11-09-2010 08:27 AM - edited 03-16-2019 01:49 AM
Hi,
I'm having problems with a voicemail.
Scenario:
CME configure with a Unity Express for voice mail. When i call from the PSTN the Auto-Attendat picks up the call,
the call is transfer to an extension. I let the extension ring until forward the call to the voicemail.
Once i hear the message "leave your message for xxxx", i leave the message but the mailbox doesn't receive any message. The phone doesn't turn on
the light indicating a new message has arrived. I have configure MWI on the CME. I also have configure the voice mail in the unity:
CME:
===
ephone-dn 61
number A61.... no-reg primary
description FOR MWI-ON
mwi on
ephone-dn 62
number A62.... no-reg primary
description FOR MWI-OFF
mwi off
dial-peer voice 5000 voip
description *** FOR VOICE MAIL ****
destination-pattern 5000
session protocol sipv2
session target ipv4:10.234.144.254
dtmf-relay sip-notify rtp-nte
codec g711ulaw
no vad
UNITY VOICEMAIL CONFIG:
====================
voicemail mailbox owner "Rigsanchez" size 5400
description "Rigoberto Sanchez mailbox"
end mailbox
Other thing i noticed is that when i'm leaving the message and press #, for more option, the system tells me that the message is too short.
Can somebody help me?
Thansk for your response
Solved! Go to Solution.
11-09-2010 01:05 PM
Can u try adding this while trying g729r8 (default) codec ?
sip-ua
g729-annexb override <--- add this (hidden so pl. just type)
11-09-2010 08:43 AM
Is this the same problem you described in another thread? If so, please do not open duplicates.
11-09-2010 08:47 AM
Hi,
Not this issue is different. In the other case when a forwarded the call from the extension to the voicemail, the call didn't connect.
Now i have connection to the voicemail. But when i leave the message it doesn't leave anything.
I solved the other problem forcing the codec in the incoming dial-peer. I was using a codec list.
thanks..
11-09-2010 09:01 AM
Does it work when calling from a local ip phone ?
11-09-2010 09:05 AM
Hi,
Not. It doesn't work if i call from an internal phone.
When i'm leaving the message the system tells me that the message is too short, when i selec the # key for more options. If i close after leaving the message and i check, there is nothing in the mailbox.
I configure the message size to 120 secs the default was 240. but it doesn't work with either of them.
thanks...
11-09-2010 09:10 AM
Is this calling AA, or the phone with VM directly ?
11-09-2010 09:41 AM
Hi,
I'am calling from the extension to the VM...
When i make an internal call test.
11-09-2010 09:23 AM
Its probably one way audio issue CME-->CUE
So nothing gets recorded.
When the call to CUE/VM is up, can u capture:
sh sip call
sh call act voice brief (3 times)
sh voip rtp connect
Also make sure you do not have ip verify unicast reverse-path CLI
on any of the interface...there is a known issue with that.
11-09-2010 09:40 AM
11-09-2010 10:05 AM
Can u share ur config so that I can make some sense out of
these IP ?
I also see that in the capture there is only one g711 call and
the Tx/Rx count is very low...
It also appears that this was for a call from external and not
from a local IP phone?
Is transcoding in use as there are lots of calls with g729br8
codec; g729br8 has builtin-vad and CUE requires no vad on
the sip dialpeer.
If u can, capture a sniffer off the CUE interface. I can send u
the procedure for using internal packet capture on IOS.
11-09-2010 10:16 AM
11-09-2010 10:39 AM
Hello,
The captures you gave are for a call over SIP trunk with xcoding invoked
for g711u<->g729br8
To simplify, can u recreate the issue with on a local IP phone (on vlan 20) and capture those?
Wondering why u r using g729br8 (dialpeer 1100) instead of g729r8 ?
DK
11-09-2010 10:58 AM
11-09-2010 11:14 AM
Thanks...this looks good...I see Tx counts going up so packets
are being sent to CUE.
Let's see if we can see the same in the sniffer....u can use following
procedure to do a packet capture on the CUE interface.......
First configure following (config t mode)....
ip traffic-export profile DKS mode capture
bidirectional
length 512
interface ISM0/0
ip traffic-export apply DKS size 2000000
Then from enable mode........
pueblo_central#trafffic-export interface ISM0/0 clear
pueblo_central#trafffic-export interface ISM0/0 start
---place your test call----
pueblo_central#trafffic-export interface ISM0/0 stop
pueblo_central#trafffic-export interface ISM0/0 copy tftp://
pueblo_central#trafffic-export interface ISM0/0 clear
Upload CUE.cap
11-09-2010 12:45 PM
Hi Dilip,
I changed the condec on dial-peer 1100 and it is working. I put the g711ulaw codec.
How can i make it work using the g729r8 codec? Because i tried using g729r8 but it didn't work either.
Thanks.
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