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O.K. I figured out that the SIP trace seems to indicate that the caller hung up before the call was actually answered because if I call and hang up I get the same type of result in the sip trace. I wonder why it keeps rimging though and why does the caller hangup in the first place? I guess I...

Greetings I have an occasional problem with my pbxiaf system and Anveo Direct SIP trunks. A call rings in and nobody is there, if they call right back they get through with no problem. I believe they just hear dead air on their end. A SIP trace shows the invite from Anveo then I believe coming...

Thanks w1ve !!

Thank you Hyksos They are at the same location behind the same router. I am having a problem where incoming calls to hunt group drops when answered (sip trace shows cancel from trunk side or pbx side) and wonder if this is related. I do not have these ports open in IPtables on the pbx side, I...

I have some phones that are remote to my PBXIAF server. One of the phones shows registered to 5060 but the rest are in the 61000 range. Can someone please explain this? Is it some type of masquerading or caused by the remote router?

Looking into my audio issue by researching on the web I kept seeing references to VoIP Optimized 1000 hz timer Linux Kernels being used for VoIP on AWS. Does the PIAF AMI use an optimized kernel? I have a working knowledge of Linux but this is above my head. I did uname command and wrote the...

Thanks for the info Ward. That kinda stinks, I though by paying more money for a small I would get better performance. If you have a system with only a few concurrent calls maybe sticking to the micro is a better idea.

This is a weird problem. I decided to take my PIAF on AWS (Wards AMI) from T1 Micro to M1 Small and after doing so I started noticing playback quality issues with system and voicemail promps, I launched new AMI's to test both the 1.8 AMI and the Ver 11 as micro's and small's and again have no...

Three of my Anveo trunks say unreachable also. Is it suppose to be like that?

I thought if there was no speech present with voice activated detection or silence suppression there would be no waveform and therefore no bandwidth used. Usually in a call one person talks while the other listens so VAD would cut the bandwidth in half right there. If there is no sound at all no...

Is VAD/silence suppression available in pbx in a flash/Asterisk? I have enabled it in my Cisco phone and added" silencesuppression=on" in asterisk sip settings but when I do sip trace it still says silencesupp=off. Im trying to find ways to reduce bandwidth, I wanted to switch to g729 but then...

Its loaded on my PBXIAF green system under the options tab and listed under third party add-ons in the module page. Whats it for? Thanks for your help

I have this third party add on called Customer DB installed and I cannot find any documentation on it and cant figure out what its for. Anybody familiar with this?

Thanks arad !

Is it possible to run multiple instances of PBXIAF as Virtual Box machines on the same server? I guess ports would be an issue, for example I do not think you can share port 5060 on the same PC. I need one instance to host trunks and 3 others to connect to the trunk pool but be independent of...

Thanks billsimon ! The second link unlock display captcha did it. I believe it didn't work for me the first time because of my pop up blocker.

Is anyone using Google Voice from an EC2 instance from Amazon? Google blocks my sign in because it looks like suspicious activity according to the email alert I received. My trunk will not connect.

I guess I will try t*f*t*p.

Im not sure how to get the presence patch into the directory. I do cd/usr/src/ wget https://issues.asterisk.org/jira/secure/attachment/46880/gareth-11.3.0.patch and it fails. Im sure Im not going about this the right way. Could you explain how to get the patch into the directory?