12-17-2004 08:03 AM - edited 03-13-2019 07:23 AM
Hi,
I´m currently setup a Cisco 2821 as ISDN gw for for using with asterisk and some 7960 phones. Now , after some trial and error finding the right command syntax for configuring the voice services(Cisco why not release some docs about the current IOS release 12.3T(11) if you change half of the voice commands from the last rel.?), i got it up and working. Now I can dial in, using fax-relay, dial out but it seems that VAD is still used on outgoing connections even if i had disabled everythere an can(2821, asterisk, 7960). Also a "show call active voice" shows that VAD is disabled but it is still active as i can hear this silly chopping sound and, if there is silence, the RX-counter from the 7960 does almost stop. But this is only on outgoing connections. On incomming, it seems to be disabled.
I also take a look at the SIP communications between asterisk and the 2821 but, even there, i cannot find any packet where SilenceSupp is anything else than off. As there is a switch between * and the router, i cannot check the communication from the IP phones to the 2821 after the connection is established.
Does somebody have an idea what´s going on?
Regards
Michael.
[Interresting sections of running config]
!
voice-card 0
dspfarm
!
!
!
voice service voip
sip
!
interface GigabitEthernet0/0
description $ETH-LAN$$ETH-SW-LAUNCH$$INTF-INFO-GE 0/0$
ip address 192.168.2.3 255.255.255.0
ip access-group 100 in
duplex auto
speed auto
no mop enabled
!!
interface BRI0/0/0
no ip address
isdn switch-type basic-net3
isdn overlap-receiving
isdn point-to-point-setup
isdn incoming-voice voice
isdn sending-complete
isdn static-tei 0
!
!
voice-port 0/0/0
no vad
compand-type a-law
cptone DE
timeouts interdigit 3
threshold noise -88
!
voice-port 0/0/1
no vad
compand-type a-law
cptone DE
timeouts interdigit 3
threshold noise -88
!
dial-peer voice 100 pots
description Incoming SESSION
application session
incoming called-number 123450
no digit-strip
direct-inward-dial
forward-digits all
no register e164
!
dial-peer voice 101 pots
description Incoming SESSION
application session
incoming called-number 12349..
no digit-strip
direct-inward-dial
forward-digits all
no register e164
!
!
dial-peer voice 2 voip
destination-pattern 123450
progress_ind setup enable 3
progress_ind progress enable 8
session protocol sipv2
session target ipv4:192.168.2.2
session transport udp
dtmf-relay rtp-nte
playout-delay minimum low
codec g711alaw
no vad
!
!
dial-peer voice 1000 pots
description Outgoing voip service
destination-pattern 0.
clid network-number 0
clid second-number strip
port 0/0/0
!
sip-ua
retry invite 3
retry response 3
retry bye 3
retry cancel 3
timers trying 1000
sip-server ipv4:192.168.2.2
!
12-17-2004 09:16 AM
Did you bounce the voice port when you made the change? If not, try bouncing it as the no vad statement won't take effect until you do that.
12-18-2004 08:15 PM
What do you mean with "bouncing"?
After a bit playing around i found that one cmd really shows me that VAD is used but a "show call active voice" still say is it not:
gw-0#show voice call summary
PORT CODEC VAD VTSP STATE VPM STATE
============== ======== === ==================== ======================
0/0/0.1 g711alaw y S_CONNECT S_TSP_CONNECT
0/0/0.2 - - -
12-17-2004 10:20 AM
I had this same problem when switching from G.711 to G.729. Voice quality went downhill to a horrible level. It turns out that G.729B will always use vad. By switching to G.729A this problem went away. It took me weeks to find how to do this though, even with the TAC's help. Simple procedure to change this in your callmanager config is below.
Please enable below parameter in Callmanager Service parameters.
Strip G.729 Annex B (Silence Suppression) from Capabilities*
Set it to TRUE.
12-20-2004 04:32 PM
it maybe you are using the default incoming ip peer which specifies vad and any codec for ip legs coming into the router. sho call active voice or show call history voice last 2 should show an incoming IP call leg with a pid of 0 if this is the case (incoming in the sense that is coming into the router from asterisk, from the users perspective this would be an offnet outbound call)
you may want to read:
http://www.cisco.com/en/US/tech/tk652/tk90/technologies_tech_note09186a008010fed1.shtml
try adding the following to set options on incoming IP legs:
dial-peer voice 9999 voip
incoming called .
session protocol sipv2
session transport udp
dtmf-relay rtp-nte
playout-delay minimum low
codec g711alaw
no vad
12-21-2004 12:00 AM
I got it. I forgot that there is also a dial-peer 0 even if i can not see it. But why is it not configurable? If if matches everything else not covered by some other dial-peers it would be like to be a default setting and as to be a default it should be configurable. In other words, why the default has VAD enabled but is not changable or even visible? I think that some devs at cisco should think about this again.
Anyway, thank you for your help.
Regards
Michael.
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