Freepbx: we have not received a valid response

Hello guys! I recently attempted to install Zoho Phone Bridge to integrate our CRM with our PBX server. Things seemed to be successful, and for 24 hours we didn’t notice any issues.

Fast forward 48 hours later, and when customers call our phone number it rings once and then says ‘We have not received a valid response’.

We did not make any changes to our FreePBX configuration during this time and I have been assured by Zoho support that their phone bridge does not modify any settings in FreePBX or Asterisk - it is merely a listening service that creates SIP calls.

It seems that roughly 1 in 15 calls get this message and I have since completely removed the phonebridge and even restored FreePBX from a backup I created before installing the phone bridge.

I contacted my phone company to see if there is a problem on their end and they have not gotten back to me.

Any help would be greatly appreciated!

Also - I am currently running an SIP Trace by using the command ‘sip set debug on’. No information is shown when I call and get the ‘we have not received a valid response’ message, but trace information is returned when the call actually goes through.

I don’t know if this indicates that the call is never even reaching the PBX server and that possibly it is an issue with our phone provider?

Update: I do not seem to get this problem when calling an inbound route that goes directly to an extension. It only seems to be happening on routes that go to a time-condition and then to an IVR.

Also - I changed the extension that one of my test routes goes to and started calling it to see if I could get the error. Mid calling it reverted back to the extension that was set before.

I am paranoid now and think something might be replacing the configurations that freepbx is setting.

I am at a loss.

Ok - it seems that roughly 1 out of 15 calls goes straight to the IVR timeout and says ‘We have not received a valid response’. Also, the IVR is setup to say the options again but it isn’t. When I try to use one of the options (Because I know what they are) it plays the hold music but still doesn’t ring the ring group for this inbound route.

I cannot figure out why some of the calls are going straight to the IVR timeout.

I have even created a new IVR and this has not fixed the issue.

Here is some output, the last call being one of the ones that goes straight to IVR timeout.

Could “[2016-03-25 14:35:02] VERBOSE[16125] asterisk.c: Remote UNIX connection” be something to do with the phonebridge? I am not seeing entries like that for the calls that do go through properly.

[2016-03-25 14:34:35] VERBOSE[16137] chan_sip.c: Extension Changed 405[ext-local] new state Idle for Notify User 401 (queued)
[2016-03-25 14:34:35] VERBOSE[22981][C-00000097] app_dial.c: SIP/402-000001f1 is ringing
[2016-03-25 14:34:35] VERBOSE[22981][C-00000097] app_dial.c: SIP/403-000001f2 is ringing
[2016-03-25 14:34:36] VERBOSE[22981][C-00000097] app_dial.c: SIP/401-000001f0 is ringing
[2016-03-25 14:34:39] VERBOSE[16137] chan_sip.c: Extension Changed 401[ext-local] new state Idle for Notify User 402
[2016-03-25 14:34:39] VERBOSE[16137] chan_sip.c: Extension Changed 401[ext-local] new state Idle for Notify User 301
[2016-03-25 14:34:39] VERBOSE[22981][C-00000097] app_macro.c: Spawn extension (macro-dial, s, 17) exited non-zero on ‘SIP/cbeyond-in-000001ee’ in macro ‘dial’
[2016-03-25 14:34:39] VERBOSE[16137] chan_sip.c: Extension Changed 401[ext-local] new state Idle for Notify User 303
[2016-03-25 14:34:39] VERBOSE[22981][C-00000097] pbx.c: Spawn extension (ext-group, 682, 11) exited non-zero on ‘SIP/cbeyond-in-000001ee’
[2016-03-25 14:34:39] VERBOSE[22981][C-00000097] pbx.c: Executing [[email protected]:1] Macro(“SIP/cbeyond-in-000001ee”, “hangupcall,”) in new stack
[2016-03-25 14:34:39] VERBOSE[16137] chan_sip.c: Extension Changed 402[ext-local] new state Idle for Notify User 401
[2016-03-25 14:34:39] VERBOSE[16137] chan_sip.c: Extension Changed 402[ext-local] new state Idle for Notify User 301
[2016-03-25 14:34:39] VERBOSE[16137] chan_sip.c: Extension Changed 403[ext-local] new state Idle for Notify User 404
[2016-03-25 14:34:39] VERBOSE[22981][C-00000097] pbx.c: Executing [[email protected]:1] GotoIf(“SIP/cbeyond-in-000001ee”, “1?theend”) in new stack
[2016-03-25 14:34:39] VERBOSE[22981][C-00000097] pbx.c: Goto (macro-hangupcall,s,3)
[2016-03-25 14:34:39] VERBOSE[16137] chan_sip.c: Extension Changed 403[ext-local] new state Idle for Notify User 301
[2016-03-25 14:34:39] VERBOSE[22981][C-00000097] pbx.c: Executing [[email protected]:3] ExecIf(“SIP/cbeyond-in-000001ee”, “0?Set(CDR(recordingfile)=)”) in new stack
[2016-03-25 14:34:39] VERBOSE[22981][C-00000097] pbx.c: Executing [[email protected]:4] Hangup(“SIP/cbeyond-in-000001ee”, “”) in new stack
[2016-03-25 14:34:39] VERBOSE[16137] chan_sip.c: Extension Changed 403[ext-local] new state Idle for Notify User 303
[2016-03-25 14:34:39] VERBOSE[22981][C-00000097] app_macro.c: Spawn extension (macro-hangupcall, s, 4) exited non-zero on ‘SIP/cbeyond-in-000001ee’ in macro ‘hangupcall’
[2016-03-25 14:34:39] VERBOSE[22981][C-00000097] pbx.c: Spawn extension (ext-group, h, 1) exited non-zero on ‘SIP/cbeyond-in-000001ee’
[2016-03-25 14:35:02] VERBOSE[16125] asterisk.c: Remote UNIX connection
[2016-03-25 14:35:02] VERBOSE[23015] asterisk.c: Remote UNIX connection disconnected
[2016-03-25 14:35:02] VERBOSE[16125] asterisk.c: Remote UNIX connection
[2016-03-25 14:35:02] VERBOSE[23017] asterisk.c: Remote UNIX connection disconnected

The Sangoma CRM Link module now supports both Zoho and SalesForce. https://wiki.freepbx.org/display/FPG/Customer+Relationship+Management