This is called overlapdial in zaptel. It works on all zaptel cards i've
tested so far, also the zapbri cards. Chan_capi supports it aswell...
(called Early B3 iirc), and with the iaxy it is no problem either...
(it starts a call when picking up the hook)
It does not correctly work with IAX.
Codecs are patentable and patented worldwide.
I'm not a lawyer --- but patents are not valid world-wide. Some countries
have mutual patent agreements, other countries haven't. Some countries
permit patents on everything, some are more restrict.
___
Hi,
-Original Message-
We use the SNOM's. They are excellent, their support is excellent and
the development of new features in the firmware is very fast.
I have one
major gripe about them, the speaker is not good enough for long
conversations.
And in the case of the Snom 105,
Hi,
-Original Message-
Try
notransfer=no in iax.conf
Hmm, I assume you mean transfer=no, but that also keeps the voice flow
through the machine. Would IAX2 support having signalling going through all
machines and voice data through the shortest path, more or less like how SIP
works,
Hi-
I have an FXO module in my TDM400P configured to receive caller*id (see
zapata.conf below). I get a curious behavior: When I call this line
with my cell phone, I see caller ID received just fine, with no
warnings or errors.. When I call from another landline, I get different
results:
Would IAX2 support having signalling going through
all machines and voice data through the shortest path
No, Signalling+Voice is tightly coupled.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Holger Schurig wrote:
Would IAX2 support having signalling going through
all machines and voice data through the shortest path
No, Signalling+Voice is tightly coupled.
To my knowledge, IAX2 will take shortest route possible. I.e.
A call from UA A to UA C through server B will switch from
Hi,
-Original Message-
No, Signalling+Voice is tightly coupled.
To my knowledge, IAX2 will take shortest route possible. I.e.
A call from UA A to UA C through server B will switch from
original path
(ABC) to (AC) and this is default behaviour unless notransfer=yes
exist in
Just dialed (or attempted to) a 800 number, still down
At 17:20 6/8/2004, you wrote:
Heh..yea, I made sure I did a search through the archives before posting
it :) (not that I'm complaining)
The weird thing though is that I _am_ able to call digium's iaxtel
number..
-Mark
Holger Schurig wrote:
Codecs are patentable and patented worldwide.
I'm not a lawyer --- but patents are not valid world-wide. Some countries
have mutual patent agreements, other countries haven't. Some countries
permit patents on everything, some are more restrict.
I didn't say one
Hi,
I have built asterisk starting from bri-stuff (latest version) and
following exactly the istructions I found at
http://www.voip-info.org/wiki-Asterisk+zaphfc+install
Unfortunately even if the building has worked ok when I do
Make loadNT
I receive an error span is not present, so it seems
tmpm wrote:
Just dialed (or attempted to) a 800 number, still down
you could always enable enum lookups and use either the freenum.org zone
or e164.org zone as they both contain IAX2 and SIP URLs for north
american and other countries toll free numbers...
--
Best regards,
Duane
You may want to take a look t.38, t.39 which are the fax/ip/smtp
standards. If Asterisk could be made to do this, then it would join the
mainstream and inter-op with cisco gw's and such handling this sort of
thing automagically for the billions of voice/fax minutes served.
-Original
On Wed, 9 Jun 2004, Duane wrote:
How's it a DNS hack when the SRV record includes the A record
Because you're having to create subdoms and use them for your SIP
addresses, rather than using the facilities that SIP provides to allow you
to use your domain, just as you would for email. Yes you
Darren Edmundson wrote:
Unfortunately, it seems from my bugreport that the powers that be are as
spit over this as we are, which is a shame - I'd have hoped that RFC
compliance was an obvious aim for any piece of software
*sigh* I have said time and time again, when it's not disabled in
Darren Edmundson wrote:
...until you need to place a call to someone who *has* followed the
standard.
I'm hedging my bets and advertising the A record, if at a later date I
introduce an SRV record the A record will still be valid, and will be
identical to the current one, oh look hasn't
Not only you would like it.
- Original Message -
From: hskim [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, June 09, 2004 2:24 AM
Subject: Re: [Asterisk-Users] E100P R2 signaling
Steve,
I'm going to use e100p for an ivr system.
Currently local telco only supports r2
It works pretty much out of the box.
On he as5300:
*Setup a user (used by asterisk for dial in)
* setup the voip and pots dialpeers
On asterisk:
* In sip.conf setup a user for the router
* In extensions.conf, setup the dialing plan, sending the # to the
router:
exten =
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Word of advice... Buy the digium card and not a clone if you want
support...
- -jwb
On Wednesday 09 June 2004 01:20 am, bino_oetomo wrote:
Dear All.
I'm very new to CT, but attracted by asterisk.
I plan to start learn to build IVR, based on
Darren Edmundson wrote:
On Wed, 9 Jun 2004, Duane wrote:
How's it a DNS hack when the SRV record includes the A record
Because you're having to create subdoms and use them for your
SIP addresses,
This is not a hack, this is standard DNS practice. The same is done for a
lot of
On Tue, Jun 08, 2004 at 07:06:22PM -0700, George Pajari wrote:
http://www.nwfusion.com/columnists/2004/0607faceoffyes.html
There are very valid arguments in the contra argument. If you have
existing equipment it's all about integration. Traditional telcos are
moving to VoIP as are enterprise
Pablo Endres wrote:
It works pretty much out of the box.
Pretty much? He has modems in that box.
I'm no Cisco expert, but aren't modems different than voice resources
(DSPs)??
Jeremy McNamara
On he as5300:
*Setup a user (used by asterisk for dial in)
* setup the voip and pots dialpeers
On
Steve Kennedy wrote:
On Tue, Jun 08, 2004 at 07:06:22PM -0700, George Pajari wrote:
http://www.nwfusion.com/columnists/2004/0607faceoffyes.html
There are very valid arguments in the contra argument. If you have
existing equipment it's all about integration. Traditional telcos are
moving to
This is the traditional view of telecoms in large organisations. However
it seems in a lot of large companies they are dumping their existing
telecoms wholesale for an IP solution, on a site by site basis, as soon
as the maintainence contract renewal comes around. It surprises me to
see
- Original Message -
From: James W. Brinkerhoff [EMAIL PROTECTED]
Word of advice... Buy the digium card and not a clone if you want
support...
So ,
Can I use Digium- X100P to start learn to build IVR ?
Or, Can I just use a ASTERISK client application for this purpose ?
Sincerely
On Wednesday 09 June 2004 02:03, brian k. west wrote:
search for app_valetparking and hope its still out there somewhere :)
How does that fix the problem? He still needs # to access ValetParking and
thus loses the use of # for remote IVR apps.
All ValetParking gets him is a known parking
Chris Bond wrote:
I think one thing * is lacking at the moment is a web interface to manage
and add users and do anything you can do via a shell interface. If it had
that but on a simplified level (oblessly you can have an advanced mode too).
The power of asterisk comes from its method of
Is anyone else having problems right now.
Only about half the times that I call my DID does it go through. I am not
getting a fast busy either, I get dead air.
When the call does go through it is VERY
choppy.
Thanks,
Steve Totaro
www.totarotechnologies.com
Daniel Jimenez wrote:
Hey all,
I have an as5300 I use for dial in customers, we have 4 PRIs on it.
Which model - 5300 or 5350.
5300 have different DSP blades for dial-up/in and VoIP
We have a few free channels on it. I'm wondering if I setup SIP on the
as5300 I can have asterisk use the free
The power of asterisk comes from its method of config. If one wraps it
with a GUI one will inherently limit the flexibility.
Then since the GUI is what gets 'seen' people ~may~ take the lack of
flexibility or even just the look and flow of the GUI to be a reflection
on the power of
On Monday 07 June 2004 20:44, Steve Underwood wrote:
A lot of people report no problems with HT turned on, but you have to
look at these reports carefully. A lot of people have no zaptel hardware
in their system. That seems OK with HT on. Some people with zaptel
hardware use it in very simple
Any chance of publishing source code as this is a good starting point for
many applications.
At 16:54 08/06/2004 -0700, you wrote:
I just uploaded a beta CallerID program.
It talks through the Asterisk Manager .
Pretty self expanatory for setup and configure.
Please Let me know what you think.
Time for Duane to start implementing DNS SRV, since it's from now on is turned on
by default in CVS head.
Thank you, Mark!
/O ;-)
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or
I think one thing * is lacking at the moment is a web interface to
manage and add users and do anything you can do via a shell interface.
If it had that but on a simplified level (oblessly you can have an
advanced mode too).
There are many of them, and most of them aren't finished.
The
Title: RE: [Asterisk-Users] NetworkWorld article on Open Source Telephony
I agree, any platform suffers when it is extremely difficult to implement. What we need is an interface that does everything we need and shows what asterisk is capable of, a lot of features will go unused because you
Olle E. Johansson wrote:
Time for Duane to start implementing DNS SRV, since it's from now on is
turned on
by default in CVS head.
Unless you're planning on breaking other standards my A records will
keep on working just fine :)
--
Best regards,
Duane
http://www.cacert.org - Free Security
I like the way the 3com NBX system works. The web interface is pretty
intuitive. Adding users and devices is a snap through the GUI but to get to
the real meat you have to edit the dial plan. To do this, you download a
text file to your desktop, edit it, then upload it again.
- Original
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Adam Hart wrote:
| Jason A. Pattie wrote:
|
| |
| | One workaround is to use Firefly, but that may not be for everyone?
|
| True. I almost got it working under Wine, though. Kept dumping files
| into C:\. Probably just means I don't have the
Hi:
Is DynExtebDB module still working??
--
JO
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
I tried some combinations of setup seen in some postings
and didn't get success on this yet.
I have grandstream phones outside the network trying to
call an * server inside my network through NAT/Iptables.
The problem that I'm facing is one-way audio.
Any suggestion?
Thanks,
Isamar
been playing around with the Pulver firmware WF.00.11/B.00.13/Apr 07
2004 and its not better in any way. Anbody made some progress with that
issue? I guess we will have to wait for ZyXEL releasing a real
production FW.
cheers
Dominique
Dominique Kull wrote:
Thanks for your replies. The hangup
On Wed, 2004-06-09 at 08:38, Andrew Kohlsmith wrote:
On Monday 07 June 2004 09:09, Steven Critchfield wrote:
So once again, have you verified with zttool where the card is getting
its timing from? If it says internal, or any non PSTN connected span,
you will have found the error. and will
Anyone actually got this working with asterisk ?
I have read posts that it is possible with capi and the diva server cards.
Regular diva and diva pro both claim to support NI-1 and NI-2 and CAPI -
will they work as well ?
Has anyone actually got it working ? Forget the should and could part, I
Steve Underwood wrote:
I didn't say one patent covered all the world. I said the patents on
codecs exist all over the world. WIPO is simplifying this a bit, but its
still pretty expensive to get a patent everywhere. I know of no country
where the key aspects of a codec cannot be patented.
Which way is the audio working?
-brian
Isamar Maia wrote:
I tried some combinations of setup seen in some postings
and didn't get success on this yet.
I have grandstream phones outside the network trying to
call an * server inside my network through NAT/Iptables.
The problem that I'm facing is
We have been getting email asking if your will be available.
We are considering publishing source.
But havnt made a decision either way.
It can be swayed to publishing it if we can get donations on the web
site to cover our time.
If some one would like to donate a Wildcard TE405P and/or TDM400P
I like the way the 3com NBX system works. The web interface is pretty
intuitive. Adding users and devices is a snap through the GUI but to get
to
the real meat you have to edit the dial plan. To do this, you download a
text file to your desktop, edit it, then upload it again.
Ditto on the
On Wed, Jun 09, 2004 at 11:24:11AM -0400, Jon Pounder said:
Also for any ISDN gurus out there - is there a simple way to loop back BRI
so I can call from one B to the other for testing with the proper
signalling for National to see if asterisk actually works without
committing to ordering a
Hi all,
I have decided to send this e-mail because you are the
developer of Asterisk .
We are developing a phone
system using Asterisk as the VOIP gateway with 1 t410 PRI card and Sip Express
Router as the proxy server but we have a problem. Our phone system setup like
this:
SIP
Anyone actually got this working with asterisk ?
Yupbut it was a year ago, so I've forgotten the specifics.
I have read posts that it is possible with capi and the diva server cards.
Regular diva and diva pro both claim to support NI-1 and NI-2 and CAPI -
will they work as well ?
I used
Hi I am trying to use firefly as an IAX client with
asterisk.
If I populate the iax.conf with the user info, I can
make calls successfuly. However if I use MYSQL and
populate the records for each users I get an error
saying
Rejected connect attempt from 8.1.2.1
I am looked in the lists to see
Ditto on Avaya...
My $75,000 Avaya Definity G3Si has a GUI that simply wrapps the CLI. If you
don't understand the CLI you can't use the GUI.
Their Java apps for their interaction center / ip office suck, I prefer the
.conf solution. Easier version control and more concrete.
TL
Hello Martin,
how would you like to integrate? PRI (E1) or BRI (ISDN)?
We have a running integration with PRI and a Hicom 150..
If you have any questions...
Bye
Felix
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Martin Mielke
Sent:
On Wednesday 09 June 2004 11:17, Steven Critchfield wrote:
Will that give you red alarms though? Clock slips, sure, but RA?
If it slips enough it will loose sync and you will get a red alarm
Yeah but you'd have to have a pretty out-of-spec 8kHz clock to do that... I'm
no expert, I'm just
I understand from the archives that * does this occassionally, but I'm trying
to figure out why.
* didn't do this at all for two days, and then it's gone and done it 3 times
in the past hour. It does not seem to be affecting calls, I'm just curious
as to the reasoning behind the B channel
*SMACK* no you don't just use the native sip transfer to park it. :)
bkw
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith
Sent: Wednesday, June 09, 2004 8:30 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users]
On Wed, Jun 09, 2004 at 11:03:50AM -0500, Rick Smith said:
You are receiving this message because of a message you sent to Rick Smith.
Rick Smith is using a new anti-spam webbased service called SpamRival.com. As such
he/she only accepts email from authenticated users. This is to insure
Hi everyone,
I've just got my X100P card installed and working but there seems to be an
issue with hang-up supervision.
If I stuff a call out over the X100P card onto the PSTN that's fine. When I
hang up the SIP phone the PSTN call ends. If I receive a call from the
PSTN, it's answered and
Actually, here in Maryland ISDN BRI is cheaper than POTS. POTS business
lines are like $20 each, and caller-id is around $8.50 per line. So two
lines with caller-id are about $57. On the other hand, a BRI, which has
awesome voice quality and includes CLID is $45. If you get a residential
ISDN
It seems that VoicePulse is down, incoming calls get busy, outgoing are
timing out as * can not register with them.
Could anyone confirm that?
Thanks,
Wojtek
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
A GUI for asterisk is really not that hard to make if what you
do is model the config files in the db. Once it's in the
DB all you have to do is the right queries to rebuild the files
when changes are made.
Then the gui can be totaly adaptable for any use.
I think it is posible to give a good
Hi,
This post may be a little off topic but I've seen lots of good ideas com
from the list, so here it goes:
I'm in the need of replacing Cisco Call Manager 3.2 that
I have working as my primary GK. I thought I could replace
it with gnugk, but it lackes some of the functionality that
I need:
Hi,
I compiled updated version from cvs, and run,
When reload the config file, it shows,
Jun 10 00:48:53 NOTICE[1200825920]: indications.c:396
ast_unregister_indication_country: Removed default indication country 'uk'
Jun 10 00:48:53 WARNING[1116941120]: acl.c:195 ast_get_ip: Unable to lookup
On Wednesday 09 June 2004 12:21, Walt Reed wrote:
You have got to be shitting me. *PLONK* goes Rick.
Yup, I got that too, and that was my response as well. Screw 'im. I won't
play that game.
Regards,
Andrew
___
Asterisk-Users mailing list
[EMAIL
ePyron Felix Deierlein wrote:
Hello Martin,
how would you like to integrate? PRI (E1) or BRI (ISDN)?
Besides of making calls with VoIP from PC to PC, we'd like that our
people abroad could dial company internal extensions through Asterisk
using a SIP client. On a second approach, the same
On Wed, Jun 09, 2004 at 11:03:50AM -0500, Rick Smith said:
You are receiving this message because of a message you sent to Rick
Smith.
Rick Smith is using a new anti-spam webbased service called
SpamRival.com. As such he/she only accepts email from
authenticated users.
Make your
I just downloaded the latest *CVS onto a freshly installed Redhat 9
system, and I noticed that the compile of asterisk-addons fails as
follows:
# make clean ; make install
rm -f *.so *.o .depend
./mkdep -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql `ls
*.c`
cc -fPIC -I../asterisk
I was having the same problem. Only about half the incoming calls were
getting through, then completely down. Now it seems to be back up.
While it was down I ugraded to today's head and it started working again.
The two are probably unrelated.
- Original Message -
From: Wojciech Tryc
Regular diva and diva pro both claim to support NI-1 and NI-2 and CAPI -
will they work as well ?
I used CAPI plus a single port Diva server card. The passive cards
don't have the required Linux CAPI support from my understanding.
From my read over the eicon site the basic difference is
On Wed, Jun 09, 2004 at 11:03:50AM -0500, Rick Smith said:
You are receiving this message because of a message you sent to Rick
Smith.
Rick Smith is using a new anti-spam webbased service called
SpamRival.com. As such he/she only accepts email from
authenticated users.
Make your
Wouldn't work on an ATA device
brian wrote:
*SMACK* no you don't just use the native sip transfer to park it. :)
bkw
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith
Sent: Wednesday, June 09, 2004 8:30 AM
To:
Hi there,
I'm going mad at this:
Asterisk with one HFC isdn card, using the zaptel driver bristuff
All works ok, but voice coming in/out of the isdn card is out of sync,
squelky and disrupted, UNTIL I PUT SOME LOAD TO THE PC, let say
launching xwindows.
I noticed this:
Strong HDD activity =
Hi there, im having some troubles with my asterisk service, sometimes when im trying to make an outbound call, to any of the phones configured on the asterisk box,it enters inmediatly to voicemail and then hungs up. After that its necessary tostopthe service and putting up again manually.
Here is
On Wed, 2004-06-09 at 11:07, Andrew Kohlsmith wrote:
On Wednesday 09 June 2004 11:17, Steven Critchfield wrote:
Will that give you red alarms though? Clock slips, sure, but RA?
If it slips enough it will loose sync and you will get a red alarm
Yeah but you'd have to have a pretty
Hello
all,
I have a fax machine
attached to one of the FXS ports on my channel bank running into one of the
spans of my TE405P. Every time I try to send a fax, I get the error "Fax
detected, but no fax extension" in asterisk. Does anyone know why this
would happen? The only other
Of course, right now things
like * do not have an adequate reputation to pick up much of that
business. There is, however, a preparedness there for radical change.
When you are able to purchase support contracts on Asterisk (E.g. Yearly
(not hourly)) * will gain a lot of momentum. There
Yet again.. *SMACK* yes it does.
bkw
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Stephen Rosebush
Sent: Wednesday, June 09, 2004 12:47 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Sending # and Asterisk Transfer
On Wed, 2004-06-09 at 11:12, Andrew Kohlsmith wrote:
I understand from the archives that * does this occassionally, but I'm trying
to figure out why.
* didn't do this at all for two days, and then it's gone and done it 3 times
in the past hour. It does not seem to be affecting calls, I'm
On Wed, 9 Jun 2004, Alessio Focardi wrote:
Asterisk with one HFC isdn card, using the zaptel driver bristuff
All works ok, but voice coming in/out of the isdn card is out of sync,
squelky and disrupted, UNTIL I PUT SOME LOAD TO THE PC, let say
launching xwindows.
Alessio,
When I was having
What I don't understand is why people think that FLASH on a SIP ATA-like
device is NOT a SIP transfer. Weird.
On Wed, 2004-06-09 at 13:09, brian wrote:
Yet again.. *SMACK* yes it does.
bkw
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On
Hi Patrick
Patrick J. Conroy wrote:
Hello all,
I have a fax machine attached to one of the FXS ports on my channel bank
running into one of the spans of my TE405P. Every time I try to send a
fax, I get the error Fax detected, but no fax extension in asterisk.
Does anyone know why this would
Jeremy McNamara wrote:
Chris Bond wrote:
I think one thing * is lacking at the moment is a web interface to
manage and add users and do anything you can do via a shell
interface. If it had that but on a simplified level (oblessly you
can have an advanced mode too).
The power of
What is the actual hdparm command you are using?
On Wed, 2004-06-09 at 12:13, Brent Franks wrote:
On Wed, 9 Jun 2004, Alessio Focardi wrote:
Asterisk with one HFC isdn card, using the zaptel driver bristuff
All works ok, but voice coming in/out of the isdn card is out of sync,
squelky
Hi,
It would be nice, if you can add the called number (DNIS) info in your
CallerID Application. So we can have both info on the screen.
Regards
Azher
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Kyle Hagan
Sent: Wednesday, June 09,
I have just finished installing all the pieces of Redhat Linux 9
(2.4.20-8), Asterisk-0.9.1, Gnophone-0.2.4 on my IBM T30.
Audio card is SoundMAX Integrated Digital Audio. Not sure what the chips
are. But everything works on both W/XP and RH9. (Machine is obviously
dual boot)
Everything starts,
On Wednesday 09 June 2004 14:00, Steven Critchfield wrote:
My suggestion still is valid though as a simple way of removing a
variable from the equation.
Oh, absolutely -- I wasn't trying to suggest otherwise. I was merely trying
to satisfy my own curiosity. :-)
-A.
Florian Overkamp wrote:
Hi,
-Original Message-
No, Signalling+Voice is tightly coupled.
To my knowledge, IAX2 will take shortest route possible. I.e.
A call from UA A to UA C through server B will switch from
original path
(ABC) to (AC) and this is default behaviour unless
I ma new to Asterisk.
I'd like to setup * as voicemail system for SER.
Let's say I have an phone number registered in ser as 5554321.
When somebody dial to ser for this number and nobody answer, the ser will
forward the call to asterisk and get into voicemail box 5554321. I already
have
Thanks for the tip. will look into that...
At 05:47 6/9/2004, you wrote:
tmpm wrote:
Just dialed (or attempted to) a 800 number, still down
you could always enable enum lookups and use either the freenum.org zone
or e164.org zone as they both contain IAX2 and SIP URLs for north american
and
I'm looking for a way to detect when a VOIP provider is unable to complete a
call and thus try another VOIP provider (failover/backup type situation).
using qualify is NOT sufficient, since the provider could very well be
reachable but not be able to complete the call for other reasons.
A
Rich,
I'm working on the tutorial agenda today. I've schedule a tutorial for you. Will need
a photograph
as well as some text for the web page that describes our tutorials.
Please read
http://www.astricon.net/astricon2004/tutorials.shtml
And you'll see what I need from you.
If you have any
Need a good document for the Manager API before a GUI can be written!!!;)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Pablo Endres
Sent: Wednesday, June 09, 2004 11:35 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] NetworkWorld
It works without a problem for me, but that does not help you.
Jason
At 17:29 09/06/2004 +0100, you wrote:
Hi everyone,
I've just got my X100P card installed and working but there seems to be an
issue with hang-up supervision.
If I stuff a call out over the X100P card onto the PSTN that's fine.
I just tried them, and I got to Versleazon just fine
BTW, ATT wireless is having a HUGE echo problem today..my cellphone was
abominable three times today
Any one got any word from Iaxtel? Did they die and fall into the switch?
All I get is fast trunk busy...Theyre still down as of this
I have a Grandstream ATA286 and still can not find a way of issuing '#'
to anything with call parking enabled.. I use call parking quite
frequently and on my ATA device I can not issue a # to anything I
encounter that might require it.
When I push flash on my ATA device it does what it should,
Hello All,
I am running Asterisk version 0.9.0 on Linux Redhat 9.0. To make Meetme
work, I load ztdummy module before running asterisk. Then I can make a
meetme conference call, but the voice prompt quality is very poor. It seems
asterisk PBX sends voice data to the phone in a very low rate. When
I've been approached to research and develop a system using asterisk. It
will be used mainly to provide voice support to about 10,000 IAX clients
operating on bank ATMs.
So was wondering if there were any financial institutions, banks etc.
using * and any comments would be much appreciated.
On Wed, 2004-06-09 at 14:15, Olle E. Johansson wrote:
Rich,
I'm working on the tutorial agenda today. I've schedule a tutorial for you. Will
need a photograph
as well as some text for the web page that describes our tutorials.
Please read
List,
Sorry for sending a private e-mail to the list. Tired...
While speaking about Astricon, we are looking to fill the last holes in the
tutorial agenda.
We agreed on two topics that we feel are missing:
* Dialplan tips and tricks
* Agent and call queues
If you are interested in teaching one of
On Wed, 2004-06-09 at 14:35, Steven Critchfield wrote:
On Wed, 2004-06-09 at 14:15, Olle E. Johansson wrote:
Rich,
I'm working on the tutorial agenda today. I've schedule a tutorial for you. Will
need a photograph
as well as some text for the web page that describes our tutorials.
1 - 100 of 139 matches
Mail list logo
