Hi Mike,
I have handytones working OK with *.
My username and name of the context set to same.
Try this;
[131]
username=131
type=friend
host=dynamic
disallow=all
allow=alaw
allow=ulaw
However I am wondering why you get destination unreacherbale from the
handytone. This is nothing to do with SI
> I had documented the Makefile modification in an email to the list. If you
> search for Sparc in the mailing list, you should be able to find it. If
> not, drop me a line and I'll see if I still have it.
>
I've got an Ultra 30 sitting here doing nothing. I'll see what I can come
up with for Li
I guess that was another thing that was strange. When I talked, I saw no
RTP coming from the handytone to *. Would there be a reason the
handytone would not send RTP until it successfully received a RTP packet
from *, but since its not accepting RTP, it would not send it either?
I do not even get
On Sun, 2004-01-04 at 21:23, Rich Adamson wrote:
> Part of the point of many of the questions is that there really are a
> lot of dependencies on devices other then asterisk, and simply going down
> a path that says clustering (or whichever approach) can handle something
> is probably ignoring seve
I had a similar problem with a Cisco phone, i.e., the "Maximum retries
exceeded on call" error.
It took three days to track down the error to buggy network hardware.
Same symptoms, too - phone registered, one way conversation was ok (had a
test extension
for music on hold)
Fixed the hardware, pho
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 ttl=2
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 con
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 and SIP running very well on it. Had some
trouble with H323. Timing was another issue as the Ultra 2 is an S
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 ring
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 (re
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 li
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
> > an
> -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 problem
> 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 sp
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 pro
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 ge
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
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. A
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.
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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
arch
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 1
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 script
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.
telnet
> -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
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
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
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 upgr
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 C
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 prima
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:
http://www.arte
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
phones
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 me
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 "johnhom
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 firs
> > -
> > ;
> > ; 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
> >
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: <[EMA
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?
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 come
up so far so i'll ask for advice no
> 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-wa
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 resolve
- 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 s
> >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 woul
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 re
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
__
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 + o
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
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 pointe
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 read
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.
>
> H
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'
>>> 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 arou
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
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 w
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:
[
>> 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'
> 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
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
___
As
> 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 tha
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
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 soft
Craig Waddington wrote:
Thanks for the info. I would like to go.
Is it in German or English?
I only speak English.
Me too.. :)
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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, setting
> 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
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
> >>
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
> 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 discus
> 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
* always records both sides of the conversation - but stores them in
separate files in
/var/spool/asterisk/monitor/. You need to combine the "in" and "out" parts
using soxmix.
Iain
--On Sunday, January 4, 2004 9:59 am -0800 Paul Mahler
<[EMAIL PROTECTED]> wrote:
Does some kind Asterisk s
> 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 i
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 netw
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
Paul,
you broke the thread! Please create your own top posting - or better,
search the list archive!
Philipp
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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 b
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
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
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 P
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 PROTECTE
Daniel,
while I am not sure whether or not you can just skip channels (my guess
would be you just skip them in /etc/asterisk/zapata.conf), you don't
want more than a total of 62 channels in your configuration.
Your 2nd span's configuration appears to be off by one channel. I would
try:
span=2,0
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
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
> 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" an
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
Runnin
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 wir
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 fo
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
is
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 ;-)
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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
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 :)
thank
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
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 Cla
> 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
On Sun, 2004-01-04 at 08:39, Matthew Bloch wrote:
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>
> 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
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
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 --- FX
You know what is strange? Both your original email to this list and the
entry in Brian's blog originated from the same IP (24.10.200.168), which
just happens to be a Comcast Cable Internet address in Utah. Can IP's
be spoofed? Sure. I highly doubt it in this case. So before you go
back-strokin
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, y
> 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
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 :)
thanks again all
john
Note
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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 sh
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 test
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