Broadcast How to transfer

[john_usc]

Hi

I made a voice broadcast campaign and it plays a pre recorded message about vote and it gives different option. But when I press 1, 2 or 3 nothing happens and it keeps playing the message as if it didn't recognize my imput.
I have this in my extensions.conf

; VICIDIAL_auto_dialer transfer script SURVEY at beginning:
exten => 8366,1,Playback(sip-silence)
exten => 8366,2,AGI(agi://127.0.0.1:4577/call_log)
exten => 8366,3,AGI(agi-VDADtransferSURVEY.agi,${EXTEN})
exten => 8366,4,AGI(agi-VDADtransferSURVEY.agi,${EXTEN})
exten => 8366,5,AGI(agi-VDADtransferSURVEY.agi,${EXTEN})
exten => 8366,6,Hangup



how could I make it go to the agent if 1 2 or 3 is pressed.

Thanks

[williamconley]

I had the same problem. My provider stopped sending the DTMF tones. I changed providers to test and ... it just started working with an otherwise identical setup (all i changed was user/pwd/host).

If that fails, try checking your SIP-INVITE via sip debug from the cli.

When I say they stopped, I mean ... at 12:32 AM it was working for testing ... and at 9AM the next morning at launch, it stopped. After diagnosing for a while, I tried another provider ... and now they have a new customer. I moved the client's box to the other provider and never looked back.

[john_usc]

Thanks for your reply. I don't think its a problem with sending DTMF as I am using their service for pbxinaflash too and dtmf works just fine there. It must be some settings somewhere in Vicidial that I am missing.
Thanks

[gardo]

This is thread is related to this: http://www.eflo.net/VICIDIALforum/viewtopic.php?t=6992

[john_usc]

I read the thread @gardo.
It didn't have a conclusive solution. Have you ever fixed it. If yes, then how. Could you please share the solution.
Thanks

[gardo]

Not yet. I might try what Williaconley suggested - try another provider again and see how it works. I've tested with two providers and still the same results.

[john_usc]

ok I will keep trying this too. Hopefully some one will come to rescue mean while. If I come up with a solution I will post here, please do the same too.
Thanks

[williamconley]

have you verified that your sip-invite contains the correct dtmf setting?

[john_usc]

Where would I find sip-invite?
I don't see it in extensions.conf

thanks

[williamconley]

sip invite is what is sent to the provider, you need sip debug to view it as a rule it's not a "vicidal" function but a basic asterisk function.

the sip.conf should be controlling its content, but in the end, the sip invite is the packet that is actually sent to the provider, if it has the appropriate information in it, the call works correctly (unless there's something wrong at the provider end).

so being able to view your sip-invite and other sip packets is essential to being able to fully understand any issues between you and the provider.

if you type "sip debug" (and later "sip no debug") at the asterisk cli you can see the sip information or you can set asterisk up to log it. try to do it in a control scenario (no other calls). even if you log it, the log files will be fairly large. and don't forget to turn it back off.

also works for iax2 debug / iax2 no debug and agi debug / agi no debug

[tsvision]

I have had similar problems with Vitelity and I switched provider thinking it was the VOIP provider. Actually, setting DTMF=auto in sip.conf file fixes the problem in most cases.

[williamconley]

Unfortunately, in all the cases i was involved with the dtmfmode was properly set, but the provider was failing to provide DTMF at all. Losing a few customers seems to resolve the issue in most cases, but not all. Most have begun to work again.

I'm not sure if there was a change in a "standard" for an interface somewhere or what, but it was definitely annoying for a while.