[Dialplan?] Call either sent to VM or phone doesn't ring at all
Hello I'm running "1.0.trunk (15841)" on Linux CentOS with a the default settings. After succesfully connecting a Windows PC running XLite (EyeBeam, really) and a GrandStream IP phone to Freeswitch, I try to make calls, but am having the following issues: 1. When calling XLite from GS, XLite rings, but when I pick up the call, the caller is sent to voice-mail right away ("the person on extension 1001 is not available") 2. When calling GS from XLite, the GS phone doesn't even ring. FWIW, the phones seem to have registered OK: freeswitch@internal> sofia status profile internal Registrations: ======================================================== Call-ID: Yzc2MzFiMjVhNGQwNjE5YWU1OGZjNGMxMTg0NDIwNDA. User: 1001@192.168.0.7 Contact: "Freeswitch" <sip:1001@192.168.0.1:41380;rinstance=0516dddfe24deef4> Agent: eyeBeam release 1104a stamp 54437 Status: Registered(UDP)(unknown) EXP(2008-01-01 03:34:00) Host: centos.workgroup IP: 192.168.0.1 Port: 41380 Auth-User: 1001 Auth-Realm: 192.168.0.7 MWI-Account: 1001@192.168.0.7 Call-ID: 3f6d4ebebd5e829f@192.168.0.9 User: 1003@192.168.0.7 Contact: "user" <sip:1003@82.237.75.54> Agent: Grandstream BT120 1.1.0.3 Status: Registered(UDP)(unknown) EXP(2008-01-01 03:44:02) Host: centos.workgroup IP: 192.168.0.9 Port: 5060 Auth-User: 1003 Auth-Realm: 192.168.0.7 MWI-Account: 1003@192.168.0.7 ======================================================== Has someone seen this type of behavior? Thanks for any hint. -- View this message in context: http://old.nabble.com/-Dialplan---Call-either-sent-to-VM-or-phone-doesn%27t-... Sent from the Freeswitch-users mailing list archive at Nabble.com.
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-... Sent from the Freeswitch-users mailing list archive at Nabble.com.
It is usually CODEC related. probably the SIP messages has the cause inside. __Yehavi: 2009/12/22 Fred-145 <codecomplete@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-... Sent from the Freeswitch-users mailing list archive at Nabble.com.
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Try tracing the calls from both sides with TCPDUMP or enable siptrace on FreeSwitch. I guess this will give you some clue. __Yehavi: 2009/12/22 Fred-145 <codecomplete@free.fr>
Hello
I'm running "1.0.trunk (15841)" on Linux CentOS with a the default settings. After succesfully connecting a Windows PC running XLite (EyeBeam, really) and a GrandStream IP phone to Freeswitch, I try to make calls, but am having the following issues:
1. When calling XLite from GS, XLite rings, but when I pick up the call, the caller is sent to voice-mail right away ("the person on extension 1001 is not available") 2. When calling GS from XLite, the GS phone doesn't even ring.
FWIW, the phones seem to have registered OK:
freeswitch@internal> sofia status profile internal Registrations: ======================================================== Call-ID: Yzc2MzFiMjVhNGQwNjE5YWU1OGZjNGMxMTg0NDIwNDA. User: 1001@192.168.0.7 Contact: "Freeswitch" <sip:1001@192.168.0.1:41380;rinstance=0516dddfe24deef4> Agent: eyeBeam release 1104a stamp 54437 Status: Registered(UDP)(unknown) EXP(2008-01-01 03:34:00) Host: centos.workgroup IP: 192.168.0.1 Port: 41380 Auth-User: 1001 Auth-Realm: 192.168.0.7 MWI-Account: 1001@192.168.0.7
Call-ID: 3f6d4ebebd5e829f@192.168.0.9 User: 1003@192.168.0.7 Contact: "user" <sip:1003@82.237.75.54 <sip%3A1003@82.237.75.54>> Agent: Grandstream BT120 1.1.0.3 Status: Registered(UDP)(unknown) EXP(2008-01-01 03:44:02) Host: centos.workgroup IP: 192.168.0.9 Port: 5060 Auth-User: 1003 Auth-Realm: 192.168.0.7 MWI-Account: 1003@192.168.0.7 ========================================================
Has someone seen this type of behavior?
Thanks for any hint. -- View this message in context: http://old.nabble.com/-Dialplan---Call-either-sent-to-VM-or-phone-doesn%27t-... Sent from the Freeswitch-users mailing list archive at Nabble.com.
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On Tue, Dec 22, 2009 at 11:35 AM, Yehavi Bourvine <yehavi.bourvine@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
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-... Sent from the Freeswitch-users mailing list archive at Nabble.com.
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-... Sent from the Freeswitch-users mailing list archive at Nabble.com.
BTW, here's the layout: http://img192.imageshack.us/img192/539/investigatingimmediater.jpg All hosts are located in the same LAN with network address 192.168.0.0/24 and are connected to the hub in the ADSL NAT router. Regardless of whether I use XLite, the GrandStream IP phone, or the analog handset connected to the Linksys 3102, I get the same error: The remote extension (target) rings, but when I pick up the call on the (target) phone, the call is terminated on the target end, and the source extension is immediately redirected to the target extension's voice-mail. -- View this message in context: http://old.nabble.com/-Dialplan---Call-either-sent-to-VM-or-phone-doesn%27t-... Sent from the Freeswitch-users mailing list archive at Nabble.com.
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-... Sent from the Freeswitch-users mailing list archive at Nabble.com.
More information: I can dial the default extensions like 9999 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-... Sent from the Freeswitch-users mailing list archive at Nabble.com.
On Wed, Dec 23, 2009 at 7:39 AM, Fred-145 <codecomplete@free.fr> wrote:
More information: I can dial the default extensions like 9999 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
Thanks for the tip. I ended up re-installing CentOS followed by compiling the latest SVN source, and I'm back to normal. The only possible cause I see is that changing the PSU sometimes requires flushing the CMOS, but as to why this would cause Freeswitch to route calls to VM, I have no idea. Sorry for the confusion. -- View this message in context: http://old.nabble.com/-Dialplan---Call-either-sent-to-VM-or-phone-doesn%27t-... Sent from the Freeswitch-users mailing list archive at Nabble.com.
participants (3)
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Fred-145
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Michael Collins
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Yehavi Bourvine