[Kevin Walsh Wrote]
Marvellous. Microsoft will bring their legendary
stability, security
and reliability to the VoIP world.
Oops - there goes my lunch.
Maybe but looking past that what the unit will bring
is a programmable touch screen GUI on a hard VOIP
phone.
And being a Microsoft product
65.39.205.111 is not local, substituting externip Check for res for
is not a local user
build_route: Contact hop: sip:65.39.205.111:5060
-- Executing Dial(SIP/fwd.pulver.com-0811c948,
IAX2/janie|20|tr)
in new stack
SIMPLE DIAL (NO URL)
-- Called janie
-- Call accepted by
Hi,
at SuSE 9.0 helped:
I am not able to compile zaptel...
Could you give me a hint?
Have you tried the following, which is suggested in the output?
'make cloneconfig make dep' in /usr/src/linux/
Felix
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
Hi,
From recent experience:
If you want to use digium hardware dont use suse 9.0. It
seems to think the E1 card is a tigerjet bri card and the
kernel hangs on ztcfg.
I have a WT405P running under SuSE 9.0 and it works great.
But I had only choosen SuSE because I also need capi...
Bye
Mike,
I've been trying to install under SuSE 9.1, but cannot compile zaptel
What's the secret incantation ??
TIA
I was helped with:
I am not able to compile zaptel...
Could you give me a hint?
Have you tried the following, which is suggested in the output?
'make cloneconfig
Tried that. Tried rebuilding kernel and rebooting. Same errors encountered.
Ah well. I've reloaded the machine with FC1.
Thanks for all the help and support anyway - it's been a great lesson. I
built my first kernel :)
Julian
- Original Message -
From: ePyron Felix Deierlein [EMAIL
I do get echo, lots of it, I am waiting until the new patch they are all
on about on the list gets into a stable release, then I will upgrade and
see if that does the trick.
I am told that some of the echo may be to do with a mismatch in the
impedance with the BT line.
I had an adsl problem a
Jeremy
It seems you misunderstood my question. I was talking about SIP not IAX. It wasn't
about access control - it was about having a problem with phones on a poor connection
that is prone to occasional packet loss or disconnection.
How much clearer do you need to be? Asterisk is telling you
I do get echo, lots of it, I am waiting until the new patch they are all
on about on the
list gets into a stable release, then I will upgrade and see if that does
the trick.
The patch didn't seem to work for me.
I am told that some of the echo may be to do with a mismatch in the
impedance
Hello all,
I'm not familiar with Asterisk at all, so any help would be appreciated.
I have an ISDN card
lspci:
07:06.0 Network controller: Eicon Technology Corporation DIVA Server PRI-30M
2.0
which has 8 channels active.
I am wondering if
a:, this card is supported/can be made to work with
Hi,
I'm using 12 fxo modules on tdm cards.
When I do 'modprobe wcfxs', the cpu usage in kernel mode
varies from 2% to 100%.
While monitoring using top, there is no process using much cpu
resource.
Is this ok?
Thanks in advance.
As a follow up to my previous post, I have now identified what is
causing the bug with the grandstream phones.
When the line
faxdetection=incoming is in the zapata.conf file, the grandstream
phones will not ring, nor connect a call to the zaptel interface.
Can anyone else confirm this bug? I'm
I can't see that any echo cancelling is going to work with a 10db difference
between rx and txgain. If the difference is due to impedance mismatch
reflections then the reflected tx signal is going to be of greater amplitude
than the callers signal.
Chris
- Original Message -
From: Chris
Hi Tobi,
I installed Asterisk with CAPI support. Everything works fine
while starting Asterisk, but when a call comes in Asterisk
hangsup the call after two times of ringing.
The output is like:
Jun 24 22:19:49 NOTICE[1082178480]: chan_capi.c:1931 capi_handle_msg:
CONNECT_IND ID=002
ePyron Felix Deierlein wrote:
Do you have DIDs (PTP-ISDN)?
Bye
Felix
yes
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I've managed to install and run Asterisk on a Fedora Core 1 server using the
fritz avm
ISDN card, and thought I'd share how it was done.
This worked for me:
Server:
Dell 6450
Quad Xeon 700 2Mb Cache
4GB Ram
2x18GB SCSI (Mirrored)
ISDN card (fritz!pci). This is packaged as a BT ISDN card.
*
hi,
I use the oh323 driver to answer H323 calls.
The connection is set up normally.
In my extensions.conf file I use:
exten = s,1,Answer
exten = s,2,Playback(demo-instruct)
exten = s,3,Hangup
So that when a call is answered i get:
*CLI -- Executing Answer(H323/ip$10.0.3.23:32782/6502, )
Hi there,
I was wondering how I can use setgroup and checkgroup for perfom incoming
and outgoing limitation checks.
I've have some users that doesn't what to be able to recieve more than 1
call at a time, and I also want to limit a users outgoing call abilities.
Any help would be greatly
Have you upgraded the firmware on the Grandstream phones?
I had the same problem on newly purchased Grandstreams until I
upgraded the firmware (currently using 1.0.5.0).
Another thing I had to do to get ringing indications, etc. working on
my Grandstream was:
- Create a separate context for the
Hi Felix,
then I guess that I have the same problem.
If I get a overlaped dial from PSTN, i get only the first did-digit as
extension
, p.e: my number 8993-12 then it goes to 89931 and that extension does not
exist
If I get a call from ISDN (or maybe mobile) with block transfer, I get
899312 and
Hmmm. Now I have another problem...
After the call goes to the extension 100 in this example, I get a jump to
the t extension for the context. I can't find a way to make the call not
time out, and asterisk acts like it needs to do something after the record
start (i.e. the 100,2 in your
Actually ACPI is enabled no matter what on the machine. I wasn't talking
about ACPI in the CMOS, but ACPI support in the Linux Kernel.
I'm using 2.4.26 (kernel.org+latest libata patches) on a Debian Sarge
box.
Kind regards,
Martin List-Petersen
On Thu, 2004-06-24 at 14:25, Isamar Maia wrote:
Hi:
I have an Asterisk server (version 0.9.0) working with a Digium
E100P card that makes an EuroISDN connection with a Siemens Hicom 300 PBX.
Hicom 300 is pri_net and Asterisk is pri_cpe.
Furthermore, makes an OH323 connection with a GNU gatekeeper
When you make a call between a Hicom
At 13:00 25/06/2004 +0200, you wrote:
Hi there,
I was wondering how I can use setgroup and checkgroup for perfom
incoming
and outgoing limitation checks.
I've have some users that doesn't what to be able to recieve more than
1
call at a time, and I also want to limit a users outgoing call
Hi there,
the latest firmware (at least 1.5.0.0) for the Grandstream HT286
HandyTone now has an option to set the FXS impedance. It appears that for
EU countries the CTR21 is the correct setting.
My question: Does changing this setting improve fax operation for you?
Cheers, Philipp
Source:
On Wed, 2004-06-23 at 17:12, Sathya wrote:
Hi,
Greatly appreciate if some one help me with the application read().
I have added a feature to reload asterisk from a phone...
it uses 'read' to get a 3 digit password
I was using '#' to end the sequence until I realised I could specify the
Hi!
Using a sip phone, x-lite, after I dialed 6601 I get the following:
Have you set Transmit silence to YES in X-Lite?
-- Reloading module 'cdr_csv.so' (Comma Separated Values CDR Backend)
-- Executing WaitMusicOnHold(SIP/666-f408, 30) in new stack
Jun 24 22:35:07 WARNING[458766]:
Hi!
Um - If my secretary transfer's a call from her BT101 to her
own number
- she looses the call. What can I do to stop this from
happening - apart from dyeing her hair from blond to brunette ???
- method a) SetGroup() and GetGroupcount() in extensions.conf
- method b) incominglimit=
Hi!
FWD only supports ULAW comment out the line allow=GSM in the general
section of the iax.conf
Nonsense - FWD *does* permit the use of GSM.
Cheers, Philipp
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At 14:48 25/06/2004 +0200, you wrote:
Hi!
FWD only supports ULAW comment out the line allow=GSM in the general
section of the iax.conf
Nonsense - FWD *does* permit the use of GSM.
Cheers, Philipp
Not in iax only with sip
Jason
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ISDN
Carlos Arnt
[EMAIL PROTECTED]
Diretor de Informática.
Divisão de Tecnologia e Desenvolvimento TI.
Intellissence do Brasil.
http://www.intellissence.com/brasil
Tel:(+55)-(21)-(3908-4667)
Tel(Direto):(+55)-(21)-(3905-1561)
Cel:(+55)-(21)-(9169-8537)
--VoIP Contact Method--
World VoIP Pin/Code:
On Fri, Jun 25, 2004, kiel hedjam wrote:
hi,
I use the oh323 driver to answer H323 calls.
The connection is set up normally.
In my extensions.conf file I use:
exten = s,1,Answer
exten = s,2,Playback(demo-instruct)
exten = s,3,Hangup
So that when a call is answered i get:
Andrew Thompson wrote:
Eric Wieling wrote:
How is this different from the way standard call waiting works
when provided from your telco?
Um, he actually has two phone lines, not just one that he's
flash-ing back and forth between.
If he hangs up the line, does the second call not
What version of asterisk-oh323 do you use?
Michael.
kiel hedjam wrote:
hi,
I use the oh323 driver to answer H323 calls.
The connection is set up normally.
In my extensions.conf file I use:
exten = s,1,Answer
exten = s,2,Playback(demo-instruct)
exten = s,3,Hangup
So that when a call is answered i
Hi,
Philipp von Klitzing wrote:
Hi!
Channel 'CAPI[contr1/**some_number**]/0' sent into invalid
extension 's' in context 'default', but no invalid handler
Look at Asterisk's standard extensions like s, i, o and so forth.
Insert this in your context [default] in extensions.conf:
exten
Rich Adamson wrote:
From: Nik Martin [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] FXO impedance matching
Date: Wed, 23 Jun 2004 11:02:00 -0500
To: [EMAIL PROTECTED]
Michael Welter wrote:
Jason A. Pattie wrote:
Robert Hajime Lanning wrote:
Echo
Hi,
I have finally tracked down this problem a bit more. The problem is not
actually related to the Grandstream phones - just they were the only
obvious exhibitors of the problem.
The problem seems to be that when faxdetect is set in zapata.conf,
asterisk does not inform the sip phones that a
Dear Philipp,
I see that you are from Germany. I would like to ask you about the
configuration for Caller ID and Zone Data
(zaptel.conf/loadzone/defaultzone).
I am asking you that because our standards in Bulgaria are similar
to German because our equipment is Siemens.
Best
I get a problem with what appears to be a slow oscillation on the line if
the rxgain + txgain adds up to more than -1db. If I use rxgain=-1.0 and
txgain=0.0, it doesn't oscillate but the levels are far too low. The card is
an X100P.
The oscillation (even on the standard built-in Asterisk echo
Hi,
I have started some users terminating calls from my
asterisk server to the PSTN through a couple of
termination providers.
The biggest problem I am having is the time it takes
to initially set the call up. It regularly exceeds
twenty seconds. I can work around this with failing
over to
hello there:
did some one compiled shadydial with asterisk on gentoo successfully, if some one plz
help me
I am getting compilation errors during asterisk compilation after replacing the files
provided with shadydial
thank you
here is my log, please help
gcc -pipe -I=/usr/local/pgsql/include
I'm running Asterisk CVS-HEAD-06/07/04.
When I try to originate a call from a SIP channel to a ZAP channel using
manager everything works up to the point
when I pickup the ringing ZAP phone. Originate ZAP to SIP works fine.
This is the error from my asterisk debug.
Jun 25 09:41:26
Does anyone know of a device that will take an SS7 link and convert it
to a PRI?
--
respectfully, Joseph - (606) 477-2355 x140
--=
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thanx that works
here is wat i want to do
sip (xlite)-*(iax2)(codec speex)-internet*(iax2)--pstn
when i do that all i get is noise in the other end..am sure am doing
something stupid...
any help would be appreciated ( i want to meassure the bandwidth will be
using the program called
OMG - I couldn't be happier to see this!
On Friday 25 June 2004 07:39, Andrew Yager wrote:
The problem seems to be that when faxdetect is set in zapata.conf,
asterisk does not inform the sip phones that a voice call has started
actually been answered.
I've been plaqued by a different problem
I do get echo, lots of it, I am waiting until the new patch they
are all on about on the list gets into a stable release, then I
will upgrade and see if that does the trick.
Not likely the patch will get applied to the Stable release since its
been stated several times that's all but
FWD only supports ULAW comment out the line allow=GSM in the general
section of the iax.conf
Nonsense - FWD *does* permit the use of GSM.
At http://www.freeworlddialup.com/advanced/iax it currently says:
Q: what codecs does FWD support?
A: All codecs will pass through FWD, but FWD
A switch ?
;-)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Joseph
Sent: Friday, June 25, 2004 4:25 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] SS7 to Pri
Does anyone know of a device that will take an SS7 link and convert it
to a PRI?
--
Kevin Walsh wrote:
Freddy Setiawan [EMAIL PROTECTED] wrote:
Hi there, linux got so many distro, but which one that have more
compability with the Asterisk?
My Asterisk server is running on Gentoo, with the 2.6.7-gentoo-r5
kernel. The Zaptel drivers work nicely too. I should think that
Try looking into FastComm. They do C7 to E-ISDN.
Kurt
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Hadar Pedhazur wrote:
Andrew Thompson wrote:
Eric Wieling wrote:
How is this different from the way standard call waiting works when
provided from your telco?
Um, he actually has two phone lines, not just one that he's
flash-ing back and forth between.
If he hangs up the line, does the
I am routing outgoing calls through a sip gateway. The calls go through
no problem, however the ringing in the callers ear begins as soon as the
last digit is dialed. This has two nasty side effects. First, the caller
hears 1-2 more rings than the callee. Second, and more importantly, if
the
hello all, I am
having a trouble with Audio using h.323 channel...
I am doing
this
Call comes into
cisco 5300 and is sent to Asterisk, asterisk catch call with h.323 driver and
send call to a SoftSwitch that routes the call, I can see log debug telling me,
CALLED XXX, and then RINGING,
BT do occasionally tweak up line gain a bit if you keep complaining that you
have a modem and are getting a very slow speed. I have had a 40k V90 come up
to 48k after this was done on my line at home (System X switch).
You have to get a sympathetic engineer though - frequently they will tell
you
We spent some time yesterday trying to understand how IAX2
authentication works, and now I'm confused...
Let's say that the receiving end has this entry in their iax.conf file:
[remote-site]
type=user
secret=foo
auth=md5
context=incoming
host=dynamic
The way I see it, there are two ways to
Caleb Kow wrote:
Here we go:
[EMAIL PROTECTED] root]# netstat -ap
Active Internet connections (servers and established)
Proto Recv-Q Send-Q Local Address Foreign Address
State PID/Program name
tcp0 0 *:32768 *:*
LISTEN
On Wednesday 23 June 2004 14:28, Joseph Finley wrote:
I have a X100P that works great for a couple days maybe even a week and
then outside callers say my phone just rings and rings. When I try to dial
out during this period, it waits dead air
You aren't alone (i've spoken to several people
I'd be willing to bet you have r in your dialout string (i.e.
something like: Dial(${TRUNK}/${EXTEN},120,r)...
Get rid of that in the outbound dialing, and you otta be ok.
Jeremy Jones
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Bruce Komito
On Fri, 2004-06-25 at 09:24, Joseph wrote:
Does anyone know of a device that will take an SS7 link and convert it
to a PRI?
I think it's called an ILEC or CLEC. 8-)
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
In a related story, the IRS has recently ruled that the cost of
On Fri, 2004-06-25 at 09:52, Bruce Komito wrote:
I am routing outgoing calls through a sip gateway. The calls go through
no problem, however the ringing in the callers ear begins as soon as the
last digit is dialed. This has two nasty side effects. First, the caller
hears 1-2 more rings
Good morning all,
I'm setting up Asterisk for the first time with no prior PBX experience.
I'm following Andy Powell's 'Getting Started with Asterisk'
(http://www.automated.it/guidetoasterisk.htm). This is my second time
through that document - as I did something weird the first time and really
I have an Asterisk PBX on the private lan, which is protected
from the public Internet with a Linux iptables machine. The
firwall is it's own seperate box running NAT with SPI.
I want to drop a SIP phone at my brothers house, and have it be an
extension off my Asterisk box. I've been looking
Thanks, that looks interesting.
On Fri, 2004-06-25 at 10:47, Kurt wrote:
Try looking into FastComm. They do C7 to E-ISDN.
Kurt
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On 2004.06.25 06:39 Andrew Yager wrote:
The problem seems to be that when faxdetect is set in zapata.conf,
asterisk does not inform the sip phones that a voice call has started
actually been answered.
Setting faxdetect=no causes everything to behave absolutely properly,
with no problems
Just checking that you have installed the proper versions of both OpenH323
and PWLib, as mentioned in ~/asterisk/channels/h232/README, and have rebuilt
asterisk after those installations as specified?
If so, then you are having the same problem I'm experiencing: no audio on
H.323. I'm also
Joseph wrote:
Does anyone know of a device that will take an SS7 link and convert it
to a PRI?
It could be * - depending which version of * you have. :-)
Regards,
Steve
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Didn't I hear a week or two ago (on this list) that someone had taken it
upon themselves to write an asterisk module for the openss7-modified
digium t1/e1 cards? Maybe soon asterisk'll do it.
Jeremy Jones
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
thanx that works
here is wat i want to do
sip (xlite)-*(iax2)(codec speex)-internet*(iax2)--pstn
when i do that all i get is noise in the other end..am sure am doing
something stupid...
any help would be appreciated ( i want to meassure the bandwidth will be
using the program called
Ed Brady [EMAIL PROTECTED] wrote:
I am about to build my first asterisk box, I want to make it Gentoo based
with a 2.4 kernel.
I'm on 2.6.7-gentoo-r6, which I installed today (upgraded from r5).
I have found the 2.6 kernel to be a lot better, in my unscientific
opinion, than the 2.4 kernel
Kevin P. Fleming wrote:
Is there a reason why Asterisk allows incoming IAX2 calls without
USERNAME specified at all?
1) host=dynamic makes no sense in a type=user
2) One sends the username to be used to the peer
On the machine you wish to dial out, you have in your iax.conf:
for question 4.
You need to register with fwd first, then use
registry = command in sip.conf.
aaron
On Fri, 25 Jun 2004 08:32:07 -0700, Fletcher Bonds
[EMAIL PROTECTED] wrote:
Good morning all,
I'm setting up Asterisk for the first time with no prior PBX experience.
I'm following Andy
On Fri, 25 Jun 2004, Fletcher Bonds wrote:
1. A general will this work? (vmware linux, same pc as phone, NAT'd
addresses,etc)
You'll probably be the first person to try it. I'd guess that it will
work, but expect call quality to be impacted because of all the extra
scheduling and
Thanks for the info Rich looks like I'll have to wait for the new FXO
module. The impedence in the UK is zcomplex(2) which looks a long way away
from a straight 600 ohms.
Here is the list of zcomplex impedences
Zcomplex(1) = 150 nF // 750 ohms + 270 ohms ( European harmonized,
France
James H. Thompson wrote:
Are there any online retailers that carry the Uniden UIP series phones? I
did a quick Froogle search to no avail.
See:
http://www.voip-info.org/wiki-Uniden
So you *must* sign up as a reseller to purchase one? What are your
opinions/problems
That would be great if * could do it.
We *have* a switch, but it will not give us
a pri by manufacture design :( .
It will give us ss7 though.
And as far as I can tell, the only way to get callerid etc is
by a PRI to *.
I can do fx trunks from the switch, but the switch will not include
Brian Weaver wrote:
I have an Asterisk PBX on the private lan, which is protected
from the public Internet with a Linux iptables machine. The
firwall is it's own seperate box running NAT with SPI.
The only way to make this work well is to run a SIP proxy of some kind
on the firewall system and
Jeremy McNamara wrote:
On the machine you wish to dial out, you have in your iax.conf:
[peer]
type=peer
host=1.2.3.4
secret=foo
and in that same machine's extensions.conf you have something that looks
like:
Dial,IAX2/[EMAIL PROTECTED]/${EXTEN}
Then on the 'peer' (other) machine
On Friday 25 June 2004 10:56, Chris Hirsch wrote:
James H. Thompson wrote:
Are there any online retailers that carry the Uniden UIP series phones? I
did a quick Froogle search to no avail.
See:
http://www.voip-info.org/wiki-Uniden
So you *must* sign up as a reseller to purchase one?
John,
Mine gives the call waiting beep also, but only when a call comes in on
the same extension that is in use. If a call comes in on another
extension on my phone, then I get no beep, just a light on the button
flashing and the sreen letting me know that there is another incoming call.
-Tor
Hello Neil/Everybody,
Yes you are correct, PostgreSQL has to be specifically configured
within the server it is hosted on to allow host calls from Asterisk so
that the socket connects.
Here is how I solved the problem through the help of everybody here:
Firstly enable the -i command in the
I've been looking for a good iax or sip ==> ptsn provider. Someone with very low cost usa calling and can offer incoming ptsn connections in most markets. The only decent providers I could find were iconnecthere and nufone. Has anyone found someone that really stood out?
Matt Hohman
New Heights
FYI
I am experiencing the same problem.
I have complied asterisk from the latest CVS
The call connects with no audio or DTMF to either end.
I tested with ulaw and g729 with no success.
-Michael
On Fri, 2004-06-25 at 10:55, Scott Stingel wrote:
Just checking that you have installed the
-
I'm here with you on this one. I've not been able to figure this out -
I triple quadruple checked that I have the right versions of pwlib
openh323, I've followed all recommendations in the README, yet I still
do not have audio in both directions.
I'm also using a cisco 5300, there is
I'm having troubles... I am new to Asterisk and SIP. I was just given
this setup and it was running fine. And somehow it stopped. I thought it
was the DID(again) But it wasn't.
All calls are getting rejected.
Called
FWIW, I just ordered a prototype for a single-line sip speakerphone from
China. If I get this to work on asterisk, I may import it -- I should
be able to sell it for about $85 to endusers and $70-$75 in quantities.
The way I look at it, $75/port is right on the dot for cheapest
possible endpoint.
And as far as I can tell, the only way to get callerid etc is
by a PRI to *.
I have EM trunks into * out of our switch. We tell the switch to pass
digits in Feature Group D DTMF format, and we are able to get ANI and
DNIS. Of course this does not allow to get the calling party's name
though.
John,
Oops! I was wrong, I do get a call waiting beep even if the call comes
in on another extension. I still would prefer the phone to ring when
another call comes in. If anybody knows how to do this, that would be great.
-Tor
Tor Roberts wrote:
John,
Mine gives the call waiting beep also, but
Joseph schrieb:
Does anyone know of a device that will take an SS7 link and convert it
to a PRI?
...
Hi,
your question is not very asterisk related. So you can
use any signal converter on the telephone device market.
I you are looking for something below 10 kEUR, there is not
much choice, e.g.
-
Is the stable branch
usable? Is there ever going to be a 1.0 release?
Should I be using
the "stable" branch or the development branch? The development branch seems to
have more fixes than the stable branch. It looks like fixes going into the
release branch aren't going into the stable
What switch do you have?
On Fri, 2004-06-25 at 11:56, Joseph wrote:
That would be great if * could do it.
We *have* a switch, but it will not give us
a pri by manufacture design :( .
It will give us ss7 though.
And as far as I can tell, the only way to get callerid etc is
by a PRI to
Hi,
there are still some questions to be answered by OpenSS7.com
in order to decide, whether E400P-SS7 is a good choice for
the asterisk SS7 support.
In the meanwhile I'm also in negotiations with another
manufacturer (whose name I currently may not tell due to
a NDA) of SS7 hardware, who gets
I have the same problem here. I have to servers working with identical
(same) configurations, the old one is working just perfect and the new one
I got, is not working (no voice in both directions). Im trying to fix this
problem with digium, we are exchanging emails so if I get a solution Im
gona
Ok, may have got to the bottom of this.
The te405P was sharing interrupts with a via82cxxx audio chip, that was
being used to generate the music on hold for our existing pbx.
Having now shut down the music on hold the system will now run correctly
with telewest as the master and the gdk as the
Andrew Thompson wrote:
Can the original poster open a bug at bugs.digium.com (if they haven't
already)?
Original poster emits heartfelt, Bwaahh
I have posted about this problem at least three times previously, and
been scoffed at each time.
I'll be glad to do a bug report,
Does it work if you try ULAW or some other codec?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sola 2000
Sent: Friday, June 25, 2004 12:08 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Speex
thanx that works
here is wat i want to do
sip
Hi Joseph,
J Does anyone know of a device that will take an SS7 link
J and convert it to a PRI?
Telesoft Technologies make the Okeford range of protocol converters and baby
switches that I have used for this purpose. Have a look at:
http://tinyurl.com/3drjp
If you are converting a number of
I've been using BroadVoice (sip based) for about 2 weeks now with no
problems at all. 19.95/month Unlimited USA plan...
-Begin Sarcasm---
And amazingly... you can use your own device with their service.. what a
novel idea! Other sip providers should get in on this!!
-End
On Fri, 2004-06-25 at 13:57, Mike Machado wrote:
And as far as I can tell, the only way to get callerid etc is
by a PRI to *.
I have EM trunks into * out of our switch. We tell the switch to pass
digits in Feature Group D DTMF format, and we are able to get ANI and
DNIS. Of course this
We have seen this on one of our polycom phones and it had something to do
with the power adapter being fried. This was the only phone on an
unprotected (non-UPS) outlet and we basically had to throw the adapter away
and use a new one then the interferance went away and all was good.
MATT---
- Original Message -
From:
Chris Hirsch
To: [EMAIL PROTECTED]
Sent: Friday, June 25, 2004 6:56 AM
Subject: Re: [Asterisk-Users] Cheap
(US$120 or less) SIP Phones
James H. Thompson wrote:
Are there any online retailers that carry the Uniden UIP series
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