CME and DTMF?

[saxon747]

Hi All,

I'm having some issues trying to get CME to send DTMF over a dialpeer voip using SIP to a VOIP telco ... have tried all versions of dtmf-relay but no joy ... anyone seen this before?

Tnxs


Config of dialpeer:

!
dial-peer voice 1 voip
destination-pattern .T
session protocol sipv2
session target ipv4:213.168.225.133
session transport udp
incoming called-number .
dtmf-relay rtp-nte
codec g711ulaw
no vad
!

 

[mishkanyc]

the command you are looking for is
dtmf-relay sip-notify

 

[saxon747]

I've tried all dtmf-relay options, should have made that clearer on my post ... none work ... the provider has set theirs to rtp-nte, which I have done ... we tried the sip-notify, it didn't work either.

 

[mishkanyc]

strange by it self the g711 codec and no vad remove any call quality (voice issues) that would give any headaches towards dtmf relay in the first place.. annd sip relay is a standard way to integrate with any other sip proxy... what kind of equipment do they have on the other side?

 

[saxon747]

hi, my end is a cisco 2611xm running cme and the other side is a asterisk i think ... not sure, it's a telco voip provider ... the provider are new and not done this on cisco before ... there setup i believe is asterisk and the phones are running sip images ... were i am running cme with sccp and the sip dialpeer to them ... new at this and sort of learning as i go ... just hit a wall on this one ... the provider has there asterisk and phones working with the rtp-nte standard ... they tried using sip and so did i, but no joy ... anyway, just decided to hit the forum to see if someone else hit this ... tnxs

 

[mishkanyc]

just out of dumn curiousity, which version of IOS are you running?
and you do have dsps?

 

[saxon747]

hi, running c2600-ipvoice-mz.123-14.T3.bin , tried it on 2 other versions, same issue ... no i don't have dsp's , could this be the problem? ... i thought u only need dsp's if u are going to compress or use codecs to go at g729 , excuse my ignore if i am wrong.

 

[mishkanyc]

I think you need to have them to support any kind of voice encoding including the voice cards such as FXO/PRI and others
but theoretically they shouldnt affect sip since its a signalling protocol


you can run a debug and see the messages... also a TAC case with cisco?

 

[saxon747]

hi ok, will look into it more ... i assumed because calls were voip they wouldn't need dsp's ... thought dsp's where used to convert analog signals to ip , perhaps i am wrong ... yes, i needed them before when i had a pra e1 connection ... but this time i am trying to use a broadband connection over ip using sip as that is all my telco ip provider will support ... will look further on cco , i don't have cisco tac support ... tnx;s