On Tue, 2005-03-22 at 17:58 -0800, Scott Wolfe wrote: > Hi there, > I have suceseffuly installed Asterisk with a TDM22B and two sip phones > (Polycom300 and Xlite). My provider is Broadvoice and I am having a heck of > a time with the audio. All of my calls are broken up and sometimes really > tough to hear. > > My CLI> 'sip show peers' has me usually at OK(128ms) connection with > Broadvoice. > > I do have the same issue with my Analog phone plugged into my TDM22B. > > Any suggestions on where to look to debug this would be great. > > This is an install from CVS-HEAD on 3-21-2005. I'll post my sip.conf just in > cast anyone sees anything odd in it.
1) Try with CVS-STABLE 2) Try calling from the sip phone to asterisk itself (ie, no internet path or PSTN) 3) Try calling from an FXS port, and to an FXO port, and between FXS/FXO and FXO/SIP and FXS/SIP and even SIP/SIP 4) What codecs are you using, I found that forcing the polycoms to use ALAW, and my PRI was ALAW also, then it improved the audio quality. Though this was with a very small number of calls, on a pretty beefy Intel PIV based server.... I never realised ALAW<->ULAW would make so much of a difference. If none of that helps, then let us all know the outcome of each test, and someone else will likely offer some additional advice... Regards, Adam -- -- Adam Goryachev Website Managers Ph: +61 2 8304 0000 [EMAIL PROTECTED] Fax: +61 2 9345 4396 www.websitemanagers.com.au _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
