> I suppose by this you mean some sort of client software installed on
> the client PC that listens to events targeted at a particular port
> this software is listening to. If this is the case, how do you make
> Asterisk communicate with this client software?
I use yac and system() with the nc com
> After I run it, I get the following:
> 99.975586% 99.987793% 99.987793% 99.987793% 99.987793% 100.00%
> 99.987793%
Just for reference, I'm running a PIII-800Mhz and I get (with no
particular load on CPU)
-Best: 100.00 -- Worst: 99.987793
100.00% 100.00% 100.00% 100.00
Is there any other way to connect multiple tables and fields to read and
write in the dialplan? (simple inserts & queries).
Perhaps via app_dbodbc or res_sqlite?
Rafal
--
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version: 7.0.308 / Virus Database: 266.11.10 - R
TC wrote:
Why not just keep it simple use dial with Macro argument
and this std macro-screen
like this
http://lists.digium.com/pipermail/asterisk-users/2005-March/098257.html
Thank you so much!
I was not familiar with this option since we only run STABLE and this
feature is only available f
Dean Collins wrote:
Yep, POE has turned out to be a real fizzer.
Whilst a great idea for Access Points (particularly ceiling mounted AP's
They are *far* more useful for simplifying phone wiring.
so you don't need to run power points) but apart from that the whole
concept has just died.
Not re
On May 15, 2005, at 10:44 PM, Waldo Rubinstein wrote:
On May 16, 2005, at 1:32 AM, Robert Goodyear wrote:
A listener on each client that calls an internet UA on significant
events.
I suppose by this you mean some sort of client software installed on
the client PC that listens to events targeted
On May 15, 2005, at 7:19 PM, Chris Mason wrote:
I need to connect up to sixteen phones per building, I can use a cheap
hub,
but POE would be useful. Is there a cheap POE hub available?
Everything I
have seen has been expensive.
Chris Mason
You could wire your own. It's simply pins 4,5,7 and 8. I
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Robert Goodyear
|Sent: Domingo, 15 de Mayo de 2005 08:42 p.m.
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [Asterisk-Users] Callerid on PC and more
|
|
|On May 15, 2005, at 5:
On Sun, 15 May 2005, Lee Howard wrote:
> Because txfax does not currently support ECM the only reason why a page of
> image data would terminate prematurely is because the receiver detected a
> carrier drop when there wasn't one.
I'm sorry, this is a bit to broad of a statement. I suppose that
Damian,
Thanks for your input. Hyperthreading is in fact enabled and now that
you mention this I will disable it.
The reason I ask is because under some load (may be 40 simultaneous
calls), voice quality degrades. We have audio problems where one
party hears the other but not viceversa and t
Nenad,
You said that the second page was terminating at some point after the
start of the fax image. There is no "handshaking" going on after page
image data starts.
Because txfax does not currently support ECM the only reason why a page of
image data would terminate prematurely is because th
On 07:57 PM 5/15/2005, Jean-Yves Avenard wrote:
Hello
I upgraded to CVS head yesterday (due to the lack of zaptel drivers
working with 2.6.10)
And noticed that now DBGet and DBPut have been deprecated in favour of the
new Set/DB one.
In the UPGRADING.txt in Asterisk it says:
* The applications D
Hi Waldo, it really depends on who you ask - Digium say that anything
less than 99.99% is going to result in problems, but ours regularly runs
at around 99.98% and we don't have any problems.
One of our boxes was running at around 99.96% and we had major issues
with the voice quality packing up
If it had been more affordable, it might have caught on.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins
Sent: Monday, 16 May 2005 1:51 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] POE hub
Yep,
Hi All I facing problem in bridging two SIP channels . I having SIP Trunk with Service Provider. Whenever I make any international call it get through and after some seconds it give error "Bridge stops bridging channels SIP/-3be3 and SI P/primus-9381"・& call drops. May 13 17:25:19 VERBOSE[7491
On Sunday 15 May 2005 09:53 pm, Paul wrote:
>
> Do you have the clout to get a handytone for evaluation and not have
> salespeople calling you every day to ask how it's going? :)
>
Why not just buy one? You can buy one for less than $100 and if you
don't like it, you can just turn around and sell
> >
> Does it work about as well as a PCI FXO interface? I have places where I
> have already linked facilities with ptp T1 or frame relay. I don't want
> to build a * server just to tie 1 or 2 pots lines to a * PBX back at the
> main location. I also have customers who buy dells with service
> con
Yep, POE has turned out to be a real fizzer.
Whilst a great idea for Access Points (particularly ceiling mounted AP's
so you don't need to run power points) but apart from that the whole
concept has just died.
Dean
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
Hello All
Any body used sangoma card A101. I have problem with this card. My system are:
Linux Redhat 8.0. asterisk 1.07 and libpri, zaptel,...
i connected E1 and MF/R2 signalling.
i configure HDLC and TDM Voice. this is my configure as belows:
#===
I was browsing the applications developed in zaptel and came across
zttest.
After I run it, I get the following:
Opened pseudo zap interface, measuring accuracy...
99.975586% 99.987793% 99.987793% 99.987793% 99.987793% 100.00%
99.987793%
99.987793% 99.987793% 99.987793% 99.987793% 99.98779
On 16/05/2005, at 1:15 PM, Paul wrote:That would be a major issue if it happened often. My primary use would be to get incoming calls handled by the * pbx. If the ata keeps the line offhook or ignores ring in those 5 minutes we have a big problem. As I said, this only happened on the French telepho
I'm using EyeBeam
from xten, and whenever I call another user, the callee phone rings but my SIP
phone immediately hangs up. The other end keeps on ringing but when the
callee answers, there is no sounds.
I have found the
"Didn't get frame from channel" error occurring in each such call.
> Does it work about as well as a PCI FXO interface?
As well if not better, actually. After reading about all the issues
with zaptel hardware (timing, interrupts, echo) I'm staying away from
it. I've installed several SPA 3000's and while the initial config
takes some time to get everything workin
Jean-Yves Avenard wrote:
Does it work about as well as a PCI FXO interface? I have places
where I have already linked facilities with ptp T1 or frame relay. I
don't want to build a * server just to tie 1 or 2 pots lines to a *
PBX back at the main location. I also have customers who buy dells
w
Rich Adamson wrote:
What is in the bug tracker helps make things clearer to people who know
what they are doing. What we need is something that makes things clear
to laymen. Saying internally and externally clocked doesn't cut it. It
needs to be made clear to laymen that clocking internally is r
The cheapest I have found was a 3COM 24 Port for $799.00.
Thank you,
Steve Maroney
On Sun, 15 May 2005, Chris Mason wrote:
> I need to connect up to sixteen phones per building, I can use a cheap hub,
> but POE would be useful. Is there a cheap POE hub available? Everything I
> have seen has bee
> I was just wondering, Do sipura 200 and linksys pap2 ATAs send their
> mac address in REGISTER message? Is their any other way to get the MAC
> address of sip peer who is trying to register?
No, the PAP2-NA does not send its MAC address in a REGISTER message.
--
Nabeel Jafferali
X2 Netwo
Last time I checked, the MAC address is not transmitted in any SIP
message. The only way I can think of getting the MAC address if the
device is remote (not on the same LAN) is by having the ATA to send a
provisioning request with its MAC address and then match them up. Not
fool proof, especially b
Does it work about as well as a PCI FXO interface? I have places where I have already linked facilities with ptp T1 or frame relay. I don't want to build a * server just to tie 1 or 2 pots lines to a * PBX back at the main location. I also have customers who buy dells with service contracts but lea
Hi All
Im facing problem in bridging two SIP channels . Im having SIP Trunk with Service Provider. Whenever I make any international call it get through and after some seconds it give error Bridge stops bridging channels SIP/-3be3 and SI P/primus-9381 & call drops.
May 13 17:25:19 VE
HelloI upgraded to CVS head yesterday (due to the lack of zaptel drivers working with 2.6.10)And noticed that now DBGet and DBPut have been deprecated in favour of the new Set/DB one.In the UPGRADING.txt in Asterisk it says:* The applications DBGet and DBPut have been deprecated in favor of functi
snacktime wrote:
I wish I could find out for sure how well the spa-3000 FXO works with *
and same for the Grandstream Handytone 488. I need FXO-SIP conversion in
places where I don't need a PC.
What in particular are you concerned about? I've done some pretty
strange stuff with the spa-3000 a
> I wish I could find out for sure how well the spa-3000 FXO works with *
> and same for the Grandstream Handytone 488. I need FXO-SIP conversion in
> places where I don't need a PC.
What in particular are you concerned about? I've done some pretty
strange stuff with the spa-3000 and it works gre
I need to connect up to sixteen phones per building, I can use a cheap hub,
but POE would be useful. Is there a cheap POE hub available? Everything I
have seen has been expensive.
Chris Mason
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.
Hi All,
I'd like to setup asterisk for a small call center. I need basic
functionality for a small call center, including but not limited to:
1) Multiple queues with different rules for each (e.g. some queues,
people may leave messages in, others people will just have to wait
until the call
follow this link
ignore the German and see the commands
http://www.vonloesch.de/node/17
for the last part be sure that you modprobe the right driver for your
particular device.
one little thing is that in Debian you shouldn't use /usr/local/bin, but
/usr/bin, if you are using the source from digi
Rich Adamson wrote:
Im looking for a zaptel type device with one (or more) FXO and one
(or more) FXS port. Basically this guy would sit in-line of your
phone line (PCI card). Any suggestions? TDM400 would be overkill.
Your only choice for zaptel type is the TDM card.
Probably the next best c
How are you popping up the screen on the users pc side?
Have you tried goldmine?
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Robert Goodyear
|Sent: Domingo, 15 de Mayo de 2005 08:42 p.m.
|To: Asterisk Users Mailing List - Non-Commercial Discuss
On May 15, 2005, at 5:29 PM, Anton Krall wrote:
Guys.
This is a good one... Is anybody doing callerid on the PC? What are you
using besides yac or things like that? And are you using some CRM like
Goldmine with it?
Good huh?
Yes, I touch userrecords in my SugarCRM implementation when an in- or
ou
there are cheapo clones of the X100P for the fxo side (up to two will be
ok),
at $20 each = $40
http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&category=61841&item=5773237792&rd=1
and you can get a refurbished sipura 2000 for the (2) fxs part of it.
($70) voipsupply
with an used compaq PIII at $50
> Im looking for a zaptel type device with one (or more) FXO and one
> (or more) FXS port. Basically this guy would sit in-line of your
> phone line (PCI card). Any suggestions? TDM400 would be overkill.
Your only choice for zaptel type is the TDM card.
Probably the next best choice is the spa
HI All:
I was just wondering, Do sipura 200 and
linksys pap2 ATAs send their mac address in REGISTER message? Is their
any other way to get the MAC address of sip peer who is trying to
register?
Thanks
Karim
___
Asterisk
Depends on which driver you are using.
Res_odbc.conf
Or
Res_mysql.conf
-Matthew
> From: Iqbal <[EMAIL PROTECTED]>
> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
>
> Date: Mon, 16 May 2005 01:14:57 +0100
> To:
> Subject: [Asterisk-Users] asterisk and realtime
>
>
> Hi
No, he wants a desk phone. He could take the phone with him but he doesn't
want to. I like the example from Robert, I'm going to try that.
Chris Mason
US Number: (646)722-0001 US Fax (815)301-9759
Skype: netconcepts
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
>
Hello all,
I have a digium TDM40B install in my Dell PowerEdge 1800. when I run
modprobe wctdm nothing happen and it does not go to the next linux prompt
util I press control c. It just like hanging at the prompt. Is anyone having
the same problem? I have try asterisk stable (wcfxs) and CVS HEAD (
You have to put entries in sip.conf
Race "the Tyrant" Vanderdecken
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michele
"O-Zone" Pinassi
Sent: Friday, May 13, 2005 6:47 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Asterisk newbie
I
Hi all,
I am new to asterisk and trying to setup clients over LAN to enable voice-chat between them. I have got two clients(IAX-phone) having extensions 4061 and 4082. I am able to call extension 600(provided with sample configuration) from both of them but when I try to call 4061 from 4082 or vice
No. Use RealTime Extensions instead.
-Matthew
> From: John Ackley <[EMAIL PROTECTED]>
> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
>
> Date: Sat, 14 May 2005 17:30:33 -0400
> To: Asterisk Users Mailing List - Non-Commercial Discussion
>
> Subject: [Asterisk-Users] DynExt
Ummm..app_dictate isn't for use with 1.0.7. You need CVS-HEAD.
-Matthew
> From: Malcolm Taylor <[EMAIL PROTECTED]>
> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
>
> Date: Sat, 14 May 2005 12:41:21 -0400
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
>
> Su
Guys.
This is a good one... Is anybody doing callerid on the PC? What are you
using besides yac or things like that? And are you using some CRM like
Goldmine with it?
Good huh?
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://list
> Message: 10
> Date: Sun, 15 May 2005 21:41:23 +
> From: Laurent Lesage <[EMAIL PROTECTED]>
> Subject: Re: [Asterisk-Users] skype channel
> To: Asterisk Users Mailing List - Non-Commercial Discussion
>
> Message-ID: <[EMAIL PROTECTED]>
> Content-Type: text/plain; charset=ISO-8859-1; for
Hi
when configuring realtime in extconfig, it allows you to set the
dbname,table for a particular conf file eg voicemail.conf, but where do
you set the hostname for the db and user/pass
Iqbal
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Asterisk-Users mailing list
Asterisk-Users@lists.digium.
On Mon, 2005-05-16 at 00:33 +0100, Tony Hoyle wrote:
> trixter http://www.0xdecafbad.com wrote:
> > does anyone have knopsterisk for download, I assume that because its GPL
> > the creator of that iso cant restrict spreading it. A friend wanted it
> > to play on a box and the only thing I can find
I don't think you have many choices for this oneOn 16/05/2005, at 9:53 AM, Tim Connolly wrote: I’m looking for a zaptel type device with one (or more) FXO and one (or more) FXS port. Basically this guy would sit in-line of your phone line (PCI card). Any suggestions? TDM400 would be over
On 5/15/05, Zen Kato <[EMAIL PROTECTED]> wrote:
> I am using CVS-v1-0-04/20/05-12:31:26. Windows Messenger5.1 works MOH
> fine. After I stop MOH on Windows Messenger, if the hungup signal could
> not send to *, the sip channel(e.g.,SIP/52001-08ca) for this MOH remains.
> Then the user trys again MO
I’m looking for a zaptel type device with
one (or more) FXO and one (or more) FXS port. Basically this guy would sit
in-line of your phone line (PCI card). Any suggestions? TDM400 would be
overkill.
___
Asterisk-Users mailing lis
Or have a small solar panel on the back of the phone. Stick it on the dash
of your car, assuming it doesn't burst into flames from heat; it should be
fully charged in an hour or two.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul
Sent: Sunday, May 15
trixter http://www.0xdecafbad.com wrote:
does anyone have knopsterisk for download, I assume that because its GPL
the creator of that iso cant restrict spreading it. A friend wanted it
to play on a box and the only thing I can find with google is the
knopsterisk.com site which wants $10 to get a c
Andres Paglayan wrote:
question about this thread,
would a wi-fi voip phone work for this guy?
meaning, he takes it to wherever he goes and it gets registered wherever
it as wireless access.
is that theoretically correct?
I like that approach. Those toys will be getting more affordable. One
con
Maybe try a version of redhat that was
released in the past 5 years? Seriously, why do you require RH7.3 over Fedora
or even RH 9?
From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
On Behalf Of Thierry Wehr
Sent: Sunday, May 15, 2005 5:58 PM
To:
asterisk-users@lists.digium.com
S
Good
evening
from the CVS of the
2005/05/14 it's impossible to build asterisk* on a redhat
7.3
i get this at
compile time
chan_sip.c: In
function `build_user':chan_sip.c:10007: parse error before
`struct'chan_sip.c:10029: `userflags' undeclared (first use in this
function)chan_sip.c
question about this thread,
would a wi-fi voip phone work for this guy?
meaning, he takes it to wherever he goes and it gets registered wherever
it as wireless access.
is that theoretically correct?
>
>
>
___
Asterisk-Users mailing list
Asterisk-Users
> You serious? I typed all that and you were asking about
> RETRIEVING vm all along? Wow, I must be really dense today.
No, I know it now :-)
> So: don't pass calleridnum to extension 8500. Or configure a
> different voicemail retrieval exten for roaming users and
> pass null to voicemailmain.
I am using CVS-v1-0-04/20/05-12:31:26. Windows Messenger5.1 works MOH
fine. After I stop MOH on Windows Messenger, if the hungup signal could
not send to *, the sip channel(e.g.,SIP/52001-08ca) for this MOH remains.
Then the user trys again MOH, a new sip channel starts. And again
the hugup signal
On Sun, 2005-05-15 at 15:55 -0600, Ira Burton wrote:
> I am curious if anybody has pointers on the best way to get the 7
> digit PSAP number for an area. I am thinking about making a '911'
> extension that will dial the PSAP number, wait for the PSAP to answer
> and play a message giving the addre
On Sun, 2005-05-15 at 17:49 -0400, Paul wrote:
> I think you would be better off to make a knoppix CD, boot it and get *
> installed and running. After that read the following and maybe you can
> create something better to share with the world.
Unfortunately that wont help my friend who wanted t
I am curious if anybody has pointers on the best way to get the 7
digit PSAP number for an area. I am thinking about making a '911'
extension that will dial the PSAP number, wait for the PSAP to answer
and play a message giving the address of the originating call, and
replay the the information e
trixter http://www.0xdecafbad.com wrote:
does anyone have knopsterisk for download, I assume that because its GPL
the creator of that iso cant restrict spreading it. A friend wanted it
to play on a box and the only thing I can find with google is the
knopsterisk.com site which wants $10 to get a c
Hi *,
I was just going to ask the same question. Does anybody have an
information about Skype and Asterisk? Any link?
Thanks in advance
Laurent
Bartek Kania a écrit :
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
I just noticed that the Skype API for linux seems to be available.
I've read before
"The truth is out there"
You just need a good shovel...
PD
On Sun, May 15, 2005 at 04:53:58PM -0400, Jeffrey Starin wrote:
> Jonathan! You don't know how much that simple explanation has helped me
> understand Asterisk. Well done. Well said. And to the point clearly.
>
> I would hope this c
does anyone have knopsterisk for download, I assume that because its GPL
the creator of that iso cant restrict spreading it. A friend wanted it
to play on a box and the only thing I can find with google is the
knopsterisk.com site which wants $10 to get a copy and does not provide
(as far as I can
On May 15, 2005, at 12:27 PM, Chris Mason (Lists) wrote:
The issue would be voicemail. He would want only one
voicemail account
to be accessed via the messages button.
Chris Mason
www.anguillaguide.com
Tel: (305) 704-7249 Fax: (815)301-9759
Infinite SIP account(s) could roll over into one VM accou
> Will this fix it or is the \@ change necessary also. If it is, I will get
> a patch in
> tomorrow.
I just checked - Dial(Local/1416967/routes/n) is not valid.
Dial(Local/[EMAIL PROTECTED]/n) is valid. Therefore, you would need the
other change as well.
--
Nabeel Jafferali
X2 Networks
www.x
@home do that for you everyday...;)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Irakli
Natsvlishvili
Sent: Sunday, May 15, 2005 2:52 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] [EMAIL PROTECTED] backup/restore question
Hello,
How
I've been running into some issues with a TE410P and lost audio and
similar which I have tracked down to some apparent incompatibilities
between the version of Linux I'm running and my hardware - which causes
zttest to not get 100% of the samples every time.
Instead of fighting this on the machin
> > What is in the bug tracker helps make things clearer to people who know
> > what they are doing. What we need is something that makes things clear
> > to laymen. Saying internally and externally clocked doesn't cut it. It
> > needs to be made clear to laymen that clocking internally is rarel
I know that part, dialing more than one extension and sending all voicemail
to the same extension, but recovering voicemail is the hard part. On the
Sipura and, I believe, on the Polycom, you configure the voicemail extensio,
e.g., 8500, and the mailbox is derived from the extension yhou are dialin
Jonathan! You don't know how much that simple explanation has helped me
understand Asterisk. Well done. Well said. And to the point clearly.
I would hope this could find it's way onto the Asterisk Wiki and be the
*first* thing someone reads when looking at the documentation about sip.
Thank
On May 15, 2005, at 12:27 PM, Chris Mason (Lists) wrote:
The issue would be voicemail. He would want only one
voicemail account
to be accessed via the messages button.
Chris Mason
www.anguillaguide.com
Tel: (305) 704-7249 Fax: (815)301-9759
Infinite SIP account(s) could roll over into one VM accou
Can you give me an example of the conguration used to do this?
Chris Mason
www.anguillaguide.com
Tel: (305) 704-7249 Fax: (815)301-9759
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Robert Goodyear
> Sent: Sunday, May 15, 2005 4:00 PM
> To:
Jeffrey Starin wrote:
I have a little confusion about the general settings (other than the
register values) in the SIP
General area.
[snip]
However, I'm confused as to the purpose of the
"general" settings -- to what or which connection do they apply? Since
the context suggested for the general
On May 15, 2005, at 12:27 PM, Chris Mason (Lists) wrote:
The issue would be voicemail. He would want only one voicemail account
to be
accessed via the messages button.
Chris Mason
www.anguillaguide.com
Tel: (305) 704-7249 Fax: (815)301-9759
Infinite SIP account(s) could roll over into one VM acc
I have a customer who wants to do large volumes of H.323 to H.323
hairpinning. We haven't tested this scenario for large volumes before;
maybe someone on asterisk-users has.
If they buy a top of the line PC, how many concurrent calls are we
likely to get? Routing logic will be simple, the machi
On Sun, 2005-05-15 at 09:27 -1000, Jean-Denis Girard wrote:
> Thanks for your reply, but unfortunately I'm aware of this, and the file
> is first created in a temporary directory, then moved to asterisk's
> outgoing directory. Move is done from a Perl script, using the
> File::Copy module; I thi
trixter http://www.0xdecafbad.com a écrit :
How do you create them?
There is a race condition with asterisk and the spool where if you
create the file or copy it into the queue directory asterisk tries to
read and parse the file before you have finished writing it. A
suggested method instead is to
The issue would be voicemail. He would want only one voicemail account to be
accessed via the messages button.
Chris Mason
www.anguillaguide.com
Tel: (305) 704-7249 Fax: (815)301-9759
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> snacktime
Thanks, at least I know it's not me
Rich Adamson wrote:
There was an issue raised about a month or so ago due to one of the
itsp's (using iax) that involved dtmf not working "after" an iax call
is considered "answered". Might that be a possibility?
I opened a bug report on it, but it was immedi
Thanks for the information Lee !
Still, something is still strange to me, since this two Panasonic Fax
machines are receiving at least 20 multi-paged faxes a day (they are in same
office as both Asterisk boxes, and me :) ). Beside that, POTS lines in those
two faxes
are from same PSTN switch as li
How do you create them?
There is a race condition with asterisk and the spool where if you
create the file or copy it into the queue directory asterisk tries to
read and parse the file before you have finished writing it. A
suggested method instead is to create it on the same partition then move
Hello,
How do I routinely backup all necessary configuration files on [EMAIL
PROTECTED] Is
there any procedure/tool/script for it? And if I need to move * with
existing configuration on a new hardware, what is the best way to do it?
Thanks
I.N.
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On 5/15/05, Chris Mason (Lists) <[EMAIL PROTECTED]> wrote:
> I have a client who works from three locations, he has a Polycom 500 at each
> location connected to the same PBX. How would you configure the phones? Is
> it practical to have the extension follow him, I don't want to rely on him
> turni
Hello,
Question #1:
I have * with g729 installed, and two phones - Cisco 7960 and Cisco 7905.
If g729 is the only available codec for 7905's configuration, then call from
7960 to 7905 goes without any problem and both phones use g729.
But if I call from 7905 to 7960 the following is displayed on
There was an issue raised about a month or so ago due to one of the
itsp's (using iax) that involved dtmf not working "after" an iax call
is considered "answered". Might that be a possibility?
I opened a bug report on it, but it was immediately closed with a
comment "that's the way * operates"
I have a client who works from three locations, he has a Polycom 500 at each
location connected to the same PBX. How would you configure the phones? Is
it practical to have the extension follow him, I don't want to rely on him
turning the phone off when he leaves each location, but he does want to
Yes, you are right. This is the code that went in on Saturday or this
morning:
$dialstr = "Local/$phone/$res->{path}|30|HL/n(" . ($maxtime * 60 * 1000) . ":6:3)";
This patch sat in the bugtracker for months as I did not have my ducks in a row. :-(
Will this fix it or is the \@ change ne
Guys.
Anybody using an open source/free Web UA that supports SIP or IAX2?
Im looking for something to create one of those click here to talk to our
Sales/tech Deps applications.
Any leads/urls?
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Asterisk-Users mailing list
Asterisk-Users@lists.dig
Hi list,
I'm using asterisk to send alerts via phone or fax, by using the spool
functionnality (writing files in /var/spool/asterisk/outgoing). The
system works, but sometimes a file in the outgoing directory is simply
ignored by asterisk (nothing on the cli); the spool file is correct.
It happe
On 15 May 2005, at 17:26, Steve Underwood wrote:
Michael Welter wrote:
What is in the bug tracker helps make things clearer to people
who know what they are doing. What we need is something that
makes things clear to laymen. Saying internally and externally
clocked doesn't cut it. It nee
I have a little confusion about the general settings (other than the
register values) in the SIP
General area. I understand that for examle in a SIP context like [FWD]
or [BROADVOICE]
the entries in those areas are ths settings that take effect in any
communication woth FWD and/or BROADVOICE. Ho
Hello, Steve and thank you for replying.
Yes, I know the DISA is going to the correct context because when a user
direct-dials into the box (not coming in via FWD) everything works fine
-- the tones are understood and acted upon. This only happens when
calls are originating from a FWD connectio
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