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external calls forwarded to unity

cbrooks
Level 1
Level 1

I have a test lab with unity 4, cm2.3 and a vg200 with 4 fxo ports.

When I call from outside, the call is transferred after no answer to unity, unity picks up and i can leave a message. However when i check the voicemail, all the voicemails are blank, there is no sound. Anyone have any clue why???

It works fine internally

1 Accepted Solution

Accepted Solutions

is the unity enabled to handle G729 calls? Your call to unity is trying to connect using G729. This is because the default codec on the gateway is G729. A quick test is to specific the G711 codec under the dial-peer and give it another try.

View solution in original post

10 Replies 10

oliviers
Cisco Employee
Cisco Employee

Sounds like a one-way audio problem with the GW connectivity to Unity. This is a common problem. Have you checked this out?

http://www.cisco.com/warp/public/788/AVVID/1way_audio.html

I tried that, but there is nothing blocking me, no firewall etc. Im an msfc hop away from the unity box. Good document though. Does cisco offer a class on how to search their website???

any chance we can have a look at the vg200's configuration? also, are you getting any error in the event log on unity when you're dialing from the gateway?

ip subnet-zero

ip tcp synwait-time 13

ip domain-name editedXXXXX

ip name-server EditedXXXX

!

call rsvp-sync

voice call send-alert

voice rtp send-recv

!

voice service pots

interface FastEthernet0/0

ip address 192.168.17.231 255.255.255.0

speed 100

full-duplex

h323-gateway voip bind srcaddr 192.168.17.231

!

ip classless

ip route 0.0.0.0 0.0.0.0 192.168.17.1

no ip http server

!

dialer-list 1 protocol ip permit

!

voice-port 1/0/0

timing hookflash-out 500

connection plar opx 5007

description 2190724

caller-id enable

!

voice-port 1/0/1

timing hookflash-out 500

connection plar opx 5005

description 5012412chris

caller-id enable

!

voice-port 1/1/0

timing hookflash-out 500

connection plar opx 5006

description 2191951-fite

caller-id enable

!

voice-port 1/1/1

connection plar opx 5008

description 5012439-jason

caller-id enable

!

dial-peer voice 2 pots

preference 2

incoming called-number .

destination-pattern 9T

no digit-strip

port 1/0/1

!

dial-peer voice 3 pots

preference 3

incoming called-number .

destination-pattern 9T

no digit-strip

port 1/1/0

!

dial-peer voice 4 pots

preference 10

incoming called-number .

destination-pattern 9T

no digit-strip

port 1/1/1

!

dial-peer voice 5 voip

destination-pattern 5005

session target ipv4:192.168.17.240

dtmf-relay h245-signal

!

dial-peer voice 10 voip

destination-pattern 5006

session target ipv4:192.168.17.240

dtmf-relay h245-alphanumeric

!

dial-peer voice 12 voip

destination-pattern 5008

session target ipv4:192.168.17.240

dtmf-relay h245-signal

Hi

We could actually start checking now the Cisco Callmanager traces to see if we are streaming correctly.

Eg (extract from a trace on my system)

Cisco CallManager|StationD: 0540e6d8 StartMediaTransmission conferenceID=0 passThruPartyID=17 remoteIpAddress=949c80a(10.200.73.9) remotePortNumber=19542 milliSecondPacketSize=20 compressType=4(Media_Payload_G711Ulaw64k) qualifierOut=?. myIP: fe48c80a (10.200.73.9)

And on the router you can check if the socket has been opened:

orbital#sh ip socket

Proto Remote Port Local Port In Out Stat TTY OutputIF

17 --listen-- 10.200.73.9 5060 0 0 89 0

17 --listen-- 10.200.73.9 67 0 0 89 0

If this is all the case we could take a sniffer trace of that conversation by spanning the ports. If you want to take a sniffer you can send it to uversave@cisco.com.

Ullie

Not sure what part of the trace you neeed, but let me know if this helps

000000822 StartMediaTransmission conferenceID=0 passThruPartyID=1001a91 remoteIpAddress=e711a8c0(192.168.17.231) remotePortNumber=17054 milliSecondPacketSize=20 compressType=11(Media_Payload_G729) qualifierOut=?. myIP: f111a8c0 (192.168.17.241)|<:STANDALONECLUSTER><:VOICESERVER><:1><:192.168.17.240><:PORT 2716="">

06/12/2003 15:55:56.652 CCM|Locations_adjustBandwidth -- cdccPID=(1.14.444) Orig=Dest=0 no entry for cdcc. No need to adjust.|<:STANDALONECLUSTER><:VOICESERVER>

06/12/2003 15:55:57.855 CCM|CMProcMon - ------Entered Router Verification|<:STANDALONECLUSTER><:VOICESERVER>

06/12/2003 15:55:57.855 CCM|CMProcMon - ----Exited Router Verification|<:STANDALONECLUSTER><:VOICESERVER>

06/12/2003 15:55:59.855 CCM|CMProcMon - ------Entered Router Verification|<:STANDALONECLUSTER><:VOICESERVER>

06/12/2003 15:55:59.855 CCM|CMProcMon - ----Exited Router Verification|<:STANDALONECLUSTER><:VOICESERVER>

06/12/2003 15:56:01.855 CCM|CMProcMon - ------Entered Router Verification|<:STANDALONECLUSTER><:VOICESERVER>

06/12/2003 15:56:01.855 CCM|CMProcMon - ----Exited Router Verification|<:STANDALONECLUSTER><:VOICESERVER>

06/12/2003 15:56:03.136 CCM|StationInit - InboundStim - KeepAliveMessage - Send KeepAlive to Device Controller. DeviceName=CiscoUM1-VI1, TCPHandle=000000822, IPAddr=192.168.17.241, Port=1933, Device Controller=[1,89,576]|<:STANDALONECLUSTER><:VOICESERVER><:1><:192.168.17.241><:CISCOUM1-VI1>

06/12/2003 15:56:03.152 CCM|StationInit - InboundStim - KeepAliveMessage - Send KeepAlive to Device Controller. DeviceName=CiscoUM1-VI3, TCPHandle=000000823, IPAddr=192.168.17.241, Port=1934, Device Controller=[1,89,577]|<:STANDALONECLUSTER><:VOICESERVER><:1><:192.168.17.241><:CISCOUM1-VI3>

06/12/2003 15:56:03.167 CCM|StationInit - InboundStim - KeepAliveMessage - Send KeepAlive to Dev

here is the show ip socket

Remote Port Local Port In Out Stat TTY OutputIF

17 --listen-- 192.168.17.231 5060 0 0 211 0

17 --listen-- 192.168.17.231 67 0 0 2211 0

17 192.168.17.241 22865 192.168.17.231 17723 0 0 21 0

forgot to say, unity and call manager, and exhcnage server show no errors in event log.

Also, if i try and listen to messages thru exchange, they are blank too.

is the unity enabled to handle G729 calls? Your call to unity is trying to connect using G729. This is because the default codec on the gateway is G729. A quick test is to specific the G711 codec under the dial-peer and give it another try.

thanks, that worked, i dont know when I changed that