http://www.voip-info.org/tiki-index.php?page=Asterisk+hardware+mini-itx
New Wiki page on mini-itx with Leo Anne's comment added to it.
Please help us collect experiences on Mini-itx configurations there.
Seems to be a lot of interest in this hardware.
/Olle
Michael Graves wrote:
Please forgive me if this is a silly question. I've been following this
thread in the hope that I could put my * server and snom 200 into
full-time service very soon. I need to find out how to have the lines
Olle Wrote:
I've opened a bug
Good day,
I want to have Asterisk as my gateway to the outside world and use
another PBX to connect my existing phones.
How do I specify the dial string to forward the original Caller ID to
over the ISDN to the second PBX?
Right now, my extensions.conf looks like this:
exten =
Let me start by saying no, this is not the normal stupid echo question :)
I currently have a SIP device and an X101P, and have had the usual echo
issues and have played around with the various solutions, none of which
are quite perfect (understandably).. however, my girlfriend is a little
more
Does anyone know what the hardware requirements would be to build an Enterprise Asterisk Universal Gateway ? I am thinking of something comprable to the Cisco AS5xxx Series of gateways.
Thanks Expand your wine savvy and get some great new recipes at MSN Wine.
Happy New Year,
I have a project to pass modem calls through * convert
them from IP to X.25 and then allow the modems at each
end to talk thru the rtp stream to each other before
calling modem terminates the call.
Datamodem --- FXS(non *) ADSL --- *
-Software Application X.25
I agree in stopping the thread, but I do have one question... What would
Qwest think of her posting to the list under a yahoo mail account
representing her company, badmouthing this community, who, in the long
run, could be VERY much worth their interest?
Hmm Just my $.02 - no flames please.
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hello there,
I'm drawing up a scheme to manage our company calls and would like to
implement it with Asterisk. In order to get moving quickly I'd like some
recommendations on what hardware to buy so I can start tinkering. Initially
we'd like to
I agree in stopping the thread, but I do have one question... What would
Qwest think of her posting to the list under a yahoo mail account
representing her company, badmouthing this community, who, in the long
run, could be VERY much worth their interest?
Hmm Just my $.02 - no flames please.
Check http://www.telappliant.com for their VoIP Starter kits or Telephony
Cards sections.
Robert
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hello there,
.
for pointing me at a friendly/knowledgeable UK supplier of such cards.
Any advice would be greatly appreciated: once I have
Matthew Bloch wrote:
Hello there,
I'm drawing up a scheme to manage our company calls and would like to
implement it with Asterisk. In order to get moving quickly I'd like some
recommendations on what hardware to buy so I can start tinkering. Initially
I seriously doubt this is actually a Quest employee. Probably just someone
trying to mess with a boss or something.
- Original Message -
From: Brian Capouch [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, January 03, 2004 8:36 PM
Subject: Re: [Asterisk-Users] New to asterisk?
Mike,
The v2.03f code (alpha/beta?) did correct the multi-line problems very
nicely, however I think the snom folks might have another tweak or two
to make to this code. If your snom 200 is running in a business
production environment, you might want to wait a little. If you're using
it in a
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Sunday 04 January 2004 12:46, rnc Info Lists wrote:
Check http://www.telappliant.com for their VoIP Starter kits or Telephony
Cards sections.
Thanks for the pointer Robert (and from Olle too). The X100P sounds like a
good deal for £60 and
I agree in stopping the thread, but I do have one question... What would
Qwest think of her posting to the list under a yahoo mail account
representing her company, badmouthing this community, who, in the long
run, could be VERY much worth their interest?
Come on guys, drop it!!! There isn't
On Sun, 2004-01-04 at 04:35, EDWARD WILSON wrote:
Does anyone know what the hardware requirements would be to build an
Enterprise Asterisk Universal Gateway ? I am thinking of something
comprable to the Cisco AS5xxx Series of gateways.
Just to prepare you, if you ask the above question, you
On Sun, 2004-01-04 at 05:33, fred alexander wrote:
Happy New Year,
I have a project to pass modem calls through * convert
them from IP to X.25 and then allow the modems at each
end to talk thru the rtp stream to each other before
calling modem terminates the call.
Datamodem --- FXS(non
I would set the Enterprise Class bar at five 9's reliability
(about 5.25 minutes per year of down time) the same
as a Class 4/5 phone switch. This would require redundant
design considerations in both hardware and software.
In our network, Linux is approaching
Enterprise Class and I don't see why
On Sun, 2004-01-04 at 08:39, Matthew Bloch wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Sunday 04 January 2004 12:46, rnc Info Lists wrote:
Check http://www.telappliant.com for their VoIP Starter kits or Telephony
Cards sections.
Thanks for the pointer Robert (and from Olle
I would set the Enterprise Class bar at five 9's reliability
(about 5.25 minutes per year of down time) the same
as a Class 4/5 phone switch. This would require redundant
design considerations in both hardware and software.
My Norstar Meridian system has nowhere near this. We get about 5
Doug Shubert wrote:
I would set the Enterprise Class bar at five 9's reliability
(about 5.25 minutes per year of down time) the same
as a Class 4/5 phone switch. This would require redundant
design considerations in both hardware and software.
In our network, Linux is approaching
Enterprise Class
Rich,
What form of config is required in * to get 1 extensions available.
Single login/registry or multiple? Do I have to specify lines per
Christiasn's earlier mail?
Thanks,
Michael
On Sun, 4 Jan 2004 07:58:21 -0600, Rich Adamson wrote:
Mike,
The v2.03f code (alpha/beta?) did correct the
Mike Jagdis wrote:
John Coll wrote:
Dave
You were right
disallow=all
allow=ulaw
allow=alaw
gave me two-way voice! Whew! Thanks a million. I wonder if I really
should
have found that for myself ... I've added it to the voip-info.org wiki
OK lets see if the next step is a bit easier :)
Hi!
Apache + Mysql + PHP ( Ming + Actionscript ) + Asterisk is good.
Dynamic effective,Easy coding and Fast response :-)
That's an excellent suggestion, I agree with Ray. Masakazu, do you think
you could provide a working sample either here on the list or in the
Wiki?
yeah.
Steven Critchfield wrote:
Just to prepare you, if you ask the above question, you are not ready to
ask the above question.
Quote added to
http://www.voip-info.org/tiki-index.php?page=Asterisk+quotes
/O ;-)
___
Asterisk-Users mailing list
[EMAIL
There was a post in the wiki for an
application to provide an outcall when there is a voicemail is left on
asterisk. I am having a problem
that this application will only work if the caller presses the pound sign at
the end of recording. As most people just hang up, this application
isnt
Andrew Kohlsmith wrote:
I would set the Enterprise Class bar at five 9's reliability
(about 5.25 minutes per year of down time) the same
as a Class 4/5 phone switch. This would require redundant
design considerations in both hardware and software.
To turn around, let's discuss what we need to
WipeOut wrote:
Asterisk would need some kind of clustering/load balancing ability
(Single IP system image for the IP phones across multiple servers) to be
truely Enterprise Class in terms of both reliability and
scaleability.. Obviously that would not be as relevent for the analog
hard
Hi,
I have two E100P boards connected to my PC. I wish to setup two
E1-ISDN-PRI lines, the first SPAN with 15 channels and the second one
with 30 channels. When I try to load zaptel modules, I get an error message:
Loading zaptel framework:
Loading zaptel hardware modules: wct1xxp wcusb
I would set the Enterprise Class bar at five 9's reliability
(about 5.25 minutes per year of down time) the same
as a Class 4/5 phone switch. This would require redundant
design considerations in both hardware and software.
In our network, Linux is approaching
Enterprise Class and I don't
Hi!
I want to have Asterisk as my gateway to the outside world and use
another PBX to connect my existing phones.
exten = ${OUTSIDEMSN},1,Dial,CAPI/${MSN2NDPBX}:${EXTEN}
How do I transfer the caller Id information initially coming in?
I have strong doubts that this can be done at all.
It is a problem - but the call recording is saved by * when you hang up.
So you need to look for new files in whichever directory the call
recordings are saved and pick them up eg with a script.
Iain
--On Sunday, January 04, 2004 12:07:35 -0500 Kevin [EMAIL PROTECTED]
wrote:
There was a
Yes, I think it should be:
span=1,0,0,ccs,hdb3
bchan=1-15
dchan=16
span=2,0,0,ccs,hdb3
bchan=32-46,48-62
dchan=47
Cheers,
Scott M. Stingel
Emerging Voice Technology Inc.
Email: [EMAIL PROTECTED]
URL:www.evtmedia.com
-Original Message-
From: [EMAIL
Does some kind Asterisk soul have an example from extensions.conf that shows
how to record both sides of a conversation?
Thanks!
Paul Mahler
mail:[EMAIL PROTECTED]
phone: 650.207.9855
fax: 877.408.0105
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Hello,
anyone from northern germany planning to go to
http://www.guug.de/veranstaltungen/telephony-summit-2004/
If yes, could you contact me off list. Maybe we can save some money by
car-pooling?!
--
Best regards
Peer Oliver Schmidt
the internet company
On Sun, 2004-01-04 at 11:07, Olle E. Johansson wrote:
Steven Critchfield wrote:
Just to prepare you, if you ask the above question, you are not ready to
ask the above question.
Quote added to
http://www.voip-info.org/tiki-index.php?page=Asterisk+quotes
While funny, it makes at least
Philipp von Klitzing worte:
I want to have Asterisk as my gateway to the outside world and use
another PBX to connect my existing phones.
exten = ${OUTSIDEMSN},1,Dial,CAPI/${MSN2NDPBX}:${EXTEN}
How do I transfer the caller Id information initially coming in?
I have strong doubts that this can
Paul,
you broke the thread! Please create your own top posting - or better,
search the list archive!
Philipp
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Mike,
What form of config is required in * to get 1 extensions available.
Single login/registry or multiple? Do I have to specify lines per
Christiasn's earlier mail?
In my implementation, the two extns are treated as though they are
separate phones with separate (independent) logins, like:
On Sun, 2004-01-04 at 10:14, Doug Shubert wrote:
I would set the Enterprise Class bar at five 9's reliability
(about 5.25 minutes per year of down time) the same
as a Class 4/5 phone switch. This would require redundant
design considerations in both hardware and software.
In our network,
There was a post in the wiki for an application to provide an outcall when there
is a voicemail is left on asterisk. I am
having a problem that this application will only work if the caller presses the
pound sign at the end of recording. As most
people just hang up, this application
WipeOut wrote:
Asterisk would need some kind of clustering/load balancing ability
(Single IP system image for the IP phones across multiple servers) to
be truely Enterprise Class in terms of both reliability and
scaleability.. Obviously that would not be as relevent for the analog
hard
Thanks for the info. I would like to go.
Is it in German or English?
I only speak English.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Peer Oliver
schmidt
Sent: 04 January 2004 18:10
To: Asterisk User List
Subject: [Asterisk-Users] OT: Anyone going
On Sun, 2004-01-04 at 12:53, Nick Bachmann wrote:
WipeOut wrote:
Asterisk would need some kind of clustering/load balancing ability
(Single IP system image for the IP phones across multiple servers) to
be truely Enterprise Class in terms of both reliability and
scaleability..
Andrew Kohlsmith wrote:
I would set the Enterprise Class bar at five 9's reliability
(about 5.25 minutes per year of down time) the same
as a Class 4/5 phone switch. This would require redundant
design considerations in both hardware and software.
To turn around, let's discuss what we
Nick Bachmann wrote:
Yes, I've played with it a bit. It's pretty simplistic... the clustering
just keeps several servers in sync with each other. I suppose that would
be easy to do with Asterisk, especially if configuration data was stored
in a RDBMS that could do replication. Even now,
Steven Critchfield wrote:
On Sun, 2004-01-04 at 10:14, Doug Shubert wrote:
I would set the Enterprise Class bar at five 9's reliability
(about 5.25 minutes per year of down time) the same
as a Class 4/5 phone switch. This would require redundant
design considerations in both hardware and
Rich Adamson wrote:
Andrew Kohlsmith wrote:
I would set the Enterprise Class bar at five 9's reliability
(about 5.25 minutes per year of down time) the same
as a Class 4/5 phone switch. This would require redundant
design considerations in both hardware and software.
To turn around,
On Sun, 2004-01-04 at 12:53, Nick Bachmann wrote:
Yes, I've played with it a bit. It's pretty simplistic... the
clustering just keeps several servers in sync with each other. I
suppose that would be easy to do with Asterisk, especially if
configuration data was stored in a RDBMS that could
Craig Waddington wrote:
anyone from northern germany planning to go to
http://www.guug.de/veranstaltungen/telephony-summit-2004/
Thanks for the info. I would like to go.
Is it in German or English?
According to the site mostly english.
rgds
pos
___
Andrew Kohlsmith wrote:
I would set the Enterprise Class bar at five 9's reliability
(about 5.25 minutes per year of down time) the same
as a Class 4/5 phone switch. This would require redundant
design considerations in both hardware and software.
To turn around, let's discuss what we need to
Iain -
First off, all of this is heavily borrowed from others. For those who see
their code embedded here, I thank you and give you full credit.
Here's how I do it. It's a bit convoluted, but I didn't want to record
everything. So, if a call comes in and I want to record it, I send it here:
On Sun, 2004-01-04 at 13:28, WipeOut wrote:
Steven Critchfield wrote:
On Sun, 2004-01-04 at 10:14, Doug Shubert wrote:
I would set the Enterprise Class bar at five 9's reliability
(about 5.25 minutes per year of down time) the same
as a Class 4/5 phone switch. This would require
Perhaps I was being somewhat ambitious by posting five 9's for
Enterprise Class
using a three tiered approach,
five 9's (5.25 min. per year) for Carrier Class
four 9's (52.56 min. per year) for Enterprise Class
three 9's (8.76 hrs. per year) for User/SOHO Class
each Class having specific
Andrew Kohlsmith wrote:
I would set the Enterprise Class bar at five 9's reliability
(about 5.25 minutes per year of down time) the same
as a Class 4/5 phone switch. This would require redundant
design considerations in both hardware and software.
To turn around, let's discuss what we need to
you nolonger need set-timestamp.agi as we have ${TIMESTAMP} in that format
by default now.
bkw
On Sun, 4 Jan 2004, John Baker wrote:
Iain -
First off, all of this is heavily borrowed from others. For those who see
their code embedded here, I thank you and give you full credit.
Here's how
This is my favourite response to this post RUN... don't walk..
WELL SAID!
John Haigh
- Original Message -
From: asterisk [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, December 31, 2003 4:24 PM
Subject: Re: [Asterisk-Users] New to asterisk? RUN... don't walk.
Here's the
Dave
I note your suggestion you probably also want to disable gsm on the GS
phones themselves (just change the 723 entry in the list on the admin page
to a repeat of a 711
My GS phone has the following codec options: PCMU, PCMA, G.723.1, G729A/B,
G726-32, G728.
Half an hour's research and
Thanks to Dave I now have two Grandstream phones with a voice path. Yippee!
Wanting to learn from the experience I compared the sip debugs from before
and after adding the disallow=all, allow=ulaw, allow=alaw lines to sip.conf
to see what I should have noticed in the debug that would have pointed
With help I got two phones communicating - PCMA/PCMU was the problem.
Next stpe is to try voicemail. VM works fine, I can leave a mesage and then
retrieve it - but no MWI on the phone and no stutter dialtone.
I promise I've spent the requisite 4 hours + on google etc. but have really
no further
Sorry for the partial post a moment ago
With help I got two phones communicating - PCMA/PCMU was the problem.
Next stpe is to try voicemail. VM works fine, I can leave a mesage and then
retrieve it - but no MWI on the phone and no stutter dialtone.
I promise I've spent the requisite 4 hours +
Hi!
The affinity table makes the RTP stuff OK, but I agree that sharing
SIP registrations is a concern.
These are stored in the Asterisk DB. Type this at your CLI:
database show SIP/Registry
Consequently it shouldn't be a a problem to sync the registry data.
Cheers, Philipp
John Coll wrote:
With help I got two phones communicating - PCMA/PCMU was the problem.
Next stpe is to try voicemail. VM works fine, I can leave a mesage and then
retrieve it - but no MWI on the phone and no stutter dialtone.
I promise I've spent the requisite 4 hours + on google etc. but have
I'd guess part of the five-9's discussion centers around how automated
must one be to be able to actually get close? If one assumes the loss
of a SIMM the answer/effort certainly is different then assuming the
loss of a single interface card (when multiples exist), etc.
I would doubt
- Original Message -
From: John Coll [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, January 05, 2004 9:07 AM
Subject: [Asterisk-Users] Newbie - MWI
Sorry for the partial post a moment ago
With help I got two phones communicating - PCMA/PCMU was the problem.
Next stpe is to
Where / how do I set DTMF payload type to 101?
- Original Message -
From: Josh Roberson [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, January 01, 2004 3:17 PM
Subject: RE: [Asterisk-Users] Re: Grandstream Early Dial
I've never had early dial working, however, I resolved my
1. Moving a physical interface (whether a T1, ethernet or 2-wire pstn) is
mostly trevial, however what signal is needed to detect a system failure
and move the physical connection to a second machine/interface? (If there
are three systems in a cluster, what signal is needed? If a three-way
does anyone have a running cisco12sp+ or 30
vip phone on their network?
and if so could you also tell me what tftp files
you actually use and if there are any special settings in skinny.conf that i
need?
(I ran several searches for setup, nothing has come
up so far so i'll ask for advice
I just signed up for Voicepulse with a DID. I
can register with Voicepulse and dialout just fine. Only problem is that
when I dial my DID from my POTS line I just get a fast busy and nothing in the
console.
Any ideas?
Thanks Paul very much!
john
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Paul Liew
Sent: 04 January 2004 22:32
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Newbie - MWI
- Original Message -
From: John Coll [EMAIL PROTECTED]
To: [EMAIL
lots of snips
-
;
; liza:/etc/asterisk/sip.conf
;
[general]
port = 5060
bindaddr = 0.0.0.0
externip = 10.0.1.198
[5702]
type=friend
host=dynamic
context=johnhome
reinvite=no
canreinvite=no
qualify=300
I am looking for common practice ideas on how to handle multiple line
phones. Is it common with asterisk to have the lines appear as
programmable buttons? Or to just have itcm like buttons and use the dial
9 approach? What I am specifically interested in, is to have my line
one appear on the
There are no guarantees that the voicemail will be in the same context
as the extension. By giving you the ability and flexibility of defining
everything independently, there's not much you can't do! Remember, the
context call in the sip.conf refers to the context in extensions.conf.
the
Does anyone know of a piece of hardware that can allow multiple earpices
to be connected directly to a server running Asterisk.
I hope I am not being to vague but basically I am looking to allow a
call center to user the server to do all of the Pickup and Hangup
functions.
The operators will
I am just starting to deploy asterisk in our office to use as our primary
phone system - we plan to use a Voicetronix OpenLine4 card as our PSTN
gateway - but one thing at a time... haven't got that far yet. Currently,
i'm trying simple IP to IP calls within the office using our Cisco 7960's
hi,
since development of dtmf caller id under * is prolly going to only be done if someone
stumps up the cash I've been looking for alternatives... Hoving found a number of
projects which turn out to be mad prototypes or unavailable details i nearly gave up..
then I found this:
what firmware are you using? is it SIP?
to check, push settings then status and firmware
you should have a load ID like this 'POS3-04-4-00'
also check the preferred CODEC
we use g711ulaw as the default
Terence Parker wrote:
I am just starting to deploy asterisk in our office to use as our
On Sun, 2004-01-04 at 17:45, Terence Parker wrote:
When I make a call between these two phones, the conversation is of a
quality so bad that it is barely audible (5% makes sense).
You must be doing something wrong (maybe codec problems), because I've
had absolutely no problems with Cisco to
Thanks for the replies.
My cisco firmware is only POS3-04-2-00, though it is SIP. It used to work
fine under vocal though - which was strange. Is this definitely nothing to
do with asterisk? I do note however that my firmware is fairly old... except
cisco aren't exactly generous with firmware
On Sun, 2004-01-04 at 17:04, Rohde wrote:
does anyone have a running cisco 12sp+ or 30 vip phone on their
network?
and if so could you also tell me what tftp files you actually use and
if there are any special settings in skinny.conf that i need?
(I ran several searches for setup, nothing has
On Sun, 2004-01-04 at 18:18, Sean Garland wrote:
I am looking for common practice ideas on how to handle multiple line
phones. Is it common with asterisk to have the lines appear as
programmable buttons? Or to just have itcm like buttons and use the dial
9 approach? What I am specifically
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Terence Parker
Sent: Sunday, January 04, 2004 8:29 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Cisco to Cisco - poor quality
Thanks for the replies.
My cisco firmware is only
see if you can upgrade to firmware 4-3 or 4-4
another point to note, are you using a full duplex 10/100 switch?
if so, you should have 'Port1 Full 100' for full duplex 100Mbit
under the 'Network Statistics'
If you like to email me your config settings, I will check them against our
phones.
Since the list community has done so much for me in my humble asterisk
beginnings I have put together a simple little script written in php that
serves as a paging reminder script. If anyone is interested in a copy of
it contact me off list and I'll forward you a copy.
The basics of the
On the config webpage, its on the bottom.
Kevin
Original Message
Subject: Re: [Asterisk-Users] Re: Grandstream Early Dial
From: Aaron Martin [EMAIL PROTECTED]
Date: Sun, January 04, 2004 3:49 pm
To: [EMAIL PROTECTED]
Where / how do I set DTMF payload type to 101?
Hi,
I'm just considering buying two Telecoms grade Sun Netra's to run a
lab-based VoIP solution. Not my immediate thoughts as a VoIP platform,
but from what I've heard, they can run Linux, and run it well.
Only thing is:
The Wiki and the Whitepaper just state that Asterisk is for the x86
Forgot to add, the Sun Netra's UltraSparc is 64bit... However, various
pointers indicate that it can run both 32bit and 64bit compiled code.
Best,
Ad.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Has anyone had any success using more than one or two X100P cards?
I have 4 in a system, and channels 2 3 and 4 all seem to work just fine.
Channel 1 however is acting up. I get random red alarms, disconnects,
etc.
I have checked the /proc/interrupts and everything is sitting on it's
own IRQ.
I got a Cisco 7960 phone recently, and have downloaded and set up
Asterisk version 0.5.0. Very nice!
I've set up the software on a test box for now and have configured
the system to route calls that start with 7 to FWD. Once I'm happy
with my various tests, I will set this all up on a dedicated
Reposted because my original
post was mangled by a nasty webmail client. I would really like some help with
this one if anyone has any ideas.
--
I am having a problem
interacting with a remote IVR system when the outbound call is going via SIP.
The only way that I have been able to
On Sunday 04 January 2004 20:25, Adthrawn wrote:
The Wiki and the Whitepaper just state that Asterisk is for the x86
architecture, but has been compiled to run on PPC architectures. No
mention of UltraSparc. If I can get it compiled, what would I be
loosing in terms of functions or what
On Sun, 2004-01-04 at 18:18, Sean Garland wrote:
I am looking for common practice ideas on how to handle multiple line
phones. Is it common with asterisk to have the lines appear as
programmable buttons? Or to just have itcm like buttons and use the dial
9 approach? What I am
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Tilghman Lesher
Sent: Sunday, January 04, 2004 9:58 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Sun Servers with UltraSparc Processors
[...]
You shouldn't face any problems with
The comments below are certainly not intended as any form of negativism,
but rather to pursue thought processes for redundant systems.
1. Moving a physical interface (whether a T1, ethernet or 2-wire pstn) is
mostly trivial, however what signal is needed to detect a system failure
and move
On Sun, 2004-01-04 at 20:42, Rich Adamson wrote:
On Sun, 2004-01-04 at 18:18, Sean Garland wrote:
I am looking for common practice ideas on how to handle multiple line
phones. Is it common with asterisk to have the lines appear as
programmable buttons? Or to just have itcm like buttons
I am trying to get the handytone 286 to make a very simple call to * and
having problems. It registers with * just fine, but when I place a call
(to echo test, for example), the RTP stream seems to have problems
opening. Here is there error I get in *:
WARNING[98311]: File chan_sip.c, Line 464
Could you explain in a little more detail about what you are trying to do
with the multi-lines? Maybe a more in depth example would help.
In my (limited) experience, I have seen two types of multi-line uses
1. The phone has a number of lines (usually) two. If the first line is busy,
the call
I had asterisk running on SuSe 7.3 on an Ultra 2 back in May 2003. Had to
make some changes to Makefiles. At that time, SuSe had no more updates planned
for Linux on Sparc. I had MGCP andSIP running very well on it. Had some
trouble with H323. Timing was another issue as the Ultra 2 is an
I seem to recall that you are only sending calls from Asterisk to the
Cisco, not sending calls from the Cisco to Asterisk. Is this correct?
On Sun, 2004-01-04 at 19:10, Jared Smith wrote:
On Sun, 2004-01-04 at 17:45, Terence Parker wrote:
When I make a call between these two phones, the
Hi Mike
I know exacty same situation about BT100 that sometimes lost any packets.
like a DoS attack for BT100? ;-(
mack_jpn
[EMAIL PROTECTED] asterisk]# ping 192.168.XX.XX
PING 192.168.XX.XX (192.168.XX.XX) from 192.168.XX.X : 56(84) bytes of
data.
64 bytes from 192.168.XX.XX: icmp_seq=0
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