I am using zaptel 1.4.12.1 and asterisk 1.4.18 - I also tried 1.4.21
lspci shows
00:00.0 Host bridge: VIA Technologies, Inc. VT8623 [Apollo CLE266]
00:01.0 PCI bridge: VIA Technologies, Inc. VT8633 [Apollo Pro266 AGP]
00:10.0 USB Controller: VIA Technologies, Inc. VT82x UHCI USB 1.1
Get a core dump.
Steve this is the stack trace still working on the dump.
Program received signal SIGSEGV, Segmentation fault.
[Switching to Thread -1229517920 (LWP 9875)]
__ast_read (chan=0x8294a68, dropaudio=0) at channel.c:2016
2016f =
Jerry Geis wrote:
Get a core dump.
Steve this is the stack trace still working on the dump.
Program received signal SIGSEGV, Segmentation fault.
[Switching to Thread -1229517920 (LWP 9875)]
__ast_read (chan=0x8294a68, dropaudio=0) at channel.c:2016
2016f
I have a small system, server, client and 2 phones. Phones are polycom
501's.
In general all is working fine. I can call the two phones, speak etc...
I can have the server call each phone and play a wave file.
However, when trying to setup a direct dial number of 1044 that
calls another machine
Are your polycom phones set up for overlap dialing or do you dial the
number then press a key to dial?
From you message I tried a couple things...
Clicking New call, then starting to dial this is when it messes up.
when I start entering the number first then click dial this successfull
Jerry Geis wrote:
I am running 1.4.22.
I am doing a simple call into the dialplan and am getting a strange
error:
[Nov 3 08:32:27] NOTICE[8022]: chan_sip.c:14316
handle_request_invite: Failed to authenticate user 404
sip:[EMAIL PROTECTED];tag=547521CB-DB0D6130
This is the only line
I am running 1.4.22.
I am doing a simple call into the dialplan and am getting a strange error:
[Nov 3 08:32:27] NOTICE[8022]: chan_sip.c:14316 handle_request_invite:
Failed to authenticate user 404
sip:[EMAIL PROTECTED];tag=547521CB-DB0D6130
This is the only line that prints on the
I am trying to setup a second asterisk box to play with console/dsp over
sip.
My sip.conf on the second box is:
[secondbox]
type=friend
username=secondbox
secret=secret
disallow=all
allow=ulaw
allow=alaw
allow=gsm
host=SERVERIP
context=consoledsp
The second box is not connecting to my asterisk
Jerry Geis wrote:
-- Attempting call on DAHDI/1ww for
[EMAIL PROTECTED]:1 (Retry 1)
[Oct 16 14:36:42] WARNING[16408]: chan_dahdi.c:8132 dahdi_request:
Unknown option 'w' in '1ww'
[Oct 16 14:36:43] WARNING[16408]: chan_dahdi.c:1481 dahdi_enable_ec:
Unable to enable echo
-- Attempting call on DAHDI/1ww for
[EMAIL PROTECTED]:1 (Retry 1)
[Oct 16 14:36:42] WARNING[16408]: chan_dahdi.c:8132 dahdi_request:
Unknown option 'w' in '1ww'
[Oct 16 14:36:43] WARNING[16408]: chan_dahdi.c:1481 dahdi_enable_ec:
Unable to enable echo cancellation on channel 1 (No
Jerry Geis wrote:
Did you check sip.conf to make sure that the port is correctly set to
5060?
Please show the output of Cli sip show peer peernumber and the
contents of your SEPMAC.cnf file.
Dave
This all ended up being CRAZY network stuff.
my server has 2 network cards
Hi Jerry,
Hm, okay. We had to use md5secret (instead of secret) in the sip.conf for
our 7970's to get them to successfully register with asterisk. However, if
you had them working before then I doubt this is the issue. You can try
anyway though,
Hi Jerry,
Hmm. We had to replace our router with one that supported SIP ALG (we went
cisco). However, since you mention all the phones are in the LAN this
shouldn't make a difference.
Does the problem go away if you go back to the old firewall?
Thanks,
Matt
unfortunately I cannot
Did you check sip.conf to make sure that the port is correctly set to 5060?
Please show the output of Cli sip show peer peernumber and the contents of
your SEPMAC.cnf file.
Dave
sip.conf has :
bindport=5060 ; UDP Port to bind to (SIP standard port
is 5060)
Are includes supported in the file /etc/dahdi/system.conf
link you can include in say a sip.conf
What about in chan_dahdi.conf?
Thanks,
Jerry
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I have a handful of cisco phones that has been working.
Today they started showing X's. looking at sip debug I see the 401
unauthorized.
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP IP:52110;branch=z9hG4bK29694d4a;received=IP
From: sip:[EMAIL PROTECTED];user=phone
To: sip:[EMAIL
Jerry Geis wrote:
I have a handful of cisco phones that has been working.
Today they started showing X's. looking at sip debug I see the 401
unauthorized.
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP IP:52110;branch=z9hG4bK29694d4a;received=IP
From: sip:[EMAIL PROTECTED];user=phone
To: sip
Did the server reboot or lose communication? This happens with our 7970's
sometimes if there's been a hiccup, usually dialing voicemail registers them
back up - occasionally we've had to do the soft reboot from the screen.
401 unauth - looks like it may be md5secret issue, or nat traversal
I just downloaded the dahdi release and installed it.
I removed anything zaptel I could find. /sbin , /lib/modules and others...
when doing a service dahdi start it is still looking to ztcfg Why?
I have look all about and cant determine why?
service dahdi start
Loading DAHDI hardware
What is the correct way to uninstall zaptel
in the zaptel directory I can do make uninstall-modules
which does just that but what about all the other files???
/etc/udev/rules/XX
/etc/init.d/XX
/sbin/ztXX
and others
doing a make uninstall gives an error.
Is there anything that removes all
I have a box running asterisk 1.4.17 that had been working.
it has 2 uniden phones connected on it.
This was working and now the phones dont ring when calling each other.
below is the sip debug. I cant see why the other phone does not ring?
I also tried changing the canreinvite for no to yes but
snip
Based on the SIP debug included here, it appears that Asterisk is not
receiving
a response to the INVITE it is sending to 522 (192.168.1.99). Since the phone
is
not ringing, it makes me suspect that for some reason the linksys is
preventing
the INVITE from reaching the phone.
I have a simple context that connects to the console dsp which works,
but then after I hangup I hear ringing on the console dsp. It rings
until I stop asterisk.
Why is that and how can I stop it?
Thanks,
Jerry
[paging]
exten = s,1,Answer
exten = s,n,Playback(beep)
exten = s,n,Dial(Console/dsp)
Are there parameters for em wink?
1) timing parameters
2) dial delay or pre dial.
Thanks
Jerry
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Register Now:
I am using centos 4.6 i586.
I have compiles zaptel 1.4.11 ztdummy.
When I load ztdummy the /proc/interupts rtc does not increment.
centos runs 2.6.9 kernel.
I'm not sure ztdummy.c uses RTC by default in this case.
Anyone using centos 4.X successfully with console/dsp and not internal
cards.
On Thu, Aug 21, 2008 at 10:29:18AM -0400, Jerry Geis wrote:
/ I am using centos 4.6 i586.
//
// I have compiles zaptel 1.4.11 ztdummy.
// When I load ztdummy the /proc/interupts rtc does not increment.
/
does ztdummy itself tick?
try zttest
If it does not stay hung there, it's working
In the past there was ztdummy - what is the new equivalent in dahdi?
Also it used to be Zap/X what is the new channel name?
searching voip-info.org for dahdi didnt show me anything about that...
Thanks,
Jerry
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This links: http://downloads.digium.com/pub/telephony/dahdi-linux-complete/
appear broken. thy just take me back to /pub
nothing downloads.
Jerry
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Tzafrir Cohen wrote:
/ That's part of Asteirsk. Not of Zaptel/DAHDI itself . Asterisk has
// renamed chan_zap.so to chan_dahdi.so . It now supports DAHDI . It also
// supports Zap/ for the moment for backward compatibility .
/
And the relevant information is in the Zaptel-to-DAHDI.txt file,
Jerry Geis wrote:
I am running centos 4.6 i386 kernel 2.6.9-67
ztdummy compiles fine, loads fine, but does not work.
modprobe ztdummy debug=1
dmesg shows
ztdummy: init() finished
however the debug is supposed to print something every 5 seconds
it does not do this. Nor does /proc
Hi
I am using centos 4.6 on an ebox 4300. Everything seems to be working
except the
/proc/interrupts rtc is always constant. On other machines the rtc
(which ztdummy uses) is always incrementing.
the uhci_hcd and ehci_hcd are both running.
What dont I have right on the system so rtc
Jerry Geis wrote:
Hi
I am using centos 4.6 on an ebox 4300. Everything seems to be working
except the
/proc/interrupts rtc is always constant. On other machines the rtc
(which ztdummy uses) is always incrementing.
the uhci_hcd and ehci_hcd are both running.
What dont I have right
I am running centos 4.6 i386 kernel 2.6.9-67
ztdummy compiles fine, loads fine, but does not work.
modprobe ztdummy debug=1
dmesg shows
ztdummy: init() finished
however the debug is supposed to print something every 5 seconds
it does not do this. Nor does /proc/interrupts rtc value
Hi all,
I am using outgoing call files to place calls. Issue is when that call
is BUSY I dont get the correct DIALSTATUS
from that call when running my AGI and the failed extension.
WHERE can I make a change in the code so that the DIALSTATUS when the
call ended can be
added as a variable in
Jerry Geis wrote:
Call files spawn a completely new channel that your AGI most likely
isn't going to be able to track. Since the call is a completely new
channel, the DIALSTATUS variable for this channel will not be visible
to your original channel. You may want to look at using
I am using asterisk 1.4.21 with outgoing call files.
I am call a line that is busy as you can see below.
How can my AGI ask what the status of the last call was
so I can tell if there was NO ANSWER or it was BUSY?
Thanks,
Jerry
-- Attempting call on SIP/401 for
[EMAIL PROTECTED]:1 (Retry
Jerry Geis wrote:
I am using asterisk 1.4.21 with outgoing call files.
I am call a line that is busy as you can see below.
How can my AGI ask what the status of the last call was
so I can tell if there was NO ANSWER or it was BUSY?
Thanks,
Jerry
-- Attempting call on SIP/401
Call files spawn a completely new channel that your AGI most likely
isn't going to be able to track. Since the call is a completely new
channel, the DIALSTATUS variable for this channel will not be visible to
your original channel. You may want to look at using the Originate
action
Assuming you have a Quad core machine, at least 4 GIG ram,
will a machine like this handle 4 Quad T1 cards?
is that advisable?
What about running AGI's on such a machine.
Will the machine handle starting/stopping all those AGI's?
Thanks,
Jerry
___
I am using 2 asterisk 1.4.21 boxes.
the main one every 20 seconds does an outgoing call file to the second
box and speaks a wav file over the console/dsp.
Works for A LONG TIME then seg fault.
I have posted a bug report but to date there has been no action.
I tried to see if there was some thing
When dialing using a T1/PRI with a outgoing call files
Like Channel: Zap/1/95551212
is there ever a need to delay or pause in there?
I have gotten feedback from a customer that instead of dialing the 95551212
it seems to have dialed 55512 which just happened to be an internal
extension.
So it
dialplan show default
There is no existence of 'default' context
Command 'dialplan show default' failed.
I am getting the same thing for default
What gives?
Jerry
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On Mon, 2008-07-21 at 16:12 -0400, Jerry Geis wrote:
/
// �[Jul 21 12:53:56] NOTICE[4881]: chan_sip.c:16416 handle_request_invite:
/ Call from 'devcentos5x64_to_ebox4300' to extension
'mediaport_audio_visual' rejected because extension not found.
Jerry--
from
I dont see any errors in the dialplan while loading.
I tried to past the whole log but it was rejected.
Again 1.2 works, 1.4 works, no on 1.6 I made no changes to the files.
I cant even dialplan show default at this time.
Jerry
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I am upgrading a box from 1.4 to 1.6 and my console/dsp stopped working.
I am getting a SIP/401 Unauthorized error and then a SIP/404 error.
I changed nothing in the configs.
Is there a particular parameter needed for 1.6 that 1.4 did not care about?
If I drop back to 1.4 it starts working
Verbosity is at least 5
[Kebox4300*CLI
--- SIP read from UDP://192.168.1.8:5060 ---
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.8:5060;branch=z9hG4bK029ea409;rport
From: Jerry Geis 204 sip:[EMAIL PROTECTED];tag=as7d1f7b71
To: sip:[EMAIL PROTECTED]
Contact: sip:[EMAIL
Do you have an extension called 'mediaport_audio_visual' in a context
called 'smvoice-mediaport'? If so, can you post that context so we can
see how it looks?
Kevin,
I mentioned that 1.4 works - 1.6 did not, going back to 1.4 works again.
Here are the pieces:
my sip.conf has context
1.4 was working fine.
I thought I would try 1.6 beta 9
from my asteirsk 1.4 server to my asterisk client 1.6beta it wont accept
the call.
[Jul 18 20:34:55] NOTICE[966]: chan_sip.c:16416 handle_request_invite:
Call from 'JJ' to extension 'jj_audio' rejected because extension not found.
I
Jerry Geis wrote:
1.4 was working fine.
I thought I would try 1.6 beta 9
from my asteirsk 1.4 server to my asterisk client 1.6beta it wont
accept the call.
[Jul 18 20:34:55] NOTICE[966]: chan_sip.c:16416 handle_request_invite:
Call from 'JJ' to extension 'mediaport_audio_visual' rejected
I am seeing the following seg fault when using a SIP connection
to Console/Dsp. It takes quite a long time to happen but it eventually
happens.
nothing else is on this box. just alsa and asterisk running sip and
console/dsp.
What should I do now?
Jerry
Program received signal SIGSEGV,
Jerry Geis wrote:
I am seeing the following seg fault when using a SIP connection
to Console/Dsp. It takes quite a long time to happen but it eventually
happens.
nothing else is on this box. just alsa and asterisk running sip and
console/dsp.
What should I do now?
Jerry
Program
I am using asterisk 1.4.21and svn-124910 and getting the chan_alsa:693
resource temporarily unavailable message.
The audio is working but I dont recall getting any error message in the
past.
Is this something to be concerned about?
Jerry
___
--
hi,
I have an AGI running after an outgoing call file starts it up.
Everything works fine except if my line has a problem.
Trying to simulate this I unplug the line. So there is no dialtone.
How do I detect this and let the AGI know so I can try line 2, 3, 4 etc...
Detecting the the AGI or some
I am using usb-audio for Console/Dsp with asterisk.
it is crashing 1.4.21 and also svn.
During the brief times its working the audio is choppy but understandable.
I have used aplay and arecord at the same time on the same wave file
and they work fine every time and I have done it MANY times.
On Fri, Jun 27, 2008 at 06:57:22AM -0400, Jerry Geis wrote:
/ I am using usb-audio for Console/Dsp with asterisk.
//
// it is crashing 1.4.21 and also svn.
/
Which channel driver? chan_alsa ? chan_oss? Or maybe chan_console ?
/ During the brief times its working the audio is choppy
When using console/dsp is that play only?
Is it play/record mode? If so how can I make it play only?
When I play wave files on a machine with aplay everything is fine. (no
record)
When I use asterisk and console/dsp I am getting seg faults in alsa-lib.
I want to make sure there is NO record
I am running asterisk from svn check out from yesterday Jun 24.
I started with 1.4.20, then 1.4.21 then svn.
I am getting:
pcm_local.h:389 snd_pcm_channel_area_addr assertion bitsofs %8 = 0 failed
segment fault.
I am running debian i386, on a 486 sx machine.
I am connecting to the Console/DSP
I managed to catch the whole trace from the seg fault. What is my next step?
Program received signal SIGABRT, Aborted.
[Switching to Thread -1224033360 (LWP 2440)]
0xb7d77947 in raise () from /lib/tls/libc.so.6
(gdb) where
#0 0xb7d77947 in raise () from /lib/tls/libc.so.6
#1 0xb7d790c9 in abort
I have compiled asterisk 1.4.20 on a 486 (sx) machine. No floating point
but math emulation is used in the kernel.
When I run asterisk -vc all I get is Illegal instruction.
I compiled as normally I do. Whats my next step. this is download source
and compiled.
Jerry
when installing zaptel it is trying to download the fw-oct6114 file.
This is a remote install and the firewall is not open to download this file.
How do I get around this???
I can put needed files on the machine - the machine just cant download
them by itself.
THanks,
Jerry
All I did was download the file separately and send it over to the
server with WinSCP. Just toss it in /usr/src/zaptel/firmware.
Chris,
I grabbed all 4 files in this location zaptel* and put them in
zaptel/firmware.
Now its trying to grab a file that is not there:
Customer with a siemens HICOM 4000 switch
they are talking about T1 OPS (I have not heard this OPS term before)
Will the digium dual T1 card work with this?
Thanks,
Jerry
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There seems to be a 2 second delay after issueing the command
/usr/sbin/asterisk -rx sip show peers
in 1.4.20
Is there a reason why the delay? It wasnt there before.
Matter of fact I just jumped into an old system and no delay. 1.4.20 has
the delay.
Jerry
Is there a method from the dialplan that I
can turn on a voicemail indicator on a polycom phone. Like a blinking
light or something.
Then I would also need to turn it off.
Is there a way to do that?
Jerry
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When doing a sip show peers I might see something like:
Name/username HostDyn Nat ACL Port
Status
devcentos5x64_to_mmfirepa 192.168.1.177 5060
Unmonitored
devcentos5x64_to_bt610tMM 192.168.1.159 5060
/ When doing a sip show peers I might see something like:
// Name/username HostDyn Nat ACL Port
// Status
// devcentos5x64_to_mmfirepa 192.168.1.177 5060
// Unmonitored
// devcentos5x64_to_bt610tMM 192.168.1.159 5060
// Unmonitored
//
Not using asterisk voicemail, is there a way I can (through a dialplan
command or manager command)
tell a polycom phone to turn on a voice mail indicator blinking light -
and then also turn it off?
Jerry
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Jerry Geis wrote:
I have xinet tftp running on centos 5.1
It seems to be running on the local network eht0 fine. My box has 2 nics.
however when I connect to eth1 for tftp I get:
in.tftpd[5084]: tftpd: read(ack): Connection refused
How can I get tftp working on BOTH eth0 and eth1 for my
I have xinet tftp running on centos 5.1
It seems to be running on the local network eht0 fine. My box has 2 nics.
however when I connect to eth1 for tftp I get:
in.tftpd[5084]: tftpd: read(ack): Connection refused
How can I get tftp working on BOTH eth0 and eth1 for my phone config files.
man
On Mon, Apr 28, 2008 at 11:07 AM, Jerry Geis geisj at pagestation.com
http://lists.digium.com/mailman/listinfo/asterisk-users wrote:
/ I have xinet tftp running on centos 5.1
//
// It seems to be running on the local network eht0 fine. My box has 2 nics.
// however when I connect to eth1
The netstat show 0.0.0.0
netstat -anp | grep :69
udp0 0 0.0.0.0:69
0.0.0.0:* 4007/xinetd
--
cat /etc/xinetd.d/tftp
# default: off
# description: The tftp server serves files using the trivial file
transfer \
#
Try having a look at the settings by running 'lokkit' or
'system-config-security-level-tui' from the command lin - ensure that
the firewall is disabled from there also, and turn off SELinux and see
if that makes any difference.
Robert
Robert,
I have turned off service iptables stop and I
Looks good to me. Try this: after doing the service iptables stop do
the following and see if there are any rules left:
iptables -L
iptables -t nat -L
iptables -t mangle -L
if there are any rules at all listed, replace the -L with -F and re-run
the commands.
-Brent
Ran each of the
I am getting an error:
chan_zap invalid tone duration 11220.
This is line 11220 in chan_zap.c and I have a toneduration of 300 in the
zapata.conf file.
I have commented it out and it is now working again.
Why is that an invalid paramter? It never used to be.
jerry
Hi all,
I have a polycom 501 phone that I rebooted today. It stopped working...
Normally the screen shows New call, Forward and that is all...
Now the screen shows New call, Forward, MyStat, Buddies.
It no longer accepts incoming calls nor can I make outgoing calls.
I have reloaded factory
try rebotting the phone. at boot up. hold down 4,8,6,* - that should reset
the local config for the phone
On Tue, Apr 15, 2008 at 5:58 PM, Jerry Geis geisj at pagestation.com
http://lists.digium.com/mailman/listinfo/asterisk-users wrote:
/ Hi all,
//
// I have a polycom 501 phone that I
Hi - Been using a TE205P for a number of months - no issues.
Today I was talking to someone and I heard click
No more phone service.
I still have data service on this T1 line. (partial phone)
zttool reports the SPAN as OK.
calls are not coming in or going out.
Does a card just go bad like
I was trying to get my polycom phone to auto answer.
I added this to the dialplan. Used a different phone to call 22
and the phone rang it did not auto answer.
Did I miss something?
exten = 22,1,SipAddHeader(Call-Info:=\;answer-after=0)
exten = 22,n,SipAddHeader(Alert-Info: Ring Answer)
exten =
When I execute the commands in my cli
pri show status
zap show status
I get errors for both commands.
I am running 1.4.19, with libpri 1.4.3, and zaptel 1.4.10.
how do I get these commands?
Jerry
--
help shows:
help
! Execute a shell command
Is there a way to inquire of the T1 link status?
I mean having cron (as example) execute a program that asks if the T1
status is OK.YEL or RED?
then on RED I can send some alert?
Thanks
Jerry
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Jerry Geis wrote:
CC [M]
/home/silentm/MessageNet/digium/zaptel-1.4.10/kernel/wcte12xp/vpmadt032.o
CC [M]
/home/silentm/MessageNet/digium/zaptel-1.4.10/kernel/wcte12xp/GpakApi.o
CC [M]
/home/silentm/MessageNet/digium/zaptel-1.4.10/kernel/wcte12xp/../voicebus.o
LD [M]
/home
Is there a way to tell the difference in an agi
between the person actually hanging up the phone
and the manager interface doing a hangup command?
Thanks,
Jerry
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asterisk-users
CC [M]
/home/silentm/MessageNet/digium/zaptel-1.4.10/kernel/wcte12xp/vpmadt032.o
CC [M]
/home/silentm/MessageNet/digium/zaptel-1.4.10/kernel/wcte12xp/GpakApi.o
CC [M]
/home/silentm/MessageNet/digium/zaptel-1.4.10/kernel/wcte12xp/../voicebus.o
LD [M]
Jerry Geis wrote:
ounds like the module got auto-unloaded due to not being in use.
I have found the most reliable way to invoke zaptel/ztdummy is using the
proper init script:
1. In your zaptel source directory, do make config. That will create
/etc/rc.d/init.d/zaptel and the rcX.d links
What does it take to get ztdummy to work correctly?
I have a new laptop HP HDX9200. I am running asterisk 1.4.19 and zaptel
1.4.9.2
Zaptel compiles fine. asterisk compiles fine. ztdummy loads asterisk runs.
Problem is playback() does not work. So then I stop zaptel, asterisk
runs and playback()
In article 47F4C604.1060301 at pagestation.com
http://lists.digium.com/mailman/listinfo/asterisk-users,
Jerry Geis geisj at pagestation.com
http://lists.digium.com/mailman/listinfo/asterisk-users wrote:
/ What does it take to get ztdummy to work correctly?
//
// I have a new laptop HP HDX9200
On Thu, Apr 03, 2008 at 01:45:09PM +, Tony Mountifield wrote:
/ Jerry, the first thing to check is cat /proc/interrupts and see if there
// is an entry for rtc on IRQ 8. There should be, and the interrupt counts
// on there should be going up at approximately 1024 per second.
/
To see it
/ uname -a shows x86_64 and Centos 5.1, 2.6.18-53.1.14.el5
/
You can try zttest, although I'd bet it will hang. See what's going
to the console (or use dmesg.) If it's a lot of rtc errors, then
you'll likely need to upgrade your kernel to at least 2.6.23.11. That
worked for me.
You
Hi Jerry,
Is that with ztdummy loaded or not? By default, Linux doesn't have
anything
that uses the RTC interrupt, so without ztdummy it will usually stay
at 1.
Once ztdummy and zaptel are loaded, then you should see it incrementing.
If not, that suggests a problem.
I have just
ounds like the module got auto-unloaded due to not being in use.
I have found the most reliable way to invoke zaptel/ztdummy is using the
proper init script:
1. In your zaptel source directory, do make config. That will create
/etc/rc.d/init.d/zaptel and the rcX.d links to it.
2.
Jerry Geis wrote:
On Tue, 2008-04-01 at 13:24 -0400, Jerry Geis wrote:
/ 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
Hi all, I seem to only be getting (1) call to sip_write() in
channels/chan_sip.c
I have a very simple setup. one server (no cards) 2 polycom IP 330 phones.
Server is 192.168.1.150 and phone is DHCP. Nothing else on the network.
No firewall is enabled.
I call into the dialplan with:
exten =
Jerry Geis wrote:
On Tue, 2008-04-01 at 13:24 -0400, Jerry Geis wrote:
/ 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
On Tue, Apr 01, 2008 at 01:44:39PM -0400, Jerry Geis wrote:
/ 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
I am using asterisk 1.4.18 with a polycom phone.
sip.conf has:
[532]
type=friend
username=532
secret=XXX
dtmfmode=RFC2833
host=dynamic
context=smvoice-sip
callerid=532
qualify=no
nat=no
disallow=all
allow=ulaw
allow=alaw
allow=gsm
canreinvite=no
I call into the dialplan and try to play
On Tue, 2008-04-01 at 13:24 -0400, Jerry Geis wrote:
/ 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
I am trying to use call files that dial and play a wave file
on 3 asterisk boxes console dsp.
This is working.
The 3 boxes are noticeably out of sync. From using 3 different call files
(time to process) I'm sure is the time delay.
Is there a way to get these audios more in sync?
Jerry
I have a cisco 7960 phone. Worked fine in the office.
I took it home. At home I have a linksys router that the phone is
plugged into.
The linksys router has DHCP enabled. I am getting the following error on
the console from the 7960.
I have tried it with nat=yes and nat=no in the sip.conf file.
MeetMe() has the K option that kills the conference,
how do I do that in app_conference() as there no kill the conference option?
Jerry
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Hi all
I am playing with meetme and I have things working in speak live mode.
What I would like to do is have a small meetme that kicks off on a
schedule (no problem there)
and just plays a wave file in the meetme and then exists.
I have played with option b of the meetme where the AGI plays
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