Noise on internal SIP calls and incoming, popping crackling

New install of trixbox 2.8 CE. This is occuring in both vm and hardware installs of the pbx. If I call ext 111 from ext 122 over SIP even the voicemail message is crackling. If I call in from a POTS line through the DID I also hear crackling on a default IVR message I’ve set. Calling outbound to a real number I hear the ringing crackling. This is via a number of SIP softphones, on iphone and android phone.

I’d like to blame the vm server but it happens on physical too. I can’t blame softphone since it happens on POTS phone calling into the pbx and listening to the voicemail or announcement message.

I’m stumped.

This is the FreePBX support forum. Suggest you post in the trixbox Forum for assistance.

posted to

Given you’re an expert on there also would ya mind replying if you have some input? Thanks much!

When the trixbox folks decided to fork FreePBX they closed the doors for their users in these forums.

I help people in the trixbox forum, as the fork is not the users faults. I support the FreePBX project and believe that anyone serious about IP telephony should learn how to install Asterisk and FreePBX from scratch and just say no to the distributions.

I am sure I will give you some direction if you post in the trixbox forums.

I’ve installed Asterisk Now 1.71 on the physical hardware. Call between two extensions over SIP on LAN. The ringing is clear no crackles in it, but while listening to the voicemail message from ext 211 i hear pops and crackles.

Hardware is p4 3.2ghz 768mb ram. realtek NIC.

With trixbox I had put the box on a new physical network/new router and the noise was still present. I can try this with AsteriskNow but?..

The only reason I went with trixbox was some reported success with installing within VMware ESX server. I read that the kernel was already complied with some necessary tidbits. (maybe it wasn’t the kernel but maybe some package portions are preexisting)

This pbx install was to be a couple hour affair, it’s turned into three days. I originally had issues with SIP resulting in ghost calls and constant busy signals. I switched to IAX2 and things worked. Well save for the crackling. So now I’m attempting Asterisk Now.

noise includes a bit of clicking too.

I notice the console is reporting the following when I make a call.

Use of uninitialized value in concatentation (.) of string at /var/www/html/panel/ line 3372

Use of uninitialized value in concatentation (m//) of string at /var/www/html/panel/ line 3374

Also… when ext 211 calls 212 it rings but there is dead air where the VM message is supposed to be… created 213 same issue… but when either 212/213 calls 211 the voicemail message plays in its full crackly glory.

Figured I’d mention this as a. its obviously a problem i’d have to fix. but b. perhaps it might give a clue.

To note this vm not being heard was not occuring on (trixbox, i said this in a hushed tone)

thanks for your help.

Those errors are from the FOP panel. Nothing to do with your issue.

Setting up a PBX for the 1st time is not a 3 hour job.

So…Do you want to troubleshoot an Asterisknow system or trixbox. I can’t possibly keep track of two discussions.

Somebody may help with Asterisknow here, but it is usually related to FreePBX issues. The problem you are having is clearly some type of Asterisk issue.

Fixed. User error.
Somewhat of a perfect storm. Most of my voice qc testing was done via an iPhone application called Adore SIP Client, the Android App played through my Dell Streak speakers so it sounded like sh!t no matter what. Adore SIP Client is total garbage. The extension to extension crackling, popping, clicking was this application. I switched to a different application(Netdial) and no playback or audio issues.
Now the reason there was similar noises during calls via the inbound DID from a POTS line was due to bandwidth. I had reserved 25k but it appears that isn’t sufficient. I provided 65k and there was no crackling/popping etc on the announcement messages on all three installations. Nor when I dial through to the voicemail prompt.
So basically my hairline thinned further due to my failure to utilize a reputable SIP client, and allocate adequate bandwidth.
Thanks for everyones time.