According to the CIA world factbook there are 800 million landlines in
use and about 6.4 billion people. This makes more sense than 800
billion. there are probably at least an equal number of cellular
telephones in use as well, but i have no idea how one would go about
getting those numbers (except
Guys.
How and which tools to use to load test an asterisk install? Say for
example, you need to see how many calls can be routed thru before losing
quality and making the cpu jump to the roof?
___
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I'm getting messages like this:
NOTICE[18552]: Scheduled event in 0
ms?
Has anyone gotten these errors before? I'm
using the get data command to try to receive input, and I'm sometimes getting
this error message.
There are also times when asterisk doesn't
recognize dtmf input and inst
I have been wanting something similar. I paid some money for a busy
detect routine from newman telecom, but it is not yet done.
We'll see what happens.
Greg
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Min Hwan
Chang
Sent: Tuesday, August 09, 2005 6
hello,
can anyone help me? im gettitng this error when i
tried runnin make on solaris 9
rm -f include/asterisk/version.h.tmpmake[1]:
`ast_expr.a' is up to date.make[1]: Leaving directory
`/export/home/fst/chris/cvs/asterisk'gcc -g -o asterisk io.o
sched.o logger.o frame.o loader.o c
On Tue, 9 Aug 2005, Fredrik Lithén wrote:
> Perhaps everything isn't as spiffy as I thought
>
> When running zttool the card still reports as internally clocked
>
> Zaptel.conf:
> # Global data
> span=1,1,0,ccs,hdb3,crc4
> bchan=1-15
> dchan=16
> bchan=17-31
Zttool still shows the card as inter
Thanks to Jean-Michel for the info. re: getting 10 calls over a 1024/256
ADSL using g729... Just the sort of info. I was looking for... Anyone
else?
I hate GSM - sounds horrible to me... but, iLBC sounds pretty good and
I think has a comparible BW to g729...
I have to take issue with some
Dean Collins wrote:
Ronald,
Why? What do you need it for?
For a power point slide.
Would the statistic or the facts are different if I would need it for a
report ? hehehehehe
bye
Ronald
Dean
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL P
On Tue, 9 Aug 2005, Eric Wieling aka ManxPower wrote:
> Panitaxx wrote:
> > yes. overlapdial=yes.
>
> You want it to be no.
What would the reasons to want overlapdial=no on a pstn pri be? Since the
pri will happily signal once the number is complete there should not be
any downside to allowing ov
On Wed, 2005-08-10 at 08:10 +0400, Jean-Michel Hiver wrote:
> I do up to 10 g.729 channels over 1024 / 256 DSL without noticeable
> difference in call statistics (i.e. avg length of calls). If you are
> using ADSL, the maximum bandwith you'll be able to use is your upload
> rate since VoIP calls
What about a PRI/BRI solution
We have a few with the voicetronix openline 4 cards and they work ok
But the PRI solution work better.We have a 4 port sangoma configured
like that.If I recall you set it to pri_net instead of pri_cpe and you
use a cross cable
On Tue, 2005-08-09 at 10:59 -0700, Edwin
Wiley Siler wrote:
Id there a benefit to which protocol I use? When asked, I told them to
set it up as NI2. The PRI is through MCI and will be used for local and
long distance with DIDs and features like CallerID, etc.
Hey Wiley,
I was a little after you. I have about 3 more weeks for mine.
Darren Wright wrote:
---
An ever better way is get some kind of SLA with guaranteed uptime and
bandwith, a symetrical link, and do some traffic shaping to ensure that
VoIP has priority. Part of the point of VoIP is to save money by
collapsing voice and data networks on
Actually, i am making this call from the phone. i can
hear the voice that says "no one is available to take
my call.
This is my SIP log. Thanks for your help.
-- AGI Script fixlocalprefix completed, returning 0
-- Executing SetVar("SIP/200-326c",
"OUTNUM=393393612") in new stack
-- Execu
I am doing traffic shaping with a open source linux firewall
http://www.ipcop.org/
and since i have traffic shaping configured my 3 VoIP lines work great.
I am not using Asterix yet but I will go to as soon as I have the time
to work myself into it.
If anybody can tell me where the best info
---
An ever better way is get some kind of SLA with guaranteed uptime and
bandwith, a symetrical link, and do some traffic shaping to ensure that
VoIP has priority. Part of the point of VoIP is to save money by
collapsing voice and data networks onto one (presumably robu
Dave Redmore wrote:
Hello All,
Wondering what sort of real world mileage people are getting out of
different internet connecions - i.e. different DSL connection speeds,
cable modems, etc... Is it reasonable to hope to carry 10 - 15
concurrent calls on a 768K DSL? I'm not talking about theo
Ronald,
Why? What do you need it for?
Dean
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Ronald_Wiplinger
> Sent: Tuesday, 9 August 2005 10:51 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-U
Hello All,
Wondering what sort of real world mileage people are getting out of
different internet connecions - i.e. different DSL connection speeds,
cable modems, etc... Is it reasonable to hope to carry 10 - 15
concurrent calls on a 768K DSL? I'm not talking about theoretical BW or
looking
> 1. How many phone lines are currently in this world (I estimate 800
> billion analog lines)
Good Guess for service lines, you would need a messurement of length
for all lines.
> 2. How many data lines are currently in this world (I estimate 1 billion)
Same as above, I would go a lot higher on
There is nothing more than figures, right?
I am looking for:
1. How many phone lines are currently in this world (I estimate 800
billion analog lines)
2. How many data lines are currently in this world (I estimate 1 billion)
3. what is the forcast of 1 & 2 ? When will it swap?
4. How many PBX a
Steve,
If I am understanding your situation correctly (i.e. you are using a SIP
client and then forcibly disconnecting/shutting it off during a call)
you may want to look at your sip.conf for a setting called rtptimeout.
This may do exactly what you want.
When on a SIP call, and you disconne
> >How much of an impact can/does local network traffic have on call quality?
> >Would opening large files on local servers affect call quality? We are
> >running QoS on the router but that will only prioritize traffic in/out of
> >the network.
>
> Sure it can. If you have a network segment that
The signal tone that we get when a second call comes in is to loud and
disruptive.
Is there a way in asterisk to change the call waiting signal tone.
If it was just not so loud it would be better.
Malcolm
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If it can pull down some sort of XML file fo rthe directory.. What else can
it "pull down"?
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|BJ Weschke
|Sent: Martes, 09 de Agosto de 2005 07:00 p.m.
|To: Asterisk Users Mailing List - Non-Commercial D
On 8/10/2005, "Irakli Natsvlishvili"
<[EMAIL PROTECTED]> wrote:
> Hello,
>
> I know that using & it is possible to dial several channels.
>
> Question is - is it possible and if yes, how to dial several channels with
> different ringing timeout?
>
> I mean the following - for example when SIP/500
Hello,
I know that using & it is possible to dial several channels.
Question is - is it possible and if yes, how to dial several channels with
different ringing timeout?
I mean the following - for example when SIP/500 is dialed, I want three
phones to be dialed simultaneously - 1000, 2000 and
Steve,
Indeed a kernel oops can happen. I am not going to in detail explain,
but I can tell you that I have seen it happen with dar (Linux backup
software) using zlib. This has been confirmed by the dar community.
Regarding a sheltered life, no I have not such a life. I work with
software sup
An interesting bug….
It may be more wide-spead than this, and there may be other
ways to reproduce it…but this is how I can produce the problem:
At the console, I type “iax2 show peer jc” and
press tab to auto-complete the peer “jcallen” that is usually
registered. But sometimes (p
Jonas Arndt wrote:
Dave,
A segmentation fault is usually caused by the program writing in a
memory area that is not allocated (it could be a result of the
optimizer sometime as well). That means that it can potentially
overwrite code that are executing there. In worst case scenario you
coul
Check out the Asterisk @ Home project. I believe they've done some
similar integration with the XML capabilities for the Cisco phones.
I'm not sure about Polycom.
The XML directory Polycom is talking about is an XML file that it can
pull down from it's "boot server" on boot time from the phone.
TNT supports caller ID
with any softswitch and any protocol.
Regards
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian West
Sent: Tuesday, August 09, 2005
4:20 PM
To: Asterisk
Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Use
Nope I do not see the DID number coming across. Are you positive the
telco has you configured for DID? This would be normal if not.
On Aug 9, 2005, at 5:20 PM, Panitaxx wrote:
Hi,
thanks for your response. here is the log of one call:
Enabled debugging on span 1
Asterisk*CLI>
< Protocol D
It did not work. thanks anyway
On 8/9/05, Eric Wieling aka ManxPower <[EMAIL PROTECTED]> wrote:
> You want it to be no.
>
> Panitaxx wrote:
> > yes. overlapdial=yes.
>
> --
> Eric Wieling * BTEL Consulting * 504-210-3699 x2120
>
> r: Generate a ringing tone for the calling party, passing no aud
The TNT can't pass callerid name as far as I know./bOn Aug 9, 2005, at 5:17 PM, Damon Estep wrote: Anyone out there have success getting caller id name from a pri, through a lucent tnt, to asterisk? What about from other media gateways? ___Asterisk-Users
Hi Jonas,
You may know this already, a codec is an algorithm for compressing and
uncompressing some signal. Often the signal was originally analog, but has
been digitized to reduce size/bandwidth and to store it in files. The GSM
codec is important in asterisk, because most or all music playbac
On Tue, 2005-08-09 at 21:56 +0200, Francesco Peeters wrote:
> Does anybody know whether somebody ever implemented (linux) drivers or
> a chan component for the Com-On-Air cards, or is the only way to make
> it work the use of a Windows box as Dect to SIP gateway for *? (Or
> does it work at all?)
I'm having this problem where if the phone is ringing from
IncomingCall #1, IC#2 will be immediately sent to VM. Is there
somethign wrong with my dial plan? I currently have 4 incoming lines
going into a TDM400 with the group set to g0.
Could it be that the way I've set this up, if any of the p
I just checked again to make sure. I am not seeing anything at all on
gateway on failed calls.
Again 2 out of 5 test calls were failed to reach gateway.
- Original Message -
From: "Paul Belanger" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Tu
You want it to be no.
Panitaxx wrote:
yes. overlapdial=yes.
--
Eric Wieling * BTEL Consulting * 504-210-3699 x2120
r: Generate a ringing tone for the calling party, passing no audio from
the called channel(s) until one answers. Use with care and don't insert
this by default into all your dial
To a telco, specifically Colombia's Telecom
On 8/9/05, Paul Belanger <[EMAIL PROTECTED]> wrote:
> Where are your calls coming from? Are you connected to the Telco or PBX?
>
> PB
>
> Panitaxx wrote:
> > Hi,
> >
> > thanks for your response. here is the log of one call:
> >
> > Enabled debugging
yes. overlapdial=yes.
On 8/9/05, Matt Fredrickson <[EMAIL PROTECTED]> wrote:
> On Tue, Aug 09, 2005 at 05:20:15PM -0500, Panitaxx wrote:
> > thanks for your response. here is the log of one call:
> >
> > Enabled debugging on span 1
> >
> > Asterisk*CLI>
> >
> > < Protocol Discriminator: Q.931 (8)
What type of client (Analog, SIP, IAX, etc??). Also, is
res_indications.so loaded?
PB
Stephen J. Wilcox wrote:
Hello,
can anyone help with my problem below, searching doesnt show any results..
thanks
Steve
On Wed, 3 Aug 2005, Stephen J. Wilcox wrote:
Hi,
I'm seeing a problem where if I
On Tue, Aug 09, 2005 at 05:20:15PM -0500, Panitaxx wrote:
> thanks for your response. here is the log of one call:
>
> Enabled debugging on span 1
>
> Asterisk*CLI>
>
> < Protocol Discriminator: Q.931 (8) len=33
> < Call Ref: len= 2 (reference 72/0x48) (Originator)
> < Message type: SETUP (5)
Where are your calls coming from? Are you connected to the Telco or PBX?
PB
Panitaxx wrote:
Hi,
thanks for your response. here is the log of one call:
Enabled debugging on span 1
Asterisk*CLI>
< Protocol Discriminator: Q.931 (8) len=33
< Call Ref: len= 2 (reference 72/0x48) (Originator)
Can you see the INVITE if you put up a trace on your gateway
(209.XXX.XXX.113)? Asterisk is not getting anything back that is why it
retransmits 5 times.
PB
OMS wrote:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 24.XX.XXX.101:5060;branch=z9hG4bK6adcaf4f
From: "512538XXX" ;tag=as532
I tried that it says
Extension 's' in context 'primario' from '915451900' does not exist.
Rejecting call on channel 0/14, span 1
thanks,
ia
On 8/9/05, Damon Estep <[EMAIL PROTECTED]> wrote:
> How many digits is your pri provider sending in the setup message? It needs
> to match your dilapl
Hi,
thanks for your response. here is the log of one call:
Enabled debugging on span 1
Asterisk*CLI>
< Protocol Discriminator: Q.931 (8) len=33
< Call Ref: len= 2 (reference 72/0x48) (Originator)
< Message type: SETUP (5)
< [a1]
< Sending Complete (len= 1)
< [04 03 90 90 a3]
< Bearer Capabili
How many digits is your pri provider sending in the setup message? It needs to
match your dilaplan, ie if they are sending 4 you need 4 digit extensions or
some other monkey business to translate.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Panitaxx
Anyone out there have success getting caller id name from a
pri, through a lucent tnt, to asterisk?
What about from other media gateways?
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Hi Rollin,
I am using SuSE's SLE 9.0, which is built for Itanium. The compiler
works for other 32 and 64 bits applications. There could still be a
problem with my environment though. I have not excluded that.
I can make it compile if I exclude the GSM codec. Now, how will that
affect the fun
Its all stable and unstable to a point. You have a stable checkout,
but like all good software you have to keep up with things.
On 8/9/05, Anderson Alves de Albuquerque
<[EMAIL PROTECTED]> wrote:
>
>
> How can I know if my Asterisk is stable?
>
> I am using:
>
> #
What does pri debug span 1 show?
On Aug 9, 2005, at 5:02 PM, Panitaxx wrote:
Hello,
I have an ISDN PRI E1. For some reason I am not receiving the did
number so every call can only go to s exten. I have tried using _X.
exten. Also I have immediate=no in zapata.conf. Any hint?
thanks in advance
Hello,
I have an ISDN PRI E1. I want to send an audio before answering, I am
using noanswer option in playback app but all the audio is muted
before the answer. I would like to play this audio.
Regards,
ia
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Hello,
I have an ISDN PRI E1. For some reason I am not receiving the did
number so every call can only go to s exten. I have tried using _X.
exten. Also I have immediate=no in zapata.conf. Any hint?
thanks in advance,
Iván Aponte
___
Asterisk-Users mai
Tom Hayden wrote:
They let you chose your protocol? Nice guys, I've never been asked -
just told. I don't know any major advantages between the different
signalling formats, though, I don't think there really are any major
differences. I've had no problems with ni1 and ni2 with Asterisk.
--
Works that way for me. IN SPA-841 for example, both lines are on the
same user/pass and the device registers once but line one rings and if I
answer it then get another call, line two rings.
-Jonathan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of jj
Se
Did you try it?
On Aug 9, 2005, at 10:25 AM, Deon wrote:
I have an ATA186, a tech just told me to set UID0 and UID1 to the same
username, and PASS0 and PASS1 to the same password. In my mind, this
would seem to have the unit registering twice under the same account,
which Asterisk wouldn't sup
Use NI2 anytime it is availabel. It will deliver calling name. NI1
will only deliver calling number. Also most COs will support NI2 with
no tweaks much better than NI1 or any of the others.
NI1 was created to solve configuration issues between systems. did a
pretty good job. But as new feat
o
(macro-record-enable,s,5) -- Executing DBget("IAX2/[EMAIL PROTECTED]/4", "RecEnable=RECORD-OUT/20") in new
stack -- DBget: varname=RecEnable, family=RECORD-OUT,
key=20 -- DBget: Value not found in
database. -- Executing SetVar("IAX2/[EMAIL PROTECTED]/4&qu
Hi Guys!
There are some phones out there that claim to be able to do xml applications
and I was wondering if anybody has actually been able to develop some for
Asterisk using phones like polycom, cisco, etc.
Polycom for example claims it can have XML apps for directories, etc.
Has anybody done
Each button on a Polycom can monitor an individual mailbox. The light
will light if any line has a message, an envelope icon will appear
next to the button with the message. to retrieve use the message
button and it will then let you select the line you wish to retrieve.
Very easy.
On Aug
Anderson Alves de Albuquerque wrote:
How can I know if my Asterisk is stable?
I am using:
# asterisk -V
Asterisk CVS-v1-0-08/03/05-15:21:21
I read files (Readme, ...) but I don´t find if it is stable or unstable.
You are using the
One must keep in mind that the config files specify how hardware is to be handled.
If config files are present, the defaults in them are adequate to keep really bad
things from happening. If not . . . . . .
By the nature of this beast, it can easily seg fault if hardware drivers don't have
proper
How can I know if my Asterisk is stable?
I am using:
# asterisk -V
Asterisk CVS-v1-0-08/03/05-15:21:21
I read files (Readme, ...) but I don´t find if it is stable or unstable.
___
Asterisk
Derek,
Thanks, is there some other unit that you might suggest that works well with
asterisk?
Jerry
--
No... The D-Link DVC series of videophones will not work with asterisk. The
unit uses H.323, which should work, but the configuration is locked to the
D-Link servers. There a
Ben,
This is an enormous help. This is exactly what I was looking for.
THANKS,
// Jonas
Asterisk wrote:
Jose,
It might help to have a look at the debian SOURCE package for Asterisk.
Here is the Debian DIFF File
http://ftp.debian.org/debian/pool/main/a/asterisk/asterisk_1.0.7.dfsg.1-2.diff.g
after editing the Make file i tried to compile
asterisk-oh323-0.6.5 but many errors displayed like
this:
[EMAIL PROTECTED] asterisk-oh323-0.6.5]# make
for x in wrapper asterisk-driver; do make -C $x build
|| exit 1 ; done
make[1]: Entering directory
`/home/ali/asterisk-oh323-0.6.5/wrapper'
./check
Jose,
It might help to have a look at the debian SOURCE package for Asterisk.
Here is the Debian DIFF File
http://ftp.debian.org/debian/pool/main/a/asterisk/asterisk_1.0.7.dfsg.1-2.diff.gz
They've obviously been successful in compiling it for Itanium - maybe something
obvious will jump out.
I wi
On Tuesday 09 August 2005 16:22, Tzafrir Cohen wrote:
> And this is still not something I could run from an init.d script (that
> has no terminal).
I use su - root -c "screen -dm asterisk -vvvgc"
that works from init.d, puts the screen somewhere root can get at it if need
be, and you can always
after editing the Makefile i tried to compile the
asterisk-oh323-0.6.5 an error massage displayed:
chan_oh323.c:37:34: asterisk/channel_pvt.h No such
file or directory
Start your day with Yahoo! - make it your home page
http:/
Dave,
A segmentation fault is usually caused by the program writing in a
memory area that is not allocated (it could be a result of the optimizer
sometime as well). That means that it can potentially overwrite code
that are executing there. In worst case scenario you could even cause a
kernel
We are having similar problems with the studder. Eventually we get a kernel
panic or the lines just get to the point it studders every few seconds.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul Traue,
Jr.
Sent: Wednesday, July 27, 2005 2:32 PM
To
On Tue, Aug 09, 2005 at 09:30:31PM +0200, Michiel van Baak wrote:
> On 22:19, Tue 09 Aug 05, Tzafrir Cohen wrote:
> > On Tue, Aug 09, 2005 at 03:06:01PM -0400, James Treleaven wrote:
> > > Is it possible to start asterisk in the foreground ("asterisk -fc") and
> > > later detach from the terminal
On Tue, 2005-08-09 at 13:55 -0600, Jonas Arndt wrote:
> I will do that. I think you are missing the point here though. If a
> program would SegFault from missing conf files, it would be a HUGE
> bug.
Why is it a _huge_ bug? The software _will not_ run without it's config
files in place. This is
On Tue, 2005-08-09 at 12:54 -0700, Edwin Lam wrote:
> i guess the way to go is using channel banks to convert those to E1 then
> connect Asterisk that way.
>
> further research, how about using these:
> http://www.welltech.com.tw/product_e_03.htm
> will that work?
Sure, that would work, all 20 of
Does anybody know whether somebody ever implemented (linux) drivers or a
chan component for the Com-On-Air cards, or is the only way to make it
work the use of a Windows box as Dect to SIP gateway for *? (Or does it
work at all?)
I can still get the cards, which would be great to retire the old DE
Hi,
I will do that. I think you are missing the point here though. If a
program would SegFault from missing conf files, it would be a HUGE bug.
The problem I am facing is most likely due to my plattform. As they
have build Debian packages for Itanium I was hoping that somebody would
have exper
Douglas Logan wrote:
With 120 Pots lines, why not get 5 T1's, then pick up a couple Digium
cards (A Quad T1, and a single T1 card).
i'd like to but unfortunately this installation is not in the US and
we have to keep the 120 phone numbers (which are not sequential) porting
over those numbers to
> I don't know if I'd recommend them. There is a lot of personal
> preference involved and the 841 certainly seems to be able to provoke
> strong dislike for people.
I switched my business from our old PBX to Asterisk 3-4 months ago.
Some folks hate "Asterisk"...I've attempted to convince them
No... The D-Link DVC series of videophones will not work with asterisk. The
unit uses H.323, which should work, but the configuration is locked to the
D-Link servers. There are no configuration parameters that can be changed...
It is all hard-coded in the firmware. Also, D-Link indicates that they
On 22:19, Tue 09 Aug 05, Tzafrir Cohen wrote:
> On Tue, Aug 09, 2005 at 03:06:01PM -0400, James Treleaven wrote:
> > Is it possible to start asterisk in the foreground ("asterisk -fc") and
> > later detach from the terminal but leave asterisk running?
>
> Start Asterisk in a screen(1) terminal ?
On Tue, Aug 09, 2005 at 03:06:01PM -0400, James Treleaven wrote:
> Is it possible to start asterisk in the foreground ("asterisk -fc") and
> later detach from the terminal but leave asterisk running?
Start Asterisk in a screen(1) terminal ?
BTW: anybody wants to wrap safe_asterisk to make it pos
Hello,
can anyone help with my problem below, searching doesnt show any results..
thanks
Steve
On Wed, 3 Aug 2005, Stephen J. Wilcox wrote:
> Hi,
> I'm seeing a problem where if I place a call, then forcibly quit or turn off
> the client the call stays active.
>
> The frames counters stop s
Was wondering if anyone else was getting these kinds of messages with
Sipuras (SPA-1001 and SPA-2100):
WARNING[26867]: Forbidden - wrong password on authentication for INVITE to
';tag=as25fb4c1a'
It's a warning, so I'm guessing it's not more critical than that. But I'm
curious about the error, wh
On 8/9/05, Dan Littlejohn <[EMAIL PROTECTED]> wrote:
> On 8/9/05, Douglas Logan <[EMAIL PROTECTED]> wrote:
> > Now that the X100P is no longer being offered by Digium, what is the
> > best solution? I seem to have run into a few posts where people talk
> > about problems they've had with their X100
Is it possible to start asterisk in the foreground ("asterisk -fc") and
later detach from the terminal but leave asterisk running?
thanks,
James
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Hi,
My asterisk box registers to broadvoice every 60 minutes without any
problems. Yet, sometimes the registration expires at the broadvoice side
and incoming calls fail to reach the asterisk server. I would like to
change the reigistration interval to a smaller value such as 10 minutes, but
I h
On 14:35, Tue 09 Aug 05, Geoff Manning wrote:
> Julio Arruda wrote:
> > Half duplex by itself doesn't hurt (depends in number of calls and etc
> > really, but anyway...)
> > What is a killer for VOIP is duplex mismatch.
> > If you have autonegotiation enabled, and your peer (the switch ?) has
> > a
Yes, I agree, the coiled handset cable is crap. It stretches out too easy.
Thank you,
Steve Maroney
On Tue, 9 Aug 2005, Peter Wemm wrote:
> On Monday 08 August 2005 11:16 am, Alvaro Parres wrote:
> > We have been using SIPURA and have no problem. With the last firmware
> > and silence supressi
Julio Arruda wrote:
> Half duplex by itself doesn't hurt (depends in number of calls and etc
> really, but anyway...)
> What is a killer for VOIP is duplex mismatch.
> If you have autonegotiation enabled, and your peer (the switch ?) has
> autoneg off, and 100/Full-duplex hard coded, you WILL have
I have had no problems with the Ambient MD3200 I bought
off ebay. It was advertised as an asterisk fxo, i didn't know which chipset
I was getting until it arrived.
Hope this helps,
Jon.
On Tuesday 09 August 2005 01:02 pm, Douglas Logan wrote:
> Now that the X100P is no longer being offered by
On Monday 08 August 2005 11:16 am, Alvaro Parres wrote:
> We have been using SIPURA and have no problem. With the last firmware
> and silence supression off.
I have one. I initially hated it, but it grew on me a lot. I got a
GXP2000 to replace it but never got around to it. I find the GXP2000
Edwin Lam wrote:
hi folks.
i'm planning to connect * to 120 POTS line. i've done some research
on FXO cards but unfortunately most manufacturers only make 4 ports/card.
the most i've found is 12 ports. so do i have to get 10 of these cards
and setup 3 Asterisk servers (assuming each have 4 free
On 8/9/05, Douglas Logan <[EMAIL PROTECTED]> wrote:
> Now that the X100P is no longer being offered by Digium, what is the
> best solution? I seem to have run into a few posts where people talk
> about problems they've had with their X100P clone cards (dropping
> calls, echos, etc) other people see
why not go back into your * src tree and 'make samples'?
On Tue, 2005-08-09 at 10:59, Jonas Arndt wrote:
> Hi Zoa,
>
> Nope, I didn't. I thought I was VERY clear on that point. What I did was
> following the guidlines in the "An introduction to Asterisk" document.It
> told me to create certai
On Tue, 2005-08-09 at 10:59 -0700, Edwin Lam wrote:
> hi folks.
>
> i'm planning to connect * to 120 POTS line. i've done some research
> on FXO cards but unfortunately most manufacturers only make 4 ports/card.
> the most i've found is 12 ports. so do i have to get 10 of these cards
> and setup 3
On Tue, 2005-08-09 at 10:59 -0700, Edwin Lam wrote:
> hi folks.
>
> i'm planning to connect * to 120 POTS line. i've done some research
> on FXO cards but unfortunately most manufacturers only make 4 ports/card.
> the most i've found is 12 ports. so do i have to get 10 of these cards
> and setup 3
Edwin Lam wrote:
> hi folks.
>
> i'm planning to connect * to 120 POTS line. i've done some research
> on FXO cards but unfortunately most manufacturers only make 4
> ports/card. the most i've found is 12 ports. so do i have to get 10
> of these cards and setup 3 Asterisk servers (assuming each ha
With 120 Pots lines, why not get 5 T1's, then pick up a couple Digium
cards (A Quad T1, and a single T1 card).
On 8/9/05, Edwin Lam <[EMAIL PROTECTED]> wrote:
> hi folks.
>
> i'm planning to connect * to 120 POTS line. i've done some research
> on FXO cards but unfortunately most manufacturers on
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