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 same issue. I wish I could go through the debug messages in the CLI, but there's so much data that I can't even see the beginning. Is there a way to reduce the amount of information, eg. only displaying the SIP messages, or only displaying the lines that start with [debug] while ignoring those that start with [notice]? Thank you. -- View this message in context: http://old.nabble.com/-Dialplan---Call-either-sent-to-VM-or-phone-doesn%27t-ring-at-all-tp26892767p26897658.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ FreeSWITCH-users mailing list 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
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: http://old.nabble.com/-Dialplan---Call-either-sent-to-VM-or-phone-doesn%27t-ring-at-all-tp26892767p26897904.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ FreeSWITCH-users mailing list 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
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 Nabble.com. ___ FreeSWITCH-users mailing list 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
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 sofia loglevel all 0 followed by sofia profile internal siptrace on doesn't do the trick, so am unable to post the whole log yet. -- View this message in context: http://old.nabble.com/-Dialplan---Call-either-sent-to-VM-or-phone-doesn%27t-ring-at-all-tp26892767p26903707.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ FreeSWITCH-users mailing list 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
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 the hook. To only keep the SIP messages in the fs_cli screen, typing sofia loglevel all 0 followed by sofia profile internal siptrace on doesn't do the trick, so am unable to post the whole log yet. If you're having this much trouble with the CLI then you might be better off just using a combination of tcpdump and rotating log files. Use this command from the shell to rotate logs: fs_cli -x fsctl send_sighup Use the info on this page to collect a pcap: http://wiki.freeswitch.org/wiki/Packet_Capture If you have Wireshark you can open the pcap and do some fun analysis. You can also follow the tcp stream and watch the messages going back and forth. Hopefully you'll see what's happening (or not happening) and then we can take it from there. -MC ___ FreeSWITCH-users mailing list 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
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 answers the call, I'm sent automatically to voice-mail. Could it be codec-related, or something like that? Thank you. -- View this message in context: http://old.nabble.com/-Dialplan---Call-either-sent-to-VM-or-phone-doesn%27t-ring-at-all-tp26892767p26893059.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ FreeSWITCH-users mailing list 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
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. 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 answers the call, I'm sent automatically to voice-mail. Could it be codec-related, or something like that? Thank you. -- View this message in context: http://old.nabble.com/-Dialplan---Call-either-sent-to-VM-or-phone-doesn%27t-ring-at-all-tp26892767p26893059.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ FreeSWITCH-users mailing list 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 ___ FreeSWITCH-users mailing list 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
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 and capture that output. If you use fs_cli it has debug output turned on by default. Pastebin that output and post the link in this thread. If you happen to look at the traces and figure it out then please let us know. :) -MC ___ FreeSWITCH-users mailing list 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
