An equally important suggestion would be a firmware check, as versions prior to 5.5.8 had two big problems for VoIP: latency issues on 10mhz channels, and incorrect WMM queuing for VoIP DSCP 46-48 (would end up in BE queue).
On May 20, 2015 8:29:16 PM AKDT, Faisal Imtiaz <[email protected]> wrote: >Ahhh, makes sense, (my apologies, I missed that part on the original >email) > > >Is the internet connection between your server and their Side over >fixed Wireless ? > >If yes, > >Is your Wireless network using Ubiquiti (AP's ) ? > >If yes, use ssh to log into your AP's, and issue dmesg command ... > >Do you see any stuck beacon errors ? > >If yes...... that is your problem for random voice cut out... > >If the answer is no to any or all of the above, then please ignore this >Trouble shooting suggestion :) > > >Faisal Imtiaz >Snappy Internet & Telecom >7266 SW 48 Street >Miami, FL 33155 >Tel: 305 663 5518 x 232 > >Help-desk: (305)663-5518 Option 2 or Email: [email protected] > >----- Original Message ----- >> From: "Nate Burke" <[email protected]> >> To: [email protected] >> Sent: Thursday, May 21, 2015 12:12:23 AM >> Subject: Re: [AFMUG] Voip analog handoff to PBX >> >> Call waiting is disabled. The customer has an existing PBX, and >didn't >> want to upgrade. It only takes POTS in, so have to convert my SIP to >> POTS for their PBX. >> >> Nate >> >> On 5/20/2015 5:56 PM, Faisal Imtiaz wrote: >> > Is it possible that what you are hearing is a beep when another >person is >> > calling that line (Call waiting ?) >> > >> > Also, If you don't mind me asking.. >> > Why are you doing pots to sip conversion ? why not just a >straight sip >> > trunk ? (i.e. sip origination ) >> > >> > Faisal Imtiaz >> > Snappy Internet & Telecom >> > 7266 SW 48 Street >> > Miami, FL 33155 >> > Tel: 305 663 5518 x 232 >> > >> > Help-desk: (305)663-5518 Option 2 or Email: >[email protected] >> > >> > ----- Original Message ----- >> >> From: "Nate Burke" <[email protected]> >> >> To: "Animal Farm" <[email protected]> >> >> Sent: Wednesday, May 20, 2015 5:24:44 PM >> >> Subject: [AFMUG] Voip analog handoff to PBX >> >> >> >> I'm handing off 4 Analog lines to a PBX via a Grandstream GXW4004, >and >> >> it's having random cutouts in the audio. The packet stream shows >that >> >> it's having DTMF Events that are causing the cutouts, DTMF Tone >'D', so >> >> I"m pretty sure that it's not actually caused by keypresses. I'm >> >> guessing something between the ATA and the PBX there's an >incompatibility. >> >> >> >> Is there another box quality wise to do 4 Analog lines before >jumping >> >> all the way into the Adtran TA904? >> >> >> >> Maybe something in like the $300 area? >> >> >> >> Nate >> >> >> >> >> >> -- Sent from my Android device with K-9 Mail. Please excuse my brevity.
