I have the same problem. My setup is:
Suse 8.0 with 2 single FXO cards with 2 lines coming in and 1 ATA186
providing the connection to two analogue phones.
What I get is an almost random hung up during calls (both incoming and
dialed). Sometimes tha call can last for 30 minutes without any probl
Muiz Motani wrote:
Google search does not work very well. For example, a search of the
keywords "opencall.org" and "asterisk-users" on Google turned up
nothing useful.
http://search.voip-forum.com/cgi-bin/htsearch?words=opencall.org+asterisk-users&config=voipsearch
/O
Christopher Lee wrote:
Out of interest is there any estimated date for the TDM400 FXO modules
receiving A-tick certification?
And has anyone compared the FXO modules with the X100P on Australian
exchanges/equipment? Do they perform any better than the X100?
Cheers,
Chris Lee
___
On Thu, 19 Aug 2004 15:53:38 -0400, James Freire <[EMAIL PROTECTED]> wrote:
>
>
> Hi All,
> I have tried searching everywhere but I cannot find a definitive answer as to if and
> how the conference button works on the BT100. Could anyone be kind enough to fill me
> in on some info on how to us
Hi,
> -Original Message-
> > Line => svip10
>
> That did it. The phone registered with * and a debug msg flys
> up when I pickup/put down the reciever.
>
> When I pick up the handset, I can hear a dialtone. But
> pressing numbers on the keypad doesn't do anything other than
> show up
On Thu, 19 Aug 2004, Kris Boutilier wrote:
> Or perhaps how to configure and refer to two parallel IAX trunks with
> different codecs?
>
> I have a situation where I'm using G.729A as my IAX trunking codec. Now I
> need to push some short duration, low bitrate modem traffic over the link (a
> c
Hey Shaun,
Because of the Isdn4Linux DTMF issue, I don't want one of those cards.
I've already spent too much time messing about with my current card.
I have a Traverse NetJet card which I got working without any problems at all. I'm pretty sure
all I had to do in order to get DTMF working all I h
Yes, they do perform better. Less echo, better busydetect.
Simon
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Christopher Lee
Sent: Friday, 20 August 2004 15:29
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Atick Certification on FXO Modules (Austr
Out of interest is there any estimated date for the TDM400 FXO modules
receiving A-tick certification?
And has anyone compared the FXO modules with the X100P on Australian
exchanges/equipment? Do they perform any better than the X100?
Cheers,
Chris Lee
___
Thanks for your advice everybody (replying collectively).
Now I'm in a bit of dilemma. I'm not reselling the system, it's all
for my home office (where my Asterisk install is).
I've sent off a couple of emails, so I'll see what happens :-)
Thanks again,
Shaun
___
Hello Shaun,
The AVM cards bought in Australia are Atick certified, hence the extra
cost (although the card is identical to overseas models). Hence the
extra cost.
If you don't care about certification, I personally would try and get
HFC-s cards off ebay. They are very cheap and should work bett
On Fri, 2004-08-20 at 14:21, Shaun Ewing wrote:
> Hello all,
>
> I was wondering if anybody knows where one might obtain a PCI ISDN
> card supporting a single BRI for use with Asterisk in Australia (and
> using something like chan_capi).
www.atp.org.au have been nothing but helpful and knowledgea
Shaun,
Contact me off list - Simon.Brown at otterson.com.au
I might be able to help you.
Simon Brown
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Shaun Ewing
Sent: Friday, 20 August 2004 14:21
To: Asterisk Mailing List
Subject: [Asterisk-Users] Wher
hi, contact ATP (www.atp.org.au) - they have digium cards and they work
well.
cheers
greg
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Shaun Ewing
Sent: Friday, 20 August 2004 2:21 PM
To: Asterisk Mailing List
Subject: [Asterisk-Users] Where to purchase I
Shaun
With chan_capi your only real choice is the fritz or an eicon diva server.
If you thought the fritz was expensive then close your eyes for the diva. I
think the 2 channel BRI card is about $1500 here.
You might want to try ebay. You can pickup a fritz for less than 20 Euros
I'm sure.
Hello all,
I was wondering if anybody knows where one might obtain a PCI ISDN
card supporting a single BRI for use with Asterisk in Australia (and
using something like chan_capi).
Because of the Isdn4Linux DTMF issue, I don't want one of those cards.
I've already spent too much time messing about
That did it - thanks.
Simon
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Darryl Ross
Sent: Friday, 20 August 2004 14:05
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] CallerId
Simon Brown wrote:
> In my zapata.conf, I have
> callerid="" <>
Tha
On Fri, 20 Aug 2004 04:51:26 +1000, Shaun Ewing <[EMAIL PROTECTED]> wrote:
> Hello all,
>
> I currently have an Eicon Diva Client isdn card using i4l. Outbound
> dtmf doesn't work (and never has), but there has been an annoying
> problem with false dtmf detection in calls (that could be triggered
Simon Brown wrote:
In my zapata.conf, I have
callerid="" <>
That doesn't look right to me. Try:
callerid="Unknown" <>
Cheers
Darryl
--
Darryl Ross
Senior Network Engineer
OEG Australia
Email: [EMAIL PROTECTED]
Phone: 08 81228363
Office: 08 81226361
If you want to live up to the whole "There is mor
More on this issue - this is what I see on the console:
-- Called 201
-- Called 202
-- Got SIP response 400 "Bad Request" back from 10.2.2.102
-- Got SIP response 400 "Bad Request" back from 10.2.2.104
== No one is available to answer at this time
I am using Cisco 7940 phones
Si
In my zapata.conf, I have
callerid="" <>
so if an incoming call doesn't set or suppresses it's callerid then my phone
will show "unknown". I have found that if the callerid on the incoming call
is suppressed, then the call goes straight to Voicemail.
Has anyone seen this problem?
Simon Brown
___
The "strange ports" mean they are probably behind a NAT device that
does PAT.
But the strange thing is that nothing has changed... their hardware
remains the same... everything had been working fine for the past 4
weeks.
--
jeremy bogan[ [EMAIL PROTECTED] ]
segment publishing - design.devel
Jeremy Bogan wrote:
For some unknown reason, two of my IAX peers started registering on
strange ports. Nothing has changed in the config, but they cannot make
calls to me, however I can still make calls to them. In my IAX2 peers,
the following is showing:
user1/user1 203.XXX.XXX.XXX (D) 255.
On Fri, 2004-08-20 at 09:42, Lee Allen wrote:
> Okay, now my script...
>
> It creates an outoing call in /var/spool/asterisk/outgoing, pulling
> information from a database (assuming I learn some perl and mysql, or
> something!)
> THAT file (outgoing call queue) would have to...
> - call the given
> That was part of my problem.
>
> I can now get the 600 to download XML, I tried using
> http://phone-xml.berbee.com/menu.xml and the phone displays "XML Error
> (1,0) syntax error". I'm guessing this is because the XML files at that
> location are formatted for the Cisco phones. Anyone have do
> "Rich" == Rich Adamson <[EMAIL PROTECTED]> writes:
Rich> If they are suggesting the sip negotiation process is trying to
Rich> negotiate something like "silence-suppression=off", and their
Rich> equipment won't handle _anything_ other then
Rich> silence-suppression=on, then that sounds like
Doesn't sound too unreasonable or unusual -- my previous PBX had message
delivery. In the sense of usability, I would probably move the prompts
around a little -- i.e. dial number, play a short prompt on answer, wait
for #, THEN play the customer message. Might also give them opportunity
to rewin
For some unknown reason, two of my IAX peers started registering on
strange ports. Nothing has changed in the config, but they cannot make
calls to me, however I can still make calls to them. In my IAX2 peers,
the following is showing:
user1/user1 203.XXX.XXX.XXX (D) 255.255.255.255 4585
> Or perhaps how to configure and refer to two parallel IAX trunks with
> different codecs?
>
...
>
> Any suggestions?
Well, push comes to shove, you can define two different entries in
iax.conf with different protocols. That is:
Box A:
[boxb-voice]
type=peer
host=boxb
disallow=all
allow=gsm
tr
On Tue, 17 Aug 2004 10:19:17 -0400, Steve Szmidt <[EMAIL PROTECTED]> wrote:
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> Hmm,
>
> My music on hold has always worked fine. But I discovered that under incoming
> IAX2 calls they don't get any MOH! All I could find was a comment saying let
>
That worked! Thanks!
-Ronan
On Wed, 2004-08-18 at 23:40, Darryl Ross wrote:
> Ronan wrote:
> > That does not work for me. I had tried that, but no luck. This is what
> > I have in there for it.
> >
> > context=darby
> > usecallerid=yes
> > musiconhold=default
> > echotraining=yes
> > echocanc
Thank you James and others for the great answers. Yes, I did try the google
site: tag but as someone else pointed out, google is not always up to date.
Is there any possibility of getting anonymous IMAP access to the archived
mbox files at lists.digium.com?
On 19 Aug 2004 at 10:52, you wrote:
> Sure, so long as that person gets the hint to hang-up when they hear the
> congestion tone...
I see what you mean... You have to be careful too, I've had a GrandStream
drop a channel (I'm assuming without sending * a BYE) and then * will keep
that channel open and there's no way (short of issui
I have been reading asterisk doc's for the past couple weeks, and
monitoring this list. I have to implement an unusual (I think)
application of asterisk. I have the beginnings of a plan, and I would
like to throw it up here for comments.
The application:
An after-hours emergency support "hotline
BTW, Ryan, Thanks for the info! :)
-Chris
___
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
Chris Shaw wrote:
This is lame, but AFAIK, the only way to do it is:
- Press Flash button & dial party to transfer to
- inform party about call & ask that party to hangup
- press flash again to return to original caller
- press transfer & dial number to transfer to
- press send
Couldn't you after t
If you don't want to have to talk through your
computer (ala X-Lite/Pro) then there's
the GrandStream BT101, it has some minor quirks
(e.g. 3-way calling does not work even though there's a button) but they should
be fixed in later firmware releases. Also GrandStream makes ATA devices (device
X-Lite by XTen works good. If you want more features, the X-Pro isn't
very expensive.
- Original Message -
From: John Williams <[EMAIL PROTECTED]>
Date: Thu, 19 Aug 2004 15:10:46 -0700 (PDT)
Subject: [Asterisk-Users] residential sip phone
To: Asterisk Users List <[EMAIL PROTECTED]>
Dear
Is anyone aware of a single FXS ATA with a built-in switch ie
2 LAN ports (other than the Cisco ATA 188)?
Grandstream 486 users: is it possible to disable the router/NAT
functionality and configure the WAN port as switched with the LAN port, effectively
giving you two switched Etherne
> This is lame, but AFAIK, the only way to do it is:
>
> - Press Flash button & dial party to transfer to
> - inform party about call & ask that party to hangup
> - press flash again to return to original caller
> - press transfer & dial number to transfer to
> - press send
>
> Ryan
Nevermind, th
Chris Shaw wrote:
I know this must have been asked before, but I was just wondering, the
manual says it can do attended transfers, has anyone gotten this to work
successfully? How did they do it?
This is lame, but AFAIK, the only way to do it is:
- Press Flash button & dial party to transfer to
- i
> I have to ask... why are you trying to get a SIP provider to work if they
> clearly aren't interested in supporting Asterisk?
>
> -A.
Most likely because they don't want to loose their DID number... A very
valid reason!
-Chris
___
Asterisk-Users mai
Is anyone successfully using the AGI script calleridnamelookup.agi (or
anything similar) ?
I get both name and number caller ID from my POTS line, but I'd save
money if I had them deliver ANI only.
I've downloaded and installed the AGI script calleridnamelookup.agi, but
I always get
-- Exe
On Thursday 19 August 2004 18:23, Dan Mahoney, System Admin wrote:
> SO what do the higher-end products use for timing?
They self-time with an internal timer.
-A.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo
On Thursday 19 August 2004 18:21, Dan Mahoney, System Admin wrote:
> Yes, that's what I've told them, too. Do you know of any software that I
> can use as a proxy which does support this?
I have to ask... why are you trying to get a SIP provider to work if they
clearly aren't interested in suppo
It sounds like you're all getting caught up in double negative language.
Is broadvox suggesting you "have to" use silence suppression? If they
are, that's not correct as that wouldn't have a clue whether you are
transmitting silence or not. Continuous voice with silence suppression
looks just exac
I believe you can do that kind of thing with SIP, using the
SetVar(${SIP_CODEC}="CODEC"}) in your extensions.conf... So, For example, if
the extension of your card machine were say '100' then you would do
something like this
[outgoing]
exten => _1NXXNXX,1,Gotoif($[${CALLERID_NUM} = '100']?2:3
Dear all.
I've placed an order for several Uniden UIP200 SIP phone to connect to
our Asterisk server but it seems that they're not going to be available
for another while.
The seller recommended the Ipdialog Siptone 2 instead which is a little
bit dearer (around $185 vs $145 for the Uniden).
Th
> >The power of Christ compels thee Not to buy Cisco...
> >
> >hehe J/K
> >
> >
> don't do that. my employer wouldn't like me for poking fun at their
> products. :-P
>
> actually, i'm planning an asterisk-based voip network and was thinking
> of using the 7940/7920 phones for the end sta
On Thu, 19 Aug 2004, Andrew Kohlsmith wrote:
SO what do the higher-end products use for timing?
-Dan
On Thursday 19 August 2004 17:59, Dan Mahoney, System Admin wrote:
Now they're complaining that asterisk is sending a Silence-Suppression OFF
request of some sort.
There's no way to turn this on i
On Thu, 19 Aug 2004, Andrew Kohlsmith wrote:
Yes, that's what I've told them, too. Do you know of any software that I
can use as a proxy which does support this?
-Dan
On Thursday 19 August 2004 17:59, Dan Mahoney, System Admin wrote:
Now they're complaining that asterisk is sending a Silence-Su
I know this must have been asked before, but I was just wondering, the
manual says it can do attended transfers, has anyone gotten this to work
successfully? How did they do it?
Is it possible to do attended transfers with the 'T' dial option? If so,
how?
-Chris
Chris Shaw
IS Manager
Water T
Or perhaps how to configure and refer to two parallel IAX trunks with
different codecs?
I have a situation where I'm using G.729A as my IAX trunking codec. Now I
need to push some short duration, low bitrate modem traffic over the link (a
credit card terminal). Obviously the modem audio isn't goin
Dear List,
Can anyone recommend a sip phone for residential use? (asterisk home pbx)
Thanks!!!
I was just on 70minute call (IAX2 -> Internet -> IAX2) and during that
time I heard several "pops", or "clicks". Each time it happened, I saw
the following message:
Aug 19 15:36:36 NOTICE[1173711792]: rtp.c:429 ast_rtp_read: RTP:
Received packet with bad UDP checksum
Any ideas what causes
On Thursday 19 August 2004 17:59, Dan Mahoney, System Admin wrote:
> Now they're complaining that asterisk is sending a Silence-Suppression OFF
> request of some sort.
> There's no way to turn this on in asterisk is there? (Yes, I know it will
> shoot call quality to shit.
Silence suppression mu
Well, in lieu of dropping us, Broadvox has transferred us to their lab
switch (keeping our DID's in the process).
Now they're complaining that asterisk is sending a Silence-Suppression OFF
request of some sort.
There's no way to turn this on in asterisk is there? (Yes, I know it will
shoot ca
Be aware that if you want to use SIP (you didn't mention you were) Park
still doesn't play nice with SIP transfers... It works, but you never hear
the announced parking slot... I think this is being addressed though... Also
there's BKW's nice valet_parking.so application which has more features tha
OK... Don't know what happened there, but I can blame it on OE's LAMENESS...
Anyway, as I was saying I have 3 BT100s and none of them do that, it must be
a firmware issue. Maybe it's a wierd isue with auto answer?
-Chris
- Original Message -
From: "Chris Shaw" <[EMAIL PROTECTED]>
To:
> Mike Schwartz wrote:
> >>>I'm experience echo on outgoing calls:
> >>> Snom 200 > Asterisk > T100P > PRI > called party
> >>>
> >>>I am getting echo on the Snom 200 phone. The called party does not
> >>>hear the echo.
> >
> >
> >
> >>Rule of thumb (i.e. a good starting point)
Couldn't see the forrest for all the fascinating tree-like applications that
are out there:
For future reference, see:
http://www.voip-info.org/wiki-Asterisk+call+parking
:-)
-Original Message-
From: Kris Boutilier [mailto:[EMAIL PROTECTED]
Sent: August 11, 2004 1:10 PM
To: [EMAIL PROT
Does Granstream BT100 Conference Button Work?Sounds like a firmware thing to
me, I have 3 of them and none of them do that..
- Original Message -
From: Kanuri, Seshu
To: [EMAIL PROTECTED]
Sent: Thursday, August 19, 2004 2:29 PM
Subject: [Asterisk-Users] Granstream BT100 Rings Once and Wait
Title: Does Granstream BT100 Conference Button Work?
Hi
Folks!
I have
another problem with BT100 Phone. Whenever someone calls me, it rings once and
stops. But the call is still on hold till I pickup.
How do
I increase the number of rings? Is this a * problem? or BT100
Issue?
Seshu
Simone Ricci a écrit :
administrator tootai ha scritto:
>
> From where you got port.h? I install tiff-3.5.7 (I'm running an RH73
> with those rpms installed) and didn't manage to create this file.
> tiffiop.h and tif_dir.h are parts of tar.gz package. I had to modify
> tiffiop.h and replace port.h
James Freire wrote:
> Sorry about that. I am in the US and using the Digium FXO TDM400 and
> I have enabled all the callerID options in my zapata.conf file.
Have you enabled verbose debugging in the console and confirmed that you're
receiving callerid from the PSTN?
What are you getting on your
> Well it does. It hangs up the connection, on my phone. Latest firmware.
: )
> - --
> Steve
lol... YAY!!!
___
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-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Thursday 19 August 2004 04:34 pm, James Freire wrote:
> Could I use the Flash button to do conferencing then??? If so.. how?
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Chris Shaw Sent:
> Thursday,
I have been adding site:lists.digium.com to my Google searches and that
seems to work well.
Bryan
Muiz Motani wrote:
This brings up a good point that has had me scratching my head for a long
time. Is there a good searchable archive of the asterisk mailing lists? I don't
particularly want to dow
> Muiz Motani wrote:
>
>> This brings up a good point that has had me scratching my head for a
>> long time. Is there a good searchable archive of the asterisk
>> mailing lists? I don't particularly want to download and keep
>> updated the full 206 MB of the asterisk-users .mbx file on my
>> lapto
Title: Does Granstream BT100 Conference Button Work?
I'm sure you could, you could also use a MeetMe
conference room...
- Original Message -
From:
James
Freire
To: [EMAIL PROTECTED]
Sent: Thursday, August 19, 2004 1:34
PM
Subject: RE: [Asterisk-Users] Does
Gra
I have an asterisk server running on Redhat 8.0 with a Digium TDM400P
w/4 FXO modules (TDM04P)
There are 2 lines going into the Digium card. One line is a Vonage
digital line, and the other line is a Comcast voice line. I have a SIP
Grandstream 100 phone connected to the Asterisk server. I als
I had no problems updating the firmware.
Is anyone able to use the MWI light on their Avaya 4602? Thats the
only thing on mine that I cant do.
On Wed, 18 Aug 2004 08:38:18 -0700, Aaron Johnson <[EMAIL PROTECTED]> wrote:
> Tenorio, Leandro wrote:
>
> >Just guessing, but 've you tried the to renam
Title: Does Granstream BT100 Conference Button Work?
Could
I use the Flash button to do conferencing then??? If so..
how?
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of Chris
ShawSent: Thursday, August 19, 2004 4:28 PMTo:
[EMAIL PROTECTED
a better google query is:
opencall.org site:lists.digium.com
nevertheless, google is not up2date.
klaus
Muiz Motani wrote:
This brings up a good point that has had me scratching my head for a long
time. Is there a good searchable archive of the asterisk mailing lists? I don't
particularly want to
Title: Does Granstream BT100 Conference Button Work?
Nope, it does nothing... It's not an * problem
either, the button just does nothing... I think they're planning on making it
work in a future release, don't quote me on that... for now it just occupies
space..
-Chris
- Original
Steven,
Spot on. I haven't checked out since June 26th. However, my bison was the
errr lagging or shall we say pre-historic (v1.28) :)
Thanks for the clue.
Best regards
Steve Beaumont
On Thu, 2004-08-19 at 13:56, [EMAIL PROTECTED] wrote:
> I have loaded the latest cvs (19/08/04). When I try
Im using 'notransfer=yes' in the iax.conf so it shouldn't happen,
What I'm doing is bridging a 2 legged call over iax using .call files.
The .call file initiates the first leg and drops the user into a context
that calles an agi script that checks against a db (like max call length)
then automatica
On Thu, 19 Aug 2004, Anton Yurchenko wrote:
> The Asterisk sends the replies to port 1031, the outbound port that
> Pingtel used to send the message. This is wrong. In the REGISTER,
> Pingtel specified a contact header field with no port, which means use a
> default port of 5060. Asterisk is vi
This brings up a good point that has had me scratching my head for a long
time. Is there a good searchable archive of the asterisk mailing lists? I don't
particularly want to download and keep updated the full 206 MB of the
asterisk-users .mbx file on my laptop. The current format is just not
s
Title: Does Granstream BT100 Conference Button Work?
Hi All,
I have tried searching everywhere but I cannot find a definitive answer as to if and how the conference button works on the BT100. Could anyone be kind enough to fill me in on some info on how to use the conferencing feature, as we
This is repetetive.
A message from yesterday from seth:
On Wed, 2004-08-18 at 14:45, David Filion wrote:
Does anyone know of an alternate source for spandsp? opencall.org is
down and all the links returned by Google just point to the dead site.
Thanks
David Filion
I threw a copy up here for y
Guten Tag,
ich bin im Moment auf einer ausreichend einsamen Inse
und werde Ihre Mail lesen, wenn ich wiederkomme.
In Notfällen bin ich unter +49-160-4439821 zu erreichen.
=
Hello,
I am currently on a reasonably remote island,
and I'll read your email when I get back.
In case of an eme
Hello,
I have a Pingtel Xpressa and trying to get it working with *. When the
phone tries to register, it sends out a REGISTER request and * replies
with PROXY AUTHENTICATION but phone never replies back with the right
info and just sends REGISTER again and again. This is what Pingtel
support t
On 19 Aug 2004 at 11:44, you wrote:
>
> On Aug 19, 2004, at 11:30 AM, Muiz Motani wrote:
> > Does anybody know what happened to the opencall.org website? I can't
> > get
> > into the home page or the ftp site.
>
> His DNS servers seem to be down. opencall.org is served by
> name[12].coppice.o
On Thu, 2004-08-19 at 13:56, [EMAIL PROTECTED] wrote:
> I have loaded the latest cvs (19/08/04). When I try to compile I receive the
> following error:-
>
>
>
> bison ast_expr.y --name-prefix=ast_yy -o ast_expr.c
> ast_expr.y:110: unrecognized: %locations
> ast_expr.y:110:Skipping to next %
I have loaded the latest cvs (19/08/04). When I try to compile I receive the
following error:-
bison ast_expr.y --name-prefix=ast_yy -o ast_expr.c
ast_expr.y:110: unrecognized: %locations
ast_expr.y:110:Skipping to next %
ast_expr.y:141: invalid @-construct
ast_expr.y:141: $. is invalid
as
Andrew Kohlsmith wrote:
Terminal -> Adit600 FXS -> * -> 1 hop -> * -> PRI
Must use ULAW and must have a decent connection (low low jitter and latency).
We have incoming and outgoing faxes doing this without issue. I can't get
faxes to go through Nufone with any kind of steady success but it's
Hello all,
I currently have an Eicon Diva Client isdn card using i4l. Outbound
dtmf doesn't work (and never has), but there has been an annoying
problem with false dtmf detection in calls (that could be triggered
easily by blowing into the receiver on the remote end).
I looked through the list an
On Aug 19, 2004, at 11:30 AM, Muiz Motani wrote:
Does anybody know what happened to the opencall.org website? I can't
get
into the home page or the ftp site.
His DNS servers seem to be down. opencall.org is served by
name[12].coppice.org, which are 202.76.92.17[23]. Neither one responds
to pin
Does anybody have any solution to get MGCP UserAgent support in *.
With the explosion of VOIP provider these days, this would allow tapping
into these providers that chosen MGCP as their protocol.
I have the CallAgent side working fine.
Thanks
MarcG.
_
Does anybody know what happened to the opencall.org website? I can't get
into the home page or the ftp site.
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Massimo De Nadal schrieb:
forget the asterisk source you have downloaded.
Zaphfc is not only a driver, it's a patch that have to be applied to
specific source version too,
You have to run the install.sh script that is included in the tarball.
This script before downloads the right asterisk version
On Thu, 2004-08-19 at 12:26, Morgan Gilroy wrote:
> Hi can someone help me, I want to do 'Dial(IAX2/bla/1234567|50|tT)'
> From an AGI script so people can dial #* to hang up (and other things) but
> when using AGI using 'EXEC DIAL IAX2/bla/1234567|50|tT' it dials ok but
> nothing happens when they
On Aug 19, 2004, at 10:00 AM, Ryan Courtnage wrote:
Andrew Kohlsmith wrote:
On Wednesday 18 August 2004 19:31, Ryan Courtnage wrote:
Theoretically, I know it's possible, but is any using multiple
tdm400ps
(fxo) in single * box? In a production environment? Any gotchas
aside
form irq sharing?
Bu
Wow.
> [00059002042b]
> context=main
> host=dynamic
> callerid = "John Doe" <123>
> nat=yes
> Line => svip10
That did it. The phone registered with * and a debug msg flys up when I
pickup/put down the reciever.
When I pick up the handset, I can hear a dialtone. But pressing numbers on
the keypad
Hi,
I finally switched (again) to chan_oh323, which
compiles without problems on opteron 64bit.
Roger.
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I have the programable button led's working properly on my snom 200
except they don't flash during a ring event. I found a post by Andre
Bierwirth saying he had a patch that he submitted but didn't make it
into CVS. I would like to get a copy of that as a starting point to
implement button flash
On Thursday 19 August 2004 13:00, Ryan Courtnage wrote:
> A Rhino channel bank with 12 fxo will retail to about $1850, plus $500
> for the t100p. It's a tough sale for a growing small company that has
> already invested in (and outgrown) 2 tdm400ps.
So give them partial (say 50%) credit to take b
James Freire wrote:
Sorry about that. I am in the US and using the Digium FXO TDM400 and I have enabled all the callerID options in my zapata.conf file.
Of course, the BT100 has only a numerical display, and will not display
CIDName, only number.
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As
Hi can someone help me, I want to do 'Dial(IAX2/bla/1234567|50|tT)'
>From an AGI script so people can dial #* to hang up (and other things) but
when using AGI using 'EXEC DIAL IAX2/bla/1234567|50|tT' it dials ok but
nothing happens when they dial #, is there something special I need to do to
escape
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