hi all,
actually i have partially connected the 2 servers but there is a problem.
2 servers A and B
server A forwards call to server B without any problem
but when i try to forward call from server B to A, server shows the
following error on the cli
WARNING[7751]: app_dial.c:1081
Hi,
what do you now get in the way of error messages?
Robert Jenkins.
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Vidura
Senadeera
Sent: 19 January 2007 23:03
To: asterisk-users@lists.digium.com
Cc: [EMAIL PROTECTED]
Subject: [asterisk-users] Re:
Well,
I have just phoned BT today who said they can increase the CPC value
on the line - however it needs to be done at the exchange - and has
been booked for Tues.
I suppose I will know wether this worked on Tues :-) - I shall post
my findings.
Regards
--
Matt Brown
On 19 Jan
Hi!
I did manage to load phone with SIP image : SIP70.8-0-3S, made
SEP-MAC.cnf.xml, but phone never read the configuration from it.
On the screen it's written Unprovisioned, and phone is not trying to
register with asterisk.
Please help!!
MihaelaMJ
Hi,
http://store.%50honiceq.com has the quadbri card for $400. We can also
offer free shipping for this card.
The card has 4 BRI ports and is based on the same main chipset
(HFC-4S) as Digium/Beronet/Junghanns cards. It does not have EC
onboard.
best regards
On 1/18/07, Cosmin Prund [EMAIL
Hi,
I tried the try version of chanskype, however, everytime that I make a call
asterisk generate an error
So you think is easy to us guess wich error you are getting?
Seriously, I think you should read this:
http://www.catb.org/~esr/faqs/smart-questions.html
Anyone has experience with this?
Recently, I got the following error messages in CLI periodically.
Jan 20 17:43:18 ERROR[8641]: chan_sip.c:11002
handle_request_subscribe: Got SUBSCRIBE for extension
[EMAIL PROTECTED] from 192.168.0.123, but there is no hint for that
extension
I have no idea what the error message tell me. I
Hello Asteriskies,
Has someone tried www.asterisknow.com ?
What is the package manager used? And what is the added value compared
to the well maintained debian based asterisk ?
Thanks,
--
Cheers,
Maxim Veksler
Free as in Freedom - Do u GNU ?
___
We have a similar system up and runing for 6 months, wiith 60 channels, and
average of simultaneas recorded calls us between 20 and 30.
We make test for recording 60 calls without any problems
We use a PIV Dual core with 3.2 Ghz with 2 mb of cache and 1Gb Ram.
regards
Mehdi
On 12/13/06, A.R.
Hello,
What's your zapata.conf and zaptel.conf?
On 1/20/07, Matt Brown [EMAIL PROTECTED] wrote:
Well,
I have just phoned BT today who said they can increase the CPC value
on the line - however it needs to be done at the exchange - and has
been booked for Tues.
I suppose I will know
Are you setting the TFTP server address in the DHCP?
Are you checking the TFTP log to see what files the phone is requesting and not
finding?
Regards
Jon
Jon Farmer
Telford, Shropshire, UK
- Original Message
From: Token PBX [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Hello
Asterisk implement only passtrough T.38, so you cant terminate calls with
asterisk using T.38.
You need T.38 gateways.
Regards
On 11/13/06, Ricardo Carvalho [EMAIL PROTECTED] wrote:
Dear all,
I'm trying to enable Asterisk to work with FAX using T38. I've tried
Asterisk 1.2.4 with the
Hi,
I'm trying to get my * server connected to a softswitch through an SBC. I
get the following error when * trys to register.
Got SIP response 423 Interval Too Brief back from xxx.xxx.xxx.xxx
Jan 20 12:43:54 NOTICE[2138]: chan_sip.c:5473 sip_reg_timeout:--
Registration for '[EMAIL
OpenPBX.org has better support, due to license issues and politial
bullshit I don't see Asterisk getting T.38 support that isnt a joke
(codec pass-thru?? LOL) for a long time. OpenPBX should have a stable
release within the month, if I am not mistaken they have a Release
Candiate #2 right now
What G723 codec do you have on Asterisk? What is your SIP.CONF? What
ATA/Phone is being used and what are the exact settings, especially
for the codec?
On 1/19/07, Phil French [EMAIL PROTECTED] wrote:
I am setting up Asterisk for use in a low bandwidth environment. As
bandwidth is precious and
On 1/20/07, Jon Farmer [EMAIL PROTECTED] wrote:
Are you setting the TFTP server address in the DHCP?
Are you checking the TFTP log to see what files the phone is requesting
and not finding?
Regards
Jon
Jon Farmer
Telford, Shropshire, UK
Hi Jon!
Yes I checked log, and phone requested
Hi,
I was wondering if someone had problems with chanskype.
Since I am wondering if they are a credible company or not.
See you
On 1/20/07, Moises Silva [EMAIL PROTECTED] wrote:
Hi,
I tried the try version of chanskype, however, everytime that I make a
call
asterisk generate an error
So
I have a system I'll be installing soon which has an ISDN30 (E1, UK) feed
and they want to hook up a fax machine and use existing analogue
conference phones (expensive polycom units)
This is something I've seen and used on legacy PBXs and would seem to be a
fairly standard offering, but an
Sounds like you need to dig into the documentation for the 7970 and
perhaps even contact Cisco TAC if that doesn't help.
It doesn't sound like your problem is related to Asterisk. The Cisco IP
phone won't register with asterisk until it's been provisioned. Those
7900 series cisco phones
you have probably something wron in config file and phone refuses to
configure,
here is my minimalistic file for 7941/61, you can try...
device
deviceProtocolSIP/deviceProtocol
sshUserIdadmin/sshUserId
sshPasswordadmin/sshPassword
devicePool
dateTimeSetting
Matt Brown wrote:
Well,
I have just phoned BT today who said they can increase the CPC value
on the line - however it needs to be done at the exchange - and has
been booked for Tues.
I suppose I will know wether this worked on Tues :-) - I shall post my
findings.
I would be keen to hear
Is func_odbc still working in trunk? I've recently (in the last few weeks)
started having a problem where my custom functions don't work. The module
loads, the configuration file is parsed fine, and the functions are even
created and visible in core show functions, but when executed from the
On 1/20/07, Pavel Jezek [EMAIL PROTECTED] wrote:
you have probably something wron in config file and phone refuses to
configure,
here is my minimalistic file for 7941/61, you can try...
device
deviceProtocolSIP/deviceProtocol
sshUserIdadmin/sshUserId
sshPasswordadmin/sshPassword
devicePool
Hi!
I have several SPA3000 devices (older versions of SPA3102) and they are
working OK, sound quality is good. It is very configurable to the slightest
details. I use it whenever I need just one or two FXO ports, like for small
scale PSTN integration, or for connecting some other equipment that
The asterisk core sounds includes a text file which gives the filename and
description of what the audio file says.
Is there a similar file for the extra sounds? I can't seem to find one.
-A.
___
--Bandwidth and Colocation provided by Easynews.com --
Ever since upgrading to 1.4 SVN, the advanced options on voicemail
have disappeared. When I press 3 for advanced options, it just
reviews the message. It used to present me with a menu to 1 = reply,
2 = call the person back, 3 = play message envelope. What gives?
If you own Aastra phones, here's a group dedicated to your specific
needs. BTW - The Asterisk-users mailing list is great but it has way
too much volume to be useful for specific problems. It needs to be
broken up into smaller more manageable lists.
Homepage:
Assuming your PRI supports timing from the remote end (CO) which I
highly suspect is the case, then you should set the asterisk machine
to be a slave to the CO timing and then set any other interfaces you
have to NOT be masters, so that the CO timing is always used. Assuming
you do this and
Perhaps someone could help you... if they actually had any knowledge
as to what your configuration is, which I doubt they do.
On 1/19/07, Eric Bishop [EMAIL PROTECTED] wrote:
On inbound calls from my SIP provider I get multiple warnings as follows:
WARNING[5351]: chan_sip.c:7086 check_via:
http://www.google.com/search?q=423+%22Interval+Too+Brief%22start=0ie=utf-8oe=utf-8client=firefox-arls=org.mozilla:en-US:official
Hint: Who develops Asterisk?
On 1/20/07, Thomas Madler [EMAIL PROTECTED] wrote:
Hi,
I'm trying to get my * server connected to a softswitch through an SBC. I
get
That's a great site! Perhaps it should be auto-sent to every poster
for the first 30 days of their membership:
Despite this, hackers have a reputation for meeting simple questions
with what looks like hostility or arrogance. It sometimes looks like
we're reflexively rude to newbies and the
I've actually found in many cases a lower bandwidth codec doesn't
improve at all and however it oftentimes makes the issue worse.
On 1/19/07, Martin Joseph [EMAIL PROTECTED] wrote:
On 2007-01-17 10:29:43 -0800, Yelson Vivas [EMAIL PROTECTED] said:
Hi Guys
I'm conecting 2 astersk servers
On Saturday 20 January 2007 6:21 pm, Rob Fugina wrote:
Is func_odbc still working in trunk? I've recently (in the last few weeks)
started having a problem where my custom functions don't work. The module
loads, the configuration file is parsed fine, and the functions are even
created and
I appreciate the response. The ATA is the linksys SPA-2102 and some of
its configured settings are below. After the ATA information I have
included the sip.conf file and packet summary of a call with garbled
audio. Regarding the G723 codec, we have compiled a g723.1 codec. This
same source is
Hi Philipp,
Thanks for the tip, but that is not what I initially meant. I'm using
IDEfisk, and I would like it when a call comes
Into IDEfisk to generate a BUSY signal, if there is already a call in the
client. Any ideas ?
Nir S
-Original Message-
From: [EMAIL PROTECTED]
Leo Ann Boon wrote:
Andrew Joakimsen wrote:
Most of the Cisco phones sold cheap are UNLICENSED (global spare)
thus you would not be able to purchase (or at least aren't supposed
to) the smartnet contracts, you need to buy the license ($100+) and
the contract ($10 or so)
I'm always surprised by
BerkHolz, Steven wrote:
Announce option for meetme - is it used?
It makes a caller record their name, but I do not see where this name recording
is ever used.
Thank You,
Steven BerkHolz
- MCSA - MCSE -
Manager of Information Systems
HIROTEC AMERICA
Fax. 248-836-5101
www.hirotecamerica.com
IDEfisk is based on iaxclient (iaxclient.sourceforge.net) which will
reject a call with the BUSY signal if there is no available line in
the softphone to take the call.
This means you need to configure IDEfisk to use only one line (call
context). I don't know if this is possible.
Somewhere in
Hi,
I was wondering if it is possible to connect a skype phone adapter, for
example:
http://zonetusa.com/DispProduct.asp?ProductID=191
http://www.actiontec.com/products/communications/ipw_usb/index.php
http://www.eradian.com/ERadianUS/staticpages/SkytoneRST301Details.htm
Some NAT problems you can solve, some you never will.
Many modern phones have NAT support in them, via STUN, or a static external IP
address. Most NATs also offer port forwarding, so you can open a hole for the
SIP port in the NAT so all outside can reach it.
(With port forwarding, you need a
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