What codec should be used to connect a * box to
a cisco router which has a t1 with 24 trunks coming in?
My router voip dial plan looks like this:
dial-peer voice 2 voip
destination-pattern [1,2,,3,5,8]..
session protocol sipv2
session target ipv4:10.x.x.x
dtmf-relay cisco-rtp
codec g711ulaw
no vad
!
The problem I have is when more than one call is on it,
sometimes the quality gets very bad.
If more than one access the conference room it starts to
blip real badly.
Thots, ideas greatly appreciated.
--
respectfully, Joseph
------=============
> What codec should be used to connect a * box to > a cisco router which has a t1 with 24 trunks coming in? > > My router voip dial plan looks like this: > > dial-peer voice 2 voip > destination-pattern [1,2,,3,5,8].. > session protocol sipv2 > session target ipv4:10.x.x.x > dtmf-relay cisco-rtp > codec g711ulaw > no vad > ! > > The problem I have is when more than one call is on it, > sometimes the quality gets very bad. > > If more than one access the conference room it starts to > blip real badly.Just a guess... ensure asterisk and the cisco (and anything else between the two) are running full-duplex ethernet. Doubtfull the problem is related to codecs, but then I don't have a cisco to test with either. Rich
I will double check. How much cpu does the MeetMe feature need per user? Or does it depend on how they connect? On Mon, 2004-05-03 at 11:54, James Sizemore wrote:> Check the duplex on your ethernet conection on both the Cisco and the > Asterisk box. Make sure neither are half duplex. > > Joseph wrote: > > >What codec should be used to connect a * box to > >a cisco router which has a t1 with 24 trunks coming in? > > > >My router voip dial plan looks like this: > > > >dial-peer voice 2 voip > > destination-pattern [1,2,,3,5,8].. > > session protocol sipv2 > > session target ipv4:10.x.x.x > > dtmf-relay cisco-rtp > > codec g711ulaw > > no vad > >! > > > >The problem I have is when more than one call is on it, > >sometimes the quality gets very bad. > > > >If more than one access the conference room it starts to > >blip real badly. > > > >Thots, ideas greatly appreciated. > > > > > > > _______________________________________________ > 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-- respectfully, Joseph --------------------
Check the duplex on your ethernet conection on both the Cisco and the Asterisk box. Make sure neither are half duplex. Joseph wrote:>What codec should be used to connect a * box to >a cisco router which has a t1 with 24 trunks coming in? > >My router voip dial plan looks like this: > >dial-peer voice 2 voip > destination-pattern [1,2,,3,5,8].. > session protocol sipv2 > session target ipv4:10.x.x.x > dtmf-relay cisco-rtp > codec g711ulaw > no vad >! > >The problem I have is when more than one call is on it, >sometimes the quality gets very bad. > >If more than one access the conference room it starts to >blip real badly. > >Thots, ideas greatly appreciated. > >