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[asterisk-users] help with no audio

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jsmith at digium.com
Guest

Posted: Tue Apr 01, 2008 12:32 pm Post subject: [asterisk-users] help with no audio

On Tue, 2008-04-01 at 13:24 -0400, Jerry Geis wrote:

Quote:

I call into the dialplan and try to play demo-congrats and I hear nothing.

Firewall is disabled.
Everything is on the 192.168.1.X network for this simple configuration.
The tftp server is giving the polycom phone the config files.

Any ideas why I dont hear audio?

Do you happen to have an unconfigured T1 card in your machine? That's
the most common problem I see for people when they get no audio at all
coming out of Asterisk. If that's not the case, I'd turn RTP debugging
on in the command-line and make sure RTP packets are coming and going
from the Asterisk box.

--
Jared Smith
Community Relations Manager
Digium, Inc.

geisj at pagestation.com
Guest

Posted: Tue Apr 01, 2008 12:44 pm Post subject: [asterisk-users] help with no audio

Quote:

On Tue, 2008-04-01 at 13:24 -0400, Jerry Geis wrote:

Quote:

/ I call into the dialplan and try to play demo-congrats and I hear nothing.

/>/
/>/ Firewall is disabled.
/>/ Everything is on the 192.168.1.X network for this simple configuration.
/>/ The tftp server is giving the polycom phone the config files.
/>/
/>/ Any ideas why I dont hear audio?
/
Do you happen to have an unconfigured T1 card in your machine? That's
the most common problem I see for people when they get no audio at all
coming out of Asterisk. If that's not the case, I'd turn RTP debugging
on in the command-line and make sure RTP packets are coming and going
from the Asterisk box.

--
Jared Smith
Community Relations Manager
Digium, Inc.

Jared,

I have no card in this unit at this time.
lsmod shows ztdummy loaded.

I turned on "rtp debug" and got a bunch of lines like:
Got RTP packet from 192.168.1.99:2226 ....

Does that help you?

Jerry

geisj at pagestation.com
Guest

Posted: Wed Apr 02, 2008 8:45 am Post subject: [asterisk-users] help with no audio

Jerry Geis wrote:

Quote:

On Tue, 2008-04-01 at 13:24 -0400, Jerry Geis wrote:

Quote:

/ I call into the dialplan and try to play demo-congrats and I hear

nothing.
/>/ />/ Firewall is disabled. />/ Everything is on the 192.168.1.X
network for this simple configuration.
/>/ The tftp server is giving the polycom phone the config files.
/>/ />/ Any ideas why I dont hear audio?
/
Do you happen to have an unconfigured T1 card in your machine? That's
the most common problem I see for people when they get no audio at all
coming out of Asterisk. If that's not the case, I'd turn RTP debugging
on in the command-line and make sure RTP packets are coming and going
from the Asterisk box.

--
Jared Smith
Community Relations Manager
Digium, Inc.

Jared,

Using the "rtp debug" I noticed that when the phone has no audio all I
see is:
Got RTP packet ...
Got RTP packet...

There are no Sent RTP packets..

under normal cases there is One sent and one Got:
Got RTP packet...
Sent RTP packet...

Why would asterisk not be sending RTP packets????

I have no hardware card in this test system. Just two polycom IP330 phones.
Once in a great while I will hear audio when calling into the dialplan
and playing demo-congrats.
95% of the time I hear NO audio though. I am using asterisk 1.4.18,
libpri 1.4.3 and zaptel 1.4.9.2 (ztdummy is loaded)

Why might asterisk NOT be sending RTP packets?

Jerry

geisj at pagestation.com
Guest

Posted: Wed Apr 02, 2008 12:42 pm Post subject: [asterisk-users] help with no audio

Jerry Geis wrote:

Quote:

On Tue, 2008-04-01 at 13:24 -0400, Jerry Geis wrote:

Quote:

/ I call into the dialplan and try to play demo-congrats and I hear

I have found the echo command. I modified the dialplan to use echo.
I turned on rtp debug and I see packets going BOTH ways.

I have looked all through the zaptel.conf (below)
everything is commented out. there are no cards in my box. zapata looks
the same everything commented out.

I am not finding a reason for not getting audio packets sent back to the
phone.

Any suggestion on something to try?

Jerry

---------------------

#
# Zaptel Configuration File
#
# This file is parsed by the Zaptel Configurator, ztcfg
#
#
# First come the span definitions, in the format
# span=<span num>,<timing>,<line build out (LBO)>,<framing>,<coding>[,yellow]
#
# The timing parameter determines the selection of primary, secondary, and
# so on sync sources. If this span should be considered a primary sync
# source, then give it a value of "1". For a secondary, use "2", and so on.
# To not use this as a sync source, just use "0"
#
# The line build-out (or LBO) is an integer, from the following table:
# 0: 0 db (CSU) / 0-133 feet (DSX-1)
# 1: 133-266 feet (DSX-1)
# 2: 266-399 feet (DSX-1)
# 3: 399-533 feet (DSX-1)
# 4: 533-655 feet (DSX-1)
# 5: -7.5db (CSU)
# 6: -15db (CSU)
# 7: -22.5db (CSU)
#
# The framing is one of "d4" or "esf" for T1 or "cas" or "ccs" for E1
#
# Note: "d4" could be referred to as "sf" or "superframe"
#
# The coding is one of "ami" or "b8zs" for T1 or "ami" or "hdb3" for E1
#
# E1's may have the additional keyword "crc4" to enable CRC4 checking
#
# If the keyword "yellow" follows, yellow alarm is transmitted when no
# channels are open.
#
#span=1,0,0,esf,b8zs
#span=2,1,0,esf,b8zs
#span=3,0,0,ccs,hdb3,crc4
#
# Next come the dynamic span definitions, in the form:
# dynamic=<driver>,<address>,<numchans>,<timing>
#
# Where <driver> is the name of the driver (e.g. eth), <address> is the
# driver specific address (like a MAC for eth), <numchans> is the number
# of channels, and <timing> is a timing priority, like for a normal span.
# use "0" to not use this as a timing source, or prioritize them as
# primary, secondard, etc. Note that you MUST have a REAL zaptel device
# if you are not using external timing.
#
# dynamic=eth,eth0/00:02:b3:35:43:9c,24,0
#
# Next come the definitions for using the channels. The format is:
# <device>=<channel list>
#
# Valid devices are:
#
# "e&m" : Channel(s) are signalled using E&M signalling (specific
# implementation, such as Immediate, Wink, or Feature Group D
# are handled by the userspace library).
# "fxsls" : Channel(s) are signalled using FXS Loopstart protocol.
# "fxsgs" : Channel(s) are signalled using FXS Groundstart protocol.
# "fxsks" : Channel(s) are signalled using FXS Koolstart protocol.
# "fxols" : Channel(s) are signalled using FXO Loopstart protocol.
# "fxogs" : Channel(s) are signalled using FXO Groundstart protocol.
# "fxoks" : Channel(s) are signalled using FXO Koolstart protocol.
# "sf" : Channel(s) are signalled using in-band single freq tone.
# Syntax as follows:
# channel# => sf:<rxfreq>,<rxbw>,<rxflag>,<txfreq>,<txlevel>,<txflag>
# rxfreq is rx tone freq in hz, rxbw is rx notch (and decode)
# bandwith in hz (typically 10.0), rxflag is either 'normal' or
# 'inverted', txfreq is tx tone freq in hz, txlevel is tx tone
# level in dbm, txflag is either 'normal' or 'inverted'. Set
# rxfreq or txfreq to 0.0 if that tone is not desired.
# "unused" : No signalling is performed, each channel in the list remains idle
# "clear" : Channel(s) are bundled into a single span. No conversion or
# signalling is performed, and raw data is available on the master.
# "indclear": Like "clear" except all channels are treated individually and
# are not bundled. "bchan" is an alias for this.
# "rawhdlc" : The zaptel driver performs HDLC encoding and decoding on the
# bundle, and the resulting data is communicated via the master
# device.
# "fcshdlc" : The zapdel driver performs HDLC encoding and decoding on the
# bundle and also performs incoming and outgoing FCS insertion
# and verification. "dchan" is an alias for this.
# "nethdlc" : The zaptel driver bundles the channels together into an
# hdlc network device, which in turn can be configured with
# sethdlc (available separately).
# "dacs" : The zaptel driver cross connects the channels starting at
# the channel number listed at the end, after a colon
#
# The channel list is a comma-separated list of channels or ranges, for
# example:
#
# 1,3,5 (channels one, three, and five)
# 16-23, 29 (channels 16 through 23, as well as channel 29
#
# So, some complete examples are:
# e&m=1-12
# nethdlc=13-24
# fxsls=25,26,27,28
# fxols=29-32
#
#fxoks=1-24
#bchan=25-47
#dchan=48
#fxols=1-12
#fxols=13-24
#e&m=25-29
#nethdlc=30-33
#clear=44
#clear=45
#clear=46
#clear=47
#fcshdlc=48
#dacs=1-24:48

#
# Finally, you can preload some tone zones, to prevent them from getting
# overwritten by other users (if you allow non-root users to open /dev/zap/*
# interfaces anyway. Also this means they won't have to be loaded at runtime.
# The format is "loadzone=<zone>" where the zone is a two letter country code.
#
# You may also specify a default zone with "defaultzone=<zone>" where zone
# is a two letter country code.
#
loadzone = us
#loadzone = us-old
#loadzone=gr
#loadzone=it
#loadzone=fr
#loadzone=de
#loadzone=uk
#loadzone=fi
#loadzone=jp
#loadzone=sp
#loadzone=no
defaultzone=us

tzafrir.cohen at xorco...
Guest

Posted: Wed Apr 02, 2008 1:05 pm Post subject: [asterisk-users] help with no audio

On Tue, Apr 01, 2008 at 01:44:39PM -0400, Jerry Geis wrote:

Quote:

I have no card in this unit at this time.
lsmod shows ztdummy loaded.

Just to make sure that this is not the problem, what's the output of:

zttest -c 3

--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.cohen at xorcom.com
+972-50-7952406 mailto:tzafrir.cohen at xorcom.com
http://www.xorcom.com iax:guest at local.xorcom.com/tzafrir

tzafrir.cohen at xorco...
Guest

Posted: Wed Apr 02, 2008 1:23 pm Post subject: [asterisk-users] help with no audio

On Wed, Apr 02, 2008 at 02:16:56PM -0400, Jerry Geis wrote:

Quote:

On Tue, Apr 01, 2008 at 01:44:39PM -0400, Jerry Geis wrote:

Quote:

/ I have no card in this unit at this time.

/>/ lsmod shows ztdummy loaded.
/
Just to make sure that this is not the problem, what's the output of:

zttest -c 3

--

When running this nothing comes back...
It says "Opened pseduo zap interface, measuring accuracy..."
and that is all.

I am using Centos 2.6.18-53.1.14.el5

I also just tried rmmod ztdummy and then starting asterisk again and the
audio works.
something is wrong with ztdummy.

I went back to zaptel-1.4.9.2 diretory and did make clean; ./configure;
make; make install
(one at a time ) I saw no errors. tail /var/log/messages after modprove
showed no errors.

I believe that this means nothing. modprobe does nothing if the module
is already loaded.

--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.cohen at xorcom.com
+972-50-7952406 mailto:tzafrir.cohen at xorcom.com
http://www.xorcom.com iax:guest at local.xorcom.com/tzafrir

eric at fnords.org
Guest

Posted: Wed Apr 02, 2008 1:40 pm Post subject: [asterisk-users] help with no audio

Yes, some kernels don't work with ztdummy. This is discussed over and
over and over again on this mailing list. Check the archives.

Tzafrir Cohen wrote:

Quote:

On Wed, Apr 02, 2008 at 02:16:56PM -0400, Jerry Geis wrote:

Quote:

On Tue, Apr 01, 2008 at 01:44:39PM -0400, Jerry Geis wrote:

Quote:

/ I have no card in this unit at this time.

/>/ lsmod shows ztdummy loaded.
/
Just to make sure that this is not the problem, what's the output of:

zttest -c 3

--

I believe that this means nothing. modprobe does nothing if the module
is already loaded.

--
Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS,
T-1, PRI, Frame Relay, Linux, and network design. Based near
Birmingham, AL. Now accepting clients worldwide.

dinesh at alphaque.com
Guest

Posted: Thu Apr 03, 2008 2:46 am Post subject: [asterisk-users] help with no audio

On Tue, 01 Apr 2008 13:32:28 -0400, Jared Smith wrote:

Quote:

On Tue, 2008-04-01 at 13:24 -0400, Jerry Geis wrote:

Quote:

I call into the dialplan and try to play demo-congrats and I hear
nothing.

Firewall is disabled.
Everything is on the 192.168.1.X network for this simple configuration.
The tftp server is giving the polycom phone the config files.

Any ideas why I dont hear audio?

Do you happen to have an unconfigured T1 card in your machine? That's
the most common problem I see for people when they get no audio at all
coming out of Asterisk.

we've seen sites where just configuring the T1/E1 card alone is not
enough, we'd need to plug the card with a loopback cable or connect it to
a live E1 for rtp to work. any clues why this is the case ?
--
Regards, /\_/\ "All dogs go to heaven."
dinesh at alphaque.com (0 0) http://www.openmalaysiablog.com/
+==========================----oOO--(_)--OOo----==========================+
| for a in past present future; do |
| for b in clients employers associates relatives neighbours pets; do |
| echo "The opinions here in no way reflect the opinions of my $a $b." |
| done; done |
+=========================================================================+

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