[Asterisk-Users] Phones were working fine - Now there is
noaudiowhen calling between extensions
Gabriel Afana
asterisk at gafana.com
Mon Mar 20 19:10:16 MST 2006
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" <asterisk at gafana.com>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at 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
> >
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