On Fri, 2004-08-27 at 16:32 +, Tony Mountifield wrote:
I have today reported a bug with the latest channel.c (1.134) that
affects music-on-hold for the first user in a MeetMe room when calling
from a Grandstream BT102. The music is broken up about 5-10 times a
second. It doesn't happen
Does this also effect 1.0-RC2? I am having a similar issue at a
customer site over a frame relay network that is having occasional
choppy sound over a fairly open line, with the jitter buffer enabled,
as well as trunk=yes enabled.
Thanks!
Brian
On Fri, 27 Aug 2004 12:47:05 -0700, Kris Boutilier
http://www.dse.com.au/cgi-bin/dse.storefront/
412ff6210573f994273fc0a87f9
c0726/Product/View/M9917
I emailed Dick Smith with the requirements but none of the power
supplies they have can do the 1500mA.
--
jeremy bogan[ [EMAIL PROTECTED] ]
segment publishing - design.develop.host
Hey all,
I've been trying to get my X101P working again as of late (it used to
work great) and before I decide to trash the card I thought I'd post up
my symptoms to see if anyone has any ideas.
My old working config was basically 1 channel running fxsks signalling.
It was working great with no
the the real power requires and read the pages...
like when can you belive any sales gimp who works for dse ??
its in their catalogue, you asked, goto a store and find it !!
also check jaycar and altronics they are good bets as well.
Gary
On Sat, 28 Aug 2004 17:08:18 +1000, Jeremy Bogan
Looking for folks experiences with ISDN BRI cards in the UK ... what's good
and what's bad and any gotchas.
Thx
--
David Gurr
Congruity Ltd.
Hemel Hempstead
UK
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BTW,
-Original Message-
I emailed Dick Smith with the requirements but none of the
power supplies they have can do the 1500mA.
I've got an IAXy running on a 1100mA powersup I found packaged with my
Lucent WLAN basestation. Be a little flexible and just try it.
Only thing: If the
Jeremy
I am in the UK but the IAXy's (x5) I just received from the states have
120VAC/9VDC 800mA supplies, I questioned this, but the supplier stated
that this was what Digium are shipping with the IAXy.
Digium state 1500mA on their docs but DigitNetworks state 1200mA on
theirs, but ship the
Hi Roy,
I am in the UK but the IAXy's (x5) I just received from the states have
120VAC/9VDC 800mA supplies, I questioned this, but the supplier stated
that this was what Digium are shipping with the IAXy.
Thanks for the info. I bought a power supply that does 1500mA at 9V DC,
but the IAXy doesn't
Jeremy
Apart from voltage and current the main thing to watch out for is that
the PSU MUST be fully regulated or switched, an unregulated PSU will
give you all sorts of weird problems.
Before I opted for 240V to 120V mains converters (they were on offer) to
use with the supplied US PSU's, I too
Hi, all!!!
What will Asterisk do in the following case:
For example, we have 4 licenses, and have 4
simultaneous calls, using G729.
Will asterisk allow incoming calls from peer,
that can talk G729 and ulaw, and will it
force it somehow to use ulaw in this case?
All phones there in LAN behind
Muiz Motani wrote:
I am using IAX soft clients [snip]
on a NATed private LAN [snip]
[snip] disconnected after 9 seconds.
8-10 seconds is roughly the time it takes for an IAX
transfer to kick in. I have seen similar cases not with
IAXtel but other IAX connections where end
I have now today also configured a 7960 to work with asterisk via chan_sccp.
I have only used SCCP firmware 5.0 (5) with the 7960 and I gotta say that I
much prefer the SIP 7.2 firmware. The real reason at this stage for going
SCCP is for support of the 7914 expansion module. This phone will be
Hi Steven,
IT is interesting you are even that far along with your AGI application
when you haven't even figured out your mail client.
Do you mean my email client? or my voicemail? Voicemail is working fine on
digim extensions, i even changed the language to spanish (btw, there are lot
of
Apart from voltage and current the main thing to watch out for is that
the PSU MUST be fully regulated or switched, an unregulated PSU will
give you all sorts of weird problems.
Maybe I have a busted IAXy, the power supply i've got is regulated and
supports multiple voltages in different mA
Hi,
I cannot seem to accept incoming calls from FWD using IAX2. I followed the
directions posted at www.fwd.pulver.com/advanced/iax I can make outgoing
calls fine using IAX via FWD. When someone calls me from FWD I get the
following message:
Chan_iax2.c:5251 socket_read: Reject connect
Hi,
I cannot seem to accept incoming calls from FWD using IAX2. I followed the
directions posted at www.fwd.pulver.com/advanced/iax I can make outgoing
calls fine using IAX via FWD. When someone calls me from FWD I get the
following message:
Chan_iax2.c:5251 socket_read: Reject connect
On Mon, 23 Aug 2004, Robert Boardman wrote:
Heartened by your that you have got x-lite working, I have been trying,
but failing to now get x-lite working, don suppose you could send me a
quick screen shot of you x-lite settings?
Not really, but it's not hard to get going.
Presuming you
Tim Robinson wrote:
Andrew, I am looking for exactly the same thing, except using the
normal Dial command via Zaptel with the c option to defer answering
til you press the # key. I am not a programmer but have briefly
looked at the code in app_dial.c and I think it is doable - just need
to
Tim Robinson wrote:
Andrew, I am looking for exactly the same thing, except using the
normal Dial command via Zaptel with the c option to defer answering
til you press the # key. I am not a programmer but have briefly
looked at the code in app_dial.c and I think it is doable - just need
to
On the extensions.conf explanation page is a mention of the switch statement
and it refers one to the connecting two * servers page. The only mention of
the switch statement there is brief and in the example.
However, the example seems to have some errors in it. It shows a sample of
what's in
Jeremy Bogan wrote:
Maybe I have a busted IAXy, the power supply i've got is regulated and
supports multiple voltages in different mA ratings, with the 9V DC at
1500mA.
Have you tried feeding it less amps at all?
--
Best regards,
Duane
http://www.cacert.org - Free Security Certificates
Try an iax2 debug at the CLI and watch the logs as you get
disconnected. See if it's something at the IAX protocol level. If you
don't get any insight from that, use Ethereal on the segment to see
what's going on.
Brad
I am using IAX soft clients (firefly, IAXComm, IAXPhone) from a Win2K
Had this problem earlier this week - ensure 'trunk=no' in iax.conf if you're
using fairly current CVS code. There is something not right w/the trunking
that causes choppy sound. See the wiki for more info.
I am using current CVS code and I have trunk=no. Still sounds crappy. I need
to
Impedance setting in the UK!? OK, I've clearly missed something along the way. A
search
on the Wiki says something about Zcomplex impedance but I have absolutely no idea
where or
what this is. If someone could point me in the right direction I'd be extremely
grateful.
That essentially
Sergey Lapin [EMAIL PROTECTED] wrote:
What will Asterisk do in the following case:
For example, we have 4 licenses, and have 4
simultaneous calls, using G729.
Will asterisk allow incoming calls from peer,
that can talk G729 and ulaw, and will it
force it somehow to use ulaw in this case?
On Aug 28, 2004, at 7:39 AM, Rich Adamson wrote:
I do a lot of work with companies throughout the US on network
performance
and we _frequently_ run into routers, switches, servers, etc, that are
allowed to auto-negotiate their half vs full duplex nic interfaces.
About
50% of the time, systems
All,
I am experiencing this problem with an IAX link to the UK provider
TelAppliant.
chan_iax2.c:5251 socket_read: Rejected connect attempt from 217.14.132.162
Not sure what is causing this, however, it seems to have started sine I
downloaded Asterisk CVS-HEAD-08/19/04-19:55:53. Not sure if
What do you have in extensions.conf for inbound from FWD?
In my default section, I have:
exten = ${FWDNUMBER},1,Goto(housemenu,s,1)
Lyle
- Original Message -
From: Storm D. J. Petersen [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, January 09, 2100 10:09 AM
Subject:
Hi,
I also have the same problem,
from memory is this not progress indicator=8 thatdeals withthe dialing indicator
?
Anyway I am also stuck with not
having any dial indication, has anybody got an idea.
MO
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Just by way of giving you some encouragement, I too run a version downloaded
on about the same day and use Telappliant successfully.
I did get a rejected error until I converted one of my accounts to IAX
from the default SIP. I don't think there is anything special about my
setup and the * PBX
to start with i am new to asterisks and i am also a telcom idiot.
with that said i have one vonage line i would like to hook up in my soon to be built
Asterisk ippbx server.
Now with the one Vonage (with call waiting) line can i receive more one call using an
auto attendant route the call the
This is something I'm after as well, what I have found is the following:
http://bugs.digium.com/bug_view_page.php?bug_id=0001082
http://lists.digium.com/pipermail/asterisk-dev/2004-February/003201.html
which pretty much does what I(you) want, the one problem with it is that
while the agent is
Hi
I know this gets covered fairly regularly, but I've had a search through the
archives and can't find anything dealing with this specifically - apologies
if I've missed it though.
I've got a TDM11B with the fxo port plugged into a standard UK BT PSTN line,
loading wcfxs with OPERMODE=UK.
Hi List,
does somebody know a SIP Provider which offers
reseller possibilities?
Moritz
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Storm D. J. Petersen wrote:
I cannot seem to accept incoming calls
from FWD using IAX2. I followed the
directions posted at
www.fwd.pulver.com/advanced/iax
I can make outgoing calls fine using
IAX via FWD. When someone calls me
from FWD I get the following message:
On Fri, 27 Aug 2004, Andrew Brown wrote:
When using the callback feature on agents I notice that when the queue calls
one of the agents and the agent picks up the call they hear nothing until
pressing the # to accept the call.
Only then does my announcement play back to the agent after
Since Thursday evening my asterisk box has been failing to register with
broadvoice. I haven't changed any of my config files in the last week.
Can anyone suggest anything?
Asterisk is reporting:
*CLI Aug 28 16:15:17 NOTICE[6150]: chan_sip.c:3914 sip_reg_timeout:
Registration for '[EMAIL
Since Thursday evening my asterisk box has been failing to register
with
broadvoice. I haven't changed any of my config files in the last week.
Can anyone suggest anything?
Asterisk is reporting:
*CLI Aug 28 16:15:17 NOTICE[6150]: chan_sip.c:3914 sip_reg_timeout:
Registration for
I had the same problem. To fix it, I had to do two things
First: I had to update to CVS head, this was as per broadvoice support.
Second: After updating, I had to change my sip.conf. Originally my
sip.conf used hard coded ip addresses for broadvoice's IP servers, so I
had to change the
One other thing that I forgot to mention, while I was hunting this
problem down. Broadvoice also reprovisioned my account, which caused me
to have to get a new password. I do not know if you will need to do
this or not.
Ed
Ed Brady wrote:
I had the same problem. To fix it, I had to do two
I had the same problem. To fix it, I had to do two things
First: I had to update to CVS head, this was as per broadvoice
support.
Second: After updating, I had to change my sip.conf. Originally my
sip.conf used hard coded ip addresses for broadvoice's IP servers, so
I
had to
I had the same problem. To fix it, I had to do two things
First: I had to update to CVS head, this was as per broadvoice
support.
Second: After updating, I had to change my sip.conf. Originally my
sip.conf used hard coded ip addresses for broadvoice's IP servers, so
I
had to
Marty Mastera wrote:
I had the same problem. To fix it, I had to do two things
First: I had to update to CVS head, this was as per broadvoice
support.
Second: After updating, I had to change my sip.conf. Originally my
sip.conf used hard coded ip addresses
Marty Mastera wrote:
I had the same problem. To fix it, I had to do two things
First: I had to update to CVS head, this was as per broadvoice
support.
Second: After updating, I had to change my sip.conf. Originally my
sip.conf used hard coded ip addresses
Hey guys,
I was wondering what POE solutions are being used ? Ive done some
searching on google and found that PowerDsine seems to be good brand.
Any comments,suggestions, and experiences on POE hubs other POE products
would be greatly appreciated.
Thank you,
Steve Maroney
Steve Maroney wrote:
Hey guys,
I was wondering what POE solutions are being used ? Ive done some
searching on google and found that PowerDsine seems to be good brand.
Any comments,suggestions, and experiences on POE hubs other POE products
would be greatly appreciated.
Thank you,
Steve Maroney
On Sat, Aug 28, 2004 at 08:39:50AM -0600, Rich Adamson wrote:
I do a lot of work with companies throughout the US on network performance
and we _frequently_ run into routers, switches, servers, etc, that are
allowed to auto-negotiate their half vs full duplex nic interfaces. About
50% of
On Sat, Aug 28, 2004 at 03:00:26PM -0400, Michael George wrote:
On Sat, Aug 28, 2004 at 08:39:50AM -0600, Rich Adamson wrote:
I do a lot of work with companies throughout the US on network performance
and we _frequently_ run into routers, switches, servers, etc, that are
allowed to
On Sat, 28 Aug 2004 15:24:01 -0400, Michael George wrote:
On Sat, Aug 28, 2004 at 03:00:26PM -0400, Michael George wrote:
On Sat, Aug 28, 2004 at 08:39:50AM -0600, Rich Adamson wrote:
I do a lot of work with companies throughout the US on network performance
and we _frequently_ run into
Edward Eastman wrote:
I've got a TDM11B with the fxo port plugged into a standard UK BT PSTN line,
loading wcfxs with OPERMODE=UK. All's working well, except if I get an
incoming call through my bt line, and the remote party hangs up, I get
approx 20secs of the bt line hungup tone before
On Sat, 28 Aug 2004, Michael Welter wrote:
Since the Cisco 79XX phones preceded the PoE standard, they are
different--polarity is reversed.
IANAE, but as I understand the PoE devices, there are two types--one
always applies -48VDC to the brown pair while the other senses (as per
the PoE
Can anyone who is
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Can anyone who is using Asterisk with Broadvoice tell of their experiences with 3-way calling and call waiting? I can't get Broadvoice to respond to my question, but I understand that there is a per minute fee (3.9 c/minute?) if you go over your use allowances.
My question is, how are 3 way
Greg Boehnlein wrote:
On Sat, 28 Aug 2004, Michael Welter wrote:
Since the Cisco 79XX phones preceded the PoE standard, they are
different--polarity is reversed.
IANAE, but as I understand the PoE devices, there are two types--one
always applies -48VDC to the brown pair while the other senses
How do I go about disallowing transfers when I am running an IAX soft
phone. Is that setting not at the * server? Obviously, I don't have control over
the configuration of the IAXTel * server.
On 28 Aug 2004 at 19:03, you wrote:
Disallowing an IAX
transfer has always solved these problems
On Saturday 28 August 2004 15:00, Michael George wrote:
The difference between that and what I'm getting from IAX/GSM is profound,
with GSM being intolerably poor quality.
That's odd; every single voice call coming in and out of the company I work
for is using the GSM codec with asterisk and
Have you tried feeding it less amps at all?
Not yet, but i'll see if I can find a power supply with less amps.
--
jeremy bogan[ [EMAIL PROTECTED] ]
segment publishing - design.develop.host
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On Saturday 28 August 2004 15:24, Michael George wrote:
I just saw a page on the wiki that mentions that running X11 or a VESA
frame buffer can cause jittery sound. I only have this problem with IAX2,
but that might be cause when I use Zap -- Zap or Zap -- SIP there is no
en/decoding
On Sat, 28 Aug 2004, Michael Welter wrote:
Greg Boehnlein wrote:
On Sat, 28 Aug 2004, Michael Welter wrote:
Since the Cisco 79XX phones preceded the PoE standard, they are
different--polarity is reversed.
IANAE, but as I understand the PoE devices, there are two types--one
always
Beierlein Moritz schrieb:
Hi List,
does somebody know a SIP Provider which offers reseller possibilities?
...
Hi,
you are from Germany?
Sorry for advertising:
http://voip.planinternet.net
We are VoIP carrier and are supporting several
German VoIP providers.
Please contact us directly, if you
I am interested in setting up an Asterisk server as my home phone system.
I ultimately want one 10 digit phone number, three extensions, and an auto attendant
My current phone service provider is Vonage, I have one line with call waiting.
My concern is will I need to add additional lines if
VoiceValley will do soon in NZ.
Quoting Beierlein Moritz [EMAIL PROTECTED]:
Hi List,
does somebody know a SIP Provider which offers reseller possibilities?
Moritz
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There are at least two ways to configure Vonage and Asterisk to
connect to each other: you can connect the FXS port on your Vonaga ATA
to an FXO port on an Asterisk box, or you can make a SIP connection to
Vonage's server that Vonage thinks is coming from a soft phone.
If you connect the ports
On Sat, 28 Aug 2004 17:44:53 -0700, Greg Broiles wrote:
There are at least two ways to configure Vonage and Asterisk to
connect to each other: you can connect the FXS port on your Vonaga ATA
to an FXO port on an Asterisk box, or you can make a SIP connection to
Vonage's server that Vonage thinks
I've been in discussions with broadvoxdirect.
[EMAIL PROTECTED] has been encouraging our company to sign up. I have
yet to do so.
Chad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Saturday, August 28, 2004 3:43 PM
To: Asterisk
We are using a Netgear FSM7326P to PoE a 7960 (with 7914 attached).
Craig
- Original Message -
From: Michael Welter [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Sunday, August 29, 2004 2:13 AM
Subject: Re: [Asterisk-Users] POE
Well, so I'm unable to get any inbound calls from Vonage but can call
outbound all day long. If you are able to get inbound calls working I
would appreciate it if you could share your configs via personal email.
Rgs,
[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
On Aug 28, 2004, at 1:13 PM, Michael Welter wrote:
Since the Cisco 79XX phones preceded the PoE standard, they are
different--polarity is reversed.
IANAE, but as I understand the PoE devices, there are two types--one
always applies -48VDC to the brown pair while the other senses (as per
the
Is there an Asterisk Assistant for linux or windows?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris Shaw
Sent: Friday, August 27, 2004 10:57 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Are there
On Sat, Aug 28, 2004 at 12:57:48PM -0500, Steve Maroney wrote:
I was wondering what POE solutions are being used ?
The dumb, cheap ($30 retail) 3com PoE injectors (3CNJPSE) are working
fine for me with my snom 200 phones.
Steve
--
Steven Pritchard - KS Pritchard Enterprises, Inc.
Email: [EMAIL
I am trying to get distinctive ring to work on my PSTN with no luck. I can get 2 different ring codes but it skips the context assigned...
here is my complete zapata.conf: [channels] signalling=fxs_ks usecallerid=yes rxgain=1.0 txgain=1.0 language=en context=default
On Sat, Aug 28, 2004 at 05:08:30PM -0400, Andrew Kohlsmith wrote:
On Saturday 28 August 2004 15:24, Michael George wrote:
I just saw a page on the wiki that mentions that running X11 or a VESA
frame buffer can cause jittery sound. I only have this problem with IAX2,
but that might be cause
Steven Pritchard wrote:
The dumb, cheap ($30 retail) 3com PoE injectors (3CNJPSE) are working
fine for me with my snom 200 phones.
And there are plenty of places to buy them cheaper as well. Provantage
has them for around $22. www.soekris.com (a maker of embedded PC
systems) has the same unit
On Saturday 28 August 2004 23:01, Michael George wrote:
It's a PII 266 (okay, not the fatest system) with 192MB RAM. X is not
running and the Framebuffer has been turned off in /boot/grum/menu.lst. I
have disabled all the servers except for sshd. I have the latest source
from CVS HEAD as of
I played around with extensions.conf and found that if you
just use the 'r'
option in Dial app and leave out the other options t, T, m,
... then you get
dial indication. Haven't got around to try out the working
option combinations
but there it is.
PK
- Original Message -
On Sat, 2004-08-28 at 14:01 -0700, Muiz Motani wrote:
How do I go about disallowing transfers when I am running an IAX soft
phone. Is that setting not at the * server? Obviously, I don't have control over
the configuration of the IAXTel * server.
Which softphone are you using?
--
Dave
Seems the opencall.org site has basically been unavailable for days/weeks.
Is there another location to obtain the current code?
Also, will spandsp install against the current * cvs?
Rich
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