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- May 1, 2018
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I have a new 3CX deployment that is Pro Licensed. (Getting ready to migrate from old PBX in office).
It is an on premise deployment with the topology looking like this:
SIPtrunk.com---Cable Modem----ASA 5506x----Ubiquiti Gear----Dell PowerEdge (Debian VM 3CX)
Cable Modem is supplying two IP's to the ASA. The 1st IP is for general traffic. 2nd IP is NAT'd to the 3CX.
The 3CX is sitting on the 40VLAN in the office, as well as all our office deskphones. Employee computers sit on 20VLAN, with a rule in the ASA to allow 20 to 40 traffic both ways.
Inner office communication (Ext to Ext) and the 3cx deployment all function great!
Before migrating to SIPtrunk.com and porting all our phone numbers, I asked for a demo line to make sure 3CX is setup and ready to work as intended.
With that out of the way, here is the reason for my post:
After setting up SIPtrunk.com and making a test call (inbound and outbound), the audio was "choppy". Where it sounds like someone is toggling the mute button every 1-2 seconds. For Example, if I say aloud my ABC's, it will sound like this "ABC...GHI...MNO...STU...YZ." And it is from either direction.
Inspect SIP is turned off in the ASA. Made no difference.
I toggled "Audio from PBX". Made no difference.
Bumped up the Codec priority to G722 & 729 and rebooted the deskphone. Made no difference.
I am not an ASA genius and I have other guys familar with Cisco, helping with configuration of the ASA. Together, we reviewed the config with SIPtrunk.com Engineers, working of a Cisco Template, and everything looked great.
Some more intense troubleshooting that I did today:
It is an on premise deployment with the topology looking like this:
SIPtrunk.com---Cable Modem----ASA 5506x----Ubiquiti Gear----Dell PowerEdge (Debian VM 3CX)
Cable Modem is supplying two IP's to the ASA. The 1st IP is for general traffic. 2nd IP is NAT'd to the 3CX.
The 3CX is sitting on the 40VLAN in the office, as well as all our office deskphones. Employee computers sit on 20VLAN, with a rule in the ASA to allow 20 to 40 traffic both ways.
Inner office communication (Ext to Ext) and the 3cx deployment all function great!
Before migrating to SIPtrunk.com and porting all our phone numbers, I asked for a demo line to make sure 3CX is setup and ready to work as intended.
With that out of the way, here is the reason for my post:
After setting up SIPtrunk.com and making a test call (inbound and outbound), the audio was "choppy". Where it sounds like someone is toggling the mute button every 1-2 seconds. For Example, if I say aloud my ABC's, it will sound like this "ABC...GHI...MNO...STU...YZ." And it is from either direction.
Inspect SIP is turned off in the ASA. Made no difference.
I toggled "Audio from PBX". Made no difference.
Bumped up the Codec priority to G722 & 729 and rebooted the deskphone. Made no difference.
I am not an ASA genius and I have other guys familar with Cisco, helping with configuration of the ASA. Together, we reviewed the config with SIPtrunk.com Engineers, working of a Cisco Template, and everything looked great.
Some more intense troubleshooting that I did today:
- Fired up a demo TWILIO eleastic SIP, adding it to 3CX next to SIPtrunk.com, to test a different SIP provider. TWILIO produces the same results of Choppy audio. [Makes it sound like its not any particular SIP provider]
- Using my Android phone 3cx app on LTE (not on WIFI), A call to an extension inside my office, produces the same results of Choppy Audio. [Confirms #1 above, by not being the SIP provider. Also points to a Cisco issue]
- Using the ANYconnect VPN app on the android phone above connected to the office VPN, I tested again...NO CHOPPY AUDIO! But, other person calling had a really bad echo.
- Calls from behind the ASA (inside the office), work GREAT! And that is across VLAN's. Where the Deskphones lie on the 40 VLAN and the employee computers and WIFI are on the 30VLAN. WIth a rule allowing 30 to see 40, and vis-a-versa. [Because of #2, #3, and #4, it really sounds like something in the ASA. All tests above, point to the ASA].