I am trying to connect Asterisk and Siemens Mod 80 using ISDN PRI ( Sangoma A101 Card) , here is my /etc/zaptel.conf
Autogenerated by /usr/local/sbin/sangoma/setup-sangoma
– do not hand edit
Zaptel Channels Configurations (zaptel.conf)
loadzone=us
defaultzone=us
#Sangoma A102 port 2 [slot:2 bus:10 span:1]
span=1,0,0,esf,b8zs
bchan=1-23
hardhdlc=24
alaw=1-23
Zapata.conf
;autogenerated by /usr/local/sbin/config-zaptel do not hand edit
;Zaptel Channels Configurations (zapata.conf)
;
;For detailed zapata options, view /etc/asterisk/zapata.conf.orig
[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 A102 port 2 [slot:2 bus:10 span:1]
switchtype=national
context=from-pstn
group=0
signalling=pri_cpe
channel =>1-23
alaw = 1-23
Now the problem I am having is the calls get through fine from Siemens to Asterisk but no audio, Changed echocancellation in /etc/wanpipe/wanpipe1.conf and /etc/asterisk/zapata.conf but no luck
SIP Device --> Asterisk --> ISDN PRI–> Mod 80 --> Phone Works fine.
but
SIP Device <-- Asterisk <-- ISDN PRI <-- Mod 80 <-- Phone no audio , thought this might be an issue with SIP Phone but no can make IAX2/SIP Calls fine to PSTN using SIP Trunk/IAX2 Trunk. No Nating involved in this scenario. Any suggesation or thoughts are highly appreciated.
Parshuram