Anyone else have problems with phones like SPA-922, SPA-921, etc? Inbound audio is perfect but the remote end reports audio quality issues on the audio the handset is sending out. It's not the network.... I've tried asterisk 1.2, 1.4. I've used ulaw, G726, G793 & G729. Ulaw seems to be the least problematic but its still an issue. Doesn't matter if we send the calls to the PSTN via IAX or SIP or PRI. I don't know it if happens all the time but about 40% of the time the remote caller reports they cannot hear me.
On Tue, 2008-01-08 at 17:26 -0500, Andrew Joakimsen wrote:> Anyone else have problems with phones like SPA-922, SPA-921, etc?If I remember correctly, the SPA-9XX phones default to sending packets every 30ms intead of every 20ms. Log in as Admin, click on the Advanced link, and go to the SIP tab. You'll find a setting labeled "RTP Packet Size". Change it from "0.030" to "0.020" and see if that makes your audio quality better. It's done wonders for me in the past. -- Jared Smith Community Relations Manager Digium, Inc.
Can you describe the issue more please? Can the remote person not hear you at all? Or is there distorted/broken voice? Cheers, Daniel Cole -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Andrew Joakimsen Sent: Wednesday, 9 January 2008 9:26 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Linksys SPA-9xx Audio Issues Anyone else have problems with phones like SPA-922, SPA-921, etc? Inbound audio is perfect but the remote end reports audio quality issues on the audio the handset is sending out. It's not the network.... I've tried asterisk 1.2, 1.4. I've used ulaw, G726, G793 & G729. Ulaw seems to be the least problematic but its still an issue. Doesn't matter if we send the calls to the PSTN via IAX or SIP or PRI. I don't know it if happens all the time but about 40% of the time the remote caller reports they cannot hear me. _______________________________________________ -- 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