Peter Zeltins wrote:
Did you check debug on cisco - I had this problem with AUEP packet on
ata-186 , cisco generate 510s error when you ask for unsupported
packages ...
try to edit chan_mgcp.c to set MGCP Version to 0.1
I'm trying to link up Cisco MGCP-enabled router (residential gateway)
AudioCodes has one 24 port fxs sip interface, i have one 8 port fxs with
SIP up and running with *
Michael Kane wrote:
The Cisco 242x (20 or 21), has a 24 port analog interface that supports 16
FXS and 8 FXO. I've delpoyed hundreds of these IAD's signaling with MGCP.
Not sure if it supports SIP
I suppose "D/[0-9#*](N)" is the thing it does not like. You should omit
D/ in the beginning and try then.
e
Peter Zeltins wrote:
I'm trying to link up Cisco MGCP-enabled router (residential gateway) with
Asterisk, and it looks like some sort of protocol mismatch, could it be MGCP
0.1 vs 1.0?
L
Hi,
Yes, each port can be addressed by *, as it behaves like a separate sip /
h323 endpoint. The "connector" is just a way to allow the voip box to have
24 connections, and they are just standard rj11 connections. Another way is
just to use 3 x 8-port gateways on separate IP addresses.
You can us
Hi all.
I have come to the conclusion that there just isn't anything out there
for allowing SIP and NAT to work together nicely. This is rather amazing
considering that as far back as March 2000 there are documents
describing how to do it.
So I've started a really simple SIP and RTP proxy project
Hi
I want to set up * as a conference bridge. I would like to be able to
conference is SIP calls (up to 12)
I am looking through all available documentation for * to get info on how it
is done. No luck so far.
Can somebody direct me to the info in this subject?
Thank you
Serge
This is covered in asterisk/channels/h323/README
RTFM
Jeremy McNamara
Peter Zeltins wrote:
I've checked everything (pwlib + openh323 + asterisk) out of CVS, compiled,
and chan_h323 module does not load with "undefined symbol
_ZTI19H323AudioCapability". What could be the problem?
Peter
It also works passably with nikotel.
Lubomir Christov <[EMAIL PROTECTED]> writes:
> Hello,
>
> Asterisk Is working with IconnectHere (SIP) and MicroTelco (H323 -
> with chan_h323) without any problems. You have to buy g729 codec
> license from digium if you haven't some hardware wich support g723
Hello,
Asterisk Is working with IconnectHere (SIP) and MicroTelco (H323 - with
chan_h323) without any problems. You have to buy g729 codec license from
digium if you haven't some hardware wich support g723 codec.
Best regards
Lubo
Chris Mason wrote:
As a newbie, can anyone advise me if Asterisk
My new company is going to enter a bid competing with 3Com and Siemens
solutions from others.
Voice service is a PRI T-1 with a block of 100 DID numbers, a second T1 that
will carry Integrated voice/data, 12 voice/768K for redundency.
I asked this a week or two ago, now I have more details.
B
I question whether this request is appropriate for the mailing list but I
don't know where else to turn. There are some vendors who monitor the list
and sometimes asterisk users have old equipment they no longer need. I'm
building out my asterisk system and need to purchase the following
components
> Does anyone know if someone makes a hard video phone for SIP.
>
> Dave
I was curious about this too as the video support has been going in.
Below are links to what I found. As with some of this stuff, I can't
really find who the manufacturer is
Leadtek is possibly the manufacturer of this d
I've checked everything (pwlib + openh323 + asterisk) out of CVS, compiled,
and chan_h323 module does not load with "undefined symbol
_ZTI19H323AudioCapability". What could be the problem?
Peter
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Hey Mark,
On Sun, Jun 29, 2003 at 17:08:00 -0500, Mark Spencer wrote:
> > I was just testing with MeetMe, which was one of the things where MEC3
> > went wrong for me in the past. After around 8 channels from an
> > E100P-connected PRI joined the conference, everything became one big
> > chaos o
I'm trying to link up Cisco MGCP-enabled router (residential gateway) with
Asterisk, and it looks like some sort of protocol mismatch, could it be MGCP
0.1 vs 1.0?
Look at this (x.x.x.99 is the router, x.x.x.98 is Asterisk):
MGCP read:
NTFY 2 aaln/[EMAIL PROTECTED] MGCP 0.1
X: 0
O: hd
from 192.
As a newbie, can anyone advise me if Asterisk can route international calls
to a US based service such as Net2Phone so we can take advantage of the
internet and save on calls?
That would be my main reason for an Asterisk based PBX.
Chris Mason
[EMAIL PROTECTED]
Box 340, The Valley, Anguilla, Br
Moshe,
I was having the same problem with my software only asterisk pbx setup. I
was using two kphones on different machines, connecting through a machine
running asterisk. They would connect just fine, but voice was not getting
routed through. I installed linphone, which can be found at
>From Archives courtesy of Don:
Email Dated: Fri, 4 Apr 2003 17:51:29 -0600
Normal RJ48 jack (some call it an RJ45 which is the same physically)
out of equipment on pins 1 & 2 (Tip & Ring)
into equipment on pins 4 & 5. (Tip & Ring)
Normally tip is the lower number so I would *guess* that you nee
Anyone try Zultys Zip2 Phones with Asterisk? Supposed to be
SIP, so it should work..
?!
Thanks,
Rich
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What is the pin layout of the E1 sockets on the E400Ps? What pins are for
the TX and RX pair?
Regards
Fredrik
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Just wanted to let everybody know that Martin not only gave us big support but
solved our core problems in getting our cardplatform working which now server
a happy customer base in 2 countries.
Man that was really great of you, I was extremely impressed.
Michael Bielicki
_
I just noticed that app_queue here
rings together all available extensions, which may not be the best for a call
center.
Is this the correct functionality or something specific from
my installation?
PauloHM
Ah, I'm wrong. It only the 2432 only supports data WICs and the 4 port
FXO VWIC. The IAD2431 supports digital voice, though.
-Original Message-
From: Michael Kane
Sent: Mon 6/30/2003 2:38 PM
To: [EMAIL PROTECTED]
Cc:
Subject: Re: [Aste
That reminds me, Cisco has a new device out, the IAD2430 that has T1
ports VWIC, and FXS ports. It will proably cost a bundle, though.
Looks like the IAD2432 will handle 24 analog ports, up to two Data T1s
and a voice T1 using the VWIC port. Supports SIP and MGCP.
-Original Message--
So, as usual, about 30 seconds after I send off the message I come to the
realization that I've done a Dumb User Trick (aka "operator error").
* noload of chan_oss.so & chan_alsa.so does disable the console.
* I can call from a softphone on the PBX to the PBX's voicemail and
transmit audio!
*
Sorry Andy, the plan on paying $1700 up for the IAD...
Michael Kane
To-Talk Communications LLC.
37 Sandusky Dr.
Wareham, Ma. 02571
508-295-2826
- Original Message -
From: "Andy Powell" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, June 30, 2003 2:51 PM
Subject: Re: [Asterisk-
The Cisco 242x (20 or 21), has a 24 port analog interface that supports 16
FXS and 8 FXO. I've delpoyed hundreds of these IAD's signaling with MGCP.
Not sure if it supports SIP yet. Hope this helps...
Mike
Michael Kane
To-Talk Communications LLC.
37 Sandusky Dr.
Wareham, Ma. 02571
508-295-2826
Steven,
As per your advice, I have tried to not load the console -- which in this
case, if I understand correctly, would be the chan_alsa.so and chan_oss.so
modules.
* I never load chan_alsa.so, because chan_alsa doesn't work for me and I
haven't figured out what the problem is just yet.
* If
I have just rebuilt my * box back to last weeks 06-20 CVS build beacuse
after getting the latest I could not hear ANY voice prompts. I have a
T1 card and a dual proc box that has been running just fine up till this
weekend. I tihnk some of the format changes affected my install.
Jun 27 16:12:38
Hi Tan,
Thanks for the reply. I'll end up asking a load more questions now...
What sort of prices are we talking about for the 24 port
VoIP gateway?
I assume that each port is individually addresable by *?
As I recall the 24 port gateways tend to be terminated at the FXS side
as some 'wierd' c
On Mon, 2003-06-30 at 13:07, Don Pobanz wrote:
> Just some clarification here.
Thank you for the corrections.
> On Monday, June 30, 2003 12:04 AM, Steven Critchfield
> [SMTP:[EMAIL PROTECTED] wrote:
> >
> > As for slips and bipolar violations...
> > T1s are just high speed serial lines. A sleep
Hi,
We provide asterisk-based solutions to customers based in the uk. One of our
customers (9 users) is trialling our low-end solution which comprises of a
box with 2 x X100P (analogue line) cards installed, and a voip carrier for
outgoing calls. This customer intends to have 13 extensions in his
Question for users of Agents and Queue.
In our queue.conf file we have entries for both agents and some
devices. We do this so that some dual pupose dispatchers can also
take a call if they have a spare moment.
We went live with a dispatch client over the weekend
a number of issues came up the mos
Just some clarification here.
On Monday, June 30, 2003 12:04 AM, Steven Critchfield
[SMTP:[EMAIL PROTECTED] wrote:
>
> As for slips and bipolar violations...
> T1s are just high speed serial lines. A sleep is when you loose sync
> with the far side and when you see a 1 come across the line, you m
If you have the console working, then you can't use a softphone on the
same machine. If you want to test with a softphone, set the console
driver to noload and try again. This would probably be the same for your
dialing to other SIP phones.
I'm guessing that ALSA doesn't allow more than one connec
I have a software-only PBX set up. I can register various softphones and
they will call each other -- but I've never succeeded in getting any
voice routed from any of the softphones. Only the console will transmit
audio.
I am writing to ask if I have missed some obvious step in configuring
the
Tim,
a good comprehensive answer to the question...certainly gave me a few things
to think about. I do have a few questions though, since I'm in Europe.
Has anyone in Europe set up something equivalent to what Tim suggested?
What sort of prices did it work out at?
How did you solve the channel
ISDN PRI E1 is enough to receive DID and CallerID (ANI).
Martin
On Mon, 30 Jun 2003, Surajee Ratnayake wrote:
> hi everybody,
>
> my question is specific to ISDN signalling,
> in my set up, i want to get cli and dnis into my asterisk box, and i am going to use
> ISDN PRI E1s coming from telco.
>
I’m getting this intermittent problem, sometimes a zap
channel gets stuck after a call. Below is a snapshot of the channel. Any ideas
what can be happening?
Name: Zap/1-1
Type: Zap
UniqueID: 1056988772.10
Caller ID: (N/A)
DNID Digits: (N/A)
Please help.
this is generally a problem with arguments that contain "," or "(", i
think.
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Auftrag von Thomas
Haeger
Gesendet: Sonntag, 29. Juni 2003 15:18
An: Asterisk User
Betreff: [Asterisk-Users] PGS
If this is for commercial use, especially if you are going to be selling
this solution, I would suggest that you don't even offer the choice of
analog lines except in the smallest of offices. Unless you like to
spend a lot of unbillable time supporting them :)
Most businesses of more than 8 to 1
Correct, not possible.
On Mon, 2003-06-30 at 10:31, Bradley Greep wrote:
> I've done a quick search through the old mailing list archives to see if I
> could dial an outside line through * using a quicknet linejack setup as an
> FXO.
> Is it posible to do so?
> It appears from the old mailinglis
I've done a quick search through the old mailing list archives to see if I
could dial an outside line through * using a quicknet linejack setup as an
FXO.
Is it posible to do so?
It appears from the old mailinglist from marko.net that this not possible?
Thanks.
Bradley Greep
> Minimum budget usually screws you for future expandability. If you buy a
> 3 cylinder car for less than a full size car you can't go towing a
> moving trailer.
It's not for my personal use: I'm trying to tailor some commercial offer for
differents kind of customers. So I need to find the compe
Does anyone know if someone makes a hard video phone for SIP.
Dave
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problem solved - forgot to update zaptel.conf
stupid me!
thanks guys :)
Dave
- Original Message -
From: "Dave Alan Caruana" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, June 30, 2003 2:55 PM
Subject: Re: [Asterisk-Users] E100P installation sheet
> yeah thanks :)
> i've co
Whilst in a call using the mec3 echo canceller today I had period of about
20 seconds of speech distortion. It's hard to describe but to me the call
sounded as though we were having the conversation in a bathroom with some
extra noise bursts and echo thrown in. I could carry on the call, with
On Mon, 2003-06-30 at 06:14, Stefano Corsi wrote:
> >To connect four analog phones to an Asterisk server, you would need
> > the TDM400P, made by Digium. That card is available for $305.00
> > (USD) and details can be found on
> > http://www.digium.com/?menu=wildcard_tdm400p .
>
> What about
yeah thanks :)
i've compiled all OK and still can't get my new installation working ..
doesn't seem to recognise the E100P board, even though the
modprobe wct1xxp command goes through OK and says
the board is found ..
anyone have any ideas ?
same board was working fine on my old server.
Dave
The AT command set on a modem is half duplex and unsuitable for
interactive work. Tie that to the multiple variations of how the AT
command set was implemented and what codecs where supported by each of
the modems and you have a pretty nasty mess.
The closest idea so far would be to try and get at
On Mon, 2003-06-30 at 00:35, Andy Hester wrote:
> Steven,
> I thought that "1" would mean that my T100P card would set the timing for
> the line. Is this incorrect? If I am reading this wrong then please set me
> straight.
>
> My carrier has their end set to be the sync source. If I set t
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Monday 30 June 2003 14:01, Dave Alan Caruana wrote:
> I seem to have lost the sheet of paper that comes
> with an E100P card and tells you how to compile
> the stuff it requires to run.
> Could someone be kind enough as to mail
> me a PDF of it ??
Hi - Jeremy
Well - I only refer to what I've been told : "Cisco CallManager use skinny
protocol communicate to their IP phones, which is a subset of H.323 - But
communication between CCM and Voicegateway like 6608-E1/T1 board is using
Skinny Client Control Protocol". But then again Cisco could say
hi ..
maybe someone can help me,
I seem to have lost the sheet of paper that comes
with an E100P card and tells you how to compile
the stuff it requires to run.
I'm trying to move my Asterisk to a different
box and at this time totally stuck.
Could someone be kind enough as to mail
me a PDF of it ?
hi everybody,
my question is specific to ISDN
signalling,
in my set up, i want to get cli and dnis into my
asterisk box, and i am going to use
ISDN PRI E1s coming from telco.
To get cli and dnis, do i need to apply for QSIG
from the telecom, or is there
some other type? and i got to know t
yes, you can connect an X100P (FXO) to an extension off the PBX
or
you can connect a TDM400P (FXS) to a trunk off the PBX
and then use FLASH/SENDDTMF to control the PBX
--
Mirza Wasim Baig | Principal Consultant | Convergence Business Systems PK
#48, St 32, Sector F-6/1, Islamabad, Pakistan 440
i read that quite all of you are using the driver for the isdn that
you are using is the digium... i use I4L and i got a problem with some
call through normal PBX, can someone tell me where i can download the
digium drivers for the ISDN card?
i didn't find it... but i'd like to try if is a driver i
What is wrong with trying to use standard modems?? Do they work at all?
> It is not worth it..Get an X100P.
>
>
> Jeremy McNamara
>
>
>
>
> Angelo Sampietro wrote:
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John Todd wrote:
I really should be doing something better on this beautiful weekend, but
I'm trying to save myself some time for later projects by documenting
some things that have been particularly troublesome in the past. That
being said...
I've written up a configuration guide for the Cisc
thanks :)
how does it works?
i get an x100P and afther i connect to a normal line to another PBX?
Monday, June 30, 2003, 10:56:28 AM, you wrote:
JM> It is not worth it..Get an X100P.
JM> Jeremy McNamara
JM> Angelo Sampietro wrote:
>>hi,
>>i need to do a demo of asterisk in a office that
>To connect four analog phones to an Asterisk server, you would need
> the TDM400P, made by Digium. That card is available for $305.00
> (USD) and details can be found on
> http://www.digium.com/?menu=wildcard_tdm400p .
What about five analog phones, or let's say, eleven? It's just a matter
It is not worth it..Get an X100P.
Jeremy McNamara
Angelo Sampietro wrote:
hi,
i need to do a demo of asterisk in a office that doesn't have an ISDN
line, can someone tell me if asterisk works also if i use a modem card
mapped on /dev/ttyl0 insted of use a isdn card?
did someone try to do somet
hi,
i need to do a demo of asterisk in a office that doesn't have an ISDN
line, can someone tell me if asterisk works also if i use a modem card
mapped on /dev/ttyl0 insted of use a isdn card?
did someone try to do something similar before?
thanks for your help!!
Angelo
__
DTMF seems to be out-of-band as a default (or, at least, it's
auto-negotiated) and LBRCodec doesn't require mucking with, so I only
change the G.711 VAD settings for each channel. More wasteful, but
sounds better when you're using cordless phones.
JT
John's guide goes into a lot more detail,
You'll note that I have a link to that guide in the "References" section.
There have been some changes in the new code that were not referenced
Shawn's site, and also a more complete explanation of each field was
required. At least, I had to learn a lot more about the ATA-186 than
is contained
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