Lists wrote:
I have an X100p cardand it is hard to hear the person on the other
end. Should I mess with these values? I have heard both yes and no to
this question in the past. If yes, how much louder should I make them?
Thanks,
MIchael
Start with 0.5 and see if its too loud or not
hi everybody,
Is there SIP client which work with Asterisk and
can be embedded in a HTML page ?
Thanks
Rattana
hello,
my questns are about few * functionality.
1) how can I make call tranfer. Not call parking.
If I'm talking with some one a I want to tramnfer call to the another
extension, to the other person.
2) how can I make call confernece. Not Meetme
If I'm talking with some one and I want to join
Should the following setup work?
SIP UA---NAT---Internet---NAT---SIP UA
If both UA's support STUN and report the external IP address in the SIP
packet..
I am trying to get away from using canreinvite=no so that traffic can go
directly between the UA's and not via the central server but I
That's for pointing out Walter Snel hack.
Adding his two additional features would not be
hard.
http://lists.digium.com/pipermail/asterisk-users/2003-October/024968.html
Any idea when these hacks will appear in CVS?
Peter
___
Asterisk-Users mailing
Peter Hudec wrote:
hello,
my questns are about few * functionality.
1) how can I make call tranfer. Not call parking.
If I'm talking with some one a I want to tramnfer call to the another
extension, to the other person.
2) how can I make call confernece. Not Meetme
If I'm talking with some one
Peter Zeltins wrote:
That's for pointing out Walter Snel hack.
Adding his two additional features would not be
hard.
http://lists.digium.com/pipermail/asterisk-users/2003-October/024968.html
Any idea when these hacks will appear in CVS?
Probably when they have been properly tested and
Hi
I have an Asterisk installation with some SIP and MGPC devices, and I also have a
TE410P on a E1 line.
If I make an outside ISDN data call to asterisk the phone rings as usual and if I
answer it, I just hear some clicks.
I've read that the D-Channel has information about the call, if its
I have problems to send faxes using a fax machine connected to a ATA186 line
2. My sip.conf is
[1151]
type=friend
username=1151
secret=
canreinvite=no
host=dynamic
dtmfmode=rfc2833
mailbox=1151
nat=1
context = optica
ATA 186 connection mode is 0x00460400 and Audio mode is 0x00150014
I
http://www.asterisk.org/index.php?menu=features
- Call features
- Call Transfer
WipeOut wrote:
Peter Hudec wrote:
hello,
my questns are about few * functionality.
1) how can I make call tranfer. Not call parking.
If I'm talking with some one a I want to tramnfer call to the another
Peter Hudec wrote:
http://www.asterisk.org/index.php?menu=features
- Call features
- Call Transfer
Yes, provided your phone supports transfer or you use the t or T
options on your dial string and then use the # key to transfer..
CLI show application dial
hello to all.
I have a PC with a E400P Card with 4 E1 with a RJ48 jack
is posible to convert any RJ48 jack to 30 phone line ?
is for example.
1 E1 come from the telco operator to PC
and with asterisk i a lot of things and before send it to a phone but, where i plug
this phone ?
Bye
Victor Sanchez wrote:
hello to all.
I have a PC with a E400P Card with 4 E1 with a RJ48 jack
is posible to convert any RJ48 jack to 30 phone line ?
yes, you need an equipment called a channel bank
is for example.
1 E1 come from the telco operator to PC
and with asterisk i a lot of things and
Victor Sanchez wrote:
hello to all.
I have a PC with a E400P Card with 4 E1 with a RJ48 jack
is posible to convert any RJ48 jack to 30 phone line ?
is for example.
1 E1 come from the telco operator to PC
and with asterisk i a lot of things and before send it to a phone but, where i plug
thanks,
you didn't make me happy :(
hudecof
WipeOut wrote:
Peter Hudec wrote:
http://www.asterisk.org/index.php?menu=features
- Call features
- Call Transfer
Yes, provided your phone supports transfer or you use the t or T
options on your dial string and then use the # key to transfer..
Hi!
1) how can I make call tranfer. Not call parking.
If I'm talking with some one a I want to tramnfer call to the another
extension, to the other person.
As it was said: This depends very much on your phone, and also on the
protocol you use (SIP, MGCP, H.323 etc). Look at the X-Pro (?)
On Wed, 29 Oct 2003 09:58:28 +0100, Rattana BIV wrote:
hi everybody,
Is there SIP client which work with Asterisk and can be embedded in a HTML page ?
Thanks
Rattana
hehe, why use a SIP client, why not a client which does IAX and bury
that in a web page ?
(yes, I haven't answered the
When I try to apply Makefile.path in the apps directory using command
patch Makefile.patch I got this error
Hunk #1 FAILED at 22.
Hunk #2 FAILED at 45.
2 out of 2 hunks FAILED -- saving rejects to file Makefile.rej
Any idea?
___
Asterisk-Users mailing
Hi,
Try this:
http://www.loligo.com/asterisk/
Ta
Senad
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Hi all,
i tried to load the tor2 driver for the E400P with
modprobe tor2
on a P4 2,7Ghz machine.
and everything seems to be ok. but after load (after the kernel messages)
the machine is freezed.
And then i have to reset the whole machine.
The same is happend when i try to load the
Hi there,
I'm very new here and would like to know if anyone has reccomendations
on fundamental reading (other than the handbook) whick might prevent me
from asking some really dumb questions (after this one of course).
What I'm trying to do:
I have a SOHO... very SO in fact. I would like to
Hello,
my next problem is with SIP device behind NAT.
First few seconds of the call are OK. Astrisk is sending the packets to
the public IP address of the FW/NAT (62.152.224.3). But this change in
10 second and packets are send to the my public addres.(192.168.1.163).
in the sip.conf for the
Alastair Maw wrote:
On 27/10/03 21:57, DUSTIN WILDES wrote:
Does anyone have any recommendations on implementing Answering
Machine detection for call generation programs?
There's obviously no nice way of doing this.
If you're doing telemarketing, and you're playing pre-recorded audio,
which
Thanks a bunch you were on the money. Do you know about when that changed?
John Todd wrote:
Grandstreams phones can't call out with the latest CVS, anyone know
what the
last good CVS date was?
You may be experiencing difficulty due to bad codec permissions, since
the latest CVS updated
I ended up viewing the Makefile.patch side by side and inserting the
new/modified lines into the apps/Makefile
This seemed to work. Doing a make created the two new apps*.so which I was
able to place into the asterisk/modules directory and start *
They, however, still don't work. My faxes turn
Ok,
meanwhile i loaded the tor2 driver with insmod (previous loaded slhc,
ppp_generic,zaptel)
this works, but when i execute ztcfg the machine is freezed!
Now any ideas ?
Thanks,
Thomas.
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Auftrag von Thomas
There's a collection of external resources here:
http://bugs.digium.com/bug_view_page.php?bug_id=434
On Wed, 2003-10-29 at 05:49, Kris Edwards wrote:
Hi there,
I'm very new here and would like to know if anyone has reccomendations
on fundamental reading (other than the handbook) whick
If you have ANY chance of sending a fax over VoIP your codec MUST be
ulaw or alaw.
On Wed, 2003-10-29 at 03:52, Manuel Marín García wrote:
I have problems to send faxes using a fax machine connected to a ATA186 line
2. My sip.conf is
[1151]
type=friend
username=1151
secret=
See
http://resource.intel.com/telecom/support/documentation/unix/SR50_linux/html
_files/vox_feat/contents.html#TopOfPage chapter 2 for a basic insight on
Dialogic does it...
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling
Sent: woensdag 29
Chris Albertson wrote:
Aha. It may be connected to this error message, then:
messages:Oct 26 21:18:15 NOTICE[137382912]: File sched.c, Line 209
(sched_settime): Request to schedule in the past?!?!
I read somewhere that I could ignore this message, therefore I just
didn't include it in my earlier
Actually,
Back in '99, Dialogic used a very simple algorithm, and it was surprisingly
accurate. You simply watch and see how long the initial greeting is. If it
is short (say, only a few seconds), then it is generally a live person.
However, if the initial greeting lasts for much longer (say
Thanks for all the info!
So I take it I would need to either build an additional APP to asterisk like
(voice_detection) or into an AGI and have that application or AGI run after the call
is Answered?
Fortunately it's not a telemarketing system! :-)
It's an appointment reminder system for some
Hi!
First few seconds of the call are OK. Astrisk is sending the packets to
the public IP address of the FW/NAT (62.152.224.3). But this change in
10 second and packets are send to the my public addres.(192.168.1.163).
in the sip.conf for the phone(X-Lite) is
I think you should check
The ATA186 is only rated to 9600 baud, it is not usable for faxing as
most faxs
are 19200. See Cisco sight for details.
http://www.cisco.com/en/US/products/hw/gatecont/ps514/products_field_notice09186a0080094af7.shtml
Eric Wieling wrote:
If you have ANY chance of sending a fax over VoIP your
Why not just ask them to press-any-key ?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
DUSTIN WILDES
Sent: Thursday, 30 October 2003 12:30 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Answering Machine Detection
Thanks for all
On Tue, 28 Oct 2003, Brian Schrock wrote:
Everyone,
Just thought I would drop a line telling everyone here I have the software
RxFAX/TxFAX up and running without any real problems. I did have to.
RH 9.0
1) Install an audio devel rpm
1) install libtiff from source, and copy over
On Wed, 29 Oct 2003, WipeOut wrote:
Lists wrote:
I have an X100p cardand it is hard to hear the person on the other
end. Should I mess with these values? I have heard both yes and no to
this question in the past. If yes, how much louder should I make them?
Thanks,
MIchael
Lists wrote:
On Tue, 28 Oct 2003, Brian Schrock wrote:
Everyone,
Just thought I would drop a line telling everyone here I have the software
RxFAX/TxFAX up and running without any real problems. I did have to.
RH 9.0
1) Install an audio devel rpm
1) install libtiff from source, and copy
Title: FW: Voice/Data mixed routing over Digium E1/T1 Card
The documentation mentions that the Digium channels can be split into some voice channels and the remainder of the channels used for routing IP traffic.
Does any one have this in use in conjunction with Asterisk? Does it work
Because I need detection for the logging functions. Otherwise I won't get accurate
logging results.
-Original Message-
From: Bryan Nolen [mailto:[EMAIL PROTECTED]
Sent: Wednesday, October 29, 2003 8:38 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Answering Machine Detection
I need connect up to 100 analog phone to a H.323 network through *. I
think use TE410P, But I need to know what channel bank is better. I use
E1 lines
Any idea?
Thanks in advance,
srsergio
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de DUSTIN
WILDES
Hi i'am again,
here my zapata.conf:
span=1,0,0,ccs,hdb3,crc4
span=2,0,0,ccs,hdb3,crc4
span=3,0,0,ccs,hdb3,crc4
span=4,0,0,ccs,hdb3,crc4
bchan=1-15,17-31
dchan=16
bchan=32-46,48-62
dchan=47
bchan=63-77,79-93
dchan=78
bchan=94-108,110-124
dchan=109
loadzone = fr
defaultzone=fr
If i comment
Might want to write a new
energy detector algorithm in dsp.c though based on a wideband/low Q
resonator approach (move the pole way in towards the origin)
as opposed to
narrow band goertzels (pole on the unit circle). More robust
for this type
of work.
Where does one go to learn this
Title: FW: Voice/Data mixed routing over Digium E1/T1 Card
I'm
currently using this setup for a channelized T1 for voice and
data.
First
9 channels of the T1 are voice - the rest are data for
internet.
Works
extremely well!
This
is being used for a production server that receives/places
Sergi,
I would say it depends of your budget. You can find on market different
channel banks. Some of them are very expensive and have all fancy features,
some of them are not so expensive and of course are missing some features.
We are using CAC and New Bridge chanell banks, they are working
FW: Voice/Data mixed routing over Digium E1/T1 CardWe are using it in three
sites where the T-1 is pure IP and calles are routed in/out over SIP IAX2
and then to a channel bank. As a router the T400/T100 works great; I would
highly recommend it. As voice server it works great. As a combined
Hi Dustin-
That's interesting! What is the physical setup that you have? IE:
Routers, etc
Also, where are you located and who is the carrier? I'm interested in
setting up a similar channelized T1 here in my office (PacBell-SBC)
Thanks
Scott Stingel
Scott M. Stingel
Emerging Voice
Why not just ask them to press-any-key ?
And if any of them get confused you can refer them
to Compaq frequently asked question #2859
http://web14.compaq.com/falco/detail.asp?FAQnum=FAQ2859
:)
___
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[EMAIL PROTECTED]
some information can be found here about algorithm and descriptions of
method being used.
http://citeseer.nj.nec.com/393112.html
Regards,
Alexander
***
XVOIP network is lunched, get your +1 777 number today. [EMAIL PROTECTED]
We are going to make pilot project for one of our customer, based on QuadT1
card.
I will let you know after we finish test.
Regards,
Alexander
Unofficial Asterisk Forums
***
URL : http://asterisk.xvoip.com
Registration
All of the setup is running on RedHat 8.0 - no other router or CSU is needed.
Don't use RedHat 9.0 yet in this setup since the ZAPTEL_NETWORK flag will not compile
with the new implementation of HDLC in the kernel.
I'm using the T100P for the interface.
They have all Cisco SIP Phones, no
We are looking to utilise Digiums QuadxE1 cards for
one of our European customer, he is looking for
16xE1. Carrier has no VOIP support currently, so we
need to organize by ourslef VOIP-TDM interface, to get them connected to our
network. Idea is to try Digium/Asterisk solution.
It will be
XVOIP network is lunched, get your +1 777 number today. [EMAIL PROTECTED]
XVOIP is lunched? Is this
Hi list,
I am facing the following problem, i need to make the
following scenerio
work
exten = _900.,1,Dial(Zap/g1/xx,25,r)
exten = _900.,2,Wait(3)
exten = _900.,3,SendDTMF(${EXTEN:1})
I am using PRI-ISDN with T400P card. Searching through
archives, i found
that we can add some 'w' s to
Does anyone has experience with real time traffic on such volumes ? What is your
experience? What kind of hardware do you run ?
We opera dual E400p (single P4, 3,0 Ghz) and dual TE410p boxes (dual
Xeon, 2,4 Ghz). Both installations can handle 240 concurrent calls.
However, I heard recoding
It happens after someone is just hungry and thinking about food ! ;-) Heh
...typoe happens..
Let's don't make from it story ...
- Original Message -
From: Linus Surguy [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, October 29, 2003 10:24 AM
Subject: Re: [Asterisk-Users] 16xE1
Did you mistype or something. That link is about power profiling the
consumption of DSPs :-)
Regards,
Steve
Asterisk online forums wrote:
some information can be found here about algorithm and descriptions of
method being used.
http://citeseer.nj.nec.com/393112.html
Regards,
Alexander
Hi Chris,
That is exactly the Dialogic implementation I was referring to that was
utterly useless. It works OK when people are demoing, as they always
follow a certain pattern. In real like it I've always found it a recipe
for screaming angry users. Depnding on your use it can get over 90% of
I am taking note of people's messages about soft fax, even if I might
appear to be ignoring them. I am getting V.27ter finished off right now,
to flesh out the facilities in the software. V.27ter is used for 4800bps
and 2400bps faxes - not critically important, but useless for lousy
lines.
Ray Burkholder wrote:
Might want to write a new
energy detector algorithm in dsp.c though based on a wideband/low Q
resonator approach (move the pole way in towards the origin)
as opposed to
narrow band goertzels (pole on the unit circle). More robust
for this type
of work.
Where
At 09:17 AM 10/29/2003 -0500, you wrote:
Might want to write a new
energy detector algorithm in dsp.c though based on a wideband/low Q
resonator approach (move the pole way in towards the origin)
as opposed to
narrow band goertzels (pole on the unit circle). More robust
for this type
of
unsbscribe
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Exclusive Video Premiere - Britney Spears
At 11:52 PM 10/29/2003 +0800, you wrote:
Ray Burkholder wrote:
Might want to write a new
energy detector algorithm in dsp.c though based on a wideband/low Q
resonator approach (move the pole way in towards the origin) as opposed
to narrow band goertzels (pole on the unit circle). More robust
No this most likely willn't work unless you have open the correct ports on
each NAT device. The problem is that NAT in general only allows packet in if
a packet has gone out first. I am assuming you have left have the fact that
* is used to setup the SIP call setup and then should drop out. If so
I spoke with someone today who is interested in an IP Centrex solution that
starts with about 3500 extensions in a multi-tenant application. And
growing from there.
I'm wondering about scalability of Asterisk. I'm trying to put my head
around how to put the whole thing together, if it can be
Great job Steve!
To share my own experiences:
I have tested using an X101P and a E1. The E1 tests
failed but that might have to do with errors am
getting on the line. The X101 worked fine many times.
I had one or two cases where my fax machine thought it
had sent the fax ok, but when I looked
did you check that there are no irq conflicts ?
On Wednesday 29 October 2003 3:13 pm, Thomas Haeger wrote:
Hi i'am again,
here my zapata.conf:
span=1,0,0,ccs,hdb3,crc4
span=2,0,0,ccs,hdb3,crc4
span=3,0,0,ccs,hdb3,crc4
span=4,0,0,ccs,hdb3,crc4
bchan=1-15,17-31
dchan=16
I can get a hold of a number of refurbished Carrier Access Bank I (T1/FXS)
units for a pretty respectable price. Will they do the job based upon your
comment that they support 'Answer Supervision'? How about the Caller ID,
MWI stutter that Asterisk provides?
Regards,
Ray Burkholder
Peter Hudec wrote:
hello,
my questns are about few * functionality.
1) how can I make call tranfer. Not call parking.
If I'm talking with some one a I want to tramnfer call to the another
extension, to the other person.
2) how can I make call confernece. Not Meetme
If I'm
Is an owen irq required ?
The card shares one irq with other devices
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Auftrag von Michael
Bielicki
Gesendet: Mittwoch, 29. Oktober 2003 17:33
An: [EMAIL PROTECTED]
Betreff: Re: AW: [Asterisk-Users] probs with
Gus,
There has obviously been a lot of postings relative to why won't nat work
and several responses say it does, and many more say it doesn't. In the
interest of using your posting as just one example, I'm going to walk
through the technical stuff for only this one example. There are many
more
I am trying to set up IAX with Voicepulse. When I turn on debugging I get the
following message when I call my PSTN number:
NOTICE[1142106560]: File chan_iax2.c, Line 4321 (socket_read): Rejected connect
attempt from 66.234.228.132, request '[EMAIL PROTECTED]' does not exist
Any help would be
--- Peter Zeltins [EMAIL PROTECTED] wrote:
That's for pointing out Walter Snel hack.
Adding his two additional features would not be
hard.
http://lists.digium.com/pipermail/asterisk-users/2003-October/024968.html
Any idea when these hacks will appear in CVS?
We should all hope never.
Title: RE: [Asterisk-Users] Software FAX
Hi,
My earlier post didn't make it to the list (over 40K and awaits moderator
decision for some time now).
We've successfully received several faxes. The problem is that all the viewers
we use show them compressed to the left side (when using fax
Is there anyway to turn off the call waiting beep in the grandstream and/or
cisco ata186?
I have a dial statement in my extensions.conf that rings 5 phones at the
same time by combining them with the in the dial statement.
i.e.) exten =
On Wed, 2003-10-29 at 10:41, Ray Burkholder wrote:
I can get a hold of a number of refurbished Carrier Access Bank I (T1/FXS)
units for a pretty respectable price. Will they do the job based upon your
comment that they support 'Answer Supervision'? How about the Caller ID,
MWI stutter that
It may not be *exactly* what you're looking for, but try:
http://fwd.pulver.com/callme.php?userid=411
In examining the source, it seems you can put any SIP address, not just
FWD ones, though there doesn't seem to be any overt SIP registration
going on.
- Original message -
From:
2003-10-22 I believe.
JT
Thanks a bunch you were on the money. Do you know about when that changed?
John Todd wrote:
Grandstreams phones can't call out with the latest CVS, anyone know what the
last good CVS date was?
You may be experiencing difficulty due to bad codec permissions,
since
The instructions they sent you (and me) are slightly faulty.
in iax.conf:
context=VPWS ;I'm not sure what VPWS stands for.
would better be
context=from-voicepulse ;(for example...this is what I use)
then in extensions.conf: (this is the simplest example)
;snip
[from-voicepulse]
I hate to say it, but I see so many people quote error messages
that
read .c line nnn and then ask what's up?. You really need
to
also post the lines of code around nnn so we can read them.
the lines numbers depend on when you did the CVS checkout. We
all have different working
Can anyone provide me with a current config for recieving calls with
Iconnecthere? I'm having some difficulty with it...
Regards,
Phillip
--
Phil Jackson, President CEO
The Jackson Group - Intelligent IT. (TM)
www.jacksongrp.com
___
Asterisk-Users
What kind of configuration would I need to simply forward all the calls from
Asterisk to a Class 5 Softswitch using SIP. All I would have is IP connectivity
to the Softswitch. Thanks - DL
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Well, my hosts hack-on-hack didn't work...internal clients could register
with * using the hosts-hacked FQDN, and * could register with (for
example) FWD and iconnecthere, but on calls in either direction, I only
got a few seconds of audio, then silence (though debugging showed what
looked like a
Sean Rodger wrote:
Is there anyway to turn off the call waiting beep in the grandstream and/or
cisco ata186?
I have a dial statement in my extensions.conf that rings 5 phones at the
same time by combining them with the in the dial statement.
i.e.) exten =
Christopher Stephens schrieb:
Is there SIP client which work with Asterisk and can be embedded in a
HTML page ?
It may not be *exactly* what you're looking for, but try:
http://fwd.pulver.com/callme.php?userid=411
[..]
Unfortunately this seem to work with Internet Explorer, only.
rgds
pos
Dear,
WhenI start asterisk -vvgrc and I
ask 'sip show peers', I don't get the ip adress in the 'Host"
field.
Name = phone1 and phone2
Host=unspecified
mask 255.255.255.255
port = 0
status = unmonitored
I can ping the two phone's and get a reply (also
from the laptop)
phone ip adres
Hi List,
I have two Cisco ATA, one of them with two phones attached, and the
other with just one phone. The ATA with two phones is behind a NAT, and
Asterisk and the other ATA have public IP addresses. I can place and
receive and blind transfer calls between them all. (Sometimes I loose
I spoke with someone today who is interested in an IP Centrex solution that
starts with about 3500 extensions in a multi-tenant application. And
growing from there.
I'm wondering about scalability of Asterisk. I'm trying to put my head
around how to put the whole thing together, if it can be put
Hi!
Thanks for the tip!
Okay, looked a little around AGI and it didn't look to hard doing a script
that read which phones that should answer which group from an external
textfile, and such file would be quite easy to modify with a CGI-script. And
I tried it with a static extensions.conf like
Why not just use appqueue?
On Wed, 29 Oct 2003, Lars Fredriksson wrote:
Hi!
Thanks for the tip!
Okay, looked a little around AGI and it didn't look to hard doing a script
that read which phones that should answer which group from an external
textfile, and such file would be quite easy to
Hi!
Is there anyone that are using a E1-channelbank and have any tips about some
type? Im looking at the TE410P and use one port for a PRI (Euro-ISDN, I
think we're using some slightly modified version here in Sweden, but I'll
check that tomorrow) and connect one port to a channelbank for 30
AFAIK, it is not the channel bank that encodes the caller id at all.
The channel bank only has to support sending on hook audio. The caller id
is sent by asterisk as inband audio, and the channelbank just passes it on.
I believe it is just a modification of a regular low speed modem format -
At 21:39 29/10/03 +0100, you wrote:
Hi!
Is there anyone that are using a E1-channelbank and have any tips about some
type? Im looking at the TE410P and use one port for a PRI (Euro-ISDN, I
think we're using some slightly modified version here in Sweden, but I'll
check that tomorrow) and connect
Just remember part of the design of the TE410P is that you can use T1
channel banks (you only get 24 ports) , if you buy these in America they
are significantly cheaper than E1 channel banks. Just assign one of the
incoming ports on the TE410P to be T1 not E1.
unfortunately you loose 6
Hi,
Is it possible to listen in on an existing call
already, say between the caller and callee? So the 3rd person would listen to
the caller and callee on an existing channel without theknowledge of
either the existing caller and callee.
Thanks,
John Haigh
Check out the ZapBarge application if you're wanting to do this on Zap
channels.
Jared Smith
On Wed, 2003-10-29 at 13:51, John Haigh wrote:
Hi,
Is it possible to listen in on an existing call already, say between
the caller and callee? So the 3rd person would listen to the caller
and
How do I get Gnophone to register to my Asterisk server? I have set up iax.
conf as follows:
[tim]
type=friend
;username=tstornes
host=dynamic
;defaultip=207.194.60.56
secret=
context=from-iax
callerid = Tim 5000
auth=plaintext
qualify=10
permit=0.0.0.0/0.0.0.0
and extensions.conf includes
Use a T-1 channel bank and a E-1 telco circuit. This is the exact reason
why Digium built the TE410P.
Jeremy McNamara
Sergio Serrano Revuelto wrote:
I need connect up to 100 analog phone to a H.323 network through *. I
think use TE410P, But I need to know what channel bank is better. I use
Sean Rodger wrote:
Is there anyway to turn off the call waiting beep in the grandstream
and/or
cisco ata186?
I have a dial statement in my extensions.conf that rings 5 phones at the
same time by combining them with the in the dial statement.
i.e.) exten =
Hi All-
I'm working on a project that will have remote (internet)access to an *
server through SIP phones, either soft or hard ones.
Does anyone have any experience to share about which SIP product they
are using under similar conditions, as well as which codec is being used
and bandwidth usage?
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