Re: [Freeswitch-users] [Dialplan?] Call either sent to VM or phone doesn't ring at all

mercutioviz wrote: Additionally, turn on debugging on the console and capture that output. If you use fs_cli it has debug output turned on by default. Thanks for the tip. I launched fs_cli, typed sofia profile internal siptrace on, and then made a call from XLite to the GS phone, with the

Re: [Freeswitch-users] [Dialplan?] Call either sent to VM or phone doesn't ring at all

I guess I can limit the amount of debug data in the CLI by choosing the right debug level: http://wiki.freeswitch.org/wiki/Sofia#Debugging_Sofia-SIP What is the recommended way to debug SIP connections like I'm having? -- View this message in context:

Re: [Freeswitch-users] [Dialplan?] Call either sent to VM or phone doesn't ring at all

FWIW, I downloaded and compiled the latest trunk (16041), and am still having this issue. -- View this message in context: http://old.nabble.com/-Dialplan---Call-either-sent-to-VM-or-phone-doesn%27t-ring-at-all-tp26892767p26902800.html Sent from the Freeswitch-users mailing list archive at

Re: [Freeswitch-users] [Dialplan?] Call either sent to VM or phone doesn't ring at all

More information: I can dial the default extensions like just fine. It's only when I call any of the IP phones (1001,1002,1003) that the call is immediately forwarded to the callee's voice-mail when the phone goes off the hook. To only keep the SIP messages in the fs_cli screen, typing

Re: [Freeswitch-users] [Dialplan?] Call either sent to VM or phone doesn't ring at all

On Wed, Dec 23, 2009 at 7:39 AM, Fred-145 codecompl...@free.fr wrote: More information: I can dial the default extensions like just fine. It's only when I call any of the IP phones (1001,1002,1003) that the call is immediately forwarded to the callee's voice-mail when the phone goes off

Re: [Freeswitch-users] [Dialplan?] Call either sent to VM or phone doesn't ring at all

I found the cause for #2: The GS phone was still configured to use NAT, even though both XLite and GS are located in the same, private LAN. Unchecking this on the GS phone solved the issue. But I'm still having issue #1, regardless of which phone is calling or being called: When the phone

Re: [Freeswitch-users] [Dialplan?] Call either sent to VM or phone doesn't ring at all

It is usually CODEC related. probably the SIP messages has the cause inside. __Yehavi: 2009/12/22 Fred-145 codecompl...@free.fr I found the cause for #2: The GS phone was still configured to use NAT, even though both XLite and GS are located in the same, private LAN.

Re: [Freeswitch-users] [Dialplan?] Call either sent to VM or phone doesn't ring at all

On Tue, Dec 22, 2009 at 11:35 AM, Yehavi Bourvine yehavi.bourv...@gmail.com wrote: Try tracing the calls from both sides with TCPDUMP or enable siptrace on FreeSwitch. I guess this will give you some clue. __Yehavi: Additionally, turn on debugging on the console