Call sent to group call terminates when group is busy instead of transferring
I set up a group call for our support team in which all their phones ring when someone needs to speak with them. If they are busy the call should be transferred to a general extension which if not answered then goes to that extensions VM. My dialplan looks like: <extension name="group_dial_support"> <condition field="destination_number" expression="^2001$"> <action application="set" data="continue_on_fail=true"/> <action application="bridge" data="group/support@${domain_name}"/> <!-- <action application="voicemail" data="default ${domain} 1000"/> --> <action application="transfer" data="1000 XML default"/> <action application="hangup"/> </condition> </extension> What is happening is a caller selects the support option from the IVR, ever phone in the support group rings, which is what should happen. If no one picks up the call Freeswitch hangs up instead of transferring the call to extension 1000. You can see that I also tried to send the call directly to voicemail but that didn’t work either. The message I see when Freeswitch hangs up is: Channel sofia/internal/sip:1002@72.38.184.18:39042 hanging up, cause: USER_BUSY The full output from cli can be seen here: http://pastebin.freeswitch.org/18696 I would like to get the call to transfer properly. Thanks -Robert
Try using loopback when you send the call to voicemail, also see the local extensions dialplan located in conf/dialplan/default.xml On Mar 19, 2012 4:43 PM, "Robert Longfield" <robert.longfield@klinsight.com> wrote:
I set up a group call for our support team in which all their phones ring when someone needs to speak with them. If they are busy the call should be transferred to a general extension which if not answered then goes to that extensions VM.
My dialplan looks like:
<extension name="group_dial_support"> <condition field="destination_number" expression="^2001$"> <action application="set" data="continue_on_fail=true"/> <action application="bridge" data="group/support@${domain_name}"/> <!-- <action application="voicemail" data="default ${domain} 1000"/> --> <action application="transfer" data="1000 XML default"/> <action application="hangup"/> </condition> </extension>
What is happening is a caller selects the support option from the IVR, ever phone in the support group rings, which is what should happen. If no one picks up the call Freeswitch hangs up instead of transferring the call to extension 1000. You can see that I also tried to send the call directly to voicemail but that didn’t work either.
The message I see when Freeswitch hangs up is:
Channel sofia/internal/sip:1002@72.38.184.18:39042 hanging up, cause: USER_BUSY
The full output from cli can be seen here: http://pastebin.freeswitch.org/18696
I would like to get the call to transfer properly.
Thanks -Robert
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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
Thanks for the tip Brian, I tried using a loopback using the example in /dialplan/default.xml and I am still experiencing the same problem. I’ve tried a loopback that looks like: <action application="bridge" data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/> <action application="bridge" data="loopback/app=voicemail:default $${domain_name} ${dialed_extension}"/> <action application="bridge" data="loopback/app=voicemail:1000@${domain_name} $1"/> <action application="bridge" data="loopback/app=voicemail:1000@${domain_name}"/> <action application="bridge" data="loopback/app=voicemail:1000@$${domain_name}"/> Only instead of dropping the call it seems to sleep... From: Brian Foster Sent: Monday, March 19, 2012 7:12 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Call sent to group call terminates when group is busy instead of transferring Try using loopback when you send the call to voicemail, also see the local extensions dialplan located in conf/dialplan/default.xml On Mar 19, 2012 4:43 PM, "Robert Longfield" <robert.longfield@klinsight.com> wrote: I set up a group call for our support team in which all their phones ring when someone needs to speak with them. If they are busy the call should be transferred to a general extension which if not answered then goes to that extensions VM. My dialplan looks like: <extension name="group_dial_support"> <condition field="destination_number" expression="^2001$"> <action application="set" data="continue_on_fail=true"/> <action application="bridge" data="group/support@${domain_name}"/> <!-- <action application="voicemail" data="default ${domain} 1000"/> --> <action application="transfer" data="1000 XML default"/> <action application="hangup"/> </condition> </extension> What is happening is a caller selects the support option from the IVR, ever phone in the support group rings, which is what should happen. If no one picks up the call Freeswitch hangs up instead of transferring the call to extension 1000. You can see that I also tried to send the call directly to voicemail but that didn’t work either. The message I see when Freeswitch hangs up is: Channel sofia/internal/sip:1002@72.38.184.18:39042 hanging up, cause: USER_BUSY The full output from cli can be seen here: http://pastebin.freeswitch.org/18696 I would like to get the call to transfer properly. Thanks -Robert _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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 -------------------------------------------------------------------------------- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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
Malfunction! Need input! Get debug log of call from start to finish and put on pastebin. -MC On Tue, Mar 20, 2012 at 8:19 AM, Robert Longfield < robert.longfield@klinsight.com> wrote:
Thanks for the tip Brian,
I tried using a loopback using the example in /dialplan/default.xml and I am still experiencing the same problem.
I’ve tried a loopback that looks like: <action application="bridge" data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/> <action application="bridge" data="loopback/app=voicemail:default $${domain_name} ${dialed_extension}"/> <action application="bridge" data="loopback/app=voicemail:1000@${domain_name} $1"/> <action application="bridge" data="loopback/app=voicemail:1000@ ${domain_name}"/> <action application="bridge" data="loopback/app=voicemail:1000@ $${domain_name}"/>
Only instead of dropping the call it seems to sleep...
*From:* Brian Foster <bdfoster@endigotech.com> *Sent:* Monday, March 19, 2012 7:12 PM *To:* FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> *Subject:* Re: [Freeswitch-users] Call sent to group call terminates when group is busy instead of transferring
Try using loopback when you send the call to voicemail, also see the local extensions dialplan located in conf/dialplan/default.xml On Mar 19, 2012 4:43 PM, "Robert Longfield" < robert.longfield@klinsight.com> wrote:
I set up a group call for our support team in which all their phones ring when someone needs to speak with them. If they are busy the call should be transferred to a general extension which if not answered then goes to that extensions VM.
My dialplan looks like:
<extension name="group_dial_support"> <condition field="destination_number" expression="^2001$"> <action application="set" data="continue_on_fail=true"/> <action application="bridge" data="group/support@ ${domain_name}"/> <!-- <action application="voicemail" data="default ${domain} 1000"/> --> <action application="transfer" data="1000 XML default"/> <action application="hangup"/> </condition> </extension>
What is happening is a caller selects the support option from the IVR, ever phone in the support group rings, which is what should happen. If no one picks up the call Freeswitch hangs up instead of transferring the call to extension 1000. You can see that I also tried to send the call directly to voicemail but that didn’t work either.
The message I see when Freeswitch hangs up is:
Channel sofia/internal/sip:1002@72.38.184.18:39042 hanging up, cause: USER_BUSY
The full output from cli can be seen here: http://pastebin.freeswitch.org/18696
I would like to get the call to transfer properly.
Thanks -Robert
ugh, I can’t believe I forgot to include the pastebin http://pastebin.com/GYLvtDB3 The termination happens also when there is a single user in the call group. The call is transferred to the extension, that extension does not pickup and FS drops the call instead of the call going to VM. When you call any extension directly and the call is not answered you end up in that extensions VM like you should. -Rob From: Michael Collins Sent: Tuesday, March 20, 2012 12:22 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Call sent to group call terminates when group is busy instead of transferring Malfunction! Need input! Get debug log of call from start to finish and put on pastebin. -MC On Tue, Mar 20, 2012 at 8:19 AM, Robert Longfield <robert.longfield@klinsight.com> wrote: Thanks for the tip Brian, I tried using a loopback using the example in /dialplan/default.xml and I am still experiencing the same problem. I’ve tried a loopback that looks like: <action application="bridge" data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/> <action application="bridge" data="loopback/app=voicemail:default $${domain_name} ${dialed_extension}"/> <action application="bridge" data="loopback/app=voicemail:1000@${domain_name} $1"/> <action application="bridge" data="loopback/app=voicemail:1000@${domain_name}"/> <action application="bridge" data="loopback/app=voicemail:1000@$${domain_name}"/> Only instead of dropping the call it seems to sleep... From: Brian Foster Sent: Monday, March 19, 2012 7:12 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Call sent to group call terminates when group is busy instead of transferring Try using loopback when you send the call to voicemail, also see the local extensions dialplan located in conf/dialplan/default.xml On Mar 19, 2012 4:43 PM, "Robert Longfield" <robert.longfield@klinsight.com> wrote: I set up a group call for our support team in which all their phones ring when someone needs to speak with them. If they are busy the call should be transferred to a general extension which if not answered then goes to that extensions VM. My dialplan looks like: <extension name="group_dial_support"> <condition field="destination_number" expression="^2001$"> <action application="set" data="continue_on_fail=true"/> <action application="bridge" data="group/support@${domain_name}"/> <!-- <action application="voicemail" data="default ${domain} 1000"/> --> <action application="transfer" data="1000 XML default"/> <action application="hangup"/> </condition> </extension> What is happening is a caller selects the support option from the IVR, ever phone in the support group rings, which is what should happen. If no one picks up the call Freeswitch hangs up instead of transferring the call to extension 1000. You can see that I also tried to send the call directly to voicemail but that didn’t work either. The message I see when Freeswitch hangs up is: Channel sofia/internal/sip:1002@72.38.184.18:39042 hanging up, cause: USER_BUSY The full output from cli can be seen here: http://pastebin.freeswitch.org/18696 I would like to get the call to transfer properly. Thanks -Robert -------------------------------------------------------------------------------- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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
Rob, Check out this pastebin: http://pastebin.freeswitch.org/18712 Look at line #424. There's no more dialplan actions parsed after the bridge. Look in the extension named 'group_dial_support' and see what's in there. Confirm that you actually have something in there after the bridge. -MC On Tue, Mar 20, 2012 at 10:00 AM, Robert Longfield < robert.longfield@klinsight.com> wrote:
ugh, I can’t believe I forgot to include the pastebin
The termination happens also when there is a single user in the call group. The call is transferred to the extension, that extension does not pickup and FS drops the call instead of the call going to VM. When you call any extension directly and the call is not answered you end up in that extensions VM like you should.
-Rob
*From:* Michael Collins <msc@freeswitch.org> *Sent:* Tuesday, March 20, 2012 12:22 PM *To:* FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> *Subject:* Re: [Freeswitch-users] Call sent to group call terminates when group is busy instead of transferring
Malfunction! Need input! Get debug log of call from start to finish and put on pastebin. -MC
On Tue, Mar 20, 2012 at 8:19 AM, Robert Longfield < robert.longfield@klinsight.com> wrote:
Thanks for the tip Brian,
I tried using a loopback using the example in /dialplan/default.xml and I am still experiencing the same problem.
I’ve tried a loopback that looks like: <action application="bridge" data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/> <action application="bridge" data="loopback/app=voicemail:default $${domain_name} ${dialed_extension}"/> <action application="bridge" data="loopback/app=voicemail:1000@${domain_name} $1"/> <action application="bridge" data="loopback/app=voicemail:1000@ ${domain_name}"/> <action application="bridge" data="loopback/app=voicemail:1000@ $${domain_name}"/>
Only instead of dropping the call it seems to sleep...
*From:* Brian Foster <bdfoster@endigotech.com> *Sent:* Monday, March 19, 2012 7:12 PM *To:* FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> *Subject:* Re: [Freeswitch-users] Call sent to group call terminates when group is busy instead of transferring
Try using loopback when you send the call to voicemail, also see the local extensions dialplan located in conf/dialplan/default.xml On Mar 19, 2012 4:43 PM, "Robert Longfield" < robert.longfield@klinsight.com> wrote:
I set up a group call for our support team in which all their phones ring when someone needs to speak with them. If they are busy the call should be transferred to a general extension which if not answered then goes to that extensions VM.
My dialplan looks like:
<extension name="group_dial_support"> <condition field="destination_number" expression="^2001$"> <action application="set" data="continue_on_fail=true"/> <action application="bridge" data="group/support@ ${domain_name}"/> <!-- <action application="voicemail" data="default ${domain} 1000"/> --> <action application="transfer" data="1000 XML default"/> <action application="hangup"/> </condition> </extension>
What is happening is a caller selects the support option from the IVR, ever phone in the support group rings, which is what should happen. If no one picks up the call Freeswitch hangs up instead of transferring the call to extension 1000. You can see that I also tried to send the call directly to voicemail but that didn’t work either.
The message I see when Freeswitch hangs up is:
Channel sofia/internal/sip:1002@72.38.184.18:39042 hanging up, cause: USER_BUSY
The full output from cli can be seen here: http://pastebin.freeswitch.org/18696
I would like to get the call to transfer properly.
Thanks -Robert
------------------------------
Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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
I think there was some confusion in the conf call and apparently all I was producing was static. When a caller is passed to extension 2001 which is our support group the extensions in the group all ring (extensions 1002 and 1003). At that point if the call is not answered the call is dropped or more recently it seems that FS is going to sleep and not dropping the call but giving only static. Here is a paste using the right one :) http://pastebin.freeswitch.org/18714 Thanks, -Rob From: Michael Collins Sent: Wednesday, March 21, 2012 2:15 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Call sent to group call terminates when group is busy instead of transferring Rob, Check out this pastebin: http://pastebin.freeswitch.org/18712 Look at line #424. There's no more dialplan actions parsed after the bridge. Look in the extension named 'group_dial_support' and see what's in there. Confirm that you actually have something in there after the bridge. -MC On Tue, Mar 20, 2012 at 10:00 AM, Robert Longfield <robert.longfield@klinsight.com> wrote: ugh, I can’t believe I forgot to include the pastebin http://pastebin.com/GYLvtDB3 The termination happens also when there is a single user in the call group. The call is transferred to the extension, that extension does not pickup and FS drops the call instead of the call going to VM. When you call any extension directly and the call is not answered you end up in that extensions VM like you should. -Rob From: Michael Collins Sent: Tuesday, March 20, 2012 12:22 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Call sent to group call terminates when group is busy instead of transferring Malfunction! Need input! Get debug log of call from start to finish and put on pastebin. -MC On Tue, Mar 20, 2012 at 8:19 AM, Robert Longfield <robert.longfield@klinsight.com> wrote: Thanks for the tip Brian, I tried using a loopback using the example in /dialplan/default.xml and I am still experiencing the same problem. I’ve tried a loopback that looks like: <action application="bridge" data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/> <action application="bridge" data="loopback/app=voicemail:default $${domain_name} ${dialed_extension}"/> <action application="bridge" data="loopback/app=voicemail:1000@${domain_name} $1"/> <action application="bridge" data="loopback/app=voicemail:1000@${domain_name}"/> <action application="bridge" data="loopback/app=voicemail:1000@$${domain_name}"/> Only instead of dropping the call it seems to sleep... From: Brian Foster Sent: Monday, March 19, 2012 7:12 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Call sent to group call terminates when group is busy instead of transferring Try using loopback when you send the call to voicemail, also see the local extensions dialplan located in conf/dialplan/default.xml On Mar 19, 2012 4:43 PM, "Robert Longfield" <robert.longfield@klinsight.com> wrote: I set up a group call for our support team in which all their phones ring when someone needs to speak with them. If they are busy the call should be transferred to a general extension which if not answered then goes to that extensions VM. My dialplan looks like: <extension name="group_dial_support"> <condition field="destination_number" expression="^2001$"> <action application="set" data="continue_on_fail=true"/> <action application="bridge" data="group/support@${domain_name}"/> <!-- <action application="voicemail" data="default ${domain} 1000"/> --> <action application="transfer" data="1000 XML default"/> <action application="hangup"/> </condition> </extension> What is happening is a caller selects the support option from the IVR, ever phone in the support group rings, which is what should happen. If no one picks up the call Freeswitch hangs up instead of transferring the call to extension 1000. You can see that I also tried to send the call directly to voicemail but that didn’t work either. The message I see when Freeswitch hangs up is: Channel sofia/internal/sip:1002@72.38.184.18:39042 hanging up, cause: USER_BUSY The full output from cli can be seen here: http://pastebin.freeswitch.org/18696 I would like to get the call to transfer properly. Thanks -Robert ------------------------------------------------------------------------------ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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 -------------------------------------------------------------------------------- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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
Okay, I have tried *really* hard and I cannot reproduce these symptoms. I've used your exact dialplan and it worked perfectly for me. I tried both group calling methods. (See default dialplan x2000 vs. x2001) I honestly think that this is a dialplan parsing issue. I suspect that you may have a rogue copy of the "group_dial_support" extension somewhere in your dialplan. The first thing I would do in your position is backup my configs, then clear out the /usr/local/freeswitch/conf directory tree. Run 'make samples' to get a fresh set of configs, then test with the stock x2000 and x2001 entries. FWIW, I've PB'd the changes I made to 2000 and 2001 and included a debug log of it working. See PB 18716, especially lines 163-165. Hope this helps... -MC On Wed, Mar 21, 2012 at 11:33 AM, Robert Longfield < robert.longfield@klinsight.com> wrote:
I think there was some confusion in the conf call and apparently all I was producing was static.
When a caller is passed to extension 2001 which is our support group the extensions in the group all ring (extensions 1002 and 1003). At that point if the call is not answered the call is dropped or more recently it seems that FS is going to sleep and not dropping the call but giving only static.
Here is a paste using the right one :)
http://pastebin.freeswitch.org/18714
Thanks, -Rob
*From:* Michael Collins <msc@freeswitch.org> *Sent:* Wednesday, March 21, 2012 2:15 PM *To:* FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> *Subject:* Re: [Freeswitch-users] Call sent to group call terminates when group is busy instead of transferring
Rob,
Check out this pastebin: http://pastebin.freeswitch.org/18712
Look at line #424. There's no more dialplan actions parsed after the bridge. Look in the extension named 'group_dial_support' and see what's in there. Confirm that you actually have something in there after the bridge.
-MC
On Tue, Mar 20, 2012 at 10:00 AM, Robert Longfield < robert.longfield@klinsight.com> wrote:
ugh, I can’t believe I forgot to include the pastebin
The termination happens also when there is a single user in the call group. The call is transferred to the extension, that extension does not pickup and FS drops the call instead of the call going to VM. When you call any extension directly and the call is not answered you end up in that extensions VM like you should.
-Rob
*From:* Michael Collins <msc@freeswitch.org> *Sent:* Tuesday, March 20, 2012 12:22 PM *To:* FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> *Subject:* Re: [Freeswitch-users] Call sent to group call terminates when group is busy instead of transferring
Malfunction! Need input! Get debug log of call from start to finish and put on pastebin. -MC
On Tue, Mar 20, 2012 at 8:19 AM, Robert Longfield < robert.longfield@klinsight.com> wrote:
Thanks for the tip Brian,
I tried using a loopback using the example in /dialplan/default.xml and I am still experiencing the same problem.
I’ve tried a loopback that looks like: <action application="bridge" data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/> <action application="bridge" data="loopback/app=voicemail:default $${domain_name} ${dialed_extension}"/> <action application="bridge" data="loopback/app=voicemail:1000@${domain_name} $1"/> <action application="bridge" data="loopback/app=voicemail:1000@ ${domain_name}"/> <action application="bridge" data="loopback/app=voicemail:1000@ $${domain_name}"/>
Only instead of dropping the call it seems to sleep...
*From:* Brian Foster <bdfoster@endigotech.com> *Sent:* Monday, March 19, 2012 7:12 PM *To:* FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> *Subject:* Re: [Freeswitch-users] Call sent to group call terminates when group is busy instead of transferring
Try using loopback when you send the call to voicemail, also see the local extensions dialplan located in conf/dialplan/default.xml On Mar 19, 2012 4:43 PM, "Robert Longfield" < robert.longfield@klinsight.com> wrote:
I set up a group call for our support team in which all their phones ring when someone needs to speak with them. If they are busy the call should be transferred to a general extension which if not answered then goes to that extensions VM.
My dialplan looks like:
<extension name="group_dial_support"> <condition field="destination_number" expression="^2001$"> <action application="set" data="continue_on_fail=true"/> <action application="bridge" data="group/support@ ${domain_name}"/> <!-- <action application="voicemail" data="default ${domain} 1000"/> --> <action application="transfer" data="1000 XML default"/> <action application="hangup"/> </condition> </extension>
What is happening is a caller selects the support option from the IVR, ever phone in the support group rings, which is what should happen. If no one picks up the call Freeswitch hangs up instead of transferring the call to extension 1000. You can see that I also tried to send the call directly to voicemail but that didn’t work either.
The message I see when Freeswitch hangs up is:
Channel sofia/internal/sip:1002@72.38.184.18:39042 hanging up, cause: USER_BUSY
The full output from cli can be seen here: http://pastebin.freeswitch.org/18696
I would like to get the call to transfer properly.
Thanks -Robert
------------------------------
Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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
------------------------------ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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
Micheal, The tip about checking for a rouge copy of the “group_dial_support” extension is exactly what the problem was. I haven’t gotten rid of default.xml and forgot to comment/remove the group_dial_support from the file. Also it seems I had to add ‘sofia/internal/’ in front of the extension. so <action application="bridge" data="1000@$${domain}"/> became <action application="bridge" data="sofia/internal/1000@$${domain}"/> I wrongly assumed that anything in default.xml would be ignored or passed over if that rule existed elsewhere. -Rob From: Michael Collins Sent: Wednesday, March 21, 2012 8:11 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Call sent to group call terminates when group is busy instead of transferring Okay, I have tried *really* hard and I cannot reproduce these symptoms. I've used your exact dialplan and it worked perfectly for me. I tried both group calling methods. (See default dialplan x2000 vs. x2001) I honestly think that this is a dialplan parsing issue. I suspect that you may have a rogue copy of the "group_dial_support" extension somewhere in your dialplan. The first thing I would do in your position is backup my configs, then clear out the /usr/local/freeswitch/conf directory tree. Run 'make samples' to get a fresh set of configs, then test with the stock x2000 and x2001 entries. FWIW, I've PB'd the changes I made to 2000 and 2001 and included a debug log of it working. See PB 18716, especially lines 163-165. Hope this helps... -MC On Wed, Mar 21, 2012 at 11:33 AM, Robert Longfield <robert.longfield@klinsight.com> wrote: I think there was some confusion in the conf call and apparently all I was producing was static. When a caller is passed to extension 2001 which is our support group the extensions in the group all ring (extensions 1002 and 1003). At that point if the call is not answered the call is dropped or more recently it seems that FS is going to sleep and not dropping the call but giving only static. Here is a paste using the right one :) http://pastebin.freeswitch.org/18714 Thanks, -Rob From: Michael Collins Sent: Wednesday, March 21, 2012 2:15 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Call sent to group call terminates when group is busy instead of transferring Rob, Check out this pastebin: http://pastebin.freeswitch.org/18712 Look at line #424. There's no more dialplan actions parsed after the bridge. Look in the extension named 'group_dial_support' and see what's in there. Confirm that you actually have something in there after the bridge. -MC On Tue, Mar 20, 2012 at 10:00 AM, Robert Longfield <robert.longfield@klinsight.com> wrote: ugh, I can’t believe I forgot to include the pastebin http://pastebin.com/GYLvtDB3 The termination happens also when there is a single user in the call group. The call is transferred to the extension, that extension does not pickup and FS drops the call instead of the call going to VM. When you call any extension directly and the call is not answered you end up in that extensions VM like you should. -Rob From: Michael Collins Sent: Tuesday, March 20, 2012 12:22 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Call sent to group call terminates when group is busy instead of transferring Malfunction! Need input! Get debug log of call from start to finish and put on pastebin. -MC On Tue, Mar 20, 2012 at 8:19 AM, Robert Longfield <robert.longfield@klinsight.com> wrote: Thanks for the tip Brian, I tried using a loopback using the example in /dialplan/default.xml and I am still experiencing the same problem. I’ve tried a loopback that looks like: <action application="bridge" data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/> <action application="bridge" data="loopback/app=voicemail:default $${domain_name} ${dialed_extension}"/> <action application="bridge" data="loopback/app=voicemail:1000@${domain_name} $1"/> <action application="bridge" data="loopback/app=voicemail:1000@${domain_name}"/> <action application="bridge" data="loopback/app=voicemail:1000@$${domain_name}"/> Only instead of dropping the call it seems to sleep... From: Brian Foster Sent: Monday, March 19, 2012 7:12 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Call sent to group call terminates when group is busy instead of transferring Try using loopback when you send the call to voicemail, also see the local extensions dialplan located in conf/dialplan/default.xml On Mar 19, 2012 4:43 PM, "Robert Longfield" <robert.longfield@klinsight.com> wrote: I set up a group call for our support team in which all their phones ring when someone needs to speak with them. If they are busy the call should be transferred to a general extension which if not answered then goes to that extensions VM. My dialplan looks like: <extension name="group_dial_support"> <condition field="destination_number" expression="^2001$"> <action application="set" data="continue_on_fail=true"/> <action application="bridge" data="group/support@${domain_name}"/> <!-- <action application="voicemail" data="default ${domain} 1000"/> --> <action application="transfer" data="1000 XML default"/> <action application="hangup"/> </condition> </extension> What is happening is a caller selects the support option from the IVR, ever phone in the support group rings, which is what should happen. If no one picks up the call Freeswitch hangs up instead of transferring the call to extension 1000. You can see that I also tried to send the call directly to voicemail but that didn’t work either. The message I see when Freeswitch hangs up is: Channel sofia/internal/sip:1002@72.38.184.18:39042 hanging up, cause: USER_BUSY The full output from cli can be seen here: http://pastebin.freeswitch.org/18696 I would like to get the call to transfer properly. Thanks -Robert ---------------------------------------------------------------------------- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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 ------------------------------------------------------------------------------ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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 -------------------------------------------------------------------------------- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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
Aha! Thanks for reporting back. This is a good learning exercise for XML dialplan editing. Also, if 1000 is a registered user on your system you can use data="user/1000" on your bridge line. -MC On Thu, Mar 22, 2012 at 9:27 AM, Robert Longfield < robert.longfield@klinsight.com> wrote:
Micheal,
The tip about checking for a rouge copy of the “group_dial_support” extension is exactly what the problem was. I haven’t gotten rid of default.xml and forgot to comment/remove the group_dial_support from the file.
Also it seems I had to add ‘sofia/internal/’ in front of the extension. so <action application="bridge" data="1000@$${domain}"/> became <action application="bridge" data="sofia/internal/1000@$${domain}"/>
I wrongly assumed that anything in default.xml would be ignored or passed over if that rule existed elsewhere.
-Rob
*From:* Michael Collins <msc@freeswitch.org> *Sent:* Wednesday, March 21, 2012 8:11 PM *To:* FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> *Subject:* Re: [Freeswitch-users] Call sent to group call terminates when group is busy instead of transferring
Okay, I have tried *really* hard and I cannot reproduce these symptoms. I've used your exact dialplan and it worked perfectly for me. I tried both group calling methods. (See default dialplan x2000 vs. x2001)
I honestly think that this is a dialplan parsing issue. I suspect that you may have a rogue copy of the "group_dial_support" extension somewhere in your dialplan. The first thing I would do in your position is backup my configs, then clear out the /usr/local/freeswitch/conf directory tree. Run 'make samples' to get a fresh set of configs, then test with the stock x2000 and x2001 entries. FWIW, I've PB'd the changes I made to 2000 and 2001 and included a debug log of it working. See PB 18716, especially lines 163-165.
Hope this helps...
-MC
On Wed, Mar 21, 2012 at 11:33 AM, Robert Longfield < robert.longfield@klinsight.com> wrote:
I think there was some confusion in the conf call and apparently all I was producing was static.
When a caller is passed to extension 2001 which is our support group the extensions in the group all ring (extensions 1002 and 1003). At that point if the call is not answered the call is dropped or more recently it seems that FS is going to sleep and not dropping the call but giving only static.
Here is a paste using the right one :)
http://pastebin.freeswitch.org/18714
Thanks, -Rob
*From:* Michael Collins <msc@freeswitch.org> *Sent:* Wednesday, March 21, 2012 2:15 PM *To:* FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> *Subject:* Re: [Freeswitch-users] Call sent to group call terminates when group is busy instead of transferring
Rob,
Check out this pastebin: http://pastebin.freeswitch.org/18712
Look at line #424. There's no more dialplan actions parsed after the bridge. Look in the extension named 'group_dial_support' and see what's in there. Confirm that you actually have something in there after the bridge.
-MC
On Tue, Mar 20, 2012 at 10:00 AM, Robert Longfield < robert.longfield@klinsight.com> wrote:
ugh, I can’t believe I forgot to include the pastebin
The termination happens also when there is a single user in the call group. The call is transferred to the extension, that extension does not pickup and FS drops the call instead of the call going to VM. When you call any extension directly and the call is not answered you end up in that extensions VM like you should.
-Rob
*From:* Michael Collins <msc@freeswitch.org> *Sent:* Tuesday, March 20, 2012 12:22 PM *To:* FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> *Subject:* Re: [Freeswitch-users] Call sent to group call terminates when group is busy instead of transferring
Malfunction! Need input! Get debug log of call from start to finish and put on pastebin. -MC
On Tue, Mar 20, 2012 at 8:19 AM, Robert Longfield < robert.longfield@klinsight.com> wrote:
Thanks for the tip Brian,
I tried using a loopback using the example in /dialplan/default.xml and I am still experiencing the same problem.
I’ve tried a loopback that looks like: <action application="bridge" data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/> <action application="bridge" data="loopback/app=voicemail:default $${domain_name} ${dialed_extension}"/> <action application="bridge" data="loopback/app=voicemail:1000@${domain_name} $1"/> <action application="bridge" data="loopback/app=voicemail:1000@ ${domain_name}"/> <action application="bridge" data="loopback/app=voicemail:1000@ $${domain_name}"/>
Only instead of dropping the call it seems to sleep...
*From:* Brian Foster <bdfoster@endigotech.com> *Sent:* Monday, March 19, 2012 7:12 PM *To:* FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> *Subject:* Re: [Freeswitch-users] Call sent to group call terminates when group is busy instead of transferring
Try using loopback when you send the call to voicemail, also see the local extensions dialplan located in conf/dialplan/default.xml On Mar 19, 2012 4:43 PM, "Robert Longfield" < robert.longfield@klinsight.com> wrote:
I set up a group call for our support team in which all their phones ring when someone needs to speak with them. If they are busy the call should be transferred to a general extension which if not answered then goes to that extensions VM.
My dialplan looks like:
<extension name="group_dial_support"> <condition field="destination_number" expression="^2001$"> <action application="set" data="continue_on_fail=true"/> <action application="bridge" data="group/support@ ${domain_name}"/> <!-- <action application="voicemail" data="default ${domain} 1000"/> --> <action application="transfer" data="1000 XML default"/> <action application="hangup"/> </condition> </extension>
What is happening is a caller selects the support option from the IVR, ever phone in the support group rings, which is what should happen. If no one picks up the call Freeswitch hangs up instead of transferring the call to extension 1000. You can see that I also tried to send the call directly to voicemail but that didn’t work either.
The message I see when Freeswitch hangs up is:
Channel sofia/internal/sip:1002@72.38.184.18:39042 hanging up, cause: USER_BUSY
The full output from cli can be seen here: http://pastebin.freeswitch.org/18696
I would like to get the call to transfer properly.
Thanks -Robert
------------------------------
Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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
------------------------------ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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
------------------------------ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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
participants (3)
-
Brian Foster
-
Michael Collins
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Robert Longfield