I am having quality problems on SIP bound calls made over the Zap channels.
All Sip only calls (Cisco phone through Asterisk to another Sip device sound
fine).
Our setup looks like this:
User --> Executone PBX --> Asterisk Server --> Router --> Internet
The user is using a legacy handset that works with the Executone PBX and
accesses the server using a button that calls up the trunk group connecting
the legacy PBX to the Asterisk server.
According to our Executone technician, the T1 card in the legacy PBX does
*not* provide timing so I have the Digium T1 card as the primary sync source
as follows:
---------------------
/etc/zaptel.conf
---------------------
span=1,1,0,d4,ami
e&m=1-24
Our /etc/asterisk/zapata.conf file looks like this:
---------------------
/etc/asterisk/zapata.conf
---------------------
[trunkgroups]
[channels]
musiconhold=default
busydetect=1
busycount=7
relaxdtmf=yes
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
echocancel=yes
echocancelwhenbridged=yes
callgroup=1
pickupgroup=1
rxgain=-5
txgain=3
immediate=yes
signalling=em_w
context=zap-incoming
group = 1
channel => 1-24
I get static/popping/clicking on all calls made over the incoming Zap
channels (that is, the calls originating from the legacy handsets that trunk
into the Asterisk server.)
I know very little about the Executone PBX we integrate into, it is under
support and these types of questions go unanswered with them as it is out of
the scope of the support contract.
On the Executone console I can see that I am getting several Blue Alarms.
There doesn't seem to be any issues on the Asterisk side, no Red/Yellow
alarms.
Are there any timing settings I can tweak that will improve the call
quality? Anyone familiar with the Executone hardware?
Any help at all would be appreciated.
Thanks,
Geoff
