Transfering call to IVR

[mohankumar]

Hi All,
I am trying to transfer my call to an IVR number, i am using DTMF option in VICIDIAL login page, but the IVR is not accepting when i dial a 10 digit number using DTMF setting.

I have also integrated audio code (VOIP) gateway instead of using soft phone,

It would be greatful if some one could help me so that i can dial the digits using HARDPHONE or any other option to dial in IVR without using the DTMF setting in Vicidial Page. Thank you in advance

Regards
Mohan

[devafree]

Hello
I had been facing a similar trouble. The thing is if I used to seperate the DTMF digits with commas (for 1 sec gap) sometimes the IVR would go thru but it was still unreliable)
However, it was resolved by this ways. I used x-lite soft phone and made the transfer-conference by dialling out from line 2 on the x-lite , it used another sip termination service provider on ulaw with rfc2833 for transmitting, then the telephone number could be entered every time from the softphone successfully.
As we had to close the transmission of DTMF with a hash (#), we had to avoid the tTo flags in extensions. Later on I had used SVN asterisk , where the transfer was made into ## instead of single # in features.conf and we could continue to define flags in the extensions.conf.

The thread I had posted was

http://www.eflo.net/VICIDIALforum/viewt ... ight=#8850

As you would see, Matt Florell advised to try and used app_conference rather than meetme, but I chosen to try tackle the root of the trouble (SIP+g729+dtmf) which sorted it out.

Hope this helps,

devafree

[mohankumar]

Hello
I had been facing a similar trouble. The thing is if I used to seperate the DTMF digits with commas (for 1 sec gap) sometimes the IVR would go thru but it was still unreliable)
However, it was resolved by this ways. I used x-lite soft phone and made the transfer-conference by dialling out from line 2 on the x-lite , it used another sip termination service provider on ulaw with rfc2833 for transmitting, then the telephone number could be entered every time from the softphone successfully.
As we had to close the transmission of DTMF with a hash (#), we had to avoid the tTo flags in extensions. Later on I had used SVN asterisk , where the transfer was made into ## instead of single # in features.conf and we could continue to define flags in the extensions.conf.

The thread I had posted was

http://www.eflo.net/VICIDIALforum/viewt ... ight=#8850

As you would see, Matt Florell advised to try and used app_conference rather than meetme, but I chosen to try tackle the root of the trouble (SIP+g729+dtmf) which sorted it out.

Hope this helps,

devafree

Thanks for the response,

Will G729 codec solves the problem?

what I mean is by installing G729 with the previous set up (meetme)

Thanks and regards,

[mflorell]

Just an update on this issue, there is a patch you can apply to asterisk to get DTMF passthru to work better in Asterisk 1.2.X

See this thread for more info:
http://www.eflo.net/VICIDIALforum/viewtopic.php?p=13287