Hi,
I made good experienes with Siemens Gigaset C610 IP. This model is about
90 Euro. Configuration via web interface. But encryption (SIPS/SRTP) is
*not* possible with this phones.
-Thorsten-
Am 11.12.2013 11:30, schrieb Mario Giammarco:
Hello,
I need to setup this configuration:
-
Hello Mario,
nice to meet you on this mailing list!
Gigaset phones are a very high quality/price ratio, so I'll suggest you to
go with the dect ip models. Then you'll need to configure asterisk to act
as IVR, configure a queue and a failover to ring all hunt list.
Drop me a phone call and I'll be
...@lists.digium.com] On Behalf Of Dean Collins
Sent: Thursday, February 17, 2011 7:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users]Newbie´s question about Asterisk...
If you already have experience with linux asterisk will be easy for you.
Other people
List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users]Newbie´s question about Asterisk...
If you already have experience with linux asterisk will be easy for you.
Other people will reply with official links but here is how I use Asterisk
in my small home office www.cognation.net
Discussion
Subject: Re: [asterisk-users]Newbie´s question about Asterisk...
i prefer to go with Elastix very easy to setup and maintain and reach UI rather
than freePBX
cheers
Dhaval
On Fri, Feb 18, 2011 at 4:14 PM, Terry Brummell te...@brummell.net wrote:
Dean's link has references to Trixbox. TB
-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *DHAVAL INDRODIYA
*Sent:* Friday, February 18, 2011 6:11 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users]Newbie´s question about Asterisk...
i prefer
-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *DHAVAL INDRODIYA
*Sent:* Friday, February 18, 2011 6:11 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users]Newbie´s question about Asterisk...
i prefer
-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *DHAVAL INDRODIYA
*Sent:* Friday, February 18, 2011 6:11 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users]Newbie´s question about Asterisk...
i prefer
...@lists.digium.com on behalf of Francisco Javier
Cintrón Olguín
Sent: Fri 2/18/2011 12:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users]Newbie´s question about Asterisk...
Is there another way to interface to 3 external and 6 internal lines??
Thank you
(Please don't top-post and please trim posts that are no longer relevant.)
On Fri, 18 Feb 2011, Francisco Javier Cintrón Olguín wrote:
I think I have 3 PSTN lines because I can connect a normal telephone to
them all and make calls between each of them. We have 5 normal
telephones and 1
First of all, thank you for your help.
I was seing Cisco and Linsys web sites and I just came across this 2
devices:
Linksys SPA8000 8 phone ports, 1 port ethernet.
Cisco SPA8800 4 phone ports, 4 lines, 1 port ethernet.
I think they could work for us, because I need maximum 10 normal phones and
If you already have experience with linux asterisk will be easy for you.
Other people will reply with official links but here is how I use Asterisk in
my small home office www.cognation.net/asterisk
Cheers,
Dean
From:
On Fri, 4 Feb 2011 10:54:56 +0330, Pezhman Lali l...@lopl.net wrote:
Meetme is a default conference application, but you can try conference or
konference
http://www.voip-info.org/wiki/view/Asterisk+cmd+Conference
http://www.voip-info.org/wiki/view/Asterisk+cmd+Conference
Dear,
Meetme is a default conference application, but you can try conference or
konference
http://www.voip-info.org/wiki/view/Asterisk+cmd+Conference
http://www.voip-info.org/wiki/view/Asterisk+cmd+Conference
http://www.voip-info.org/wiki/view/Asterisk+cmd+Konference
On Mon, 31 Jan 2011, Piotr Górski wrote:
I have 4 PSTN lines connected to TDM410. I can make exact 60 minutes of
free calls from each of 4 pstn lines... Can I configure Asterisk to call
thru pstn line that has free minutes? For example
Outgoing calls are going through PSTN 1 for 60 minutes.
UIT DEVELOPMENT wrote:
Sorry for such a silly question but I am VERY new to Linux, Asterisk,
and so forth. I just downloaded and burned the AsteriskNOW ISO to CD
and installed it. Everything went great. I removed the CD and
rebooted and there is a prompt for me to login.
I hate to ask
John, Thank you! That was it. I was trying admin, login...I
should have searched google more. Thank you and thanks for the link -
I've got lots of reading ahead of me this evening!
Mike
On Fri, Dec 25, 2009 at 6:00 PM, John Novack
jnov...@stromberg-carlson.org wrote:
UIT DEVELOPMENT
Hi,
I'd say Linphone configuration. I suggest to check Linphone
configuration and also asterisk debug - one of those will give you an
answer (most likely asterisk debug as it shows you what it receives ...
if it doesn't receive anything then Linxphone fails)
--
razu
On 11/19/2009 11:48 PM,
You can tee your CLI screen (google for it) so your output is in a file
that you can use more|less|vi or some other controlled viewing method on.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bill Shaw
Sent:
On Tue, Nov 17, 2009 at 11:33:45AM -0500, Bill Shaw wrote:
Hi All,
When typing 'help' on the command line (* console) is there a way to
keep it from just scrolling most of the information off the top of the
screen? I can't hit ctrl-s fast enough so I miss most of the info. This
makes
On Tue, Nov 17, 2009 at 11:33:45AM -0500, Bill Shaw wrote:
Hi All,
When typing 'help' on the command line (* console) is there a way to
keep it from just scrolling most of the information off the top of the
screen? I can't hit ctrl-s fast enough so I miss most of the info. This
makes
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] newbie question
On Tue, Nov 17, 2009 at 11:33:45AM -0500, Bill Shaw wrote:
Hi All,
When typing 'help' on the command line (* console) is there a way to
keep it from just scrolling most of the information off the top of the
screen
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Bill Shaw
Sent: Tuesday, November 17, 2009 11:34 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] newbie question
When typing 'help' on
On Tue, Nov 17, 2009 at 09:09:39AM -0800, Steve Edwards wrote:
On Tue, Nov 17, 2009 at 11:33:45AM -0500, Bill Shaw wrote:
Hi All,
[snip]
2. Run from the external shell prompt:
asterisk -rx 'help whatever' | less
Or, you can use the script command to capture the output to a file
When typing 'help' on the command line (* console) is there a way to
keep it from just scrolling most of the information off the top of the
screen? I can't hit ctrl-s fast enough so I miss most of the info. This
makes 'help' be not much help.
my default scroll back buffer is set to around
On Tue, 17 Nov 2009, Noah Miller wrote:
You could also make it much simpler and just set your verbosity very
low or just turn it off, so there are very few messages coming across
your screen. Unless you're on a really busy machine, you should be
able to read most of the help screens.
core
On Wed, 19 Aug 2009, Lee, John (Sydney) wrote:
I was copying tracks from CD into mp3 files so that I could use it in
Asterisk 1.4.21.2 MOH.
Are there any Asterisk+Audio expert that can offer me some advice?
Don't use MP3. Why would you want to burn CPU cycles decompressing the
same stuff
Steve Edwards wrote:
On Wed, 19 Aug 2009, Lee, John (Sydney) wrote:
I was copying tracks from CD into mp3 files so that I could use it in
Asterisk 1.4.21.2 MOH.
Are there any Asterisk+Audio expert that can offer me some advice?
Don't use MP3. Why would you want to burn CPU cycles
Yep, agreed.
Convert the file to the native codec(s) in which it will be played.
Alex, could you please elaborate on this? I am no audio guy.
On Media player, I can rip it into mp3 or wav or windows media audio.
Which one should I use?
___
--
Probably none of the ones you list, though I believe wav files are
uncompressed. Use SOX http://sox.sourceforge.net/ under Linux, Windows or
OSX and RIP/Convert the files to match the codec you are using for calls.
If you are accepting calls that use the GSM codec then have a set of MOH
files
On Thu, 20 Aug 2009, Lee, John (Sydney) wrote:
Convert the file to the native codec(s) in which it will be played.
Alex, could you please elaborate on this? I am no audio guy.
On Media player, I can rip it into mp3 or wav or windows media audio.
Which one should I use?
Neither.
If your
Lee, John (Sydney) wrote:
Thanks Tilghman.
I learnt it the hard way - I never imagined I need to jot down the
serial number of a PCI card :-(
I've had a linecard that's been unregistered now for 4 years or more,
because it's in a production server.
It does of course mean that I didn't get
Sent: Monday, 17 August 2009 1:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Newbie: How to find the serial number
ofDigium card?
On Sunday 16 August 2009 19:59:50 Lee, John (Sydney) wrote:
Does anyone know how to find the serial number of Digium
On Sunday 16 August 2009 19:59:50 Lee, John (Sydney) wrote:
Does anyone know how to find the serial number of Digium card without
opening the machine?
I was trying to call for support at Digium and they asked me for the
serial number.
You cannot. The serial number is not anywhere in the
:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Newbie: How to find the serial number
ofDigium card?
On Sunday 16 August 2009 19:59:50 Lee, John (Sydney) wrote:
Does anyone know how to find the serial number of Digium card without
opening
On Sat, 20 Jun 2009, C. Savinovich wrote:
Let me see if I get you: you inserted the installation CD, then you
restarted the computer, and now you want to know what to do next?
How about:
1) Turn off the computer.
2) Read the installation guide for the CD.
3) Install the software.
4) Read
to help, by getting you to try a better way.
Good luck.
Cary Fitch
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Shiva Kumar
Sent: Tuesday, June 16, 2009 12:27 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk
On Mon, 15 Jun 2009, Shiva Kumar wrote:
Hello Asterisk-users,
I am new to Asterisk. I got SIP Calls to work between two computers using a
soft phone and asterisk in the middle. Since then, I have been trying to get
my soft phone to make a PSTN call with terrible failure for about two days
now.
Need help pls..Anyone?
On Mon, Jun 15, 2009 at 10:58 PM, Shiva Kumar shi...@gmail.com wrote:
Hello Asterisk-users,
I am new to Asterisk. I got SIP Calls to work between two computers using a
soft phone and asterisk in the middle. Since then, I have been trying to get
my soft phone to make a
On Mon, Mar 30, 2009 at 5:16 PM, Bruce Thayre br...@mipscomputation.comwrote:
Up to this point, all i have set up are two SIP phones, my POTS phone,
and 1 ring group. My POTS line is connected to channel 1, and my POTS
phone is connected on channel 3. Perhaps my understanding of how the
Show us your dialplan.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bruce Thayre
Sent: Monday, March 30, 2009 4:16 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Newbie trying to make calls
Thank you for the prompt input! My extension.conf can be viewed here:
http://dpaste.com/21356/
I'm currently doing the configuration through the GUI bundled with the
trixbox distro, and i'm not entirely sure where it stores all of the
changes as i haven't seen the changes to extension.conf that i
Geoff Lane wrote:
On Thursday, February 5, 2009, Mark Michelson wrote:
I've tried it and you're correct. So it looks like the docs need a
bug report - any idea how I go about that?
If you're using the 2nd edition of the book, check the preface, page xix for
contact information.
Thanks
On Thu, 2009-02-05 at 22:09 +, Geoff Lane wrote:
Thanks. For info, *TFOT says:
PrivacyManager() sets a channel variable named PRIVACYMGRSTATUS to
either SUCCESS or FAILURE. If Caller ID is received on the channel,
PrivacyManager() does nothing.
I've tried it and you're correct. So it
On Thu, 2009-02-05 at 22:01 +0100, Philipp Kempgen wrote:
Mark Michelson schrieb:
Actually, jumping to priority n + 101 is a thing of the past
And in addition extensions.conf is a thing of the past. ;-)
extensions.ael is cleaner and easier to maintain for most
purposes.
In the same
On Thu, 2009-02-05 at 22:01 +0100, Philipp Kempgen wrote:
Mark Michelson schrieb:
Actually, jumping to priority n + 101 is a thing of the past
And in addition extensions.conf is a thing of the past. ;-)
snip
How about .. dialplan.conf .;-)
Geoff Lane wrote:
Hi All,
Asterisk 1.4.12 on CentOS 5
Sorry for a question that I'm guessing is obvious to most of you.
I'm trying to revamp my dialplan. When I first created it, I had
something like:
exten = s,1,Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})})
exten =
On Thursday, February 5, 2009, Mark Michelson wrote:
Actually, jumping to priority n + 101 is a thing of the past, and
this will only occur now if you pass the 'j' option to Dial. Dial
will just go to the next priority on a timeout now, and the
DIALSTATUS channel variable will be set to
Mark Michelson schrieb:
Actually, jumping to priority n + 101 is a thing of the past
And in addition extensions.conf is a thing of the past. ;-)
extensions.ael is cleaner and easier to maintain for most
purposes.
Philipp Kempgen
--
AMOOCON 2009, May 4-5, Rostock / Germany -
On Thursday, February 5, 2009, Philipp Kempgen wrote:
And in addition extensions.conf is a thing of the past. ;-)
extensions.ael is cleaner and easier to maintain for most purposes.
Oh-oh ... I don't think I can keep up with the rate of change ;-)
BTW, on a related note, I'm having some
Philipp Kempgen wrote:
And in addition extensions.conf is a thing of the past. ;-)
extensions.ael is cleaner and easier to maintain for most
purposes.
*gack*
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither
On Thursday 05 February 2009 15:37:19 Geoff Lane wrote:
On Thursday, February 5, 2009, Philipp Kempgen wrote:
And in addition extensions.conf is a thing of the past. ;-)
extensions.ael is cleaner and easier to maintain for most purposes.
Oh-oh ... I don't think I can keep up with the rate
On Thursday, February 5, 2009, Tilghman Lesher wrote:
The correct string is FAILED, not FAILURE.
Thanks. For info, *TFOT says:
PrivacyManager() sets a channel variable named PRIVACYMGRSTATUS to
either SUCCESS or FAILURE. If Caller ID is received on the channel,
PrivacyManager() does nothing.
Tilghman Lesher schrieb:
On Thursday 05 February 2009 15:37:19 Geoff Lane wrote:
BTW, on a related note, I'm having some trouble with Privacy Manager
that I'd appreciate some insight with. In one priority, I'm calling
PrivacyManager(2,8). In the next priority, I've got:
...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Philipp Kempgen
Sent: Thursday, February 05, 2009 16:01
To: Asterisk Users
Subject: Re: [asterisk-users] Newbie query: how to write priority n+101
Mark Michelson schrieb:
Actually, jumping to priority n + 101
Geoff Lane wrote:
On Thursday, February 5, 2009, Tilghman Lesher wrote:
The correct string is FAILED, not FAILURE.
Thanks. For info, *TFOT says:
PrivacyManager() sets a channel variable named PRIVACYMGRSTATUS to
either SUCCESS or FAILURE. If Caller ID is received on the channel,
On Thursday, February 5, 2009, Mark Michelson wrote:
I've tried it and you're correct. So it looks like the docs need a
bug report - any idea how I go about that?
Thanks again,
If you're using the 2nd edition of the book, check the preface, page xix for
contact information.
Thanks -
'
Subject: Re: [asterisk-users] Newbie in Cisco Phone
Hi
I am no expert in the cisco phone
Do you have time to help
Sam
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Federico
Santulli
Sent: Saturday, January 24
...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michiel van
Baak
Sent: Friday, January 23, 2009 3:35 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Newbie in Cisco Phone
On 05:39, Fri 23 Jan 09, Sam Tam wrote:
Yes I know too.
Is there anyway
- Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Friday, January 23, 2009 8:56 AM
Subject: Re: [asterisk-users] Newbie in Cisco Phone
Well does it matter if the asterisk server is not located in the same
network?
I am willing to spend a bit of cash to get someone help me
- Non-Commercial Discussion
Cc: tam...@gmail.com
Subject: Re: [asterisk-users] Newbie in Cisco Phone
you can try chan_sccp at www.chan-sccp.org
it supports most of ccm features and all kind of cisco phones with skinny
firmware.
Take a look ;)
If you need support you can write me back.
Federico
The 7936G/ 7937G Data Sheet says SCCP only which is a shame. It really
is a great sounding phone. I have several customers with them as SCCP.
http://www.cisco.com/en/US/prod/collateral/voicesw/ps6788/phones/ps379/p
s8759/product_data_sheet0900aecd806e021a.html
From:
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Newbie in Cisco Phone
The 7936G/ 7937G Data Sheet says SCCP only which is a shame. It really is a
great sounding phone. I have several customers with them as SCCP.
http://www.cisco.com/en/US/prod/collateral/voicesw
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jason Aarons
(US)
Sent: Friday, January 23, 2009 5:10 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Newbie
List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Newbie in Cisco Phone
Asterisk's Skinny support is very rudimentary and doesn't include the
CCM provisioning stuff.
Short answer - not really. Not unless you want to go through a *whole*
lot of work.
Sam Tam wrote:
Yes I know too
the phone.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jason Aarons
(US)
Sent: Friday, January 23, 2009 5:10 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michiel van
Baak
Sent: Friday, January 23, 2009 3:35 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Newbie in Cisco Phone
On 05:39, Fri 23 Jan 09, Sam Tam wrote:
Yes I know too.
Is there anyway to make it work with asterisk
Lee, John (Sydney) wrote:
Calling all Polycom gurus:
I am using Polycom IP601 phones with Asterisk 1.4.21.2
In all Polycom phones, I set the following in sip.cfg.
dialplan dialplan.impossibleMatchHandling=2
/dialplan
(I leave the digitmap unchanged because I thought setting
on the console.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Steve Murphy
Sent: Thursday, 11 September 2008 2:13 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Newbie AEL2: Syntax for Hint
On Thu, 2008-09-11 at 17:41 +1000, Lee, John (Sydney) wrote:
Steve, I downloaded the latest Asterisk version (see below).
*CLI core show version
Asterisk 1.4.21.2 built by root @ machine1 on a i686 running Linux on
2008-09-11 06:10:06 UTC
If I code:
Hint(Custom:light1)
It will pass
context BLF {
hint(Sip/1000) 1000 = NoOp();
};
Works for me
Thanks Eric.
I did not experience any problem in hint with SIP. The problem is if you use
it with Custom.
winmail.dat___
-- Bandwidth and Colocation Provided by
Lee, John (Sydney) wrote:
I am struggling to find out how to code hint in AEL2.
I did hint(Custom:light1) and it keeps complaining about the : (colon).
It works fine for SIP device like hint(SIP/439).
Anyone who has tried it before?
I just whipped this up to test and it works for me in
On Wed, 2008-09-10 at 18:10 +1000, Lee, John (Sydney) wrote:
I am struggling to find out how to code hint in AEL2.
I did hint(Custom:light1) and it keeps complaining about the : (colon).
It works fine for SIP device like hint(SIP/439).
Anyone who has tried it before?
Yes, a while back I
: Thursday, 11 September 2008 2:13 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Newbie AEL2: Syntax for Hint
On Wed, 2008-09-10 at 18:10 +1000, Lee, John (Sydney) wrote:
I am struggling to find out how to code hint in AEL2.
I did hint(Custom:light1
I played with the Polycom login/logout function about a year ago, and it
looked brilliant.
I could never get it to work, but at the time I had both Polycom and
Digium agree that it would be worth getting running.
I ran out of time on that project, and have never re-visited it. But it
would
?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Faraz Khan
Sent: August 13, 2008 8:20 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Newbie: Queue and CDR Reporter and Analyser
queuemetrics
Lee, John
Doesn't Queuemetrics run on a license basis?
Anything else that's probably open source and free?
Does anyone have any comments/experience about using asteriskguru queue
statistics?
http://www.asteriskguru.com/tutorials/installation_guide.html
___
One of the Asterisk people down here in Melb set it up for the company
they used to work for, and I played with it once and it seemed to be usable.
PaulH
Lee, John (Sydney) wrote:
Doesn't Queuemetrics run on a license basis?
Anything else that's probably open source and free?
Does
There's actually a document included with the source code which will
take you through setting up an agent callback system. You can find it
in 'doc/queues-with-callback-members.txt'.
The 'AgentCallBackLogin' application has some issues, and since you
can
do the same thing with your
I am really grateful to all the experts on the mailing list who gave me
some very good advice on this problem which I experienced in China. I
think we have fixed the problem and the card is no longer reporting any
problems. We are able to dial out successfully and we will continue to
test.
Here
Excelent!!
but may be better if you send to the list the zaptel.conf and zapata.conf
Regards,
Luis Morales
On Thu, Aug 21, 2008 at 10:19 PM, Lee, John (Sydney)
[EMAIL PROTECTED] wrote:
I am really grateful to all the experts on the mailing list who gave me
some very good advice on this
queuemetrics
Lee, John (Sydney) wrote:
I am trying to look for a software (open source or proprietory) that
could do reporting on both queue and CDR in Asterisk 1.4.*
Could someone give me some suggestions?
: Re: [asterisk-users] Newbie: Queue and CDR Reporter and Analyser
queuemetrics
Lee, John (Sydney) wrote:
I am trying to look for a software (open source or proprietory) that
could do reporting on both queue and CDR in Asterisk 1.4.*
Could someone give me some suggestions
On 7/31/08, Jay R. Ashworth [EMAIL PROTECTED] wrote:
On Thu, Jul 31, 2008 at 05:36:14PM +1000, Lee, John (Sydney) wrote:
Yes, I tried all sorts of cables and ended up getting the local contact
to complain to NETCOM. An engineer came and swapped the Fast Ethernet
to E1 converter.
Hmmm.
if after you tried both straight through crossover cables and
it still give you RED alarm. just tell them you can't get any
clocking signal. they'll probably send someone on site and test
the line.
Yes, I tried all sorts of cables and ended up getting the local contact
to complain to NETCOM.
On Thu, Jul 31, 2008 at 12:31 PM, Uros Djokic [EMAIL PROTECTED] wrote:
Hi,
Ensure that in file indications.conf you have
[general]
contry=cn ; not usa ! or if you are in Australia shortcut for Australia
Regards,
Uros
--
Use Free Software http://www.fsf.org/
Hi,
Ensure that in file indications.conf you have
[general]
contry=cn ; not usa !
Regards,
Uros
--
Use Free Software http://www.fsf.org/
---
Four essential software freedoms:
1) To study source code
2) To copy program
3) To modify source code
4) To
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Lee, John (Sydney)
Sent: Thursday, July 31, 2008 3:36 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Newbie in China: Red alaram in
Zaptel for E1
Sounds like you're making progress. I would try the above span
definition without the crc4. That might do the trick.
Thanks Brad.
I already tried it without crc4 but it makes no difference.
___
-- Bandwidth and Colocation Provided by
Ensure that in file indications.conf you have
[general]
country=cn ; not usa ! or if you are in Australia shortcut for Australia
Uros, that was a good reminder. However, I don't think it is related to this
problem.
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-- Bandwidth and
Make experiment.Make loopback Rj-45. (wire 1 from pin 1 to pin 4 wire 2 from
pin 2 to pin 5). Then put it in card and if card is OK you should see green
led.You should also see dozens of ALARMS notices or warnings on asterisk
CLI.
Also check pinout http://www.goonda.org/archive/docs/pinout.html
On Thu, Jul 31, 2008 at 05:36:14PM +1000, Lee, John (Sydney) wrote:
Yes, I tried all sorts of cables and ended up getting the local contact
to complain to NETCOM. An engineer came and swapped the Fast Ethernet
to E1 converter.
Hmmm.
Whose side is Fast Ethernet, and whose side is E1?
Are you
Dan Austin wrote:
John wrote:
Thanks Steve for your suggestions.
In China you will generally get either MFC/R2 or EuroISDN. MFC/R2 is
much more common.
This is exactly my current problem.
NETCOM in Shanghai just told my local contact it is an E1 and that's it.
I
You don't need to install it. Just run kernel/xpp/utils/genzaptelconf
directly from the source directory.
Thanks Tzafrir.
My local contact is away today and so I could not get him to plug the
line to port 4. So, it is still in port 1.
Here is the output after running genzaptelconf.
#
emist wrote:
My best guess from looking at that is that its a driver bug. The last
thing that happens before the lockup seems to be an ioctl call to the
device.
That was a bug that should have been resolved by 1.4.11 (he subsequently
updated and it was resolved).
Matthew Fredrickson
Lee, John (Sydney) wrote:
The test for that is simple:
head -n 1 /proc/zaptel/*
Let's look at all four spans. Not just the first one.
Thanks Tzafrir.
# head -n 1 /proc/zaptel/*
== /proc/zaptel/1 ==
Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 (MASTER) HDB3/ RED
== /proc/zaptel/2
There's actually a document included with the source code which will
take you through setting up an agent callback system. You can find it
in 'doc/queues-with-callback-members.txt'.
The 'AgentCallBackLogin' application has some issues, and since you can
do the same thing with your dialplan,
Lee, John (Sydney) wrote:
i've installed several Asterisk systems in Shanghai Beijing.
Thanks Edwin.
The remote site is in Shanghai and NETCOM is the telco.
Do you know if their E1 line is MFC/R2 or EuroISDN?
i'm not sure if they provide MFC/R2. but we always
ordered PRI from them. as far
Lee, John (Sydney) wrote:
I am trying to build a simple queue with several agents using
AgentCallBackLogin.
From what I read on the Internet and tried briefly, it seems to suggest that
I should be coding my own queue system for AgentCallBackLogin using AEL2
instead of using the
I think it can't hurt to try a different release. Let me know how it
goes.
Thanks Igor.
I just upgraded zaptel to 1.4.11.
However, I am still seeing red in the alarm in zttool and the LED on
port 1 also shows red.
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