C F wrote:
You could, under programming section 1.3.4 in the http interface to
configure the GW card enable DTMF Detection, that will enable Out of
Band DTMF. In the TDE they renamed this to DTMF signalling.
Believe me, I spent a great deal of time on this including Ethereal
captures and
Jonn R Taylor wrote:
Install a T1 between the Panasonic and Asterisk and program the T1 in the
Panasonic as a other custom PBX. VOIP card would be the best.
Jonn
One thing to beware of with the Panasonic VoIP card, is that I have
found no way of getting it to pass out of band DTMF,
Sema Arca wrote:
Hi Richard,
I could not succeed to make my ooh323 work somehow. I can see the peers
and the users but although my exten definition states that the call
should be forwarded to a GK, Asterisk does not send it out. I also have
the same problem with registration.
Do you
Sema Arca wrote:
Can you still send the config files? Maybe I can come up with an idea? :(
extensions.conf entry
exten = _1XX,1,Dial(OOH323/[EMAIL PROTECTED])
exten = _1XX,2,Congestion
ooh323.conf
[general]
h323id=ObjSysAsterisk
e164=100
callerid=asterisk
context=default
tos=lowdelay
Tony Mountifield wrote:
I don't know whether Objective Systems have abandoned chan_ooh323 and
the ooh323c stack, but it would be great to see them moved from -addons
into the main Asterisk tree.
This was always the plan from the beginning.
I have a post from one of the Objective Systems
Hi,
I have the following setup, with asterisk on a dual homed box:
PolyIP500(SIP)--192.168.4.0--Asterisk--192.168.0.0--Panasonic(H323)
It is running a recent SVN version of Asterisk 1.2 and ooh323.
The problem I have, is that despite having canreinvite=no in the
sip.conf, asterisk still
Dovid B wrote:
I doubt this will help but try also nat=yes.
Thanks for the reply but no, I have spent some time trying quite a
number of variations of NAT related configuration changes in various places.
I don't know why it does not honour the canreinvite=no entry.
Regards,
Richard
Steve Totaro wrote:
I guess I am just lucky to have 24 hour manned data centers with staff
that walk around looking for flashing LEDs.
I am sure there is some error thrown in /var/log/messages about a
failure that could be used to trigger a notification quite trivially.
Both smartd
Can someone comment why only Digium cards still under warranty are
eligible to use this EC at no cost, versus older cards?
Regards,
Richard
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C F wrote:
Which panasonic system?
I'm assuming you are talking about the TDA line. If so get a IP
Gateway card on the TDA system, that card uses h323, then configure it
with asterisk as h323, or my favorite, get a PRI card on the TDA sysem
(unless it's a TDA50 then the option is not
Yuan LIU wrote:
I just didn't want to accept fxotune.c's claim about working only with
TDM. Several other users indicated that they were not able to tune
X100P. There's also a README.debian note that specifically indicated
exclusion of X100P.
fxotune is written to change register values
Jesus Mogollon wrote:
Hi all
Does anyone know of any motherboards with PCI slots that can take the
TE412P card? Is there such a MB for Athlon 64 or P4 procs?
I have no experience of it, but you could look at the Asus M2N32 WS
which has 2 x PCI-X (3.3V) slots. It is a socket AM2
Dave Fullerton wrote:
I just verified it here as well. Running Asterisk 1.2.11 and two polycom
I'll throw in a me too here, with the addition that it also occurs
with canreinvite=no.
Regards,
Richard
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antonio wrote:
I have a problem: when i make i call from a device h323 to sip, i have no
cdr, and i don't see cdr variables for the channnel ooh323.
Anyone can help me ??
Thanx
On my system, this lives in /var/log/asterisk/cdr-csv/ast_h323.csv.
Regards,
Richard
For those enquiring last week about ooh323 not compiling with the svn
version of asterisk, the module loader changes have just been checked
into the svn version of asterisk-addons and so should now work with svn
asterisk.
Have not yet tested this.
Regards,
Richard
Martin Joseph wrote:
Do you just mean the tar balls of 1.2.9 and latest addon?
Yes. I believe the svn addons package will be updated soon.
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Tetsuya Yamamoto wrote:
I can't makel asterisk addon, asterisk-ooh323.
I use Asterisk and addons svn version.
The current svn version of asterisk has had the module loader code
redesigned and to date, the svn addons have not been updated to match
this change.
You will need to use the latest
Bruce Reeves wrote:
I'm trying to add ooh323c to my asterisk 1.2.7.1 http://1.2.7.1
install and did an svn update of asterisk-addons and followed the readme
in asterisk-ooh323c and I get through the .configure with no errors. But
make causes:
rpath /usr/local/lib -L./ooh323c/src
Bruce Reeves wrote:
I'm trying to add ooh323c to my asterisk 1.2.7.1 http://1.2.7.1
install and did an svn update of asterisk-addons and followed the readme
in asterisk-ooh323c and I get through the .configure with no errors. But
make causes:
rpath /usr/local/lib -L./ooh323c/src
Daren J. Howell DTCommunication wrote:
I have restricted the asterisk server to G711 to match the choice on the
PBX, and still same result.
I have read that either endpoint have to be either a master or slave to
communicate to each other. I see in the logs that both are shown to be a
Daren J. Howell DTCommunication wrote:
Have Asterisk connected to a H323 compatible legacy PBX using QSIG
protocol and IP trunks.
I can call to Asterisk, and from Asterisk using X-Lite softphone but
whenever either end picks up, the calls disconnects.
Try restricting both ends to one
Giordano Grandis wrote:
Hi all,
I have to bought a PCI with 4 PRI but on digium site I saw that there a
re two different kind (3,3V and 5v). What’s the difference?
33MHz 32 bit PCI slots are 5V.
PCI-X slots MAY support 5V and 3.3V depending on the age of the board.
My understanding is
Eric ManxPower Wieling wrote:
There are 2 issues here.
1) Asterisk does not have a RTP Jitter Buffer.RTP is what is used to
transport audio for SIP (and other protocols). This means that ANY
jitter on the SIP Phone - Asterisk link will cause audio problems.
This is only an issue if
Kenige Ho wrote:
the ooh323 is from Asterisk-addon-1.2.1. Is there a bug on this version
for the ooh323 and also how can i get the newer version of the
ooh323(0.8.1) to compile with? Many thanks to you all.
You will find 0.8.X in the asterisk-addons svn branch.
Regards,
Richard
Morgan Gilroy wrote:
As far as I know there will be no difference.
32bit runs natively on AMD64 chips.
The only advantage of 64bit is the extra address space and huge integers
:)
But I could be wrong, iv not done any benchmarking myself just what i
have read on the net.
I have no idea if
Mike Fedyk wrote:
Matt Riddell wrote:
I would instead recommend the SuperMicro 1U servers - we have had a
really
great run with these.
Do you use Opteron or Intel?
I would not suggest that Supermicro are in Intel's pocket, so they must
have had their fingers in their ears going,
Dean Collins wrote:
Lol, so Dell must be doing the same thing.
Did you ever consider that Supermicro are an enterprise setup to make
money, and that possibly their financial interests are served by
sticking with Intel?
Absolutely. However, it looks as though their lack of AMD product is
Steve Kann wrote:
Richard Scobie wrote:
My SVN asterisk systems use the following topologies:
1) PolycomSIP - *1 -IAX- *2 - H323 Gateway
2) PolycomSIP - *1 -IAX- *3 - Zap TDM400 Analog
3) H323 Gateway - *2 -IAX- *3 - Zap TDM400 Analog
There's a few points in here so far:
1) the new
Using a TDM400P with an FXO module and an FXS module, and a zapata.conf
with echocancel=yes above both channel definitions, is echo cancelling
applied individually to each module when a call is made out to the PSTN?
Regards,
Richard
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Kevin P. Fleming wrote:
Individually? Yes... but I don't know how else you are thinking it
would be applied.
Apologies for breaking the thread.
Just trying to get an idea of how things work together. I had considered
that in this scenario, the echo can on the FXS only has to deal with a
Is there a way to configure the IAX jitterbuffer to get the benefit of
trunktimestamps, while not having any jitterbuffering (reducing delay)?
My SVN asterisk systems use the following topologies:
1) PolycomSIP - *1 -IAX- *2 - H323 Gateway
2) PolycomSIP - *1 -IAX- *3 - Zap TDM400 Analog
3)
Adam Rybak wrote:
Hello,
i have succesfully ipgraded my system to asterisk 1.2 with OOH323C channel
driver, today i got hangup of my asterisk after this messages:
Nov 25 21:03:22 WARNING[24395] channel.c: Avoided initial deadlock for
'0x8198118', 10 retries!
This issue is currently
Today I updated a couple of TDM400 based asterisks to the latest CVS
head and started seeing the following messages. The update prior to
today was a couple of weeks ago.
-- Starting simple switch on 'Zap/6-1'
-- Executing Dial(Zap/6-1, IAX2/[EMAIL PROTECTED]/0) in new stack
--
Jon Reynolds wrote:
Hello,
I have an AAH-1.5 with a TMD400P with four lines, 8 Grandstream GXP-2000
phones, I am having echo issues on the GXP-2000 side.
Here is what I have tried so far:
The server has everything in the bios turned off except what is needed,
USB, LPT, Serial etc,etc.
It looks like there is a PE-68624 chip near each RJ-45 connector
now. Google says that it's a frequency control filter
Looking at the data sheet for this chip, it is being used as an EMI
filter, (preventing RF interference generated on the card being radiated
back out the cable).
It would
[EMAIL PROTECTED] wrote:
Put in your zapata.conf for the channel:
busydetect=yes
busypattern=1500,500
busycount=4
callprogress=no
Steve, is this a better solution than the COMPARE_TONE_AND_SILENCE
busydetect option that can be enabled in the Makefile?
Regards,
Richard
Of possible interest to people having various issues with TDM400 cards,
is that a fix has just been submitted to CVS for the issue where CPU
usage would regularly spike up to 100% with the wctdm driver loaded.
Regards,
Richard
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Matt wrote:
Am I correct in thinking that at this time the CVS-HEAD supports
Jitter Buffer for SIP on Asterisk?
No, but attached to issue 3854 you will find patches you may be able
to apply to the current CVS-Head to acheive this.
Regards,
Richard
jennyw wrote:
I've never heard about IO-APIC before, so I just did a Google search.
The articles I found say that it's an Intel thing, and, since I have an
AMD processor w/ ASUS motherboard, it's unlikely it'll work, right?
Even so, it sounds interesting. But does it apply to
Frank Tarczynski wrote:
What is the easiest way to check if the H.323 code is working? I've
edited the h323.conf and extensions.conf files but I'm sure that things
aren't right. I've tried connecting to my asterisk box via netmeeting
but I'm having much success. I don't know if my conf
I am about to add h323 to my system and although I have found
information on the Wiki, comparing the asterisk implementation to oh323,
I have not found anything about the new ooh323, which is included in the
addons.
Can anyone please compare this to the other two?
Thanks,
Richard
Matt Darnell wrote:
These phones are mentioned in the Sip 1.5 manuals, anyone know what
the differences are?
Where are you getting SIP 1.5 from?
When I log into the Polycom download area, all I can find is 1.4.1.
Regards,
Richard
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Paul Hales wrote:
It now works - but only in the latest (1.5+) firmware releases.
Where are the 1.5 releases? I see only 1.4.1 on all the Polycom sites.
Regards,
Richard
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Paul Hales wrote:
And my dreamthat one day Polycom phones will support Australian Daylight savings...
But it's only a dream.
Unless I am missing something, you don't need to dream about it - set it
in ipmid.cfg.
Look at the Sip Admim PDF for an explanation of:
Jason Leach wrote:
About every 24-48h the Zaptel FXO port crashes. If I pick up my phone
and try to make a call on the FXS port I get a hissing and squealing
sound. Seems to be right where Asterisk makes the bridge. Also
Asterisk does not answer an inbound call on the FXO port; does not
even
[EMAIL PROTECTED] wrote:
Is there a list of these anywhere? This is now the third one I've heard
of, with no documentation: lowpower (IIRC), robust and now boostringer.
Do I have to go diving in the source, or is there a Wiki I can't find?
I have only ever found the information in the driver
Ian Pattison wrote:
ringing. If I connect the GE cordless models I've been asked to use (0.1
REN) they normally will not ring... they light up and indicate there's
an incoming call and once or twice I've received 1/2 - 3/4 of a ring but
never a complete ring or multiple rings. When connected
David Brodbeck wrote:
-Original Message-
From: Scott Nelson [mailto:[EMAIL PROTECTED]
Perhaps you have an earlier hardware revision than I do; I also have
never rebooted the system. I have two TDM04Bs.
If so, they must have sold me old stock. I bought the cards less than two
months
Wilson Pickett wrote:
Now here's a thread I've been waiting to see! I have had issues with
what is considered to be a decent phone, the siemens DECT line.
Fortunately, the problem is just callerID which although annoying,
isn't mission critical, and we are in Europe. Still, the USA isn't the
Carlos Chavez wrote:
I have 2 Asterisk servers that communicate with IAX2 between them and
support multiple SIP clients each. Only one of them has Zap channels to the
PSTN. I've been having problems because the Zap channels do not hang up when
a sip client of the external server makes a
Setup:
POTS phone1 - Panasonic Analog PBX - Digium FXO - Asterisk1 - IAX2 -
Asterisk2 - Digium FXS - POTS phone2
I am attempting to balance the Digium FXO (shown above), analog audio
levels using ztmonitor -v, which the information I have found means
getting the TX and RX indicators to hit
administrator tootai wrote:
If you're telling that I have to pass parameters to module when loading,
I checked with modinfo wctdm (at office I have head version) and options
I have are those:
[EMAIL PROTECTED] asterisk]# /sbin/modinfo -p wctdm
debug int
loopcurrent int
robust int
_opermode int
Goutam Shaw wrote:
Hi
I have a strange situation. Once in a while (non-deterministic) the 2 TDM04B
cards lock up at the same time and stop processing incoming and outgoing
calls even though * shows that it is trying to communicate to ZAP channels
(at least on the outgoing). The only cure is to
Jeremy SALMON wrote:
Hi,
Just a question,
For you, what is the more reliable kernel for an asterisk prod
server...
The following 2 recent quotes from kernel developers may be worth
considering when making your decision:
After 2.6.9-ac its clear that the long 2.6.9 process worked very
Kavit Munshi wrote:
Hi,
Has anyone in australia got asterisk running on FreeBSD? how would i
pass the opermode=AUSTRALIA parameter to the wcfxs.ko module as kldload
doesnt let you pass parameters to the module like modprobe in Linux.
I tried to get the sysctl variable using sysctl -a it might
Andrei (MPI) wrote:
Richard Scobie wrote:
It is a simple one liner.
...
Index: wctdm.c
...
+ reset_spi(wc,card);
...
This is exact same patch that Digium support tried before sending me new
fxo modules. That wctdm.c patch did not help in my case.
Interesting, thanks Andrei. I have run
Steven Critchfield wrote:
Okay, link this to my rambling above and you would see that by thrashing
the disk, you are actually keeping the spindle spooled up and not
measuring the spool up draw. My guess is a spooled down machine getting
a random incoming call that then must generate ring and
Victor Rini wrote:
This has been an interesting discussion. I'll chime in with my
experience here.
I have two servers. One with the cheapest motherboard and athlon
processor I could find on Newegg.com. The other is a 1999 era
motherboard with a Via C3 processor, again a bargain basement
Rich Adamson wrote:
Have you noticed that a TDM with fxo modules is more/less stable then
a TDM with only fxs modules?
Gut feeling (no reasonable analysis at all) from various postings tend
to suggest the TDM with fxo's is less stable. Would you agree or not?
Yes.
Also, could you share the driver
Steven Critchfield wrote:
I would suggest something in a serverworks board. So far we have had a
PIII 850 on a serverworks chipset and SCSI drive running for a long
time. Our main PSTN gateway has a 418 day uptime and asterisk has been
running non-stop for nearly 20 weeks. We take nearly 500
I'm really discouraged at Digium's disinterest in this problem. I
understand they have limited resources with lots to do, but it only
takes a minute to reply to my email to say either they are aware or
not aware of the problem.
Digium are aware and have acknowledged the issue.
See my replies to
Rich Adamson wrote:
The tdm card does have some unusual issues that appear to be driver
oriented, but there are lots of folks using the card in production.
Usually in situations where the client knows how to and tolerates having
to reload drivers and/or reboot his PBX periodically...
Regards,
Rich Adamson wrote:
Is their an open Bug # that we can track against for those of us that
watch the -cvs list and have a vested interest?
I tried, see Bug 2901. Seems probable driver issues don't count as bugs.
Reported it to support, who were aware of the issue, requested login to
my machine
Steven Critchfield wrote:
On Wed, 2004-12-01 at 13:03 -0700, Michael Welter wrote:
Steven Critchfield wrote:
On Wed, 2004-12-01 at 13:36 -0600, Rich Adamson wrote:
So, isn't the issue he/I are chasing after essentially 'why is cpu consumption
jumping 30% (or 100%) every ten seconds when zaptel
Damon Estep wrote:
I'm having the same problem on my TDM40B (FXS). Unloading and
loading the modules seems to fix it temporally. Digium is
sending me a replacement.
Hopefully that will fix it.
I plan to call tech support and see what they have to say, hopefully it
is just defective and not
Gabriel Gunderson wrote:
I've installed a TDM04B and a TDM40B. I haven't plugged any lines
into them yet but I'm starting to see this in my logs...
[EMAIL PROTECTED] asterisk]# grep alarm /var/log/messages
Sep 20 09:13:22 webster kernel: Power alarm on module 1, resetting!
Sep 22 11:07:07
Luis Vazquez wrote:
Hello all
We have tested for a mounth or two an asterisk PBX using one T1 channel
bank with 24 fxs and one TDM400P digium card with 4 FXO modules.
This worked with minor problems, the most notorious being some sporadic
static noice or failure in the first FXO module on the
[EMAIL PROTECTED] wrote:
On 15 Sep 2004 at 1:52, San Singhania wrote:
Hello everyone,
I am having big problems trying to detect dtmf tones while a IVR
prompt is playing on zap channels. Sometimes the detection only starts
4-5 seconds into the prompts. Other times it works very well for the
1st
David wrote:
It sounds like my lockups may be related since my TDM422b card has the FXS FXS FXO
FXO configuration and doesn't have an FXO in position 1 either.
My card is identified in software as Rev E/F and has the wire jumper on the back.
Further investigation shows that my TDM cards have the
Maciej Kietlinski wrote:
Are the FXOs on the 2x on ports 1-2 or 3-4? Maybe it has to
do with *any* FXO on port 1...
Please get back with the list with your findings.
My experience led to a replacement from Digium, but the card is a
TDM400P with 4 FXO...now that I think of it, during
Michael George wrote:
To follow up on this, I heard back from Digium and they asked the
configuration of my TDM. It was: FXO,FXS,FXS,FXS. They said they have had
report of this configuration being a problem and that I should change it to
FXS,FXS,FXS,FXO.
Before the change the system would
Edward Eastman wrote:
I've got a TDM11B with the fxo port plugged into a standard UK BT PSTN line,
loading wcfxs with OPERMODE=UK. All's working well, except if I get an
incoming call through my bt line, and the remote party hangs up, I get
approx 20secs of the bt line hungup tone before
Leif Madsen wrote:
Would you mind maybe expanding upon the hardware configuration you are
using and why? I, and I'm sure others, are curious as to what you are
using. I haven't had to roll out any systems yet that require
multiple Digium cards, but I'm sure the information would be quite
useful
Michael George wrote:
Well, we only want 3 TDM400s: 4 FXO and 8 FXS. That will fit in nearly any
desktop PC. That's not the scale that should require multiple boxes.
But the question is where does the IRQ sharing instability creep in? I would
think that *someone* out there would have a * box
Mike Benoit wrote:
I have a P3-800 with two IDE drives in a software RAID1 configuration.
Each drive is on a separate IDE channel. Now anytime there is HD
activity, I hear beeps and cutting out on a call using the X100P
card.
I ran the zttest program, and discovered HD activity would drop the
William R Sowerbutts wrote:
I have a TDM22B (TDM400 PCI + 2xFXO + 2xFXS). One FXO is connected to the
PSTN.
When Asterisk places a call, it dials using DTMF. If I listen in on the line
during the dialing, there is a roughly one second pause between the
penultimate and the final digit --
Brian K. West wrote:
per peer
bkw
- Original Message -
From: Michael Graves [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, July 11, 2004 9:25 PM
Subject: Re: [Asterisk-Users] Stopping reinvite with IAX2?
Is this set on a per peer basis, or in the general section?
Michael
Dr. Rich Murphey wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Arjan
On Sun, 11 Jul 2004 at 15:39 -0500, Dr. Rich Murphey wrote:
You might check login class in login.conf for the user that invokes
asterisk. Setting cputime=unlimited may
Steven Critchfield wrote:
On Mon, 2004-07-12 at 02:38, Richard Scobie wrote:
A slightly similar observation, which I assume is normal as the boxes
work fine, is both my P4 2.4GHz Linux asterisks spike up to 100% load,
about every 30 seconds, with no calls being handled.
You don't mention
Martin Pycko wrote:
Well first of all if you're outside of US or callprogress-supported zones
then you can use only busydetect. And that will only work if after the
remote hangup your telco gives the fast-busy or any type of busy. You can
tweak the duration of tone/pause and increase the count
[EMAIL PROTECTED] wrote:
Does this only work with the new fxo modules?
Yes.
Richard
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Chris Stenton wrote:
Is opermode set via asterisk or do you need to do
modprobe wcfxs opermode=UK
You need to do modprobe wcfxs opermode=UK
This will only work if you have the TDM400 FXO modules. The X101P is a
600 ohm US/JATE card only.
Richard
___
Chris Stenton wrote:
Thanks for the info Rich looks like I'll have to wait for the new FXO
module. The impedence in the UK is zcomplex(2) which looks a long way away
from a straight 600 ohms.
Here is the list of zcomplex impedences
Zcomplex(1) = 150 nF // 750 ohms + 270 ohms ( European
Adam Goryachev wrote:
I might add that I has similar problems on a very frequant basis,
finally I 'accidentally' found a version of asterisk + zaptel modules
that was stable for more than 6 weeks. Eventually I asked for (and got)
a replacement card from digium with the internal power connector.
Peter Zeltins wrote:
I do not have a hardphone to play around with, but the echo is there both
with built-in audio card (SigmaTel) and Bluetooth headset. There are no
mixer settings than I can adjust as well. I'll try disabling AGC and/or
lowering mike sensitivity.
Peter
According to the
Thanks for all the help and I found the different cadences in chan_zap.c.
Richard
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I am somewhat unsure as to the definition of Distinctive Ring.
What I am trying to achieve is to have Zap connected phones (TDM400P)
ring with different cadences depending on whether the call is incoming
on the PSTN context or an IAX2 context.
Googling, I find this from Mark:
I've added
Louis-David Mitterrand wrote:
Snip
The main problem with ipsec packets is the lack of TOS support: data and
voice traffic are agregated in one stream which is opaque to external
routers.
This is not the case with FreeS/WAN, below is an excerpt from the website:
Can I use Quality of Service
Steven Critchfield wrote:
On Sat, 2003-10-04 at 12:02, Lists wrote:
But would you then not be able to use caller id from the telco?
CallerID on an analog line is sent between the first and second ring
normally. So if the requester wants callerid and first ring answer, he
will have to move to
Martin Pycko wrote:
take out usecallerid=yes in zapata.conf
Martin
Thanks Martin, but my zapata.conf is :
[channels]
echocancel=yes
echocancelwhenbridged=yes
busydetect=yes
busycount=6
context=incoming
signalling=fxs_ks
group=1
channel = 1-2
Perhaps I need a usecallerid=no in there. I'll
After some months of Make updates, I have just deleted my Zaptel and
Asterisk source directories and done cvs checkout 's of asterisk and
zaptel, in order to clean up the trees.
After re-installing, I am finding that when dialling into an X100P, that
Answer is now answering on the second
Andy Powell wrote:
Can't find the message in a search.. but below is a msg retreved from my
archive..
this is what Mark sent a little while ago
I have no idea if it actually does anything to the card, but on a modprobe I
do get a msg saying it's using CTR21
Andy
I'm in Paris right now
Andy Powell wrote:
FCC mode is for the US. CTR21 is for Europe - you even pasted the info
in your message!
Exactly, the question really is how do you change it?
modprobe wcfxo opermode=1
HTH
Andy
This switch (opermode=1) is redundant with the current X100P cards, as
it changes register
I have been running busydetect=yes, using BUSYDETECT_MARTIN and am
having hangups during calls.
The busycount=6 workaround seems to be doing the job, but I was
wondering if there was any value in using BUSYDETECT_TONEONLY or
BUSYDETECT_COMPARE_TONE_AND_SILENCE, as well as BUSYDETECT_MARTIN or
Knowing very little about Basic Rate ISDN and having spent the last
couple of hours educating myself, I thought I would seek some more
informed comment. Please go easy if this is blindingly obvious :)
I have a ZyXEL Prestige 100 ISDN Router, a stand alone relic from when
we used to access the
Martin Pycko wrote:
Do 'iax2 debug' to see more.
Martin
Thanks Martin,
As it seems as if it may be a bug, I'll get IAX2 debug output from both
*'s and put them in the bug tracker to save list clutter.
Richard
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A call is placed via IAX2 from one asterisk to another, to a TDM400
channel whose extensions.conf entry is
exten = 502,1,Dial(${COLIN})
exten = 502,2,Congestion
If this channel is already busy when called, the call is instantly
hungup, without the caller hearing the congestion tone.
The log
When starting *, I get the following when the chan_iax2.so loads:
[chan_iax2.so] = (Inter Asterisk eXchange (Ver 2))
== Manager registered action IAXpeers
== Parsing '/etc/asterisk/iax.conf': Found
WARNING[16384]: File chan_iax2.c, Line 4980 (set_config): Ignoring port
for now
== Registered
Is anyone on the list running an Asterisk system with 2 x X100P and 2 x
TDM40 (4 port) cards? I am interested in your hardware setup.
Regards,
Richard
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ProSLIC powerup on module 2
Module 2: Not installed
Module 3: Initialized
Found a Wildcard FXS: Wildcard S400P Prototype (4 modules)
Any help appreciated.
Richard Scobie
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