Call Rejection Drops Call
I am having an issue with calls coming from another SIP server into FS. Everything works fine with the call getting to its destination. And also works if the call is ignored and goes to voicemail. However if the call is rejected by a polycom endpoint the call is dropped completely and disconnects without going to voicemail. Here is a sample of the dialplan <action application="answer"/> <action application="bridge" data="sofia/internal/57331@192.168.100.25 "/> <!--<action application="answer"/>--> <action application="voicemail" data="default ${domain_name} 57331"/> <action application="hangup"/> If someone could point me in the right direction
Check this https://wiki.freeswitch.org/wiki/Variable_continue_on_fail On Tue, Jul 22, 2014 at 6:36 PM, Joel White <joelewhite@gmail.com> wrote:
I am having an issue with calls coming from another SIP server into FS. Everything works fine with the call getting to its destination. And also works if the call is ignored and goes to voicemail. However if the call is rejected by a polycom endpoint the call is dropped completely and disconnects without going to voicemail.
Here is a sample of the dialplan
<action application="answer"/> <action application="bridge" data="sofia/internal/ 57331@192.168.100.25"/> <!--<action application="answer"/>--> <action application="voicemail" data="default ${domain_name} 57331"/> <action application="hangup"/>
If someone could point me in the right direction
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
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Immediately before the bridge set hangup_after_bridge=true and continue_on_fail=true The first will hangup when the bridge ends normally, avoiding going to voicemail. The second will continue dialplan processing so voicemail is reached on error (eg call rejection). You also don't need to answer the call before bridging. The bridge will do that for you when required. Delaying it will have advantages to the caller, since they'll get charged while the call is ringing if you 'answer' early, as well as making the call time on their phone incorrect. Steve On 22 July 2014 16:36, Joel White <joelewhite@gmail.com> wrote:
I am having an issue with calls coming from another SIP server into FS. Everything works fine with the call getting to its destination. And also works if the call is ignored and goes to voicemail. However if the call is rejected by a polycom endpoint the call is dropped completely and disconnects without going to voicemail.
Here is a sample of the dialplan
<action application="answer"/> <action application="bridge" data="sofia/internal/ 57331@192.168.100.25"/> <!--<action application="answer"/>--> <action application="voicemail" data="default ${domain_name} 57331"/> <action application="hangup"/>
If someone could point me in the right direction
_________________________________________________________________________ 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
Thank you On Wed, Jul 23, 2014 at 6:34 AM, Steven Ayre <steveayre@gmail.com> wrote:
Immediately before the bridge set hangup_after_bridge=true and continue_on_fail=true
The first will hangup when the bridge ends normally, avoiding going to voicemail. The second will continue dialplan processing so voicemail is reached on error (eg call rejection).
You also don't need to answer the call before bridging. The bridge will do that for you when required. Delaying it will have advantages to the caller, since they'll get charged while the call is ringing if you 'answer' early, as well as making the call time on their phone incorrect.
Steve
On 22 July 2014 16:36, Joel White <joelewhite@gmail.com> wrote:
I am having an issue with calls coming from another SIP server into FS. Everything works fine with the call getting to its destination. And also works if the call is ignored and goes to voicemail. However if the call is rejected by a polycom endpoint the call is dropped completely and disconnects without going to voicemail.
Here is a sample of the dialplan
<action application="answer"/> <action application="bridge" data="sofia/internal/ 57331@192.168.100.25"/> <!--<action application="answer"/>--> <action application="voicemail" data="default ${domain_name} 57331"/> <action application="hangup"/>
If someone could point me in the right direction
_________________________________________________________________________ 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)
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Joel White
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Steven Ayre
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Vladimir Ralev