[cisco-voip] Problem with E1 Calls

Kevin Thorngren kthorngr at cisco.com
Wed Mar 8 20:54:31 EST 2006


The first place to look is to make sure that you don't have any IP 
connectivity issues during the one way audio.  Although your diagram 
shows a pretty simple network but the phones may be on a different 
subnet from the GW.  You can hit the "i" or "?" button twice quickly 
while on the call.  This will show you the call stats along the RxCnt.  
If the RxCnt is increasing then the RTP stream is getting to the phone 
but there is apparently blank audio if you are not hearing anything.  
If the counter is not increasing then the problem is some sort of IP 
connectivity issue between the GW and the Phone.

You can use the "show voice dsp" command to see if there are any DSPs 
that are having transmit or receive issues.  Just look in the 
TX/RX-PAK-CNT columns.

It doesn't look like dial peers 2,3,4 or 102 are needed.  There are no 
voice ports assigned to 2,3 or 4 and no session target assigned to 102.

Kevin
On Mar 8, 2006, at 6:37 PM, new look wrote:

> Dear All,
> I have a performance problem in my IP Phone system with the E1 calls 
> (Some times I can't hear the person I called him), my system already 
> configured before. (My network Diagram is attached)
> So first can any one help me to know if the existing configuration has 
> any wrong or unneeded commands? (The configuration of the VGW is 
> attached)
> Note: My IP Phone DN#'s are 11XX
>  
>
> 1-I want to know if the interface Serial0/0/0:15 configurations have 
> any wrong commands or unneeded commands?
> 2-I want to know if I need dial-peers 2, 3 and 4, or not?
> 3-I want to know if I need dial-peers 102, why?
> 4-Is the Codec g711ulaw is right codec for E1 calls or I need to use 
> another Codec? which one?
> 5-is any one has the URL for Cisco example for this solution ?
>
> Yahoo! Mail
>  Bring photos to life! New PhotoMail makes sharing a breeze.isdn 
> switch-type primary-net5
> !
> voice-card 0
>  no dspfarm
> !
> voice translation-rule 1
>  rule 1 /^84/ /11/
> !
> voice translation-profile incom
>  translate called 1
> !
> controller E1 0/0/0
>  framing NO-CRC4
>  pri-group timeslots 1-31
> !
> translation-rule 1
> !
> !
> interface Serial0/0/0:15
>  no ip address
>  isdn switch-type primary-net5
>  isdn overlap-receiving
>  isdn not-end-to-end 64
>  isdn incoming-voice voice
>  isdn guard-timer 20000 on-expiry accept
>  isdn map address * plan unknown type unknown
>  isdn map address .* plan unknown type unknown
>  isdn map address transparent
>  isdn T309-enable
>  isdn T306 400000
>  isdn sending-complete
>  isdn incoming alerting add-PI
>  no isdn gtd
>  no cdp enable
> !
> voice-port 0/0/0:15
> !
> dial-peer voice 1 pots
>  translation-profile incoming incom
>  destination-pattern .T
>  direct-inward-dial
>  port 0/0/0:15
> !
> dial-peer voice 2 pots
>  preference 1
>  destination-pattern .T
> !
> dial-peer voice 3 pots
>  preference 2
>  destination-pattern .T
> !
> dial-peer voice 4 pots
>  preference 3
>  destination-pattern .T
> !
> dial-peer voice 100 voip
>  destination-pattern 1...
>  session target ipv4:172.16.80.24
>  dtmf-relay h245-alphanumeric
>  codec g711alaw
> !
> dial-peer voice 102 voip
>  dtmf-relay h245-alphanumeric
>  codec g711ulaw<Problem.ppt>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: not available
Type: text/enriched
Size: 4070 bytes
Desc: not available
Url : https://puck.nether.net/pipermail/cisco-voip/attachments/20060308/62a84f2a/attachment-0001.bin 


More information about the cisco-voip mailing list