On Wed, 8 Dec 2004, Steve Underwood wrote:
Andrew Kohlsmith wrote:
Are you using RH's stock kernel or a plain-vanilla kernel? I have heard
nothing but bad things with Asterisk and RH's custom kernels. If you can,
try a stock 2.6.9.
It is just the kernels supplied with FC2 that have
On Tue, 7 Dec 2004, Bartosz Wegrzyn - asterisk wrote:
So besides the Budgettone 100(or any other), there is not way to force
asterisk to play a message. What about if the phone will be connected to
tdm400 port?
See immediate=yes in the zapata.conf file.
Peter
On Tue, 7 Dec 2004, Lee Howard wrote:
On 2004.12.07 10:06 Matthew Boehm wrote:
Here is the setup:
POTS - PRI - Asterisk - ATA (Fax)
The ATA is set to only 711. Asterisk's sip.conf sets this device to
only
711. Yet, faxing works less than 50% of the time.
I have a couple of
On Tue, 7 Dec 2004, John Harragin wrote:
What I have in mind is a pci card with zap-like-driver that supports digital
phones. This eliminates (is compairable to using channel bank) additional
delay and a primary echo source when both haves of a conversation are carried
on the same pair as
On Mon, 6 Dec 2004, Jerry Glomph Black wrote:
Kris, thanks for the thoughful helpful response! This makes sense in the
same way that a dialplan on a SIP phone would behave.
But... If I remove the 3-digit number (224) from the asterisk
dialplan, I have no problem dialing 2246
On Mon, 6 Dec 2004, Kris Boutilier wrote:
The originating PRI system passes the entire dialed number in the d-channel
setup frame, thus the concept of a wait time for additional digits is
meaningless. Progressive digit gathering implies that the signalling is
occuring 'in-band' as would be
On Tue, 7 Dec 2004, el Flynn wrote:
John Harragin wrote:
Are there any digital phones that run on asterisk yet? I'm talking about
non-IP phones here...
Asterisk can work with ADSI phones, more info on the wiki at
http://www.voip-info.org/wiki-ADSI
You can use isdn phones, if you want
On Sat, 4 Dec 2004, Tracy R Reed wrote:
I have created hint priorities in my dialplan:
exten = l00,hint,SIP/100
exten = 100,1,Macro(stdexten,100,SIP/100)
^
I guess it may just be a typo during retyping, but you have 'l' (lower
case L) in the hint line and a '1' (one) in the macro
On Sat, 4 Dec 2004, Rich Adamson wrote:
The mind boggles -- PRI is *always* out of band.
Looks like the command is documented in the current config samples.
I'm not knowledgable/experienced (as yet) on where it is actually used,
but just reading the comments in the config sample led me
On Sat, 4 Dec 2004, Kevin Blackham wrote:
Yeah, proper crossover cable. I've eliminated all cabling issues with
the T1 analyzer. I get a full and accurate pattern back when I test
from the cable end where it would have been connected into the T100P,
with the channel bank in loopback. The
On Wed, 1 Dec 2004, Brian C. Fertig wrote:
You can setup recording by default. This is how I have mine setup. I
don't believe the way app_queue is now you can have the agent press
something to have it start recording.
Maybe the patch in
On Wed, 1 Dec 2004, Dave Cotton wrote:
On Wed, 2004-12-01 at 12:07 +0100, Tomasz Chmielewski wrote:
What I found on voip-info.org was that I didn't have a working timer -
and I had to load ztdummy module. So I did (modprobe ztdummy), started
asterisk again, but I'm still getting the
On Wed, 1 Dec 2004, Enoch Root wrote:
I'm diagnosing a problem related to PRI card. I would
like to know the following: assuming I've got a
working PRI card and correctly installed Linux drivers
and a PRI line connected to the card, even without
starting asterisk, shouldn't I hear a ring
On Wed, 1 Dec 2004, Steve Underwood wrote:
Patrick wrote:
I'm running an Asterisk 1.0 server with 4 HFC cards and bri-stuff behind an
Anlagenanschluß with 8 B-channels in Germany. It worked fine with Deutsche
Telekom, but since we switched to Arcor nothing works at all.
After some debugging,
On Wed, 1 Dec 2004, Steve Underwood wrote:
Peter Svensson wrote:
Maybe he has NFAS (Non Facility Associated Signalling) where the D channel
on one of the BRI lines handles the signalling for the B channles on all 4
BRIs.
I think NFAS would be a pretty unusual thing for BRI. However, he
On Wed, 1 Dec 2004, Brian West wrote:
Or he has a Channelized T1 with inband signaling.
Not on four BRIs he isn't, not a T1. :)
I wonder if someone runs voice channels with inband (or robbed bit!)
signalling on an bri-interface? Now that would be a weird thing.
Peter
[moved to asterisk-users]
On Wed, 1 Dec 2004, Chris A. Icide wrote:
currently asterisk requires that you have one D channel per PRI, and that D
channel must be channel 24.
Is it possible to support one D channel for multiple spans?
It seems that you would need a bonding definition.
On Wed, 1 Dec 2004, Brian C. Fertig wrote:
But now in this instance it drops them into voice mail. Is there a way
to have them punch in there phone number so they can keep there space in
the
system? Like if they are #20 in queue when they left their # for call
back
that when they get to
On Tue, 30 Nov 2004, Karl Brose wrote:
Why don't you just set up an extension that calls the system application
to execute a Linux script
Then just make a call to that extension, perhaps use disa to
authenticate and done.
The application Authenticate() may be more suited.
Peter
On Mon, 29 Nov 2004, Matthew Marlowe wrote:
Is anyone successfully using directed call pickup with asterisk?
*8exten to only pick up that persons extension if the phone is
ringing.. It says in the wiki asterisk supports it but I can not get
it to work..
You could use app_intercept from
On Mon, 29 Nov 2004, Mark F. Vickers wrote:
According to the FAQ When you load the module and have no
circuit/channel bank the LED's should flash red
I get the knight rider lights before the module loads, but after the
modules are loaded I don't get any lights, other equipment plugged in
On Mon, 29 Nov 2004, Andrew Kohlsmith wrote:
Checking our fax logs, almost *every* company we fax (several hundred) all
connect at 14.4kbps and have the ECM or whatever it's called turned on.
This is our experience as well. Most companies here in Sweden seem to have
moved to laser faxes and
--
Peter Svensson ! Pgp key available by finger, fingerprint:
[EMAIL PROTECTED]! 8A E9 20 98 C1 FF 43 E3 07 FD B9 0A 80 72 70 AF
Remember, Luke, your source will be with you... always
On Sun, 28 Nov 2004, Brian West wrote:
I don't agree with this patch yet... It's the distro's fault for doing this
wrong and I don't feel we have to work around it. The few people I talked
to have Symlinks the build to /usr/src/linux or the like. Then again I
may be wrong anyone know what
On Sun, 28 Nov 2004, Chad Scott wrote:
On Nov 28, 2004, at 9:45 AM, Peter Svensson wrote:
Fair enough. If my unserstanding is correct perhaps someone can add a
note
to the wiki? It is not totally obvious.
Peter, why don't *you* add a note to the Wiki?
This is a community-supported
On Sun, 28 Nov 2004, Bob Goddard wrote:
On Sunday 28 November 2004 19:25, Steven P. Donegan wrote:
Well - if 2.6.etc did adopt this it isn't reflected in actual make/make
install world - i.e. nothing gets installed in /lib/modules/anywhere...
And this is with kernel source from kernel.org
On Sun, 28 Nov 2004 [EMAIL PROTECTED] wrote:
This looks like a config issue, class of service barred but getting
config information out of verizon is nearly impossible. I compared what
the Mitel is sending to asterisk (since the mitel does work with the PRI)
with what asterisk is sending and
On Sun, 28 Nov 2004, Lee wrote:
On Sat, 27 Nov 2004 20:53:24 -0500, Steve Totaro
[EMAIL PROTECTED] wrote:
Only way that I know is to open the case and look at the slot to see if
there are two dividers. I would be interested in knowing this as well.
I've seen many motherboards that
On Sun, 28 Nov 2004, Lee wrote:
So my question remains: Is PCI 2.2 a requirement to use the TDM400P
card? If so, where is this specified? If not, is there a performance
difference when using PCI 2.1?
PCI 2.2 is mostly a clerification on the 2.1 specification. One difference
that may be
On Sat, 27 Nov 2004, Rob Emanuele wrote:
I've got a pretty easy question here I can reconfigure my configs
pretty easily when I'm storing everything into a MySQL database. In the
case of using the zaptel cards and zapata.conf how would I reload the
config of an individual channel? In
On Sat, 27 Nov 2004, Roy Sigurd Karlsbakk wrote:
Change this into SetVar(_SIP_CODEC=g726) and it will work.
you sure?
sipgw1:/usr/src/asterisk # grep -r _SIP_CODEC .
sipgw1:/usr/src/asterisk #
The leading underscore means the variable will be inherited by the
outgoing channel. Did you
On Sat, 27 Nov 2004, Roy Sigurd Karlsbakk wrote:
How to implement some of the function into asterisk like *67 call
number blocking
exten = _*67*X.,1,CallerPres(32)
exten = _*67*X.,1,Dial(Zap/g1/${EXTEN:4},${TIMEOUT},${DIALOPTS}))
Do you mean CallingPres? There is more information on
On Sat, 27 Nov 2004, Peter Svensson wrote:
On Sat, 27 Nov 2004, Roy Sigurd Karlsbakk wrote:
How to implement some of the function into asterisk like *67 call
number blocking
exten = _*67*X.,1,CallerPres(32)
exten = _*67*X.,1,Dial(Zap/g1/${EXTEN:4},${TIMEOUT},${DIALOPTS}))
Do
On Sat, 27 Nov 2004, Rich Adamson wrote:
True. However, you want to distribute the clocking to _all_ your
downstream peripherials to avoid the equivalent of frame-slips. If your
cards are not clocked the same exactly you will need to invent/drop a
freme efery now and then. That is why
On Sat, 27 Nov 2004, Rich Adamson wrote:
There is a
buffer but the buffering can only handle jitter, not compensate for
frequency difference.
No, you're assuming a one-byte (or very small) buffer, and that's not
what's going on in asterisk.
You misunderstand me. I know that the
On Sat, 27 Nov 2004, Rich Adamson wrote:
You misunderstand me. I know that the buffers are larger. However, even if
they are 1 second deep they will eventually empty / overrun. There is no
way about this except to either allow data to be invented/dropped or to
keep the source and sink
On Fri, 26 Nov 2004, Francois Fernandes wrote:
- Caller checking:
If someone calls the number of the Asterisk server it should be able to
check if the guy is allowed to call this number. That means, that
asterisk should pass the number to a third parity program which decides
if the number
On Fri, 26 Nov 2004, Andrew Kohlsmith wrote:
On November 26, 2004 11:06 am, Patrick wrote:
Doesn't sync source mean that the card is generating its own clocking?
If your telco provides the clocking, the card should not.
0 = don't use the remote clock for sync (use internal clock)
1 = use
On Fri, 26 Nov 2004, Patrick wrote:
On Fri, 2004-11-26 at 11:36 -0500, Andrew Kohlsmith wrote:
[snip]
No.
0 = don't use the remote clock for sync (use internal clock)
1 = use remote clock as card's primary clock source
2 = use remote clock as card's secondary clock source
3 = ...
On Fri, 26 Nov 2004, Voip Business wrote:
Guys is there any EM available?
thought it was only fxo and fxs.
EM signalling is supported on the T1/E1 cards. There are no cards from
digium supporting analog 4-wire EM. You need to hook up a channel bank
for that at the moment.
Peter
On Fri, 26 Nov 2004, Andrew Kohlsmith wrote:
There can be only one clock and you must engineer your system such that
everything is synchronized properly. For simple systems like what we are
describing it's not difficult but when you have multiple spans coming from
multiple providers it
On Sat, 27 Nov 2004, Steve Underwood wrote:
Peter Svensson wrote:
Most providers should be synchronized to a traceable time source derived
from UTC. I.e. they should all tick exactly the same even if they are not
directly interconnected.
Uh? UTC? I think you mean derived from
On Fri, 26 Nov 2004, Dr. Fernando Macías Garza wrote:
It seems to me that if not all cards are clocked from the same source,
then each one should be able to get its own external clock. However,
card 0 has an external clock, but card 1 does not. Look at this:
[snip]
I am sure the line
On Fri, 26 Nov 2004, Rich Adamson wrote:
I've read the early posts relating to this and there still seems to be
a misunderstanding on this clock sync issue. This stuff has been around
for a long time in the telephony business, but it seems like not
many people understand it on this list.
On Thu, 25 Nov 2004, TinKoon wrote:
However, for the Asterisk implementation, unless you have a huge ups, you
will not be able to make and receive any call during power failure, since
there will be no power to the Asterisk server. And since all the incoming
lines, be it analog lines or T1/E1
On Thu, 25 Nov 2004, Alex Barnes wrote:
Sorry I dont have any answers, however I do have a question.
I was told that ISDN-30 lines do not work during power failure. Can
anyone with some better knowledge confirm or deny this?
Is this because the ISDN-30 box on the wall requires power (and
On Thu, 25 Nov 2004, Ashling O'Driscoll wrote:
So basically if I want to support approx 100 calls, I would have to
purchase a digium PRI card and then pay eircom (or whoever my service
provider is) approx 3000 a year for the PRI ISDN connection??
100 simultaneous calls would require 4 E1
On Thu, 25 Nov 2004, Rich Adamson wrote:
However, zttool reports card as Internally Clocked. No matter how I've
tried, I cannot get card 1 to clock from the external source:
Sync Source:Internally clocked
First span on card 0 is configured just the same:
On Thu, 25 Nov 2004, Colin Anderson wrote:
I have 4 gig in my * box. I'm tuning for performance and I'd like to ask
opinions:
1. asterisk -p == renice -20 ??
The -p option sets asterisk to realtime priority if possible. This is
different from the traditional unix nice levels. A program
On Wed, 24 Nov 2004, Michael Vogel wrote:
Soren Rathje schrieb:
Note: The Wildcard X100P/X101P only have FCC approval.
What does that mean for me? Is it illegal to use it in germany or do
they don't work in germany?
The X100 only support the US line impedance (600 ohm resistive). Most
On Wed, 24 Nov 2004, Andrei (MPI) wrote:
David Boyd wrote:
On Wed, 2004-11-24 at 04:14, Mike Dent wrote:
Hi,
I've recently set Asterisk up, 1.0.2 version. With 1 x X100P card and
1 SIP phone.
I've noticed some horrible buzz/rasping type of sounds! These seem to occur
when
* is trying to
On Tue, 23 Nov 2004, Ben Merrills wrote:
Is there a way to log all PRI events to a logfile?
Maybe pri intense debug span ??? is what you are after? If you set up a
logging file in /etc/asterisk/logger.conf that logs everyting you should
get all the pri events.
Peter
On Tue, 23 Nov 2004, Chad Sawyer wrote:
I have a pri comming into a t100p in my asterisk box. I have a second
t100p configured as pri_net connected to a nas server. I can route
modem calls to the NAS with no problem, but I am concerned about isdn
data connections.
Will asterisk route 64k
On Tue, 23 Nov 2004, Asterisk wrote:
At the moment, I have the following working scenario:
isdn30B-(1)te410p(2)-merdian(B)
isdn30A-meridian(a)-te410p(2)
IOW, two isdn30 lines, one going to * span 1, the other going to the
meridian (pri card a), which then is connected to * by pri card
On Mon, 22 Nov 2004, Jason Williams wrote:
I recommend you use Iax trunking rather than TDMoE this would scale better.
Using iax trunking will also loose the advantage of being tdm all the way,
i.e. low latancies. If the rest of the setup is tdm there is a lot of
value in not going to voip
On Mon, 22 Nov 2004, Kevin Brennan wrote:
Using iax trunking will also loose the advantage of being tdm all the way,
i.e. low latancies. If the rest of the setup is tdm there is a lot of
value in not going to voip for one hop.
This is what I was thinking, FAX would be more reliable (low
On Mon, 22 Nov 2004, Steve Prior wrote:
Michael Welter wrote:
echocancel=yes
echocancelwhenbridged=yes
Steve Underwood says not to use echo cancel on a fax line.
Oops, you're right. I knew I was not supposed to use echocancel, but
somehow got these two lines backwards.
Shouldn't
On Mon, 22 Nov 2004, Nick Bachmann wrote:
You know you shouldn't (can't?) use the same interface for regular IP
networking and TDMoE, right? The TDMoE should have an address-less NIC
to itself and _really_ shouldn't run through a hub (an xover would be
ideal). Bonding seems possible,
On Sat, 20 Nov 2004, Brian Roy wrote:
I would look at putting a dual monitor on her desk. You can pick up a
15 flat panel and a video card for about the same cost as the SNOM.
Not to mention, you get quite a bit more benifite from the FOP
controls than you do busy lamp fields. It's a a new
On Fri, 19 Nov 2004, Michael Devenijn wrote:
We are located in Belgium and just ordered a PRA line, the telco asked
the following questions :
- 120 or 75 ohm ?
120 ohm is delivered over two balanced twisted pairs and normally
terminated in an rj45. This is what you need for the
On Thu, 18 Nov 2004, Rich Adamson wrote:
Examples:
1. two-wire analog pstn lines: as soon as current draw is sensed by
the central office, answer supervision is generated by that central
office, period. It has nothing to do with whether * handled it or
whether an analog phone is hanging on
On Wed, 17 Nov 2004, Steven Critchfield wrote:
On Tue, 2004-11-16 at 23:34 -0700, Chris Modesitt wrote:
Thanks for your feedback, after I restarted Asterisk the card came up as
expected. However I am still seeing these WARNINGS when I reload *, to be
clear I have not made any additional
On Wed, 17 Nov 2004, Jason Becker wrote:
On our current phones (Iwatsu) we have a button on the
phones for each extension that lights up when that
extension is ringing or is in a call, so I can see at
a glance if one of my coworkers is on the phone before
I go barging into his office.
On Wed, 17 Nov 2004, Thomas Hutton wrote:
Question: Does anyone know of a lightweight popup method to put an
incoming call ID string on a client machine? Something as simple as
winpopup would work great- for example: I have a call coming in on Zap/4
but the phone on Zap/4 doesn't have a call
On Wed, 17 Nov 2004, Joe Greco wrote:
I don't think this is really a key system. AFAIK a traditional key system
has a one-to-one mapping between lines and the buttons. Some pbx:es offer
a mode where each *extension* is / can be represented by a button. This is
called a Busy Light Field
On Thu, 18 Nov 2004, Daniel wrote:
On Thu, 2004-11-18 at 12:05, Chad Scott wrote:
You *can* play a welcome message without answering the line, however,
this doesn't always work. eg, I tried this config on my PRI in Australia
(Telstra) and:
a) Calling from a standard analog line I got my
On Mon, 15 Nov 2004, Jim Dossey wrote:
I have a client who currently has a Toshiba PBX. We are trying to
replace it with an Asterisk system. One of the features that they have
on their current PBX is the ability to select a POTS line by pressing a
button on their phones. They have 10 POTS
On Tue, 16 Nov 2004, Tobias Jönsson wrote:
On Mon, 15 Nov 2004, Jason Williams wrote:
After the Authenticte why not do a Playtones(Dial) this will give
dialtone
The dialtone won't stop after pressing first digit then. If course you can
have an X extension that will do a StopPlaytones
On Wed, 17 Nov 2004, Matt Riddell wrote:
Régis MARTIN wrote:
When I first read the answer, I look at it like another quick answer with no
understanding of my problem.
Aha! But you didn't notice that it was Brian West (bkw) who gave you
the answer!
He is one of few able to give
On Wed, 17 Nov 2004, Matt Riddell wrote:
Peter Svensson wrote:
I guess you just have to know that Brian is a bit trigger happy sometimes.
It has it's ups and downs. Things get fixed quickly, but sometimes his
instinct is wrong.
I was beginning to think he wasn't human. Thanks
On Mon, 15 Nov 2004, Peter Osborne wrote:
I am using the Asterisk Manager API to originate calls and it is working
well,
when a call is placed the local phone rings, once you pick it up you can here
the call ringing the other end. Now, I am using Polycom IP 300 and I have
them setup to
On Mon, 15 Nov 2004, Brian West wrote:
Ok to cut confusion here
Its:
Variable: _ALERT_INFO
Value: somevalue
Its always var/val via manager.
Not in the Originate action it isn't. This is what both the help
show manager command originate
say and what reading the source indicates.
On Tue, 9 Nov 2004, Kristian Kielhofner wrote:
G.711 is a standard that defines Ulaw and Alaw, commonly called Ulaw
and Alaw. But last I checked Meetme transcodes all codecs to Ulaw for
the purposes of the conference. So, I suppose G.711u would be your best
bet for low processor
On Tue, 9 Nov 2004, Henry Devito wrote:
HI I am trying to use the outcall going by the wiki.(
http://www.voip-info.org/wiki-Asterisk+auto-dial+out) But I keep getting the
errors below. Here is a sample of a callout file. What am I doing wrong?
Begin Outgoing.call
Channel: sip/2075
On Fri, 5 Nov 2004, Ryan Thrash wrote:
What about an expensive Supermicro dual Xeon PCI-X system with 1GB ECC
RAM and a hardware RAID controller (it was SATA, though)?
Echo was noticeable even on SIP-to-SIP calls internally with the
system, with all sorts fo tweaks to tx/rx gain.
On Sat, 6 Nov 2004, William M. Sandiford wrote:
Hello All:
I need some help. I am trying to configure * so that users that are
placed in a call are able to break out of the queue and go to voicemail
if they no longer wish to wait in the queue. I read the cmd options for
the Queue command
On Sat, 6 Nov 2004, William M. Sandiford wrote:
Excuse the newbie nature of the question, but can you elaborate a little
further. Sorry...I am pretty new
There is a block in the queues.conf.sample file in the Asterisk
distribution that reads:
; A context may be specified, in which if the
On Sun, 7 Nov 2004, Reid A. Forrest wrote:
Currently, our office phone systems have 6 outside lines
coming in. The
actual phones have lights ( indicators ) for these lines, so matter
where you are in the office, you can look at the phones and see that
someone is on line #2 ( for
On Fri, 5 Nov 2004, Kurt Bauer wrote:
--On Thursday, November 04, 2004 04:41:53 PM +0100 Peter Svensson
[EMAIL PROTECTED] wrote:
On Thu, 4 Nov 2004, Kurt Bauer wrote:
connection is to a Ericsson MD110 wich is set as network, * is set as
CPE.
Have you set the span as the timing
On Fri, 5 Nov 2004, Matthew Marlowe wrote:
This seem to be fixed in CVS 11/05 - Altho ALERT_INFO is still broken
in CVS 11/05
Isn't this an effect of the new automatic variable inheritance? Since
ALERT_INFO is used in the called channel you would have to set _ALERT_INFO
instead of
On Thu, 4 Nov 2004, Kurt Bauer wrote:
Hi list,
every now and then I get the following message in my * logs:
chan_zap.c:7379 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary
D-channel of span 1
As this is only a notice and voice worked quite well, despite the messages,
I didn't
On Thu, 4 Nov 2004, Kurt Bauer wrote:
Is your timing source set correctly? If you are connecting to the pstn
the pstn connection should be the primary timing source.
connection is to a Ericsson MD110 wich is set as network, * is set as CPE.
Have you set the span as the timing source?
On Thu, 4 Nov 2004, Nate Carlson wrote:
Area you using a PRI line, or what?
If a PRI, you need your provider to allow you to set the outgoing CallerID
to whatever you'd like, instead of just one of your own numbers.
If BRI, Analog, etc, I don't think there is a way to set your own
On Tue, 2 Nov 2004, steve szmidt wrote:
It is quite true for some classes of batteries. E.g. some Li-ion batteries
will explode if charged (or in the case of rechargeable batteries charged
with the wrong voltage / polarity). They pack quite a punch as well. The
normal household alkaline
On Mon, 1 Nov 2004, Luís Palma wrote:
I've been digging around /zaptel/zonedata.c file which has the
different frequency tones per country, and I would like to know the
purpose of the following fields in the struct data defined there.
For example in US data we have:
{ 0, us, United States
On Mon, 1 Nov 2004, Jon Lawrence wrote:
There isn't a digium solution to connect to POTS lines in the UK other than
X100P's, and I for one can't live without callerID - I'm even considering
going across to ISDN so that callerID continues to work with future *
versions.
There are a lot of
On Sun, 31 Oct 2004, Remco Barende wrote:
I will probably order the base station, it seems like an almost ideal
solution to connect phones to a voip pabx. I would not prefer a pci card
solution personally, anything connected to the network doesn't cause irq
headaches :)
On the other hand
On Fri, 29 Oct 2004 [EMAIL PROTECTED] wrote:
Does anybody have the miracle setting required to get the video portion of
eyebeam from Xten to actually work.
All I get is blank screen.
Last time I looked it seemed that Asterisk did not allow the addition of
the video stream after the call
On Fri, 29 Oct 2004, Derek Conniffe wrote:
I've been wondering about this too. I've now got two telephone systems
side by side - my old system is an analogue PBX connected to ZyXel
routers (Prestige 100s) which give me POTS lines from the ISDN NT1 boxes
and its only since I've started
On Fri, 29 Oct 2004, Derek Conniffe wrote:
I'm telephone company connections only (due to only having a 64Kbps
fixed internet connection).
Its definitely relating to the far end because it only happens when I'm
talking to a person using an analogue line on the far end but the
question
On Thu, 28 Oct 2004, Steve Underwood wrote:
The original poster is asking about 2-way telephony. All the normal
forms of telephony on T1 can support 2-way operation, and Asterisk
supports them. However, ISDN and SS7 are more robust than the robbed bit
signalled forms, like wink start.
On Thu, 28 Oct 2004, Ashish Shinde wrote:
I need to interface the wildcard t100p with the Simens HiPath 3000
PBX's T1 interface. I tried all the possible options for framing and
signalling, but could get the card to interface correctly. The LED on
the card always shows error. I tried
I really don't know who supplies the clocking. How to find that out? I
did use a T1 cross - over cable and I tried all possible options for
framing and coding in zaptel.conf. Tried ztcfg too. It doesn't
complain. Is there any way to find out the framing and coding
Who is the network end of
On Thu, 28 Oct 2004, Steve Underwood wrote:
Stephen David wrote:
i don't have a specific bug in mind, i was just wondering WHY call progress doesn't
work so well -- in particular, on analog lines. ie. is it a hardware or software
problem (or both). with more info, i'd like to help to work
On Thu, 28 Oct 2004, Nicolás Gudiño wrote:
Asterisk detects hangups with busydetect and busycount just fine. At
least for me. The problem is ANSWER detection for billing purposes.
Does asterisk support polarity reversal detection for answer/disconnect
supervision? For a quick look at the
On Fri, 22 Oct 2004, Neill Wilkinson wrote:
All,
newbie to Asterisk and just trying to get a load of bits together
including PSTN interface using Digium Quad E1 interfaces using EuroISDN.
Question can I/how do I get access to the ISDN reason codes for call
disconnect? The purpose is to
On Fri, 22 Oct 2004, joachim wrote:
I was thinking of the answered statuses. That g was not working for me last
time i checked.
Can you post your Dial line (and preferably the lines after that as well)?
The 'g' option should work. It does for us, but we are a bit behind HEAD.
Peter
On Tue, 19 Oct 2004, Michael Loftis wrote:
We figured it out. Well I did. You pretty much have to use
pridialplan=unknown in zapata.conf it looks like, with the others libpri
seems to try to get stupid with the actual digits sent/coded to the remote
switch.
Also, your telco may
On Tue, 19 Oct 2004, David H Hickman wrote:
This tends to be a religious issue. I guess I am an older admin. :)
I come from the school of thought that it is a good idea to
reboot a server that is not meant to be used interactivly (console or
terminal) on a schedule. Most software does
On Tue, 19 Oct 2004 [EMAIL PROTECTED] wrote:
I'm at work now and don't have my access to my asterisk box (which isn't
much use as I can't post debug data or other lines from the config files).
Just wondered if anyone had done this and where I was going wrong (I have
tried different number
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