How to Handle Extension Misdials

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mantalope

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How do I set up the Digital Receptionist to NOT hang up if an extension is dialed incorrectly?

For example, if a caller tries to dial an extension that does not exist, the system plays a prompt indicating that the extension does not exist (which is good), but then it disconnects the call. I want to change it so that if an extension is dialed incorrectly, 3CX will restart the digital receptionist instead of hanging up. Can someone point me in the right direction here? It is probably a simple thing to configure.
 
I just tested mine and if I misdial an extension it says "call transfer failed" and goes back to the digital receptionist. I dont believe I ever made any change to enable this.
 
KerryG said:
I just tested mine and if I misdial an extension it says "call transfer failed" and goes back to the digital receptionist. I dont believe I ever made any change to enable this.

Probably because as far as I can tell there are no changes to allow this. I get the same message Kerry, Mantalope I can double check tomorrow in the office but something can't be right in your setup.

Regards,
Nate
 
On my system, when an incoming call dials an extension that does not exist, it tells me "your call cannot be completed at this time" and then hangs up. I don't know why it behaves differently on my system.
 
What does the server activity log show when that happens?
 
The Server Activity Log is below after dialing an unassigned extension (299)

16:24:35.216 [CM503007]: Call(151): Device joined: sip:[email protected]:5060
16:24:35.204 [CM505003]: Provider:[Nextiva] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [] PBX contact: [sip:xxx:5060]
16:24:35.204 [CM503002]: Call(151): Alerting sip:[email protected]:5060
16:24:34.724 [CM503025]: Call(151): Calling VoIPline:299@(Ln.10000@Nextiva)@[Dev:sip:[email protected]:5060]
16:24:34.666 [CM503004]: Call(151): Route 2: VoIPline:299@(Ln.10001@SkypeConnect)@[Dev:sip:[email protected]:5060]
16:24:34.666 [CM503004]: Call(151): Route 1: VoIPline:299@(Ln.10000@Nextiva)@[Dev:sip:[email protected]:5060]
16:24:34.664 [CM503010]: Making route(s) to <sip:[email protected]:5060>
16:24:34.395 Currently active calls - 1: [151]
16:24:27.733 [CM503007]: Call(151): Device joined: sip:[email protected]:40600;rinstance=d14de6fdee7d26bc

It looks like it is trying to call out on my SIP Trunk lines.
 
Looks like a problem with your outbound rules.
 
I have the same issue ;(
invalid extensions dialed will just disconnect, instead of returning to DR.
Any ideas? Server log below...

08:56:13.022 [CM503008]: Call: Call is terminated
08:56:12.388 [CM503016]: Call: Attempt to reach <sip:[email protected]:5060> failed. Reason: Not Found
08:56:12.388 [CM503014]: Call: No known route to target: <sip:[email protected]:5060>
08:56:12.386 [CM503010]: Making route(s) to <sip:[email protected]:5060>
08:56:06.285 [MS211000] C:14.1: 208.73.146.95:51546 is delivering DTMF using RTP payload (RFC2833). In-Band DTMF tone detection is disabled for this call segment.
08:55:52.188 [CM503007]: Call(): Device joined: sip:[email protected]:40600;rinstance=bca8c22f598a15ee
08:55:52.184 [CM503007]: Call(): Device joined: sip:[email protected]:5060
08:55:52.176 [CM505001]: Ext.8000: Device info: Device Identified: [Man: 3CX Ltd.;Mod: 3CX IVR;Rev: General] Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CX IVR] PBX contact: [sip:[email protected]:5060]
08:55:52.175 [CM503002]: Call(): Alerting sip:[email protected]:40600;rinstance=bca8c22f598a15ee
08:55:52.029 [CM503025]: Call(): Calling Ext:Ext.8000@[Dev:sip:[email protected]:40600;rinstance=bca8c22f598a15ee]
08:55:51.995 [CM503004]: Call(): Route 1: Ext:Ext.8000@[Dev:sip:[email protected]:40600;rinstance=bca8c22f598a15ee]
08:55:51.994 [CM503010]: Making route(s) to <sip:[email protected]:5060>
 
same with you Dave, it seems you have an issue with outbound rules.

08:56:12.388 [CM503014]: Call: No known route to target:
 
Thank you. I had an outbound rule that was a "catch all" to prioritize our SIP Trunk providers. When I removed the catch all rule, the mis-dials routed back to the digital attendant.
 
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