Page Comments since: 2008-12-30 10:00 http://www.voip-info.org Copyright 2005-2008 Arte Marketing Inc. I Installed asterisk 1.4.22 and lateset Dahdi an Slackware 12.2 machine . But when i hear the messages from asterisk , for example voicemail , the quality of sound is very poor . Why do I have this quality of sound ? Thank you in advance . <br>spooky (Nektarios Sourligas) at 2008-12-31 09:47 GMT http://www.voip-info.org/boards/view/board/3?view_comment_id=61661#comment_61661 Wed, 31 Dec 2008 09:47:41 GMT We have new Linksys Pap2 NA, linksys spa2002, Linksys WRT54GP2A-AT Wireless-G Broadband Router with 2 Phone Ports, Linksys WRT54GP2 （PAP2+Wireless Router） and RT31P2 (PAP2+Router). If u are interested, please contact our sales by MSN or by email, we also have ebay store for sale. MSN: sales@ez2telecom.com Email:sales@ez2telecom.com <br>ez2telecom2005 () at 2008-12-31 07:56 GMT http://www.voip-info.org/boards/view/board/7?view_comment_id=61660#comment_61660 Wed, 31 Dec 2008 07:56:56 GMT We have the cheapest quality a-z termination in the market for sale. Please contact us MSN: sales@ez2telecom.com <mailto: sales@ez2telecom.com > Email: sales@ez2telecom.com <br>ez2telecom2005 () at 2008-12-31 07:56 GMT http://www.voip-info.org/boards/view/board/7?view_comment_id=61659#comment_61659 Wed, 31 Dec 2008 07:56:32 GMT Hi, I have 20 users does calls trough my asterisk server. what when sometimes the voice become silent for few second with all the users. but other end users are still connected after 10-15 sec they again resume their conversation. Please help immediately. Thanks Sanjay <br>me_sanjay1986 () at 2008-12-30 22:45 GMT http://www.voip-info.org/wiki/view/voip-info.org?view_comment_id=61653#comment_61653 Tue, 30 Dec 2008 22:45:47 GMT I'm running AsteriskWin32 (0.66), asterisk ver 1.2.14. I'm using Cisco 7941 phones, and with a SIP trunk for my lines. Everything is working fine... Only exception is I'm getting a RTP error when I play a message to a incoming caller before passing the call on to a phone. If i pass a incoming call directly to the phone, everything works fine! I have a public IP and ALL incoming (all ports, UDP & TCP) traffic for that IP is pointed directly to the PC hosting Asterisk. All out going traffic also goes out on the same public IP. I've tried almost every NAT configuration and various different "caninvite" scenarios for both the SIP connection and the phone device. Nothing seems to correct the error. ;sip.cong [general] context = incoming bindaddr = 0.0.0.0 insecure = very videosupport = yes tos = lowdelay srvlookup = yes defaultexpiry = 600 nat = route bindport = 5060 externip = 66.225.35.115 register => RFGI:w2tgb88z@stl04a.netlogic.net:5060 [4000] type = friend context = default subscribecontext = localhint callgroup = 1 pickupgroup = 1 host = dynamic port = 5060 dtmfmode = rfc2833 nat = never mailbox = 4000 disallow = all allow = ulaw canreinvite=yes [provider] _register = yes type = friend context = incoming host = stl04a.netlogic.net port = 5060 outboundproxy = stl04a.netlogic.net secret = w2tgb88z dtmfmode = rfc2833 nat = route username = RFGI insecure = very disallow = all allow = ulaw canreinvite=no ;extensions.conf ;the following works GREAT, call rings to phone, phone can be awnsered and no problems [incoming]; working example exten => _.,1,Answer exten => _.,2,NoOp(icoming call, ${SIP_HEADER(To)}, ${SIP_HEADER(From)}, ${CALLERID(name)} ) exten => _.,3,Macro(stdexten,4000|4000|SIP/4000) ;if i swapp the names of incomingx and incoming to use the following instead ;it runs throug, plays the inomingwelcome wav file, the phone rings, and then once awnsered the call is dropped and the asterisk console shows RTP Read error, connection reset by peer [incomingx]; exten => _.,1,Answer exten => _.,2,NoOp exten => _.,3,Playback(incomingwelcome) exten => _.,4,NoOp exten => _.,5,Macro(stdexten,4000|4000|SIP/4000) Any ideas? Anyone!? <br>jroozee () at 2008-12-30 20:46 GMT http://www.voip-info.org/boards/view/board/3?view_comment_id=61648#comment_61648 Tue, 30 Dec 2008 20:46:40 GMT Speakers and all other VoIP Periipherals can be purchased at WestCon 888VoIPstore Neobites Anixter <br>billmajerczak () at 2008-12-30 18:33 GMT http://www.voip-info.org/wiki/view/CyberData?view_comment_id=61643#comment_61643 Tue, 30 Dec 2008 18:33:07 GMT Does anyone know why this now displays with the keys in columns rather than rows? They used to be in rows, which is much neater. Best wishes Adrian <br>Moonrakre (Adrian Birt) at 2008-12-30 13:40 GMT http://www.voip-info.org/wiki/view/Asterisk%40home+end+user+manual?view_comment_id=61633#comment_61633 Tue, 30 Dec 2008 13:40:57 GMT