Audio goes blank for a few seconds during the conversation, and then comes back on

#1

audio goes blank for a few seconds during the
conversation, and then comes back on in asterisk 1.4 with trunk as PRI line’

with the sangoma PRI card.

/etc/dahdi/system.conf

#Sangoma A101 port 1 [slot:13 bus:3 span:1]
span=1,1,0,ccs,hdb3
bchan=1-15,17-31
echocanceller=mg2,1-15,17-31
hardhdlc=16

and the chan_dahdi.conf is below

[trunkgroups]

[channels]
context=default
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
relaxdtmf=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1
immediate=no

;Sangoma A101 port 1 [slot:13 bus:3 span:1]
switchtype=euroisdn
context=from-pstn
group=0
echocancel=yes
signalling=pri_cpe
channel =>1-15,17-31

#2

In my experience, usually it’s only for a fraction of a second, although you’d hear it repeatedly over the course of only a few seconds, but it sounds kind of like what happens when the connection between your SIP client and the SIP server is losing packets.

This could be some kind of firewall/network issue.

#3

in case you want to monitor the rtp traffic for debuging purposes, use this command " rtp set debug on" . I dont know your network structure , but as you are using dahdi for outbound calls instead of SIP this could be more related to the Telcom company or card issue than a network issue.

If the SIP clients are losing connection with the server this cause call drop issues and you will see on the logs unreachable messages or lagged all depend on the network issue