Hello all. I have a problem with audio issues con my the company PBX system. I have a FreePBX 2.11.0.11 with Asterisk Ver. 11.4.0. I have a dedicated Dell Server for this. My setup is like this. I have a Quintun Tenor DX with 4 PRI T1’s that is connected directly to my FreePBX server. I have my own VLAN for the phone’s system and QoS on all the Cisco Switches on the company. We are not using SIP trunking. We have the Quintum configured like a SIP Trunk like this:
PEER Details:
username=xxxxxxxx
type=peer
secret=xxxxxxx
insecure=very
host=x.x.x.x
disallow=all
canreinvite=no
canredirect=no
allow=ulaw
qualify=yes
keepalive=45
We can make and receive calls all the time.
We have the issue that sometime we get words that repeats itself (Example: is everything Okay, Okay, Okay) Sometimes the call comes back and sometimes it stays on the loop.
The other thing is we get audio drops on calls for a couple of seconds, when this happens we are the party that does no here the other end. The other end always get the audio.
I have recorded some calls on the PBX itself. And the call are perfectly fine no Audio Drops and no echo or repetitions. So I assume that from the Quintum to my PBX is OK. and the problem is the audio from my PBX to my Phones.
I have various types of phone:Cisco SPA504G, CP-7971G, Polycom IP 331 IP, 601 , IP 335 with the later SIP firmware.
I have tested the latency from the FreePBX to some of the phones and I’m getting:
icmp_seq=3 ttl=64 time=0.383 ms
icmp_seq=11 ttl=64 time=0.373 ms
and another phones I get:
icmp_seq=3 ttl=64 time=0.405 ms
icmp_seq=11 ttl=64 time=0.515 ms
Intervals
So is less that 1ms.
Where do you think I should start Looking:
Heres My sip_general_additional.conf if needed
vmexten=*97
faxdetect=yes
context=from-sip-external
callerid=Unknown
notifyringing=yes
notifyhold=yes
tos_sip=cs3
tos_audio=ef
tos_video=af41
alwaysauthreject=yes
useragent=FPBX-2.11.0(11.4.0)
disallow=all
allow=ulaw
allow=alaw
allow=gsm
callevents=no
jbenable=yes
jbresyncthreshold=500
jbforce=yes
jbimpl=adaptive
jbmaxsize=300
jblog=yes
minexpiry=60
allowguest=yes
defaultexpiry=120
srvlookup=no
maxexpiry=3600
registerattempts=0
registertimeout=20
rtpkeepalive=0
g726nonstandard=no
videosupport=no
maxcallbitrate=384
canreinvite=no
rtptimeout=30
rtpholdtimeout=300
checkmwi=10
notifyhold=yes
notifyringing=yes
nat=no
externip=0.0.0.0
Any help or suggestion I will appreciate.
Thanks to all