Hello, Thanks for your replies.
We checked our sip.conf and we have canreinvite=no already. I agree it could be a firmware issue. I will get another vendors phone hooked up to the pbx before going crazy with support. Thanks, Neal On Sun, Oct 12, 2008 at 6:14 AM, Vieri <[EMAIL PROTECTED]> wrote: > > --- On Sat, 10/11/08, Eric "ManxPower" Wieling <[EMAIL PROTECTED]> wrote: > > > Try setting canreinvite=no in each of the device sections on > > a couple of > > phones, reload chan_sip.so and see if that fixes things. > > It has fixed > > the issue when I've tried it. > > > > [EMAIL PROTECTED] wrote: > > > Hello, > > > > > > We are using asterisk 1.6, sangoma pri card, and Cisco > > 7960 phones. When we > > > make or receive calls there is a delay before voice is > > heard. Anyone have > > > any ideas on where to start to debug or has anyone > > seen this before. We > > > have played with settings on pri, asterisk, and phones > > with no change. > > I'm having the same problem but with ATA-connected analog phones. The ATAs > are Grandstream GXW4008 with firmware v. 1.0.1.15. The "canreinvite" > option in sip.conf doesn't change anything for me. Downgrading the GXW4008 > solves this issue so this is obviously a firmware bug in my case. I had a > vague report once of a user in another installation having this 1-second > delay on call connection. That user had a Cisco phone but I don't remember > which one. I suggest you check this with Cisco Support if you can. > > Vieri > > > > > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
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