Hi list!
I asked this in this list some time ago, and now I was searching for
evolution about this subject, but I found nothing.
Nowadays, what is the state for H.323 video support?
Is there support in the 1.6 beta brunch?
If not, is this in the roadmap for 1.6 brunch?
Regards.
Diego.
will an ata directly connect to another remote ata thus emulating a long
phone cord? also most of the ATA's I've seen drive a phone rather than
accepting a line from the telco.
Eric
On Thu, May 22, 2008 at 10:17 PM, Alex Balashov [EMAIL PROTECTED]
wrote:
Eric Fort wrote:
I'm looking for a
On Fri, May 23, 2008 at 10:04 AM, Eric Fort [EMAIL PROTECTED] wrote:
will an ata directly connect to another remote ata thus emulating a long
phone cord? also most of the ATA's I've seen drive a phone rather than
accepting a line from the telco.
Depending on the reliability needed (is this a
Eric Fort wrote:
will an ata directly connect to another remote ata thus emulating a long
phone cord? also most of the ATA's I've seen drive a phone rather than
accepting a line from the telco.
Good, higher-end ATAs and IADS will be able to trunk to each other, and
do FXO and FXS
Just before 12 Noon EDT, call :
- SIP/[EMAIL PROTECTED],60,D(22622#1#)
-or- (724) 444-7444 and enter 22622# 1#
there is also a DNS: TS.X2Z.EU if you're too lazy to type 66.212.134.192
--- If you have a PIN, please use it rather than the 1#
I didn't want to use Big and Fast as the subject and
C F wrote:
Then there is basicly no way to do this besides for cracking it? I
Not that I am aware of, no. This subject went around several years
back. They also talk about brute forcing the password as well. As far
as I recall, nobody came back saying they were successful.
have
On Fri, May 23, 2008 at 4:05 AM, Diego Moreno [EMAIL PROTECTED] wrote:
Hi list!
I asked this in this list some time ago, and now I was searching for
evolution about this subject, but I found nothing.
Nowadays, what is the state for H.323 video support?
Is there support in the 1.6 beta
On Fri, May 23, 2008 at 5:49 AM, VIAGRA (R) Official Site
asterisk-users@lists.digium.com wrote:
About this mailing:
You are receiving this e-mail because you subscribed to MSN Featured Offers.
Microsoft respects your privacy. If you do not wish to receive this MSN
Featured Offers e-mail,
On Thu, May 22, 2008 at 9:51 PM, Sam Tam [EMAIL PROTECTED] wrote:
Why if you have 50 operator then I would even consider using dual server
running backup
So the idea of using vmware may really be very risky, let alone not talk
about performance issue
well vmware will not be installed on a
On Fri, May 23, 2008 at 3:19 PM, Steve Totaro
[EMAIL PROTECTED] wrote:
On Fri, May 23, 2008 at 5:49 AM, VIAGRA (R) Official Site
asterisk-users@lists.digium.com wrote:
About this mailing:
You are receiving this e-mail because you subscribed to MSN Featured Offers.
Microsoft respects your
Two grandsreams, a 4008 and a 4108 would inexpensively do this for you.
Instructions are on their site.
-Original Message-
From: Alex Balashov [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: 5/23/08 1:21 AM
Subject:
We are trying to get our first B410P installation working but unable to get
any L1 or L2 links.
Connecting 3 ports to BRI lines via S/T interfaces on telco provided NT
equipment.
Using Debian etch 2.6.18-6
Ports are configured as TE
Have tried both PTP and PTMP modes
All 4 red
Sorry to jump in on this but I am also interested in this topic.
In my scenario I have about 10 POTs lines brought into the front of a
facility and the only infrastructure connecting the back of the facility
is a 3000ft fiber backhaul. I've been asked to bring the POTs lines to
the back of the
On Fri, May 23, 2008 at 8:41 AM, Chris Curtis
[EMAIL PROTECTED] wrote:
We are trying to get our first B410P installation working but unable to get
any L1 or L2 links.
Connecting 3 ports to BRI lines via S/T interfaces on telco provided NT
equipment.
Using Debian etch 2.6.18-6
Ports
As usual its rather urgent to get it running. If anyone can provide
suggestions ,I'd be very grateful.
Chris
Is your card compatible with BRIStuff?
Thanks,
Steve Totaro
Don't forget to modprobe qodah ;-)
Steve T
___
-- Bandwidth and
There are a couple of companies out there that make 24 port fxo and fxs boxes.
If you have some unused fibers you cout do this very reliably with two channel
banks... One with fxs ports and the other with fxo ports and t1 media
converters.
The grand stream solution mentioned in an earlier
Thanks :-D change the context to default and everithing works fine.
I assigned the sip context because that was the context on the example.
Thanks :-)
Nomar
Alex Balashov wrote:
Nomar Mora wrote:
Alex Balashov wrote:
Do you have dial plan routes for internal extension calls?
This is an email to all New York based Asterisk users.
For some time it's been bugging me that we don't have a local contact
point/user community. If you are involved in Asterisk and in NY/NJ shoot
me an email, I'm going to try and revitalize either meetup.com or some
other shared environment
Yes, you are right... sorry for my fast and poor English.
I rewrite my questions:
Nowadays, what is the state for H.323 video support?
Is there support in the 1.6 beta branch?
If not, is this in the roadmap for 1.6 branch?
Regards.
2008/5/23 Steve Totaro [EMAIL PROTECTED]:
On Fri, May 23,
I don't really know as I am unfamiliar with BRIstuff. If fact the whole ISDN
world is a new one for me as its not that common for small business here in
Australia. We are using the Digium B410P. A quick Google of B410P and
BRIstuff is inconclusive.
-Original Message-
From: [EMAIL
Dear Randulo,
Thanks for your suggention.
Now i am able to communicate between 2 computers.
Regards,
Baskar
--- randulo [EMAIL PROTECTED] wrote:
On Mon, May 19, 2008 at 8:44 AM, bas karan
[EMAIL PROTECTED] wrote:
[May 19 12:02:29] NOTICE[2559]: chan_sip.c:13879
handle_request_invite: Call
Quoting Doug Lytle [EMAIL PROTECTED]:
C F wrote:
Then there is basicly no way to do this besides for cracking it? I
Not that I am aware of, no. This subject went around several years
back. They also talk about brute forcing the password as well. As far
as I recall, nobody came back
On Fri, 2008-05-23 at 22:41 +1000, Chris Curtis wrote:
We are trying to get our first B410P installation working but unable
to get any L1 or L2 links.
Which mISDN and kernel version are you using? There was a problem with
very recent kernels mISDN so you might want to check the mISDN mailing
Friends,
I will be spending a few days in Porto, Portugal in the beginning of
June.
Any Asterisk and/or OpenSER users there that wants to go out and have
dinner and Open Source Voip talk?
Respond off list, and we'll see if we can meet.
Have a nice weekend!
/Olle
Do you mean the city or the state of New York?
I'm in NY, but a long ass way from NYC.
This is an email to all* New York* based Asterisk users.
For some time it’s been bugging me that we don’t have a local contact
point/user community. If you are involved in Asterisk and in NY/NJ
shoot me
In the setup tutorial @
http://www.voip-info.org/wiki/view/CentOS+5+and+Asterisk+1.4.x+installation
it states the potential issue regarding setting up UniqueID
as the primary key, but doesn't state how to rectify this?
What is the proper way to make sure this is done right?
Also, has anyone
Remind me to pick on your poor Spanish next time I see you for a
mid-morning meal. :)
Steve Totaro wrote:
On Fri, May 23, 2008 at 4:05 AM, Diego Moreno [EMAIL PROTECTED] wrote:
When and where is the 1.6 brunch? ;-)
___
-- Bandwidth and
:)
Same here
On Fri, May 23, 2008 at 10:13 AM, Adam Moffett [EMAIL PROTECTED] wrote:
Do you mean the city or the state of New York?
I'm in NY, but a long ass way from NYC.
This is an email to all* New York* based Asterisk users.
For some time it's been bugging me that we don't have a local
Hey Adam,
Yes I was thinking NYC - basically I was surprised at the lack of
response about Ming from Voiceroute wanting to organize a physical
meeting event (btw it got moved to the 2nd of June)
This bugged me as when you look at other opensource community groups in
NY I belong to they
On Fri, May 23, 2008 at 01:20:58AM -0400, C F wrote:
serial, I don't know the IP address.
Loopback cable.
nmap -sP 0/1
nmap -sP 1/1
Cheers,
-- jra
--
Jay R. Ashworth Baylink [EMAIL PROTECTED]
Designer The Things I Think
Also what do I do if I see deadlocks all over the CLI?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark Hamilton
Sent: May 23, 2008 12:13 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] reload stopping
Hi All,
In my old telco days (SS7), if I was wanting to hand back a call to the
network for transfer to a different PSTN number, there was a specific
SS7 action I could take, which send the call back to the network, which
in turn then routed the call appropriately. It added a transfer-number
Hello! Iam configuring chan Skype on my asterisk box, doing some test calls
I saw that asterisk answer the calls but hungs up before the call are
stablished. Is this a license problem?
Gustavo A. González
Dto. de Infraestructura
Despegar.com, Inc.
[EMAIL PROTECTED]
yes that's how I figured out the username. since it returned incorrect
login before the password prompt on the wrong username.
I a don't know the password however.
On 5/23/08, Shane Young [EMAIL PROTECTED] wrote:
Quoting Doug Lytle [EMAIL PROTECTED]:
C F wrote:
Then there is basicly no way
seen that thread, it doesn't help me much since I only have seriel
access and no linux machine with a seriel port
On 5/23/08, Doug Lytle [EMAIL PROTECTED] wrote:
C F wrote:
Then there is basicly no way to do this besides for cracking it? I
Not that I am aware of, no. This subject went
Will fax and dial-up internet work through the gateway?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joe
Carroll
Sent: Friday, May 23, 2008 8:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion; Asterisk
Users Mailing List -Non-Commercial
Hello,
I've been annoyed quite some time by behavior of queue timeout
(specified as argument to Queue app). Basically if I specify timeout
for queue 5 minutes,
and ring time to agent for 15 seconds, and ring to agent starts at
4:59, agent will receive ring only for 1 second, after which call
the subject of this thread has been on this list way too many times
just search the archives.
On 5/23/08, Joseph L. Casale [EMAIL PROTECTED] wrote:
In the setup tutorial @
http://www.voip-info.org/wiki/view/CentOS+5+and+Asterisk+1.4.x+installation
it states the potential issue regarding
I am in Philadelphia, keep me updated and will try to make time to attend.
On Fri, May 23, 2008 at 10:25 AM, Dean Collins [EMAIL PROTECTED] wrote:
Hey Adam,
Yes I was thinking NYC - basically I was surprised at the lack of response
about Ming from Voiceroute wanting to organize a physical
Nevermind. Attached atleast two backtraces, one with Asterisk running and
not coredumping, and two with Asterisk built coredumps using DO_CRASH.
If anyone is interested, please check
http://bugs.digium.com/view.php?id=12709
Thanks.
-Original Message-
From: [EMAIL PROTECTED]
In my Asterisk CLI I get Ring/Off-hook in strange state 6 when i make
a call into the system, the system claims to answer the call, and do
the things in the dial plan, but I just hear ringing on the phone I'm
calling in from.
I am using a Sangoma A200 4 Port Analog card.
my wanrouter
Mark Hamilton wrote:
Also what do I do if I see deadlocks all over the CLI?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark Hamilton
Sent: May 23, 2008 12:13 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
In my Asterisk CLI I get Ring/Off-hook in strange state 6 when i
make a call into the system, the system claims to answer the call,
and do the things in the dial plan, but I just hear ringing on the
phone I'm calling in from.
I am using a Sangoma A200 4 Port Analog card.
my wanrouter
Daniel Lockard wrote:
In my Asterisk CLI I get Ring/Off-hook in strange state 6 when i make
a call into the system, the system claims to answer the call, and do
the things in the dial plan, but I just hear ringing on the phone I'm
calling in from.
I am using a Sangoma A200 4 Port
Mark Hamilton wrote:
Nevermind. Attached atleast two backtraces, one with Asterisk running and
not coredumping, and two with Asterisk built coredumps using DO_CRASH.
If anyone is interested, please check
http://bugs.digium.com/view.php?id=12709
Thanks.
-Original Message-
From:
Has anyone tried using ooh323 in Asterisk to talk H.323 to an Avaya S8500
running Communications Manager 4 software?
I have a potential customer who has such a system, and wants an Asterisk
box to talk to it. Apparently they don't have SIP installed.
I've successfully got ooh323 talking between
Well, then it might not be that that is causing me issues? I have no idea
why I would be able to call in, hear ringing on my phone, and then have the
CLI tell me that it has answered...
Daniel Lockard
On 5/23/08, Sherwood McGowan [EMAIL PROTECTED] wrote:
Daniel Lockard wrote:
In my Asterisk
Any more information on this?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of faraz
Sent: Friday, April 18, 2008 6:30 PM
To: Watkins, Bradley
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom LDAP
I posted about this exact same problem about a month or two ago and got no
replies. The problem was an A200D with Wanpipe 3.2.1. We've since made some
changes to this installation and the problems have gone away. The changes were
specifically getting a new local loop from the telco on the
Adrian Marsh wrote:
Hi All,
In my old telco days (SS7), if I was wanting to hand back a call to
the network for transfer to a different PSTN number, there was a
specific SS7 action I could take, which send the call back to the
network, which in turn then routed the call appropriately. It
On Fri, May 23, 2008 at 01:25:43PM -0400, Donny Kavanagh wrote:
This is getting downright abusive, and is totally uncalled for, this
is not a list for personal attacks.
You thought that Steve suggesting JT step in was abusive?
If that's not what you meant, then you need to either a) be
sorry for not replying to this sooner!
but the canreinvite=no trick worked a treat.
thank you
--
Alan Williamson
Try our free registrationless email/sms reminder
http://yourli.st/
b: http://alan.blog-city.com/
Steve Davies wrote:
If the two phones attempt to refer to each other
2008/5/22 C. Chad Wallace [EMAIL PROTECTED]:
When it says FXS only, I think it's reasonable to assume that FXO is
excluded.
FXS is the signalling of FXO cards.
I have only FXO cards.
--
PicoStreamer - the real WEB live streaming software
vinz486.com
Hi,
If Asterisk doens't suffer a virtualization, a service virtualized on
a solid infrastructure is more scalable and hardware independent
at the beginning of my project I was thinking to do so too (with Xen),
but I was told that delays etc. in a virtualized environment will be a
significant
This does not do the trick, because while the voice path is not created
until the digit 1 is dialed, when the first extension picks up the others
stop ringing. What is needed is something where all extensions continue
ringing until the digit is dialed.
On Mon, May 12, 2008 at 10:54 AM, Andreas
I'm trying to set the outgoing caller id to the DID number, but only if
the extension is greater than 140. MAINSTUB is simply the first 7 digits
of the main number. sip.conf sets the CALLERID(num) to the extension.
exten =_1NXXNXX,n,Set(CALLERID(num)=${MAINSTUB}${CALLERID(num)})
works. But
I used to do lots of Asterisk, but got an offer I couldn't refuse, and
went SysAdmin. Well, now I'm trying to bring Asterisk in-house, and want
to set up a test system. One thing I'd really like to get my hands on is
recent firmware, etc., for SoundPoint IP 430's. Freedomphones.net, my old
On May 23, 2008 11:25:55 am Dennis P. Clark wrote:
Will fax and dial-up internet work through the gateway?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joe
Carroll
Sent: Friday, May 23, 2008 8:51 AM
To: Asterisk Users Mailing List -
On May 23, 2008 05:27:49 pm Ken D'Ambrosio wrote:
I used to do lots of Asterisk, but got an offer I couldn't refuse, and
went SysAdmin. Well, now I'm trying to bring Asterisk in-house, and want
to set up a test system. One thing I'd really like to get my hands on is
recent firmware, etc.,
On Friday 23 May 2008 16:27:49 Ken D'Ambrosio wrote:
I used to do lots of Asterisk, but got an offer I couldn't refuse, and
went SysAdmin. Well, now I'm trying to bring Asterisk in-house, and want
to set up a test system. One thing I'd really like to get my hands on is
recent firmware, etc.,
Thanks for the tip on the 400x, 401x, and 4024 grandstreams. They will work
quite nicely (and they'll do t.38 fax if necessary). My application is that
I help move business offices from place to place and during the move
period. This solution is helpful so the move can be done during the phone
Hi,
when i use Dial(SIP/201SIP/202SIP/,60,r) the ring sound is very strange.
This happens after an IVR menu.
Instead of
ing..ing..ing
is
On Fri, May 23, 2008 at 05:08:28PM -0400, sean darcy wrote:
This doesn't work:
exten =_1NXXNXX,n,Set( CALLERID(num) = ${IF ( $[${CALLERID(num)}
140] ? ${MAINSTUB}${CALLERID(num)} : ${MAINNUMBER} )})
Change IF ( to IF(.
___
-- Bandwidth and
Remove the r. Asterisk will provide the proper ringing by default.
If it is not doing so then something is wrong in the config.
Compressed codecs (any codec other than ulaw/alaw) do not handle
non-voice very well (i.e. ringing, MoH, etc).
Vinz486 wrote:
Hi,
when i use
Hello,
Do you redirected the rtp ports to your phone?
usually 1 - 2 defautl rtp ports
Best Regards
Carlos Rojas
On Thu, May 22, 2008 at 8:48 AM, Phibee Network Operation Center
[EMAIL PROTECTED] wrote:
I have a problem connecting a Grandstream ipphone to an asterisk.
The
Barry Miller wrote:
On Fri, May 23, 2008 at 05:08:28PM -0400, sean darcy wrote:
This doesn't work:
exten =_1NXXNXX,n,Set( CALLERID(num) = ${IF ( $[${CALLERID(num)}
140] ? ${MAINSTUB}${CALLERID(num)} : ${MAINNUMBER} )})
Change IF ( to IF(.
Thanks for the response.
Tried it this
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