No sound when making calls

:confused: Hello everyone,

I’m new on this forum and I really tried to look for a solution to the problem I have at hand but didn’t find something that could help me. I used to have a virtualized Asterisk box and was having loads of problems with it, it basically froze every now and then, so I migrated the Asterix box to a dedicated server. I used CentOS on both boxes and the same Asterisk version and I even copied the configuration from one box to the other using the command. I shutdown the old virtualized box and put the other one into action with the same IP so I wouldn’t have to change anything on the phones. All the extensions are working well except for those which have Cisco routers at the end, everything was working perfectly before I did the change but with the new box it doesn’t.

I use 2611 cisco routers to connect to remote sites using pots at the other end (The Cisco router has FXS ports so it converts the voice from digital to analog to send it through this ports). At the beginning I thought this was a codec problem so I’ve been trying to change codecs on the Cisco routers but it hasn’t worked.

I’m really lost here, I don’t know where the problem is. I attach the configurations of the extensions and the cisco dial-peer configuration to see if it clarifies what I’ve sadi.

/etc/asterisk/sip.conf

;UIO 1
[1123701]
type=friend
insecure=very
host=10.250.4.224
permit=10.250.4.224
disallow=all
canreinvite=no
allow=g729
allow=ulaw
allow=alaw
context=uio1
qualify=3000
dtmfmode=rfc2833
;UIO 2
[1123702]
type=friend
insecure=very
host=10.250.4.224
permit=10.250.4.224
disallow=all
canreinvite=no
allow=g729
allow=ulaw
allow=alaw
context=uio2
qualify=3000
dtmfmode=rfc2833
;Ambato
[1123706]
username=1123706
secret=1123706
host=dynamic
nat=yes
type=friend
qualify=yes
insecure=port,invite
;context=merus-sipphone
context=ambato

/etc/asterisk/extensions.conf
[uio1]
exten => 1123706,1,Dial(SIP/${EXTEN},1123706) ; to Ambato
[uio2]
exten => 1123706,1,Dial(SIP/${EXTEN},1123706) ; to Ambato
[ambato]
exten => 1123701,1,Dial(SIP/1123701/${EXTEN},1123701) ; to uio 1
exten => 1123702,1,Dial(SIP/1123702/${EXTEN},1123702) ; to uio 2

CISCO

voice-port 2/0
!
voice-port 2/1
!
!
mgcp profile default
!
dial-peer cor custom
!
!
!
dial-peer voice 1123706 voip
description to Ambato
destination-pattern 1123706
session protocol sipv2
session target ipv4:192.168.100.221:5060
session transport udp
codec g711alaw
no vad
!
dial-peer voice 1123701 pots
destination-pattern 1123701
port 2/0
!
dial-peer voice 1123702 pots
destination-pattern 1123702
port 2/1
!

Everyone claims to be a newbie and to have searched the web. To be useful you need to say what lines of enquiry you followed.

You have two peers with the same host address and no fromuser overrides; that really isn’t helpful.

You have one option value (“very”) which is no longer supported and one option name that is deprecated (“canreinvite”), but you haven’t identified the version of Asterisk. Most people do not need anything looser than “insecure=invite”.

Why are you setting the call setup timeout to the numerical value of the called extension number?