CISCO 2811, FXO call connection issue

[oschenk112]

Hi All,

Everyone keeps talking about "call disconnection" issues, but I have a bit of a call connection issue.

I have a CISCO 2811 with a 4 port FXO card and this is connected to a Windows server running FreeSWITCH PBX. I wrote a program which triggers FreeSWITCH to make an outgoing call to my desk phone. My phone rings, but as soon as I pick up the CISCO does not "see" that I picked up. Here is the strange thing:

as soon as I make a loud noise (whisle or snap my fingers) into the receiver it suddenly connects and the FreeSWITCH IVR starts playing.

In other words the "200 OK" sip message is not sent to FreeSWITCH until it "hears" a noise coming through the line...

My question is ... what the hell? lol

My voice port is configured as shown:

voice-port 0/3/2

supervisory disconnect dualtone pre-connect

supervisory answer dualtone

no battery-reversal

output attenuation -3

cptone AU

timeouts call-disconnect 5

timeouts wait-release 5

impedance complex1

I tried battery reversal, but that doesn't work. What happens if I use "battery-reversal answer" is is that the CISCO sends a "200 OK" message right away and does not even wait for me to pick up my ringing phone.

Please see link below where I was trying to solve the problem.

The first post in the thread of conversations:

http://lists.freeswitch.org/pipermail/freeswitch-users/2012-January/078941.html

Thank you,

Oliver

[oschenk112]

I don't know if this is relevant but here are some more settings in my CISCO 2811:

voice service pots

fax rate disable

!

voice service voip

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip

no supplementary-service h450.2

no supplementary-service h450.3

supplementary-service h450.12

no supplementary-service sip moved-temporarily

no supplementary-service sip refer

fax protocol cisco

sip

  registrar server expires max 3600 min 3600

  no update-callerid

  no call service stop

voice register global

mode cme

source-address 192.168.x.1 port 5060

max-dn 10

max-pool 10

authenticate register

hold-alert

tftp-path flash:

voice-card 0

dsp services dspfarm

sip-ua

retry invite 3

retry response 3

retry bye 3

retry cancel 3

timers trying 1000

sip-server ipv4:192.168.x.50

presence enable

[oschenk112]

I can try to play with it, but why should it matter if someone skeaks or not?? That's what I don't get...

The problem is that because it's an automated IVR outgoing call, if someone sees the number on their phone/mobile they will just listen and not say anything so there may not be any user voice and the call won't get answered. I mean I can tell them: "If you see this number, make sure you say hi to the nice text-to-speech lady, otherwise she won't talk."  Not exactly a neat solution...

Is this by design or why does there need to be audio down the line before the CISCO tells FreeSWITCH that the phone has been picked up?

I've just read things about grounding the chassis properly and polariy all those sorts of things, so I'll go to site tomorrow and have a look, but I don't know if that's related to this problem.

[Gajanan Pande]

lol Oliver.

Besides tweaking output attenuation on port, did you try tweaking " input gain " values ? Increasing it might result in amplification of user voice & serve the purpose ( which is done by your whistle now ).

Lemme know how it goes.

GP.

[paolo bevilacqua]

That is normal.

Using FXO, the only mean to know if a phone is being answered or not, is by tone monitoring, because telco does not sent signalling information on analog lines. Better said, they could using "battery-reversal", but few if any telco of the world actually support that.

So, the only way to detect if a phone has been answered or not, isusing some sort tone detection, that apparently the router is doing. In addition, the router also need to detect if a max or modem has answered the call, to switch to appropriate fax or modem relay modes.

You can try disabling fax relay, but the only reliable way to have answer confirmation, is using ISDN or SIP trunks, not FXO.