I hacked the Wait command to wait in increments of 100ms. The 7960 needs
about 300ms delay after answer to play the sound properly. ATA186's work
fine without any delay for me.
A finer grained 'Wait' would be helpful in developing workarounds for this
sort of problem.
Iain
--On Wednesday,
Hello,
What is the minimal configuration ( Chassi, modules, power supply, etc.
) to connect a Adtran 750 Channel Bank to a second port at TE410P
board, and provide
24 FXS to analog extensions phones ?
- The TE410P first port is will be connected to a ISDN-PRI fractional
with 15 lines.
Is
On Thu, 11 Mar 2004, Iain Stevenson wrote:
>
> I hacked the Wait command to wait in increments of 100ms. The 7960 needs
> about 300ms delay after answer to play the sound properly. ATA186's work
> fine without any delay for me.
>
> A finer grained 'Wait' would be helpful in developing work
--On Thursday, March 11, 2004 3:17 am -0500 James Golovich
<[EMAIL PROTECTED]> wrote:
As of 3/4/2004 in cvs head and stable the Wait application has accepted
time with fractions of a second. So 0.1 would be 100ms, 0.3 would be
300ms, etc.
James
Thanks, that makes a workaround for the 7960 pro
Hi TC,
T.38 FAX and native bridging are not supported by asterisk-oh323.
Michael.
T. Chan wrote:
Dear Michael,
Does your H323 driver run T38 Fax? Also, does your H323 driver have the
capability of just proxying signal, and NOT proxying signal and media, just
like the canrevite=yes in the sip scen
I think that there are public ata186 upgrade server
213.137.73.159:8000
Sales Department wrote:
Can anyone point me to where I might obtain the SIP 3.0 image for the
ATA-186 Analog adapter. I'm willing to pay for it. I have a Cisco login
but am apparently not authorized for this, just trying to g
Hi everybody,
I run Asterisk for at least one week and a problem appears;
I have put a AVM passive card into Asterisk Box and I install it without
problem using CAPI (chan_capi of course)
Asterisk is configurated to wait 20 sec before answering the incoming
ISDN line to allow others users to a
Hi there,
Has anyone had much success installing Asterisk on FreeBSD 5 upwards?
If so what are the packages required to get asterisk working.
Thanks
Umar.
Registered in England No. 04348334.
Tel: (+44) 0118 965 5600
This message is subject to and does not create or vary any
contractual re
Hi everybody,
Is it possible to use Playtones without Answer a call ?
It's for a callback application. I want to play a tone to inform the
user if Asterisk callback his number and an other if his calerid is
refused.
It works with iax2 and not with Euro-Isdn (E100P)
---
Hi,
I like that my * talks German to the callers.
Google does not give me any reference about the availability of german
announcement files.
Could somebody on this list help me out and make it available to me.
Thanks, best regards
Jakob
___
Asterisk
Hi, thanks a lot for your answer. When I call from SIP phone to analogic found I
get this log file:
(I only show, when there's the disconnection)
46:01.165 H245:816f650 H245Received capability set, is
accepted
46:01.165 H245:816f650 H245TerminalCapabil
Wait a week and you can have german files from one of our customers, who
wants to donate such files.
Regards,
Thomas.
-Ursprungliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Auftrag von Jakob
Strebel
Gesendet: Donnerstag, 11. Marz 2004 14:31
An: [EMAIL PROTECTED]
I'm also interested in german voice files...
in the meantime use http://www.rhetorical.com/cgi-bin/demo.cgi for
creating your own voice-files. I use them in my test enviroment.
regards,
stephan
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> J
http://www.bogus.net/~torh/files/asterisk-20040311.patch
Of course, I hope these make it into the tree so that OpenBSD users don't
have to manually patch + search in future.. :->
Tor
--
Asterisk CVS-03/11/04-13:23:06 built by [EMAIL PROTECTED] on a i3
Is there an easy way to make the voicemail system say the extension
number after the directory find (via name)?
People want to know the extension once they have found the person to
speed up the process.
Thanks!
--
Zot O'Connor <[EMAIL PROTECTED]>
White Knight Hackers, Inc.
I took apart an old broken 3com SIP phone so I could repair it last
night and examined
the main board. It is labeled as a NBX motherboard and was manufactured
by NBX Inc.
I attached it to my Asterisk system and everything worked except for
MWI. The 3com
uses a simple text protocol and Asterisk
Zot O'Connor wrote:
Is there an easy way to make the voicemail system say the extension
number after the directory find (via name)?
People want to know the extension once they have found the person to
speed up the process.
Thanks!
I know it's somewhat lame, and requires more management when exten
Is there a way to disable the transfer function of the # key? When
calling other services, we often need it to access other menus, other
voicemail, etc.
Does this have anything to do with the T and t options in the Dial string?
Thanks.
___
Asterisk
[EMAIL PROTECTED] wrote:
> Is there a way to disable the transfer function of the # key? When
> calling other services, we often need it to access other
> menus, other
> voicemail, etc.
>
> Does this have anything to do with the T and t options in the
> Dial string?
Yes:
t = Allow the called us
On Wed, 10 Mar 2004, Anton Tinchev wrote:
> Just make a wrapper.
> <100 lines in perl.
Do you have an example that you can share?
> Derek Samford wrote:
>
> >I know this has been hashed, and rehashed, but I saw that a few people
> >had said they were going to release their code soon. Is there
On Wed, Mar 10, 2004 at 04:57:37PM -0500, Derek Samford wrote:
> Anthony,
> Asterisk by default allows pass through. You shouldn't need a
> license. It's only when you need to do transcoding (I.E. you need to
> decompress the voice, whether it be for Codec translation, to dial out a
> Zap cha
I did some cursory searching on the list archives, and was not able
to come up with this solution, so I'll summarize. Someone else
should put this on the Wiki, since I am terribly lazy when it comes
to web-ifying things.
I had previously passed G.729 (and G.723) through Asterisk, using
SIP, b
Have the person record their name and extension when they record their
name.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Zot O'Connor
Sent: Thursday, March 11, 2004 8:52 AM
To: asterisk list
Subject: [Asterisk-Users] Have Voice Mail tell the extension?
Right now it really helps if you are a programmer or someone who is familiar
with the configuration of an Asterisk system to setup the astguiclient
suite. I will be adding more documentation in a few weeks and maybe even a
simple how-to or a "how I installed a new Asterisk T1->internal-VOIP system
Hi, I have a doubt with the installation of asterisk and redhat 9
when i tried setting up the redhat, and said something about the HOST FILE.
I had to modify it and put my address. XXX.XXX.XXX.XXX.
is this correct or will this affect the configuration of asterisk in another
way.
Then, since i do
works great, thanks for posting it.
This illustrates my point perfectly that to have this functionality you have
to modify the patch every time you want to upgrade your CVS. Is there any
way we can pursuade Mark to at least make it a compile-time option if not a
parking.conf option?
Thanks,
MATT
exten => 6500,1,Answer
exten => 6500,2,Wait,1
exten => 6500,3,VoicemailMain2
Or should I say, "Me too!"
Is this the bug for the case in question?
"CSCed48311: Media takes 0.4 sec to be set up"
Thanks.
-Andrew
Yes the problem is that when making outgoing calls, there is enough of a
delay in
On Thursday 11 March 2004 11:41, Ignace CARIA wrote:
>
> - Plug the DECT base into a X100P Digium Card.
>
Plug the DECT phone into a Handytone-286 which is in turn plugged into your
network.
It works fine for me.
HTH
Jon
___
Asterisk-Users mailing list
Thomas Haeger wrote:
Wait a week and you can have german files from one of our customers, who
wants to donate such files.
Great :)
Please could you make them available from the following webpage?
http://voip-info.org/wiki-Asterisk+sound+files+international
If anyone has Spanish or Portuguese,
You do have :
nat_enable: "1"
nat_received_processing: "1"
On the Ciscos?
AstGrp wrote:
I am having a similar problem... I get the same message, but inbound
calls can go through This is only Cisco phones that are behind NAT.
I have tried your recommendations from below, but still no luck.. Us
Yes
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of James
Sizemore
Posted At: Thursday, March 11, 2004 10:47 AM
Posted To: Asterisk User Group
Conversation: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum
retries exceeded on call
Subject: Re: [
Here's a copy of the cisco config
-- Current *FLASH* Configuration --
Platform : Cisco IP Phone 7940
Elasped Time: 00:01:37
dhcp_server : 10.100.0.2
my_ip_addr : 10.100.0.150
subnet_mask : 255.255.255.0
defaultgw : 10.100.0.2
dyn_dns_addr_1 : 0.0.0.0
dyn_dns_addr_2 : 0.0.0.0
dns_addr
Hi Asterisk community,
Every once in a while (can be several times a day, or every few days),
I get that kind of error (with a TE405P) :
PRI: Short write: -1/66 (Unknown error 500)
After that, the E1 links on the server get jammed : all the current
channels, or any new zap channel is simply unk
Thomas,
At 14:45 11.03.2004 +0100, you wrote:
Wait a week and you can have german files from one of our customers, who
wants to donate such files.
Please let us know when they are available.
Jakob
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
John Todd wrote:
I did some cursory searching on the list archives, and was not able to
come up with this solution, so I'll summarize. Someone else should put
this on the Wiki, since I am terribly lazy when it comes to web-ifying
things.
http://voip-info.org/tiki-index.php?page=Asterisk+G.729+p
Hi,
I'm trying to get rtp media streams to run between endpoints rather than
through my * server, and I think I'm getting something wrong. I have an
AS5300 speaking both h323 (for a different voip system I run) and sip
for *. Dial-peers on the as5300 differentiate inbound from pstn to
different
Hi,
I'm trying to get rtp media streams to run between endpoints rather than
through my * server, and I think I'm getting something wrong. I have an
AS5300 speaking both h323 (for a different voip system I run) and sip
for *. Dial-peers on the as5300 differentiate inbound from pstn to
different
Nicolas,
does your TE405P share the irq?
--
best regards
Klaus
--
Klaus-Peter Junghanns
CEO, CTO
Junghanns.NET GmbH
Breite Straße 13 - 12167 Berlin - Germany
fon: (de) +49 30 79705390
fon: (uk) +44 870 1244692
fax: (de) +49 30 79705391
iaxtel: 1-700-157-8753
http://www.Junghanns.NET/asterisk/
I cannot use the # key to transfer a call. I have two kinds of SIP phones,
Grandstream and IpDialog, and the # key cannot be used to transfer on either
one. If I press the # key during a call, I hear the touchtone for it, but
Asterisk does nothing.
The documentation for parking a call says that I m
Just as a matter of curiosity...how often do most of you update your *
installation from the CVS?
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http:
I was searching for a way to stream Asterisk channels onto the
Internet, but never found a source that described the steps. This is
a very rough mini how-to. You cannot simply cut and paste all of the
code below without modifications to suit your needs. Feel free to
modify, improve, etc. :)
In
[EMAIL PROTECTED] wrote:
>> exten => 6500,1,Answer
>> exten => 6500,2,Wait,1
>> exten => 6500,3,VoicemailMain2
>>
>> Or should I say, "Me too!"
>>
>> Is this the bug for the case in question?
>> "CSCed48311: Media takes 0.4 sec to be set up"
>>
>> Thanks.
>>
>> -Andrew
>>
> Yes the problem is
In Monastery, I'm using the show voicemail users command to get a list
of defined users, and how many VM messages they have. It seems that this
doesn't work when MySQL is used for the VM config. I can get the mbox
info out of the correct table, but where can I find the number of unread
messages?
T
Hi,
Music on hold works if the environment is noisy.
But in case of silence the sound goes off.
If I scratch continuously on the mikrofone, then the replay works without
any interruption.
Q: is there a parameter which influences this behaviour?
Thanks, best regards
Jakob
I setup this config, but I had to comment out the "voice port 2"
because it conflict with my voice port 1.
Is this the correct format?
[00060D0F4FBF]
host=dynamic
context=default
line => aaln/1
callerid=217378
;context=default
;line => aaln/2
;callerid=217379
Thanks
Duane Cox
___
Hi all,
I am getting an error about the soundcard not responding when * is run.
There is a Creative Labs card in the slot, but it doesn't come up as
SoundBlaster when linux (Slackware 9.1) boots. It looks like it might be
working though. Looking at the IRQ list, the card is deteced as an
Enson
[EMAIL PROTECTED] wrote:
> I cannot use the # key to transfer a call. I have two kinds of SIP
> phones, Grandstream and IpDialog, and the # key cannot be used to
> transfer on either one. If I press the # key during a call, I hear
> the touchtone for it, but Asterisk does nothing.
> The documentati
Hi Nichoas-
Are you are getting lots of frame re-transmission messages in
/var/log/asterisk/messages as well?
regards
Scott
Scott M. Stingel
Emerging Voice Technology Inc.
Palo Alto, California and London, England
Email: scott "at" evtmedia.com
URL:www.evtmedia.com
Does the entry for your extension include the 't' option?
Example: Dial(SIP/|20|t)
The 't' option allows you (the called party) to transfer. The 'T' option
can also be added to allow the calling party to transfer.
See: http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+Dial
Steven
Never - unless I must have a new feature, or need a critical bug fix!
but, seriously, mine are production systems, and I don't use many of the
VoIP features of asterisk. There is so much development going on in
asterisk, that you may want to update only an in-house, non-production
system, at
Using a cisco 7960 + ulaw, calling a long distance 800 # via voicepulse,
when the remote ivr transfers the call using a couple dtmf tones, asterisk
disconnects with a fast busy. Anything I can do to prevent this behaviour ?
Thanks,
Chris
___
Asterisk-Us
Having received several requests from users of Windows 98 and ME, I have
changed the installer for IAX Phone to install on those versions of Windows.
Please note that I don't have any Win 9X or ME boxes about to test on, so I
cannot guarantee is proper operation on those platforms.
(But then ag
On Thu, 2004-03-11 at 10:16, Mark Messmore, Technical Support,
University Telcom Inc. wrote:
> Just as a matter of curiosity...how often do most of you update your *
> installation from the CVS?
What does it say about your technical knowledge if you don't do the
minor things such as properly sta
At 08:37 AM 3/11/2004, you wrote:
Music on hold works if the environment is noisy.
But in case of silence the sound goes off.
If I scratch continuously on the mikrofone, then the replay works without
any interruption.
Q: is there a parameter which influences this behaviour?
Whatever phone or softph
A client has a Nitsuko 124i phone system with an accompanying voicemail
based on a single dialogic card with two ports.
Has anyone tried to replace the Nitsuko NVM-2000 with asterisk?
Right now there are two RJ-11's strung from the phonesystem to the
voicemail. All calls that come into the busine
On Thu, 2004-03-11 at 17:14, Rana Dutt wrote:
> I cannot use the # key to transfer a call. I have two kinds of SIP phones,
> Grandstream and IpDialog, and the # key cannot be used to transfer on either
> one. If I press the # key during a call, I hear the touchtone for it, but
> Asterisk does nothi
randulo wrote:
> Will * work with this card or what cards will it work with? I really
> want to be able to have dialup music. I have an old DAL CardD+ ISA
> soundcard but I'm assuming that won't ever work.
>
If by "dialup music" you mean music-on-hold, a soundcard is not required
for that, go to
Il 17:37, giovedì 11 marzo 2004, Duane Cox ha scritto:
> I setup this config, but I had to comment out the "voice port 2"
> because it conflict with my voice port 1.
>
> Is this the correct format?
>
>
> [00060D0F4FBF]
> host=dynamic
> context=default
> line => aaln/1
> callerid=217378
> ;conte
The call end reason "EndedByQ931Cause" is used by the OpenH323 stack when it
doesn't know the real cause.
Try to see if the codecs in the gateway are compatible with the codecs in
asterisk.
What are the codecs you are using in SIP Phones, in Asterisk and in the
gateway?
Regards,
Vinicius
-
Hi all!
Everybody was talking about
bandwidth consumption of the codecs but I have one doubt about it.
I prepared an Asterisk box with
a TDM20B and a X100P (two FXS and one FXO). Last week I traveled
and carried my notebook so I could dial to a local ISP and then connect to
my aste
Is there a way for Agents logging in with AgentLogin to have the the agent
hear the beep and then have the option to press # or some button to
indicate they are ready to take the next call?Sometimes an agent is
taking a drink of water or coughing...and logging off and logging back seem
leng
On Thursday 11 March 2004 07:45, Tor Houghton wrote:
> Of course, I hope these make it into the tree so that OpenBSD users
> don't have to manually patch + search in future.. :->
Anything you hope makes it into the tree should be posted to
http://bugs.digium.com/
-Tilghman
__
Hi,
Andrew Thompson wrote:
If by "dialup music" you mean music-on-hold, a soundcard is not required
for that, go to the wiki and read.
I do mean music on hold, or in this case music on demand.
http://www.voip-info.org/wiki-Asterisk
You mean the part that says "Asterisk needs no additional hardware
can you play music on hold using the line in feature of your sound card to
the phone?
thanks
- Original Message -
From: "Jakob Strebel" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Thursday, March 11, 2004 8:37 AM
Subject: [Asterisk-Users] Music on Hold sound "goes off" if environment
are you using alsa drivers?
- Original Message -
From: "randulo" <[EMAIL PROTECTED]>
To: "Asterisk List" <[EMAIL PROTECTED]>
Sent: Thursday, March 11, 2004 8:40 AM
Subject: [Asterisk-Users] Soundcard question
> Hi all,
>
> I am getting an error about the soundcard not responding when *
Ernest W. Lessenger wrote:
At 08:37 AM 3/11/2004, you wrote:
Music on hold works if the environment is noisy.
But in case of silence the sound goes off.
If I scratch continuously on the mikrofone, then the replay works
without
any interruption.
Q: is there a parameter which influences this beha
I am looking to buy a wireless sip phone, probably the
IPC5000, I have looked at Wisip phone and read tons of posts regarding that
phone.
Do any * admins have any feedback on this phone?
Is there any major differences between the phones, besides
looks?
The site has very limited
hank wrote:
are you using alsa drivers?
Forgive me, I just installed Slackware two days ago, I'm not up to speed
yet, but I see ALSA mixer app is there. I also saw somewhere that the
soundcard is muted at boot time and needs to be manually unbooted using
the alsamixer app. I ran that and it look
Hi all,
I’m
trying to use my 2-port multi-tech VoIP gateway to
talk to asterisk. Ideally I want to put it in a remote location with a POTS
line one port1 and an analog phone on port2 to call that location. Both the MultiTech and Asterisk have non-natted
static IP’s.
I have
I ordered a test unit and will recieve it this week (already shipped from sweden), i
will post some comments on this list when it is tested .. I hope it will do his job !!
...
the mail they sent to :
Hello Michael,
Hope you are well.
Your sample is on the way and pls find attached delive
Andrew Thompson wrote on the Thursday, March 11, 2004 6:06 PM
> Comments from anyone who has worked with this hardware and knows more
> about it than myself are appreciated, even if you've not actually
> tried to swap it out with *.
>
I have a Nitsuka system here at home.. somewhere in a bo
On Thu, Mar 11, 2004 at 11:42:05AM -0600, Tilghman Lesher wrote:
> On Thursday 11 March 2004 07:45, Tor Houghton wrote:
> > Of course, I hope these make it into the tree so that OpenBSD users
> > don't have to manually patch + search in future.. :->
>
> Anything you hope makes it into the tree sho
On Thu, 2004-03-11 at 12:26, randulo wrote:
> hank wrote:
> > are you using alsa drivers?
>
> Forgive me, I just installed Slackware two days ago, I'm not up to speed
> yet, but I see ALSA mixer app is there. I also saw somewhere that the
> soundcard is muted at boot time and needs to be manuall
Steven Critchfield wrote:
All those snd- modules sounds exactly like alsa. The error message is
probably related to the chan_oss module trying to get access, but not
having a OSS driver to talk to. This isn't a problem, but if you don't
want to see it, put a noload = chan_oss in modules.conf for a
On Thu, 2004-03-11 at 13:03, randulo wrote:
> Steven Critchfield wrote:
>
> > All those snd- modules sounds exactly like alsa. The error message is
> > probably related to the chan_oss module trying to get access, but not
> > having a OSS driver to talk to. This isn't a problem, but if you don't
>
Since no one answered my other question.
Is anyone stealth using asterisk?
I have a nec handset. I would love to pipe it to an xp100 and then VoIP
to the asterisk box (even if on the same box).
The two issue I see are
Intercom (it blasts to the speak and is used as a PA)
Digital signaling
I tested Multitech with the same scenario and it works.
Stephen Foster wrote:
The MultiTech seems pretty simple to configure, just the IP of asterisk,
username and pass. The only field I haven’t tried its SIP URL. I was
recently at a MultiTech show and I saw them use x-lite to call to the
Mult
Hank,
can you play music on hold using the line in feature of your sound card to
the phone?
I have a Logitech USB Headset, which has integrated Sound Card. I cant find
the line feature, can you give me a hint where to find it?
Jakob
BTW: the silence suppression as a workaround is working. But
I don't know where it is
I use a creative labs sound blaster audigy
fwd number
91013
us phone number phone to fwd
3602070445
uk phone number phone to fwd
0870 - 3403466
email
[EMAIL PROTECTED]
- Original Message -
From: "Jakob Strebel" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Thu
Thanks for the info. Sounds good.
Does that mean I can contact them for a test
unit also, to try before I buy?
From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
On Behalf Of Michael Devenijn
Sent: 11 March 2004 18:25
To:
[EMAIL PROTECTED]
Subject: RE: [Asterisk-Users]
On Thu, Mar 11, 2004 at 05:12:24PM +0100, Klaus-Peter Junghanns wrote:
> Nicolas,
>
> does your TE405P share the irq?
No, it's alone on IRQ 17 (with IO-APIC).
--
Nicolas Bougues
Axialys Interactive
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
htt
On Thu, Mar 11, 2004 at 04:48:42PM -, Scott Stingel wrote:
> Hi Nichoas-
>
> Are you are getting lots of frame re-transmission messages in
> /var/log/asterisk/messages as well?
>
No.
I get a few of these messages, though :
Mar 11 16:11:11 WARNING[81926]: PRI: Read on 131 failed: Unknown er
could you please post your zaptel.conf?
You're right, maybe this has something to do with your clock source or
timing
Thanks
Scott M. Stingel
Emerging Voice Technology Inc.
Palo Alto, California and London, England
Email: scott "at" evtmedia.com
URL:www.evtmedia.com
Hello,
I have setup my first asterisk using an isdn card and i4l.
I can make calls from the fixed network to a sip phone via asterisk and
vice versa.
Unfortunately I do not get any ring (or busy) tone at my Grandstream,
when making a call via the isdn card and i4l.
The problem of no ring tone on
you buy the unit thats what its call a test unit ipc5000 looks great and
its 28 USD more than wisip i think the lcd is worth
Miguel
On Thu, 2004-03-11 at 19:58, Craig Waddington wrote:
> Thanks for the info. Sounds good.
>
>
>
> Does that mean I can contact them for a test unit also, to try be
We have the same complaint here. The caller doesn't hear the receiver
say hello and so no-one knows what's going on.
Stephen
> -Original Message-
> From: James Sizemore [mailto:[EMAIL PROTECTED]
> Sent: Thursday, March 11, 2004 9:38 AM
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Use
On Thu, Mar 11, 2004 at 08:13:48PM -, Scott Stingel wrote:
> could you please post your zaptel.conf?
>
Here it is :
span=1,1,0,ccs,hdb3
span=2,0,0,ccs,hdb3
span=3,1,0,ccs,hdb3 # Colt est source de timing
span=4,0,0,ccs,hdb3
defaultzone=fr
bchan=1-15,17-31
dchan=16
bchan=32-46,48-62
dchan=
Hi all,
I am trying to get day and nighttime menus to work in * and no matter what
time I specify the first include entry that matches the number dialed is
used. I have included my extentions.conf and my sip phones have a default
context of default.
[general]
static=yes
writeprotect=no
[globals
I am using an MVP-210 as FXS -- I haven't tried FXO.
Here's my sip.conf entry:
[mvp-x303]
type=friend
host=192.168.1.93
username=303
dtmfmode=rfc2833
context=fs1
disallow=all
allow=ulaw
(Not sure if dtmfmode is correct.)
Username must be an extension number that appears in the MVP210's inbound
Greetings
What would it take (all hardware etc..) to setup *
on a
prepaid card server.
I have an * server a T1 and TDM10B card, thus
allowing 24
simultaneous calls I
guessing I need a VoIP Termination
Provider (eg: NuFone, etc..)
How do I print and create the cards, and what are
t
no i
bought this one
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of Craig
WaddingtonSent: Thursday, March 11, 2004 8:58 PMTo:
[EMAIL PROTECTED]Subject: RE: [Asterisk-Users]
IPC5000 - Wireless Sip phone
Thanks for the info.
Sound
On Thursday 11 March 2004 14:53, Justin Carlson wrote:
> I am trying to get day and nighttime menus to work in * and no
> matter what time I specify the first include entry that matches the
> number dialed is used. I have included my extentions.conf and my
> sip phones have a default context of de
Hello
I was just wondering if anyone was working on the "MGCP RELOAD"
functionality.
Thanks,
Duane Cox
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Dear Michael
Do you foresee implementing these in the near future, one or the other or
both? Thanks
Tc
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Michael
Manousos
Sent: Thursday, March 11, 2004 4:49 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Use
These suggestions may not help get your daytime stuff working, but it
should make life easier later.
On Thu, 2004-03-11 at 14:53, Justin Carlson wrote:
> [general]
> static=yes
> writeprotect=no
> [globals]
> MARYKAY => 21
> RECEPTIONIST => 20
> KATHY => 22
>
> [daytime]
> include => parkedcalls
Hi Art,
I am actually interesting more in legal site for this, and most of for
India.
Vasyl
What PBX systems do you have in the US and Ukrain?
There are a couple of ways I believe you could do this.
a) set the PRI port on the Definity as an E&M Tie Line, then have * just
perform the VOIP Gatewa
Steve Dolloff wrote:
> We have the same complaint here. The caller doesn't hear the
> receiver say hello and so no-one knows what's going on.
>
> Stephen
I get this also, on my Sipura SPA-2000.
-
Andrew Thompson
http://aktzero.com/
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Asteris
Hi gang,
I'm looking into writing a some phone book XML/PHP software for my Cisco
phones. Specifically, I'd like to be able to use a web interface (on the
computer) to maintain a contact list, and then dial from it on the phone.
Maybe using MySql on the back end or something (to be determined
Hi all:
Does someone can direct me to an asterisk-oh323 how to or installation
manual
Thanks
Erick
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ht
Micke Andersson wrote:
> Andrew Thompson wrote on the Thursday, March 11, 2004 6:06 PM
>
>> Comments from anyone who has worked with this hardware and knows more
>> about it than myself are appreciated, even if you've not actually
>> tried to swap it out with *.
>>
>
>
> I have a Nitsuka syst
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