The initial seconds in voicemail are no recorded

[RBL]

Hi everybody,

I am having a problem with the voicemail this problem start around a month go. The initial seconds of a voicemail are not recorded, you can call to extension or receive incoming call to extension when voicemail pickup the call you ear the greeting perfectly and the beep and after that you start talking for recording but the firs 6-8 second are not record, any one is having this problem or had this problem before

Thanks.

 

[aitp/nadmin]

Hi RBL,

I'm super green here, but I too noticed this issue once. I never figured out what the root cause was, but I had this issue when i was first testing fusionpbx on centos 7. When I used the debian script on 9.9, i didn't have any more trouble so I ignored the issue. Now wishing that I had done some basic troubleshooting.

I also noticed that I didnt hear the "silence" loop on the endpoint leaving the message until the recording started actually "happening". Does this make sense? It was like FS was hung up or waiting for something else before it played the silence loop and began recording...

I'm sure someone will have something more helpful or insightful. I just thought I'd leave my experience with this issue here.

Respectfully,

/nadmin

 

[RBL]

Thanks nadmin,

I have 3 system and only in one is happening this phenomenon, this system have like 1 year running fine and now is when start happening.

Regards,

Roberto

 

[cp6183]

Did anyone have a resolution for this dilemma?

 

[cp6183]

Solved: port number for each number was going to 5080 and I changed the port number for the number to 5060.

 

[aitp/nadmin]

@cp6183 , Super cool that you got that fixed up! Could you possibly elaborate a little tiny bit more for me so I can understand? I have a pea brain and not much coffee yet this morning.

 

[cp6183]

@aitp/nadmin

So routing for DIDs were to port 5080, because the SIP trunk end point is configured to IP address:5080 so each DID was pointing to port 5080. I changed the routing of each number to SIP url:5060.

 

[DigitalDaz]

The port should not matter at all, if that has fixed it it is pure luck or a setting in the other profile.

 

[cp6183]

@DigitalDaz

I am a VoIP carrier. This configuration wasn't changed in my hosted platform in fusion. This was changed in my SBC.