Follow-me to cell phone oddness

Hi everyone,

I’m trying to setup follow-me from a ring group to a cell phone aliased as an extension… and it’s not going so well.

I have the ring group set to call a set of extensions and one of those extensions has a follow-me setup that bounces to a cell. With follow-me setup as ringall, it appears as though the handoff is processing as expected. The hard extension rings for INITIAL_RING_TIME and then the call bounces to the cell extension and is forwarded out a SIP trunk to my cell phone. But then it gets weird…

The call rings on the cell for 10 seconds and then drops with the last messages in asterisk showing:

[Jan 3 21:26:41] – SIP/FlowRoute-SIP-00000c03 is ringing
[Jan 3 21:26:41] – SIP/FlowRoute-SIP-00000c03 is making progress passing it to Local/[email protected];2
[Jan 3 21:26:41] – Local/[email protected];1 is ringing
[Jan 3 21:26:52] == Spawn extension (macro-dial, s, 22) exited non-zero on ‘Local/[email protected];2’ in macro ‘dial’
[Jan 3 21:26:52] == Spawn extension (followme-sub, 202, 37) exited non-zero on ‘Local/[email protected];2’
[Jan 3 21:26:52] == Spawn extension (macro-dial-one, s, 54) exited non-zero on ‘Local/[email protected];2’ in macro ‘dial-one’
[Jan 3 21:26:52] == Spawn extension (macro-exten-vm, s, 14) exited non-zero on ‘Local/[email protected];2’ in macro ‘exten-vm’
[Jan 3 21:26:52] == Spawn extension (macro-dial, s, 22) exited non-zero on ‘SIP/2200-00000c01’ in macro ‘dial’
[Jan 3 21:26:53] == Spawn extension (ext-local, 755, 2) exited non-zero on ‘Local/[email protected];2’
[Jan 3 21:26:53] – Executing [[email protected]:1] Macro(“Local/[email protected];2”, “hangupcall,”) in new stack
[Jan 3 21:26:53] – Executing [[email protected]:1] GotoIf(“Local/[email protected];2”, “1?theend”) in new stack
[Jan 3 21:26:53] – Goto (macro-hangupcall,s,3)
[Jan 3 21:26:53] == Spawn extension (ext-group, 610, 14) exited non-zero on ‘SIP/2200-00000c01’
[Jan 3 21:26:53] – Executing [[email protected]:1] Macro(“SIP/2200-00000c01”, “hangupcall,”) in new stack
[Jan 3 21:26:53] – Executing [[email protected]:3] ExecIf(“Local/[email protected];2”, “0?Set(CDR(recordingfile)=)”) in new stack
[Jan 3 21:26:53] – Executing [[email protected]:4] Hangup(“Local/[email protected];2”, “”) in new stack
[Jan 3 21:26:53] == Spawn extension (macro-hangupcall, s, 4) exited non-zero on ‘Local/[email protected];2’ in macro ‘hangupcall’
[Jan 3 21:26:53] == Spawn extension (ext-local, h, 1) exited non-zero on ‘Local/[email protected];2’
[Jan 3 21:26:53] – Executing [[email protected]:1] GotoIf(“SIP/2200-00000c01”, “1?theend”) in new stack
[Jan 3 21:26:53] – Goto (macro-hangupcall,s,3)
[Jan 3 21:26:53] – Executing [[email protected]:3] ExecIf(“SIP/2200-00000c01”, “0?Set(CDR(recordingfile)=)”) in new stack
[Jan 3 21:26:53] – Executing [[email protected]:4] Hangup(“SIP/2200-00000c01”, “”) in new stack
[Jan 3 21:26:53] == Spawn extension (macro-hangupcall, s, 4) exited non-zero on ‘SIP/2200-00000c01’ in macro ‘hangupcall’
[Jan 3 21:26:53] == Spawn extension (ext-group, h, 1) exited non-zero on ‘SIP/2200-00000c01’

What’s odd is that if I answer the cell phone within this short 10s window, all seems ok. But, if I dont, it fails as shown above. But… the cell phone gets an odd recording. As soon as I see the last “hangup” message in the asterisk console, the message on the cell phone says:

“Please enter your area code and phone number followed by pound. If you make a mistake press blah blah blah…”

If I enter a number and area code and press pound, it then asks me to enter my PIN. I have no idea where this message is coming from. My guess is it’s something on the cell phone side of things b/c there are no more asterisk logs after the final “hangup” shown above.

But… this 10s drop only happens when an outside call is routed to an outside cell phone. If I call the follow-me extension from an internal extension, it rings through and keeps ringing.

At this exact 10s mark, I see debug messages showing SIP 102 CANCEL coming from the Linksys SPA3201 that’s handling the inbound POTS line. The outbound line to the cell is over a SIP trunk. Finally it all closes down with a SIP 103 CANCEL.

I have other inbound SIP trunks that auto forward to outbound SIP trunks to cell phones and they work without issue, but I’m not trying to use ring-groups or follow-me with those.

The precise and consistent 10s drop time is very suspicious to me. I just don’t know where to look for it.

This leaves me with a few perplexing questions…

  1. If the odd recording is something on the cell phone side, what is it about the asterisk “hangup” thats triggering it, and how do I stop it. It does this with Consumer Cellular and Verizon cell phones. (so far)

  2. How do I increase the allowed ring-time once the call is handed to the cell phone so the cell will ring long enough for the cell phone email to answer the call for calls that pass through from outside to outside? I have the RING_TIME set to 33 seconds. A native call allowed to ring on the cell goes to voicemail at 30s. But, no matter the setting, it all crashes at 10s.

It’s almost like there’s another timer somewhere that I’m missing that’s causing asterisk to drop the call after it’s handed to the cell phone. What’s odd is that no matter what I set the RING_TIME to, the hangup happens at the same 10s mark.

  1. Could this be some kind of weird interaction between the SPA3201 POTS line and the outbound SIP Trunk? Is the precise 10s drop an indicator of some sort?

  2. Is this a POTS-to-SIP issue and has nothing to do with ring groups and follow-me at all, or are these apps some how causing the problem?

I’m stumped. I thought I had it working with the calls being passed to the cell, caller ID being passed right along and then boom… calls drop right away. If I call the cell directly, or through the extension alias, there is no problem; so I don’t think there’s a routing issue or anything like that. I only have trouble when bouncing the cell through one of the apps (ring-groups / follow-me).

If anyone has any idea what this might be, or where I could look for answers, please let me know.

Thanks to you all in advance.

This is the strangest set of symptoms I’ve ever seen.

I assume that you have a typo and mean SPA3102 – if you have some other device, post make and model.

If the phone line it’s connected to is other than a copper pair fed from a Central Office, please post details (cable MTA, VoIP provider with locked ATA, POTS from fiber ONT, etc.)

Immediately after a failed call, what does the Info page of the SPA show for Last PSTN Disconnect Reason? If a disconnect tone was the cause, see whether temporarily setting Detect Disconnect Tone to no works around the problem.

If your POTS carrier sends caller ID as 10 digits and you are not prefixing a 1 before passing it to Flowroute, you are sending an international caller ID, which might affect either Flowroute’s or the MNO’s handling of the call. In Follow-Me for the relevant extension, try temporarily setting Change External CID Configuration to Fixed CID Value and set Fixed CID Value to a valid 11-digit number starting with 1.

Report whether answering the POTS line immediately, e.g. by temporarily setting Play Music On Hold for the Ring Group to default, works around the problem.

I am guessing that the robot you hear on your cell phone may be ‘access to voicemail from external’. However, since it might be something malicious, to test this do:

  1. Change your voicemail password to something you don’t use anywhere.
  2. Answer the follow-me call after 10 seconds, enter the area code and number of your mobile, enter the temporary password and report what happens.
  3. Immediately change your voicemail password back to the original value.

Report any non-voice equipment connected to your POTS line (fax machine, alarm system, etc.)

Post any special settings in your Ring Group or Follow Me (announcements, music, confirmations, etc.)

Post firmware version of your SPA3102 and the value set for Disconnect Tone.

Hi Stewart1,

It is a SPA3102 – sorry for the typo. It is connected to a copper pair from the telco CO.

After a call fails, the reason is: VoIP Call Ended

POTS CID is 10 digits. When I send to the flowroute trunk, it is sent with correct CID. ie. When the call bounces to the cell phone, the correct caller ID shows up on the cell phone.

I don’t understand what you meant by “Report whether answering the POTS line immediately, e.g. by temporarily setting Play Music On Hold for the Ring Group to default, works around the problem.”

If I answer the incoming POTS line, it wouldn’t bounce to the FMFM extension since it had been answered, would it?

I don’t have VM configured on this line. I think you’re right in that it sounds like it’s going to some VM system, but if it was the asterisk VM, wouldn’t it show up in the logs as going to VM? After the hangup, there are no more logs at all.

There are no non-voice devices connected to this POTS.

Nothing special about the ring group. It’s set to “ringall”, send call progress, ring for 35 seconds, and if no answer terminate/hangup.

Nothing special about the FMFM settings. It’s set to “ringall” and the external cell extension is the only number in the follow-me list.

The SPA3102 code versions are:
Software Version: 5.2.13(GW002)
Hardware Version: 1.4.5(a)

Thank you

I’m fairly sure that I understand what’s happening:

The POTS line once had voicemail (from the carrier or a third party). When it was cancelled, the account on the voicemail server was removed, but the ‘call forward no answer’ accidentally remained in place. So, if the POTS line is allowed to ring long enough (SPA’s PSTN Answer Delay + Initial Ring Time + setup time calling cell phone + 10 seconds), it forwards to the VM system, which doesn’t recognize the ‘redirected dialed number’ and prompts for it.

To confirm this, set (on the SPA3102 PSTN Line page in Advanced mode) Off Hook While Calling VoIP to yes, or set (for the Ring Group) Play Music On Hold to default. Either will cause the SPA to ‘pick up’ the POTS line before the follow me actions take place.

However, the above is likely not acceptable for production, because:

  1. Any analog phones connected to the POTS line will ring only once.
  2. Caller will be charged (if applicable to their plan) for calls not actually answered.
  3. Caller’s logs will show all calls as answered, even those that weren’t.

So, if the above test passes, the long term fix is to remove the unwanted POTS forwarding. On some systems, you can do this yourself by dialing a code. Otherwise, you’ll have to contact customer service.

I think you hit the nail on the head…

I disconnected the SPA and called the external number. It rang for exactly 30s and then went to that odd voicemail prompt.

I’m working with the telco now to try and fix it.


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