Silence at end of voicemail greeting & skip instructions

[ou812]

Hi All

I am new to Fusionpbx but a long time user of Freepbx, I am looking to move on from Freepbx and thought I would give Fusionpbx a try, so far I like it a lot, I followed the instructions from here to install a server in the cloud on Debain 8. I was hit with bad people trying to register almost right away so I installed Travelingman3 and it stopped.

The system is up and running with 2 additional domains, I have a trunk registered and used by both domains for in & out calls, I have extensions registered on both domains, there is a lot to learn but for now I have 2 questions about voicemail.

1. when I record a greeting it says I can press any key to stop the recording or stop talking, I press the # key but it does not react it just waits and then stops, so greetings have 5 seconds of silence at the end, to get buy I downloaded it then edited it and uploaded, so how do I stop the recording.
2. after the greeting is played to a customer if gives instructions before the record beep, I would like to go straight to the beep.

Thanks,

Gary,

 

[EasyBB]

when I record a greeting it says I can press any key to stop the recording or stop talking, I press the # key but it does not react it just waits and then stops

Try adjusting DTMF parameters on the profile you are connecting through.
https://wiki.freeswitch.org/wiki/Sofia.conf.xml#DTMF

1. after the greeting is played to a customer if gives instructions before the record beep, I would like to go straight to the beep.

https://wiki.freeswitch.org/wiki/Mod_voicemail#skip_instructions

Where you add this entry in the dialplan depends on how the caller gets to voicemail.

 

[ou812]

EasyBB, thanks for the reply, using fusionpbx I'm not sure how to make the adjustments you mentioned above in the gui, how do I assign a sip profile to a user, I created a copy of the internal profile and assigned a host name but it looks like my extensions in that domain are still using internal sip profile and not the new one created and assigned to that hostname ?
Is this something that needs to be adjusted at the command line.

Gary.

 

[EasyBB]

how do I assign a sip profile to a user

You don't; please use the internal profile and adjust DTMF parameters in that profile. Every profile should have a unique IP : Port combination, in case you want to create multiple profiles.

 

[Andrew Byrd]

I am having the same issue with "press any key to stop recording" and it waits another 4 - 6 seconds and does not recognize when I press a key. I looked at the sip profile settings. They are set to dtmf mode rfc2833 and the duration is set to 2000. I tried duration 1000 and 3000, no noticeable change. What did you do to fix this?

 

[Incubugs]

Hi guys, i also have this issue i think it may be a bug, are you accessing the vm via *98 or *97 ? if i use *98 i get the issue as you have reported if i use *97 i don't just a thought..

 

[Andrew Byrd]

I agree - must be a bug. I tested it on my end. Here are the results

1) You dial *98, record your voice mail. It says to press any key to stop recording. Pressing keys has no effect, therefore you greeting has 5 seconds of dead air at the end. The caller starts to leave their message not realizing it is not ready yet.

2) You dial *97, record your voice mail, and pressing any key works perfectly.

I have to be at the actual ext to use *97 where *98 would allow me to record greetings from any extension - that's the only disadvantage

 

[DigitalDaz]

Which version of fusionpbx is this??

 

[Andrew Byrd]

4.4

 

[sebastian hache]

exact same issue here, on 4.2 *97 work perfectly, as soon as i enter a dtmf, the wav stop and process it. but *98, it process the dtmf, but after the whole wav has played... i made a trace when doing *97 and *98. *97 has this when entering a dtmf :
2018-05-07 09:07:43.768245 [INFO] switch_cpp.cpp:1360 [voicemail] dtmf digit: 5, duration: 1120
2018-05-07 09:07:43.768245 [INFO] switch_cpp.cpp:1360 [voicemail] dtmf digits: 5, length: 1 max_digits: 20
but not *98.

 

[hailthemelody]

Having the same issue here. Has anyone found a solution to this?

Edit: my FusionPBX version is 4.2.5, Branch: 4.2 Commit: 72ff086be098598a3b463bfdb26bc71b27a0f539

 

[bcmike]

Running into this exact issue, did anyone ever solve it?

Running 4.4.3

This is with *98 (Not working):

2019-07-25 09:47:15.704635 [DEBUG] switch_ivr_play_say.c:1941 done playing file /usr/share/freeswitch/sounds/en/us/callie/voicemail/vm-record_greeting.wav
2019-07-25 09:47:15.704635 [DEBUG] switch_ivr_play_say.c:1497 Codec Activated L16@16000hz 1 channels 30ms
2019-07-25 09:47:16.724636 [DEBUG] switch_ivr_play_say.c:1941 done playing file tone_stream://L=1;%(1000, 0, 640)
2019-07-25 09:47:16.844633 [DEBUG] switch_cpp.cpp:905 getDigits dtmf_buf:
2019-07-25 09:47:16.844633 [DEBUG] switch_ivr_play_say.c:567 Raw Codec Activated, ready to waste resources!
2019-07-25 09:47:16.844633 [DEBUG] switch_ivr_play_say.c:681 Raw Codec Activated
2019-07-25 09:47:16.844633 [DEBUG] switch_core_codec.c:223 sofia/External-NAT2/100@nwv.pbx.crimxen.com Push codec L16:100
2019-07-25 09:47:21.044607 [DEBUG] switch_core_io.c:780 Engaging Read Buffer at 960 bytes vs 160
2019-07-25 09:47:21.764619 [DEBUG] switch_rtp.c:7789 RTP RECV DTMF #:1680
2019-07-25 09:47:21.764619 [INFO] switch_channel.c:515 RECV DTMF #:1680


This is with *97 (working)

2019-07-25 09:57:32.144631 [DEBUG] switch_ivr_play_say.c:1497 Codec Activated L16@16000hz 1 channels 30ms
2019-07-25 09:57:37.004648 [DEBUG] switch_ivr_play_say.c:1941 done playing file /usr/share/freeswitch/sounds/en/us/callie/voicemail/vm-record_greeting.wav
2019-07-25 09:57:37.004648 [DEBUG] switch_ivr_play_say.c:1497 Codec Activated L16@16000hz 1 channels 30ms
2019-07-25 09:57:38.024638 [DEBUG] switch_ivr_play_say.c:1941 done playing file tone_stream://L=1;%(1000, 0, 640)
2019-07-25 09:57:38.144597 [DEBUG] switch_cpp.cpp:905 getDigits dtmf_buf:
2019-07-25 09:57:38.144597 [DEBUG] switch_ivr_play_say.c:567 Raw Codec Activated, ready to waste resources!
2019-07-25 09:57:38.144597 [DEBUG] switch_ivr_play_say.c:681 Raw Codec Activated
2019-07-25 09:57:38.144597 [DEBUG] switch_core_codec.c:223 sofia/External-NAT2/100@nwv.pbx.crimxen.com Push codec L16:100
2019-07-25 09:57:44.044598 [DEBUG] switch_rtp.c:7789 RTP RECV DTMF #:960
2019-07-25 09:57:44.044598 [INFO] switch_channel.c:515 RECV DTMF #:960
2019-07-25 09:57:44.044598 [INFO] switch_cpp.cpp:1443 [voicemail] dtmf digit: #, duration: 960

The only diff i see is with the highlighted line. Not sure why it would change

 

[bcmike]

Ok found a workaround.

If you dial*98[extension_number] the greeting recording menu will work.

Seems the issue with dialing *98 then inputting the extension number is DTMF byte length is somehow set different and that confuses the buffer. Not sure why or how but don't have the time to chase it down.

 

[NBT Teknoloji]

after the greeting is played to a customer if gives instructions before the record beep, I would like to go straight to the beep

Gary,

https://docs.fusionpbx.com/en/latest/applications/voicemail.html
variables can be set in advanced -> variables

from the link above, I made a Voicemail section
https://ss.nbt.host/20/4mawu13v2rea84c.png

and now after greeting the voicemail only beeps

 

[DigitalDaz]

Most people would not want these set server wide, indeed not even domain wide. The best place for these variables is actually in the dialplan that you are using it with.

 

[NBT Teknoloji]

Most people would not want these set server wide, indeed not even domain wide. The best place for these variables is actually in the dialplan that you are using it with.

Thanks for your answer,
I am very pleased that you can give an example, if it is possible specific for 101 extension in abc.com domain.

 

[DigitalDaz]

Well most people only usually want to skip when they are doing something custom, if you really do want it generically then leave it as you have it now.