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388 Views 1 Reply Latest reply: Apr 23, 2012 11:42 AM by Brian Jonson RSS

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Help Bad call qaulity on Cisco gateway Incoming

Apr 20, 2012 3:03 AM

Treschen1 58 posts since
Apr 15, 2010

Dear All please assist

 

I work for a VOIP provider.

 

We run asterisk, we have a customer who is has issues with incoming voice. For example when we call in to the customer he cannot hear us but i can hear him perfectly. Please assist me the customer conects to us via serial interface X.21 on a Cisco C2600 this is then linked to another cisco C2600 via ethernet which has a vwic card. See config below. From our cisco voip gateway the call goes to the customers avaya which then sends it to the avaya phones in the network

 

On Cisco VOIP Gateway

 

!

controller E1 1/0

framing NO-CRC4

clock source internal

pri-group timeslots 1-31

!

!

!

!

!

interface FastEthernet0/0 (THIS INTERFACE IS CONNECTED TO THE ROUTER WITH THE SERIAL X.21 CONNECTION)

ip address 10.50.0.2 255.255.255.252

duplex auto

speed auto

!

interface FastEthernet0/1

no ip address

shutdown

duplex auto

speed auto

!

interface Serial1/0:15

no ip address

encapsulation hdlc

isdn switch-type primary-net5

isdn protocol-emulate network

isdn incoming-voice voice

no cdp enable

!

ip route 0.0.0.0 0.0.0.0 10.50.0.1

!

!

ip http server

no ip http secure-server

 

!

!

control-plane

!

!

!

voice-port 1/0:15

input gain 12

output attenuation -6

no comfort-noise

!

!

!

!

!

dial-peer voice 1 pots

destination-pattern .T

direct-inward-dial

port 1/0:15

forward-digits all

!

dial-peer voice 2000 voip

description Local

max-conn 15

destination-pattern 0T

progress_ind setup enable 3

modem passthrough nse codec g711ulaw

session protocol sipv2

session target ipv4:10.1.1.3

dtmf-relay rtp-nte

codec g711ulaw

fax-relay ecm disable

fax rate disable

ip qos dscp cs5 media

ip qos dscp ef signaling

no vad

!

dial-peer voice 4602 voip

modem passthrough nse codec g711ulaw

session protocol sipv2

incoming called-number 4602

codec g711ulaw

fax-relay ecm disable

fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711ulaw

no vad

!

dial-peer voice 4600 voip

incoming called-number 4600

fax-relay ecm disable

fax rate disable

no vad

!

dial-peer voice 4601 voip

incoming called-number 4601

fax-relay ecm disable

fax rate disable

no vad

!

dial-peer voice 4603 voip

incoming called-number 4603

fax-relay ecm disable

fax rate disable

no vad

!

dial-peer voice 4604 voip

incoming called-number 4604

fax-relay ecm disable

fax rate disable

no vad

!

dial-peer voice 4605 voip

incoming called-number 4605

fax-relay ecm disable

fax rate disable

no vad

!

dial-peer voice 4606 voip

incoming called-number 4606

fax-relay ecm disable

fax rate disable

no vad

!

dial-peer voice 4607 voip

incoming called-number 4607

fax-relay ecm disable

fax rate disable

no vad

!

 

Asterisk config below:

 

[Customer]

accountcode=customer

type=friend

qualify=yes

insecure=port,invite

host=10.50.0.2

echotraining=yes

echocancel=yes

dtmfmode=rfc2833

disallow=all

t38pt_udptl=yes,none

context=customer

canreinvite=no

allow=alaw

allow=ulaw

allow=g729

rtptimeout=60

call-limit=15

deny=0.0.0.0/0.0.0.0

permit=10.50.0.2/255.255.255.255

  • Brian Jonson 82 posts since
    Jun 25, 2008
    Currently Being Moderated
    1. Apr 23, 2012 11:42 AM (in response to Treschen1)
    Re: Help Bad call qaulity on Cisco gateway Incoming

    If you can only hear one side then this is likely an SDP issue creating one way audio.  Using Wireshark, or a similar packet capture tool, check the SDP messages from your external interface of whatever your session border controller is.  Look for "sendonly" or "recvonly" within SDP.  Let me know if you see anything...we can go from there.

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