Hi,
We have freepbx hosted by one of the freepbx vendors.
We have a problem and am not sure if its a pbx issue or phone issue.
We are on FreePBX 2.8.1.0 version and use Cisco SPA525g and Cisco SPA504G phones.
What happens is that when we dial a conference bridge we also need to dial a code after it connects. A lot of time the code we dial is not accepted. eg. if the code is 2821009, it says we dialed 288211009 and therefore cannot connect to the conference call.
We tried with different setting in “DTMF Tx Method:” on our phone ext. eg. “AVT”, “INFO”, “INBAND”, “INFO+INBAND”, etc… nothing works. There is something or the other which is not compatible.
I will appreciate any help. Please let me know if you have any question or need more information.
Avt=rfc1483 just match the extension and the phone.
What Asterisk version, what type of trunks are you using?
Thanks For your ersponse:
We have free pbx FreePBX 2.8.1.0, Asterisk version is 1.6.2
On our ext. we have dtmfmode=rfc2833
On our phone here is the configuration.
Audio Configuration
Preferred Codec: G711u
Use Pref Codec Only: no
Second Preferred Codec: Unspecified
Third Preferred Codec: Unspecified
G729a Enable: Yes
G722 Enable: Yes
G726-32 Enable: Yes
Release Unused Codec: Yes
DTMF Process AVT: Yes
Silence Supp Enable: no
DTMF Tx Method: InBand+INFO
DTMF Tx Volume for AVT Packet: 0
Use Remote Pref Codec: no
Codec Negotiation: Default
So are you saying to change DTMF Tx Volume for AVT Packet = rfc1483 on the phone
and dtmfmode=rfc1483 on the ext.
Do we need to change DTMF Tx Method also?, currently set as InBand+INFO.
Appreciate your help.
Sorry missed to mention that we are using HostPBX SIP trunking.
Levels don’t matter with data.
I had my fingers on wrong keys. It’s RFC2833 and AVT are same thing.
You should have your extension set to rfc2833 and the phone set to AVT.
Thanks for a prompt response. I had that set up but we still have that issue.