'Call Transfer Failed' when transferring to external number

[Aegis PS]

We have a series of menu options set up with Digital Receptionist. One of the options on an IVR is set to dial a Ring Group with a Ring Time of 1 second that is set to Forward to Outside Number upon no answer. Before upgrading to v16, this number seemed to work fine but now it does not.
Calling the external number directly works fine. It does not work when I go through the menu options and choose the option that dials the external number, does not work when I dial our main number then the IVR's extension and choose the option, and does not work when I dial the IVR's DID and choose the option. All Firewall tests passed and calling is working fine otherwise.

Is there an attribute/flag that needs to be set? Is there an easier way to set an option to forward to an external number?

 

[JohnS_3CX]

Hi,

In V16 we introduced some tighter security measures, such that the system extensions cannot make outbound calls unless there is an outbound rule that covers this.

Slide 9-10 should explain this in more detail:
https://www.3cx.com/3cxacademy/videos/advanced/configuring-outbound-routing/

 

[Aegis PS]

Thank you for your response, JohnS. I added another Outbound Rule but the transfer still isn't working.

Currently, I have it at the top of our rules list. I tried adding extension group and length, but it did not work with either. All 5 Routes are configured with our SIP provider.
Am I missing something?

 

[JohnS_3CX]

I think you need to revise the rule to look like the one in the tutorial like "Allow ALL Calls"
I see you added 899 as the only match and that may not work as intended.



Just add a route, and a prefix like the screenshot so that 1 thing matches when the outgoing attempt is made.

 

[Aegis PS]

I added a rule at the top with only the Length field set to 10 and configured routes 1-5 to use our SIP trunks. I am still receiving the message "call transfer failed, no answer" when dialing that option. We already had rules set for outbound calls that should have caught dialing from that extension, so I don't think the issue is with Outbound Rules.

 

[Saqqara]

Create a new rule with prefix containing the outside number in question - move it to the top.

 

[Aegis PS]

Tried a rule with just the prefix set and routes for our SIP. Still giving me the same "call transfer failed" message.

 

[JohnS_3CX]

Switch to verbose logs from the Activity Log page, and see what you get

 

[Aegis PS]

I enabled verbose logging and searched for logs pertaining to the external number, 8442004067.
I'm seeing some messages about "Not Acceptable DMTF not allowed" and "Unsupported Media Type." Calling the external number directly works fine, but not from IVR.

03/19/2020 2:52:03 PM - ~Target=Unknown:8442004067@(Ln.10000@Redacted-SIP)

03/19/2020 2:52:03 PM - Leg L:2791.7[Line:10000>>8442004067] is terminated: Cause: 415 Not Acceptable DMTF not allowed/INVITE from 216.147.191.155:5060

03/19/2020 2:52:03 PM - L:2791.7[Line:10000>>8442004067] got Terminated Recv 415/INVITE from 216.147.191.155:5060 tid=0c87a62e84093572 Call-ID=4lC0vA9Aa-fWHsfkUnv4ww..: SIP/2.0 415 Not Acceptable DMTF not allowed Via: SIP/2.0/UDP Redacted-Internal-IP:5060;received=Redacted-Public-IP;rport=5060;branch=z9hG4bK-524287-1---0c87a62e84093572 To: <sip:[email protected]-SIP:5060>;tag=a9da2ef085dcd86bf818ee7348f29716-d04b From: "5139480066"<sip:[email protected]-SIP:5060>;tag=dd02fa54 Call-ID: 4lC0vA9Aa-fWHsfkUnv4ww.. CSeq: 2 INVITE Server: VSIP RTE (2.4.0-notls (x86_64/linux)) Content-Length: 0

03/19/2020 2:52:03 PM - Terminated from <sip:[email protected]-SIP:5060>;tag=a9da2ef085dcd86bf818ee7348f29716-d04b to "5139480066"<sip:[email protected]-SIP:5060>;tag=dd02fa54; reason: Error

03/19/2020 2:52:03 PM - L:2791.1[Line:10000<<Redacted-Calling-Number] failed to reach Line:10000>>8442004067, reason Unsupported Media Type

03/19/2020 2:52:03 PM - Call to T:Line:10000>>8442004067@[Dev:sip:[email protected]-SIP:5060] from L:2791.1[Line:10000<<Redacted-Calling-Number] failed, cause: Cause: 415 Not Acceptable DMTF not allowed/INVITE from 216.147.191.155:5060

03/19/2020 2:52:03 PM - L:2791.7[Line:10000>>8442004067] got Failure: Failure Recv 415/INVITE from 216.147.191.155:5060 tid=0c87a62e84093572 Call-ID=4lC0vA9Aa-fWHsfkUnv4ww..: SIP/2.0 415 Not Acceptable DMTF not allowed Via: SIP/2.0/UDP Redacted-Internal-IP:5060;received=Redacted-Public-IP;rport=5060;branch=z9hG4bK-524287-1---0c87a62e84093572 To: <sip:[email protected]-SIP:5060>;tag=a9da2ef085dcd86bf818ee7348f29716-d04b From: "5139480066"<sip:[email protected]-SIP:5060>;tag=dd02fa54 Call-ID: 4lC0vA9Aa-fWHsfkUnv4ww.. CSeq: 2 INVITE Server: VSIP RTE (2.4.0-notls (x86_64/linux)) Content-Length: 0

03/19/2020 2:52:03 PM - [CM503003]: Call(C:2791): Call to <sip:[email protected]-SIP:5060> has failed; Cause: 415 Not Acceptable DMTF not allowed/INVITE from 216.147.191.155:5060

03/19/2020 2:52:03 PM - Session 387911 has failed in leg L:2791.7[Line:10000>>8442004067] ; Cause: 415 Not Acceptable DMTF not allowed/INVITE from 216.147.191.155:5060

03/19/2020 2:52:03 PM - Failure from <sip:[email protected]-SIP:5060>;tag=a9da2ef085dcd86bf818ee7348f29716-d04b to "5139480066"<sip:[email protected]-SIP:5060>;tag=dd02fa54

03/19/2020 2:52:02 PM - Leg L:2791.6[Line:10000>>8442004067] is terminated: Cause: 415 Not Acceptable DMTF not allowed/INVITE from 216.147.191.155:5060

03/19/2020 2:52:02 PM - L:2791.6[Line:10000>>8442004067] got Terminated Recv 415/INVITE from 216.147.191.155:5060 tid=cb351e193ea0f636 Call-ID=tLQiIWQfAbscYKQUH-cF_Q..: SIP/2.0 415 Not Acceptable DMTF not allowed Via: SIP/2.0/UDP Redacted-Internal-IP:5060;received=Redacted-Public-IP;rport=5060;branch=z9hG4bK-524287-1---cb351e193ea0f636 To: <sip:[email protected]-SIP:5060>;tag=1dc6d9d6d4a5f41990738bd252562e74-a8bf From: "5139480066"<sip:[email protected]-SIP:5060>;tag=9ec3e218 Call-ID: tLQiIWQfAbscYKQUH-cF_Q.. CSeq: 2 INVITE Server: VSIP RTE (2.4.0-notls (x86_64/linux)) Content-Length: 0

03/19/2020 2:52:02 PM - Terminated from <sip:[email protected]-SIP:5060>;tag=1dc6d9d6d4a5f41990738bd252562e74-a8bf to "5139480066"<sip:[email protected]-SIP:5060>;tag=9ec3e218; reason: Error

 

[Aegis PS]

I still get the "Call Transfer Failed" response when calling the Ring Group from the Menu. The strange thing is calling the Ring Group's extension directly transfers the call correctly. I tried rebuilding the Ring Group, using just an extension, using a Call Queue, but no methods work. Using a 3CX Call Flow with a Transfer component just ends the call when it gets to it.

Is there a way to set up forwarding to an external number from the Digital Receptionist menu?

 

[leejor]

03/19/2020 2:52:03 PM - L:2791.1[Line:10000<<Redacted-Calling-Number] failed to reach Line:10000>>8442004067, reason Unsupported Media Type

It would certainly seem to indicate a Codec incompatibility issue. Which Codecs are you using in the sets, have 3CX set to use, and are used by your provider? Have you set restrictions so that some Codecs are being excluded?

 

[JohnS_3CX]

Also the Ring Group with a Ring Time of 1, which extensions does it include and are they registered?

I wish to know if there are any registered phones in that RG that ring at least once before the number forwards to an external line.

 

[Aegis PS]

The Ring Group originally just had the x000 extension as a member. It does not have any assigned phones. This configuration worked in v15.
I tried configuring the Ring Group to use my extension, which does have phones assigned and registered. With a ring time of 1 second, it briefly rings my extension, but the caller still gets the message "call transfer failed." I set the ring time to 5 seconds and I still get the same message after the 5 seconds.

 

[JohnS_3CX]

Ok for the call that involves your extension (which has a registered phone) can we see anything in the logs showing the flow of the call? You can set filters to show the last call so you can see the whole thing including your phone ringing, and the attempt to forward

 

[Aegis PS]

After going through the menus I see my call is routed to the Ring Group

03/31/2020 11:00:58 AM - [CM503025]: Call(C:2440): Calling T:RingAll:898@[Dev:sip:[email protected]:5483;rinstance=a7b50777dbca533d ...

And it does ring my extension

03/31/2020 11:00:58 AM - Reported(Insert): SE:2440.4{Ringing}

03/31/2020 11:00:58 AM - StatusNotification: SE:2440.4{Ringing}; Skip=0

03/31/2020 11:00:58 AM - UacSession 607354 has formed leg L:2440.4[Extn:142]

There is no response from the members of the ring group (just my extension)

03/31/2020 11:00:59 AM - [CM503016]: Call(C:2440): Attempt to reach <sip:[email protected]:5060> from Line:10000<<redacted-caller has failed. Reason: No Answer

03/31/2020 11:00:59 AM - L:2440.1[Line:10000<<redacted-caller] failed to reach RingAll:898, reason No Answer

A few instances of these logs show up (there are verbose logs saying the same thing as these)

03/31/2020 11:00:59 AM - L:2440.4[Extn:142]: Terminating targets, reason: SIP ;cause=487 ;text="Request Terminated"

03/31/2020 11:00:59 AM - Leg L:2440.4[Extn:142] is terminated: Cause: 487 Request Terminated/INVITE from redacted-phone-ip:11217

I see a few logs involving our SIP provider like the one below

03/31/2020 11:01:00 AM - L:2440.1[Line:10000<<redacted-caller]: SIPReq destination is forced to redacted-sip-ip:5060/UDP fk=14 tgt= (endpoint forced)

I'm not sure what the logs below mean

03/31/2020 11:01:00 AM - Call(C:2440), Ivr:800 on entry: DlgInfo(2440-1525/Confirmed / R)

03/31/2020 11:01:00 AM - Notify dialog-info: Ivr:800: sip:[email protected]:5483;rinstance=998f518cf406118c, Call(C:2440)

03/31/2020 11:01:00 AM - L:2440.2[Ivr:800] answering in state GotOffer

03/31/2020 11:01:00 AM - L:2440.2[Ivr:800] providing answer in state GotOffer

Eventually, I see a "BYE" message

03/31/2020 11:01:05 AM - L:2440.2[Ivr:800] got Terminated Recv Req BYE from 127.0.0.1:5483 tid=54653b4974e7903c Call-ID=kUIg_fMIZCy8EGzXb2_kXg..: BYE sip:[email protected]:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5483;branch=z9hG4bK-524287-1---54653b4974e7903c;rport=5483 Max-Forwards: 70 Contact: <sip:[email protected]:5483;rinstance=998f518cf406118c> To: "AEGIS PROTECTIV:Main Number to Menu"<sip:[email protected]:5060;nf=e>;tag=1ab5e961 From: <sip:[email protected]>;tag=40bd1d4e Call-ID: kUIg_fMIZCy8EGzXb2_kXg.. CSeq: 6 BYE User-Agent: 3CX IVR Reason: SIP;text=User Hung Up Content-Length: 0

03/31/2020 11:01:05 AM - L:2440.2[Ivr:800] Sending: OnSendResp Send 200/BYE from 0.0.0.0:0 tid=54653b4974e7903c Call-ID=kUIg_fMIZCy8EGzXb2_kXg..: SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:5483;branch=z9hG4bK-524287-1---54653b4974e7903c;rport=5483 Contact: <sip:[email protected]:5060> To: "AEGIS PROTECTIV:Main Number to Menu"<sip:[email protected]:5060;nf=e>;tag=1ab5e961 From: <sip:[email protected]>;tag=40bd1d4e Call-ID: kUIg_fMIZCy8EGzXb2_kXg.. CSeq: 6 BYE Content-Length: 0

And the call ends

03/31/2020 11:01:05 AM - Removing leg L:2440.2[Ivr:800]

03/31/2020 11:01:05 AM - L:2440.2[Ivr:800]: Terminating targets, reason: SIP ;cause=200 ;text="User hangup"

03/31/2020 11:01:05 AM - Leg L:2440.2[Ivr:800] is terminated: Cause: BYE from 127.0.0.1:5483

03/31/2020 11:01:05 AM - Call(C:2440), Ivr:800 on entry: DlgInfo(2440-1525/Terminated / R)

03/31/2020 11:01:05 AM - L:2440.1[Line:10000<<15138853077]: Terminating targets, reason: SIP ;cause=200 ;text="Call terminated on user request"

Oddly enough, there is no mention of the external number to which I'm trying to forward.

 

[Aegis PS]

It would certainly seen to indicate a Codec incompatibility issue. Which Codecs are you using in the sets, have 3CX set to use, and are used by your provider? Have you set restrictions so that some Codecs are being excluded?

There are no codec restrictions and I tried setting various codecs for the member extensions of the Ring Group and none worked (GSM-FR, G729, G722, etc.). We did not change the codecs when updating to v16.

 

[mdc2]

I ended up having to change the process after our version 16 upgrade. I created extensions that rang out to "Mobile" numbers or forwarded out to a number from the extension. Put the extension in a group that had rights to make outbound calls. And then set the options to transfer to those extensions instead of directly out. That allowed me to duplicate what the pre-version 16 system used to allow directly from the digital receptionist.

 

[Aegis PS]

I ended up having to change the process after our version 16 upgrade. I created extensions that rang out to "Mobile" numbers or forwarded out to a number from the extension. Put the extension in a group that had rights to make outbound calls. And then set the options to transfer to those extensions instead of directly out. That allowed me to duplicate what the pre-version 16 system used to allow directly from the digital receptionist.

Could you elaborate how you accomplished this?
I tried to set up forwarding with an extension and, just like the Ring Group, it forwards correctly if I call the extension directly but not if the call goes through the Digital Receptionist.
Checking the option "Announce Caller (Rebound)" gives me the prompt "Please wait while we locate [extension number]"

 

[mdc2]

I set the outbound number in the "Mobile Number" and then on the forwarding rules set it to "Forward to Mobile"

 

[Aegis PS]

I set the outbound number in the "Mobile Number" and then on the forwarding rules set it to "Forward to Mobile"

I tried using that method but got the same result with "call transfer failed" when going through the Digital Receptionist.