Hope you're doing alright.
I was wondering if you could shed some light on this issue since I've doublechecked my config and found nothing unusual so far.
My scenario contains a callmanager express and cisco phones (3905 and 7841), I managed to make 'em register and calls between extensions and to/from the PSTN are working fine.
The thing that I just can't get working is the dtmf generation or sending, you see, when a call between two extensions is made, there's good quality 2 way audio, but when one party presses a key in order to send a dtmf tone, the other party is not able to listen to it, this happens in every direction.
When dialing to the pstn, calls sound just fine, but the phone in the pstn doesn't listen to the tones the local phone sends.
I've already tried kpml, rfc 2833, and sip Notice (and every possible combination between my extensions) as possible values for the dtmf-relay setting in all my voice register pool configuration figures.
Same result every time, when I call ticketmaster or some line with an IVR, of course I'm not able to navigate trough the menus and my call gets ultimately torn down.
I felt free to upload my configuration in case one of you more experienced guys could take a look at it and suggest a change in order to make it work or at least contribute to narrow down the problem.
Any comment and support are priceless to me since I've been dealing with this for several days and still looking for the solution.
I thank you in advance for your valuable support.
Thank you Suresh!
I applied as adviced, each command at the time in all of my dial-peers (voice-class sip dtmf-relay force rtp-nte & dtmf-relay rtp-nte), and all my voice register pools (dtmf-relay rtp-nte only).
Unfortunately no dtmfs were recieved douring the calls, I'm attaching an output from a test call between 2 extensions, once the call is stablished, both extensions send dtmf (at least I think they try since I push a number button and the tone is heard in the originating phone) but the other party is not able to hear the tone sent.
The attached file contains outputs of both debug voip rtp session named-event and debug ccsip message.(first the debug commands are entered and then the test call takes place, once the call ends, the debugs are deactivated).
Do you think of any othe configuration move that should be tried for this scenario?
I once again thank you for your comments!
Thank you Suresh and Ayodeji,
Right now I've tested 3509 sip phones with releases 9.2 and 9.4 (different devices, 'cause the phone running 9.2 was not able to pick up the 9.4 image loaded to the spiad) and so far tones have not been succesfully detected by any IVR.
The only scenario where the dtmf config was succesful was using a 7861 sip phone with dtmf-relay sip-notify as the relay method, meanwhile the dial-peer config used sip info as dtmf configuration:
dial-peer voice 1028 voip
translation-profile outgoing PSTN_Outgoing
session protocol sipv2
session target sip-server
ip qos dscp cs5 media
ip qos dscp cs4 signaling
This way, the phone can dial trough an IVR but only if the head set is being used, no tones are recognized if the speakers are used during the calls (really weird, right?).
Regarding exploring with sccp, is there much of a change in configuration? is there a sugested method of obtaining and configuring the propper software image for the 3905 phones?, if this change, will they interwork with the 7861 (sip) or theese must change protocol as well?,
I thank you once again for your comments!
Everything in the logs look okay however there is a bug with the firmware you are running on the 3905..You may wan to open a TAC case for this