UC560 AA not working

[czampelas]

Hello,

I have setup a UC560 with AA. When I dial the AA number from the internal extensions everything works. When a call comes from the SIP the AA answers the call but it cannot recognize any options that I send through the phone.

For example I have "0" to transfer the call to the operator and "1" to dial by extension but when I press the button on the phone nothing works. It just keep playing the welcome message for 3 times and then it transferred the call to the number I have under "No Options Transfer To:".

Any suggestions will be very helpful.

Regards,

CZ

[linuxchild]

Hello

this sems to be a DTMF issue , can you give the exact call flow , you say from sip , is it your profider or an internal sip phone ??

Regrads

[czampelas]

Hello and thanks for your response.

I have checked the DTMF issue and I have added under dial-peer of the pilot number the config:

voice-class sip dtmf-relay force rtp-nte

Do I need anything else?

[Christopher Graham]

It's not enough to just change the dtmf method under the voip dial peer to AA.  You need to make sure that whichever dtmf method you are using on CME also matches CUE.  What dtmf method do you have configured under ccn subsystem sip on cue? 

For simplicity, assuming you are using rtp-nte for dtmf to the telco, just configure 'dtmf-relay rtp-nte' on the AA pilot dial peer (no need to use the 'voice-class sip dtmf-relay force rtp-nte), and configure 'dtmf-relay rtp-nte under 'ccn subsystem sip' in cue.

[czampelas]

I have checked all the dtmf configuration and is correct. However when I run the command

show voip rtp connection i get the following:

VoIP RTP active connections :

No. CallId     dstCallId  LocalRTP RmtRTP     LocalIP                                RemoteIP                              

1     49         50         18274    27896    192.168.0.230                          xxx.xxx.xxx.xxx                        

2     50         49         17236    16898    xxx.xxx.xxx.xxx                            xxx.xxx.xxx.xxx                           

Found 2 active RTP connections

What I cannot figure out is the IP 192.168.0.230. This is my IP towards my LAN. I do not use the UC as my router but only as my Call Manager. My SIP Trunk connection is directly connected on the WAN port of the UC. So any traffic towards my SIP provider should go through the IP that I have configured on my WAN port and not through the IP that I connect the UC on my LAN.

I have checked and is not routing issue.