@telemaco069 wrote:
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=45We 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=0.0.0.0Any help or suggestion I will appreciate.
Thanks to all
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Participants: 2