att flex cut over getting no caller ID and major line noise incoming call

[Song]

Hi,

Anyone know experience this before with ATT flex.? I know it is not ceritified with 3cx as a siptrunk vendor. But i seen people set this up without any issues. Now the pointing begin?? ATT is saying that the 3cx is not configure correctly and or a network issues on our end. I say BS to that because the 3cx system is working fine with siptrunk.com . We are not even getting caller ID fixed. This is something on ATT end and not a 3cx end. But the line quality is so bad!! i wish the client have never switched to it. BAD ideas!!

Any codec setting or changes I can try to fix this? going to have a call meeting with ATT tech and see what we can do..
please advise.. the only setting i see is the inbound advance option on the sip trunk setting. everything else is normal.

 

[Matt Scanzano]

Verify if there's loss or timeouts between the PBX and the SIP trunk provider for one. That could definitely cause issues. Is the trunk provided over the internet directly or did AT&T provide a router with a dedicated port for VoIP signalling so that you can route locally?

 

[Song]

Verify if there's loss or timeouts between the PBX and the SIP trunk provider for one. That could definitely cause issues. Is the trunk provided over the internet directly or did AT&T provide a router with a dedicated port for VoIP signalling so that you can route locally?

ATT ip flex have a dedicated cisco device in house connected directly to our switch. and from 3cx we point to their IP address locally. So this should have no issues.. I just got another call saying when someone called in and the operator pick up. the end user hear a dead silence. Maybe its a code issues? I was reading online ATT ip flex needed to be set codec G729 first. I looked at the ATT setting on 3cx and inbound advance setting the codec was using G711 U law first. so i changed it. Hopefully this will fix it..

Anything else I need to be looking for? maybe I need to ask them to unblock SIP (port 5060) and turn off SIP ALG on their cisco device?

 

[nub]

I would check the activity log on Verbose and make some inbound test calls from a known number and see if you can find the caller id, it maybe just malformed. if you do find it in the activity log, you will have to work with the sip trunks > sip trunk in question > inbound parameters tab, and test away I like to start with all default values and then change as needed for testing.

 

[Matt Scanzano]

ATT ip flex have a dedicated cisco device in house connected directly to our switch. and from 3cx we point to their IP address locally. So this should have no issues.. I just got another call saying when someone called in and the operator pick up. the end user hear a dead silence. Maybe its a code issues? I was reading online ATT ip flex needed to be set codec G729 first. I looked at the ATT setting on 3cx and inbound advance setting the codec was using G711 U law first. so i changed it. Hopefully this will fix it..

Anything else I need to be looking for? maybe I need to ask them to unblock SIP (port 5060) and turn off SIP ALG on their cisco device?

I would try capturing one of these calls with wireshark or even direct from the server to see if RTP is being transmitted both ways. Are your users able to hear the caller?

 

[leejor]

The noise could be a Codec issue, but not simply the setting on the trunk, but what your sets are using. Ideally, you don't want 3CX doing transcoding (converting from one Codec to another) which will show in the Activity Logs. If the sets are using G711 U (first choice), then , for the best quality, the trunks should be using the same.

Is the noise on both incoming and outgoing calls?