autoattendent is not able to recognise digits when we press them.
sometime it will but most of the time its not.
if it recognises then it will dial but both the sides will not hear anything.
thx for ur reply.
First do you have the correct information on your dial-peer for the sip connection to CUE. Do you have the configuration you can post?
Sounds like you have a couple issues. Pressing digits is probably something to do with dtmf-relay. Make sure that's enabled on your dial-peers.
For your audio problem, your router config would be helpful.
ip unnumbered GigabitEthernet0/0.3
service-module ip address 220.127.116.11 255.255.255.0
service-module ip default-gateway 18.104.22.168
ip routeoute 10.1.1.0 255.255.255.252 10.1.1.2
!ip route 22.214.171.124 255.255.255.255 Service-Engine1/0
ip route 192.168.5.0 255.255.255.0 10.1.1.2
this is router config..i hope this is the one ur asking for.
dial-peer voice 555 voip
session protocol sipv2
session target ipv4:126.96.36.199
! codec g711ulaw
this is for dial-peer.
Is this just for auto attendant (i.e. do digits work fine for voicemail)? Is the CallManager Express? If so, are the phones registered to the same router that contains the CUE? For the DTMF problem, you could probably do a 'debug ccsip message' on that router and see if it even looks like the router is sending the digits to CUE. Otherwise, if this is urgent, you may want to open up a case with TAC.
ya its just for autoattendant voicemail works fine. and this is CME.
phones are registered to same router.
router is sending digits to router as voicemail is fine and during prompt its recognising .
the matter is only when i use autoattendant script its not able to recognise digits .
it will just go off only couples of number its able to read.
"suppose i dial number 777 then its saying dialing 777 and call will go it will ring on other side also but then u wont be able to listen".
thx for ur reply.
do you see
voice service voip
allow-connections h323 to sip
allow-connections sip to h323
in your config.
Try that... Otherwise you may have a bug in your code. Check bug tracker...
this is wht m getting after
sh ccsip messages
rest of the things are there.
the other file with name transfer can forward call to some of the extensions but the calling party cannot hear even dialing or any voice.
the cue version is 2.3.2
thanks for your support.
were you able to solve this issue. I am having the same problem of phones having no audio with CUE AA. Often times, it would ring and drop the call. It appears to be a problem with with codec, because when I did debug on ccsip messages, it showed no codec negotiated. I am sure the dial-peers are all good.
yaa i am able to solve this..
one thing is that always set the codec to g711 for autoattendant.
also have you defined the codec in the dial-peer??