This is driving me nuts!! After unplugging all the phones, restarting the router and the modem, and reconfigurating my * boxes, I was finally able to communicate between both phones only when they were both registered to the same server.
If I try to call between phones between two different servers trunked with IAX, there is no sound (but the call rings and connects perfectly). This was working last week *nooooo problem*, all of a sudden its dead!!! Its killing me because it *was* working and now its not and I cannot figure out. - Gabe ----- Original Message ----- From: "Gabriel Afana" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Sent: Monday, March 20, 2006 4:39 PM Subject: Re: [Asterisk-Users] Phones were working fine - Now there is noaudiowhen calling between extensions > I just did a little RTP debug and this is what it shows: > > == Spawn extension (304, 301, 1) exited non-zero on 'SIP/304-d9f8' > -- Accepting AUTHENTICATED call from 216.152.244.81: > > requested format = ulaw, > > requested prefs = (), > > actual format = ulaw, > > host prefs = (), > > priority = mine > -- Executing Dial("IAX2/to_80-1", "SIP/301") in new stack > -- Called 301 > -- SIP/301-1fec is ringing > -- SIP/301-1fec answered IAX2/to_80-1 > Got RTP packet from 24.50.66.128:2228 (type 0, seq 16810, ts 344311448, len > 160) > Got RTP packet from 24.50.66.128:2228 (type 0, seq 16811, ts 344311608, len > 160) > Got RTP packet from 24.50.66.128:2228 (type 0, seq 16812, ts 344311768, len > 160) > Got RTP packet from 24.50.66.128:2228 (type 0, seq 16813, ts 344311928, len > 160) > Got RTP packet from 24.50.66.128:2228 (type 0, seq 16814, ts 344312088, len > 160) > Got RTP packet from 24.50.66.128:2228 (type 0, seq 16815, ts 344312248, len > 160) > Got RTP packet from 24.50.66.128:2228 (type 0, seq 16816, ts 344312408, len > 160) > Got RTP packet from 24.50.66.128:2228 (type 0, seq 16817, ts 344312568, len > 160) > Got RTP packet from 24.50.66.128:2228 (type 0, seq 16818, ts 344312728, len > 160) > Got RTP packet from 24.50.66.128:2228 (type 0, seq 16819, ts 344312888, len > 160) > Got RTP packet from 24.50.66.128:2228 (type 0, seq 16820, ts 344313048, len > 160) > Got RTP packet from 24.50.66.128:2228 (type 0, seq 16821, ts 344313208, len > 160) > Got RTP packet from 24.50.66.128:2228 (type 0, seq 16822, ts 344313368, len > 160) > .................. > > > that goes on for ever while the call is in progress. This is a call between > phones that go between two * servers. If I make a call between phones both > registered to the same asterisk server, this is my RTP stream: > > -- Executing Dial("SIP/304-c211", "SIP/301|30|r") in new stack > -- Called 301 > -- SIP/301-b2c8 is ringing > -- SIP/301-b2c8 answered SIP/304-c211 > -- Attempting native bridge of SIP/304-c211 and SIP/301-b2c8 > Got RTP packet from 24.50.66.128:2234 (type 0, seq 39729, ts -1972065425, > len 160) > Sent RTP packet to 24.50.66.128:2232 (type 0, seq 24819, ts 0, len 160) > Got RTP packet from 24.50.66.128:2234 (type 0, seq 39730, ts -1972065265, > len 160) > Sent RTP packet to 24.50.66.128:2232 (type 0, seq 24820, ts 160, len 160) > Got RTP packet from 24.50.66.128:2234 (type 0, seq 39731, ts -1972065105, > len 160) > Sent RTP packet to 24.50.66.128:2232 (type 0, seq 24821, ts 320, len 160) > Got RTP packet from 24.50.66.128:2234 (type 0, seq 39732, ts -1972064945, > len 160) > Sent RTP packet to 24.50.66.128:2232 (type 0, seq 24822, ts 480, len 160) > Got RTP packet from 24.50.66.128:2232 (type 0, seq 46694, ts 1105329892, len > 160) > Sent RTP packet to 24.50.66.128:2234 (type 0, seq 42690, ts 0, len 160) > Got RTP packet from 24.50.66.128:2232 (type 0, seq 46695, ts 1105330052, len > 160) > Sent RTP packet to 24.50.66.128:2234 (type 0, seq 42691, ts 160, len 160) > Got RTP packet from 24.50.66.128:2232 (type 0, seq 46696, ts 1105330212, len > 160) > Sent RTP packet to 24.50.66.128:2234 (type 0, seq 42692, ts 320, len 160) > Got RTP packet from 24.50.66.128:2232 (type 0, seq 46697, ts 1105330372, len > 160) > Sent RTP packet to 24.50.66.128:2234 (type 0, seq 42693, ts 480, len 160) > Got RTP packet from 24.50.66.128:2232 (type 0, seq 46698, ts 1105330532, len > 160) > Sent RTP packet to 24.50.66.128:2234 (type 0, seq 42694, ts 640, len 160) > [THE END] > > Once I anser the call, the RTP string starts and then stops right where I > put [THE END]. > > - Gabe > > > > > > > > > > Hey group, > > I have a Polycom 501 and a 301 together in my office. Each phone is > > registered to a different server. When I call one of the phones from the > > other, the other phone rings no problem (the calls are passed between > > servers via IAX). However, when I answer it, there is absolutely no audio > > in either direction. This just started happening today. > > > > It was working great before - I just plugged them in, got them > > registered and I was calling between phones no problem. Now I dont know > > what is happening and I cannot figure it out. It seems like a NAT issue, > > but I have qualify=yes, nat=yes and insecure=port,invite. > > > > - Gabe > > > > _______________________________________________ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > Asterisk-Users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users