Hello All -
I've got an asterisk setup using 3 broadvoice lines and 5 Polycom IP300 phones. We have 1.5Mbit up and down via cable. 40ms (ave) pings to the broadvoice proxy and no packetloss.
The phones sound like cell phones. The person on the other end complains about it cutting in and out. On our end, it cuts in and out as well.
Within the office, we can call from one IP300 to another with absolutely no problems at all. Sounds great.
We are connected through a Linksys (Firmware v1.05.0). Wired QoS is enabled with the asterisk box's mac being highest priority, and everything else being low. Upstream bandwidth is set to Auto. [I doubt these settings are the problem as the choppy/cell-phone-sounding effect also occurs when there is minimal network traffic.]
Any help troubleshooting this plz?
Snippets from sip.conf:
[210]
username=210
type=friend
secret=*******
record_out=On-Demand
record_in=On-Demand
qualify=no
port=5060
nat=never
[EMAIL PROTECTED]
host=dynamic
dtmfmode=rfc2833
context=from-internal-bbp
canreinvite=no
callerid="Mike" <210>
username=210
type=friend
secret=*******
record_out=On-Demand
record_in=On-Demand
qualify=no
port=5060
nat=never
[EMAIL PROTECTED]
host=dynamic
dtmfmode=rfc2833
context=from-internal-bbp
canreinvite=no
callerid="Mike" <210>
[bbpbv1]
username=949743####
user=phone
type=peer
secret=**********
nat=yes
insecure=very
host=sip.broadvoice.com
fromuser=949743####
fromdomain= sip.broadvoice.com
dtmfmode=inband
context=from-bbp-pstn
canreinvite=no
authname=949743####
username=949743####
user=phone
type=peer
secret=**********
nat=yes
insecure=very
host=sip.broadvoice.com
fromuser=949743####
fromdomain= sip.broadvoice.com
dtmfmode=inband
context=from-bbp-pstn
canreinvite=no
authname=949743####
[949743####]
username=949743####
user=949743####
type=user
nat=yes
insecure=very
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
dtmfmode=inband
dtmf=inband
context=from-bbp-pstn
username=949743####
user=949743####
type=user
nat=yes
insecure=very
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
dtmfmode=inband
dtmf=inband
context=from-bbp-pstn
Test call:
# asterisk -vvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvrx "sip show channels"
Peer User/ANR Call ID Seq (Tx/Rx) Format
147.135.8.128 1949##### 5855b260439 00103/00000 ulaw
192.168.1.100 210 f8c9ee5e-9f 00101/00002 ulaw
2 active SIP channel(s)
Peer User/ANR Call ID Seq (Tx/Rx) Format
147.135.8.128 1949##### 5855b260439 00103/00000 ulaw
192.168.1.100 210 f8c9ee5e-9f 00101/00002 ulaw
2 active SIP channel(s)
_______________________________________________ --Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
