having problems with the "press 1" outbound campai

[ycfinancial]

we are having trouble with the press 1 xfers. they are able to here the message but when they press 1 to speak to a live agent, nothing happens. even if pressing all the numbers. we have it configured so when they hear that message and press 1, it will get xfer to an extension. what am i missing?

campaign/details/survey

Survey DTMF Digits: 1
Survey Method:agent_xfer
Survey Survey Xfer Extension: 8366

[mflorell]

admin.php version and build?

carrier type?

codecs used?

loadavg of the server when running a campaign?

[ycfinancial]

VERSION: 2.0.5-173
BUILD: 90320-0424

[fibacom]
disallow=all
allow=ulaw
type=peer
host=67.215.238.146
dtmfmode=rfc2833


exten => _1NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _1NXXNXXXXXX,2,Dial(${FIBACOM}/${EXTEN},,tTor)
exten => _1NXXNXXXXXX,3,Hangup

where can i find the rest?

[mflorell]

you can get the loadavg by typing 'w' on the Linux CLI

have you tried inband for dtmf?

[rishuk]

ello,

I have setup a broadcasting camp[aign with press on option.

When i press 1 it says thank you for your responce, i will tansfer to you to our call center, your call is very important to us.

But call never comes to agent.


here is portion of extensions.conf

exten => 8366,1,AGI(agi://127.0.0.1:4577/call_log)
exten => 8366,2,AGI(agi-VDADtransferSURVEY.agi,${EXTEN})
exten => 8366,3,AGI(agi-VDADtransferSURVEY.agi,${EXTEN})
exten => 8366,4,AGI(agi-VDADtransferSURVEY.agi,${EXTEN})
exten => 8366,5,Hangup

[gardo]

Based from experience this is usually more of a DTMF issue with your carrier. It doesn't mean that if it works with a regular IVR (pressing "1" to get to an option) DTMF should work with the broadcast/survey campaign. You can also try switching codecs first. If it doesn't work use a different carrier. I tried more than 3 carriers before getting the one that works with broadcast/survey campaign.

[rishuk]

I have changed DTMF to inband but still same

[mflorell]

Might want to try upgrading to Asterisk 1.4.21.2. Asteirsk 1.4 has a lot of SIP improvements.

[rishuk]

Might want to try upgrading to Asterisk 1.4.21.2. Asteirsk 1.4 has a lot of SIP improvements.

i haved upgrade to asterisk 1.4.21.2 but still same.

can anyone explain me procedure,how to do this.i might be doing something wrong.

i have not found this in manager manual

[mflorell]

The manager manual does not cover issues with Asterisk or SIP related DTMF issues. It only covers ViciDial operations.

[rishuk]

i am not complating about manager manual.

I am not sure this issue is with DTMF. as when i press 1 i hear next message that says that "we are transfering your call to our call center".it means asterisk accepting this DTMF.

Its just a thought, i mnot sure.

Is there anyone explain me?