fosia] cause: > [Invalid_number_format] ... > > > Would someone please give me some help to set this up. I am a bit confused > with these. > > Thank you > > _______________________________________________ > Freeswitch-users mailing list > [email protected] > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com _______________________________________________ Freeswitch-users mailing list [email protected] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
