Hello,
It does appear to be an issue with the colon, as I ran
this test:
exten = _9X.,2,SetVar(REC_FILE_NAME=test)
exten = _9X.,3,Monitor(wav|${REC_FILE_NAME}|m)
and it worked fine. Indeed a colon is a valid
filename under Linux. So is this a bug?
Jason
--- Jim Van Meggelen [EMAIL
Asterisk wrote:
I've got a test * server (hppbx) where I install CVS-HEAD as often as
possible, with my extension registered to this, talking through IAX to
our production server which then channels out to the PSTN.
After completing a call just now, the following appeared on the CLI of
hppbx
Peter Svensson wrote:
On Wed, 16 Feb 2005, Rob Scott wrote:
Why is it that Asterisk can't cope with silence suppression?
All the clients seem to be able to but not Asterisk.
What would be needed to get it to work with silence suppression?
What is the problem?
Asterisk clocks outgoing rtp data to
Hey!
I installed V0.5 and i was suprised: Good job, i love it!
Is there a plan to include drivers for HFC-S Cards (zaphfc / bristuff)??
Greets from germany
Michael
[EMAIL PROTECTED] schrieb:
New features include Festival text to speech and a new
Web Conferencing GUI. There are also numerous small
Chris St Denis wrote:
I am using mysql sipfriends and can't seem to get the MWI to work. From what
I've read it seems this is not supported with that dynamic system, and
probably never will be.
In the 1.0 stable release, you can not send MWI for database peers.
In CVS head, the base for the future
[...] In the meantime, get a Sipura 2100, supports 2 729 calls and
has both WAN/LAN ports.
I was told that the Uniden DTA200 also supports 2 g729 calls. I'm
buying one to test. Street price around US$ 90.
Another one with dual g729 channels is MTA V102. Street price US$ 100.
Also will test this
On Feb 16, 2005, at 10:34 AM, Steve Underwood wrote:
BTW, Steve, if you're still reading, what is the RADIO_RELAX option
intended to be for in dsp.c?
It is something someone else added to the code to make the detection
criteria in relaxed mode even more relaxed. If setting that helps,
Hello,
I have this configuration
Cisco 2600 SER Asterisk
When I receive a call on asterisk from ser then I dial
2 different extensions a${EXTEN} and b${EXTEN} but I can not set correctly the
caller id number.
When I make a dial asterisk set caller id name and
number to
I wouldn't recommend the grandstreams, I had very bad experience using
the grandstream 102, It kep locking up on me. The buttons are very bad
buttons. The sound quality is just as bad.
grandstream barbie^H^H^H^H^Hudgettone phones really sucks. they're
cheap, and that's it
roy
I wonder what makes the difference between inserting 4 HFC-S cards
(cca. 120
EUR) and using 1 QuadBRI card (approx. 700 EUR) ?
What makes such difference ? Is it possible to do first configuration
?
With what drivers ? Is it stable ?
1 HFC-S card - lots of interrupts
4 cards - interrupt havoc
I've been considering doing a web based database system, where you can
post your termination offerings or wanted, then search by location,
price, minimum volumes, etc.
I'd probably make it free, supported by advertising my consulting
company, or Google Adwords, or something like that.
I've
On 16/02/2005 at 09:00 Michael Graves wrote:
Andy Powell has prepared a CF image at www.automated.it/asterisk. I
have been able to get this booted on a testbed system.
Sadly, I'm a Linux newbie and not skilled at command line
administration, thus I'm stuck at the moment. I can get the existing
Hi,
I have a problem when configuring Asteriskand SER,
using SER as a simple SIP gateway. SER connects to
another third party SIP server. I want to call a user
that is registered in the third party SIP server, from
asterisk. In order to achieve this, I defined a peer
in sip.conf, as follows:
Yesterday I asked about a user manual - ie a user guide to actually
using asterisk (now on how to set it up) the doc project (v2) isn't
anywhere near complete and is the closest thing I could find.
Does anyone know of such a doc? The reason I ask is that while a lot of
this may be obvious to
Is there an easier way to cancel the echo ?
Is there a way to use chan_capi with Teles cards ?
Hi,
If your cards are supported by i4l, the odds on support in
mISDN are good. mISDN provides a CAPI interface for the
cards. Maybe you should check that out.
My experience with the echo on i4l
Kim Daeyong wrote:
I downloaded asterisk to use cvs to checkout the release version.
After installing, I would like to load module chan_h323.so but there is some
error :
*CLI load chan_h323.so
Feb 17 15:22:38 WARNING[2865]: loader.c:258 ast_load_resource:
/usr/lib/asterisk/m
odules/chan_h323.so:
I have successfully installed and configured the
asterisk, the incoming and the outgoing calls are working fine, its a tremendous
solution :)
Now i want to enable trunking between two asterisk
boxes, in the iax.conf i have put:
[karachi]
...
...
...
trunk=yes
...
...
...
everything seems
Hi everybody,
I have aproblem with voicemail:
I have two TDM boards for a total of 5 fxs and 3
fxo. One of thefxo is connectedto the local tel provider and is
redirected to a voicemail box.
When I call asterisk from outside, I leave my
message, but, after hanging on, voicemail continues
Hello,
I recently instaled an asterisk 1.0.3, libpri 1.0.1 and zaptel 1.0.3 withuot errors.
When i try to load the modules, i get,
modprobe zaptel - load zaptel without errors.
modprobe wcfxs - can't locate wcfxs
I search for wcfxs location, and it is on /lib/modules/2.4.20/misc/ like zaptel.
hello,
i have a problem on started asterisk, when try to start
asterisk a get the fowlling error:
chan_misdn.so] = (Channel driver for mISDN Support
(Bri/Pri))
Feb 17 11:34:01 WARNING[3104]: config_old.c:27 ast_load:
ast_load is deprecated, use ast_config_load instead!
== Parsing
Good day
Dean,
I am interested in developing the video conferencing capability. I am
going to look over the request during the following two days in order
decide defnitively. Can you tell me if your offer still stands?
Herman Webley
___
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Dear friends,
Can i use the Asterisk functions (call recognition for example), using
conventional telephony (in Brazil) ?
Thanks in advace
Pablo Fernandes
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Whats up to VoipJet.com? Their DNS servers are not reachable. Both primary and secondary
are on the same subnet - weird setup.
Thanks,
David
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The problem was this line at the end of modules.conf
alias wcfxs wctdm
Ismael.
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To UNSUBSCRIBE or update options visit:
Whats up to VoipJet.com? Their DNS servers are not reachable.
Looks like their provider is maybe having problems. AS3728, onr.com, Onramp.
Both primary and secondary
are on the same subnet - weird setup.
While that might be true, it also might not be.
206.55.64.64 and 206.55.64.65 are
Thomas == Thomas RULMONT [EMAIL PROTECTED] writes:
Thomas When I call asterisk from outside, I leave my message, but,
Thomas after hanging on, voicemail continues to record the busy tone
Thomas that the provider sends. How can I avoid this behaviour?
First of all, try to isolate the problem by
Haven't had it since, so it's hard to try debug :(
Julian.
Olle E. Johansson wrote:
Asterisk wrote:
I've got a test * server (hppbx) where I install CVS-HEAD as often as
possible, with my extension registered to this, talking through IAX to
our production server which then channels out to the
Florian Lefeuvre wrote:
Hi Steve,
I was the one who post a question about the RADIO_RELAX option.
In fact when I set it , I remark some better result in the detection
of the DTMF...
after a few more tests, It appears I was wrong.
I did a record of samples used by the DTMF_detect function.
I
Any analog modem (fax or pc) is going to be limited to 9600 baud or
slower,
and will only achieve that speed if g711 is used through the entire path
(including asterisk). If a modem call comes in one T1 (or PRI) and goes
out another, asterisk is still handling the pcm packets. The
On Thu, 17 Feb 2005, Rich Adamson wrote:
In the post that I was responding to, the writer hinted his understanding
was that T1 to T1 channel connections didn't involve any asterisk code.
His impression seemed to suggest that codec selection, etc, wasn't a
factor since the analog fax modem
No, it don't. If i call from outside to an inside phone, when I hang up the
outside phone, I hear the busy tone on the inside phone.
- Original Message -
From: Samuel Tardieu [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Thursday, February 17, 2005 1:22 PM
Subject:
Anyway, they're not reachable since yesterday evening.
-D
Joe Greco wrote:
Whats up to VoipJet.com? Their DNS servers are not reachable.
Looks like their provider is maybe having problems. AS3728, onr.com, Onramp.
Both primary and secondary
are on the same subnet - weird setup.
While that
On February 17, 2005 06:08 am, Muhammad Muzzamil Luqman wrote:
Feb 17 10:59:14 WARNING[18726]: chan_iax2.c:7536 build_user: Unable to
support trunking on user 'karachi' without zaptel timing Feb 17 10:59:14
The answer's pretty simple -- do you have a zaptel timing source? i.e. X100P,
T100P,
Peter Svensson wrote:
What is c-ourcallstate set to at this time? Can you provide a debug log
(pri intense debug span xxx) of the call?
it's Q931_CALL_STATE_ACTIVE - that's what it should be after a call is
established.
Asterisk only expects INFORMATION elements when expecting overlap digits
Hi Dan.
' - audio delay when IAX bridging inside Asterisk
Will it cover that problem of long delays that we talked before!?
Regards,
Denis Galvão.
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How do I test if the card is working or not ? Is there something that I can
do
to get a response from the card ?
Ive put the card in, installed drivers ect but can't dial out and can't see
a
response when I try dial in from external number.
Any ideas ?
Thanks
Shaun
---
Outgoing mail is
In the post that I was responding to, the writer hinted his understanding
was that T1 to T1 channel connections didn't involve any asterisk code.
His impression seemed to suggest that codec selection, etc, wasn't a
factor since the analog fax modem signals were coming in one T1 channel
Hi Herman, yes the offer still stands but I really need to see something
soon otherwise I'm going to go out and buy the macromedia communications
server solution and run it is as a separate standalone application to my
Asterisk voice conferencing server.
I have had one other email 2 days ago from
Roy Sigurd Karlsbakk [EMAIL PROTECTED] wrote:
I wouldn't recommend the grandstreams, I had very bad
experience using
the grandstream 102, It kep locking up on me. The
buttons are very bad
buttons. The sound quality is just as bad.
grandstream barbie^H^H^H^H^Hudgettone phones really
sucks.
Sounds very interesting, would providors be willing to insert pricing
or would you need to enter all the data?
I would suggest a set of rules like pricewatch.com uses to keep people honest.
Keep us informed,
Cheers,
Jonathon
On Thu, 17 Feb 2005 10:29:54 +, Alistair Cunningham
[EMAIL
No, it don't. If i call from outside to an inside phone, when I hang up the
outside phone, I hear the busy tone on the inside phone.
- Original Message -
Thomas When I call asterisk from outside, I leave my message, but,
Thomas after hanging on, voicemail continues to record
Welcome to the Asterisk users community!
Asterisk is the leading Open Source Telephony platform,
with support both for classical telephony and IP telephony.
Asterisk.org is a fast moving project. New code is added every
day.
Our community is also growing
Folks,
I've been running asterisk successfully using the
extensions.conf and voicemail.conf.
Now that I've got asterisk happily looking up MySQL
tables for the VM configuration, I decided to try out
the contributed script
/usr/src/asterisk/contrib/scripts/retrieve_extensions_from_mysql.pl
I
On Feb 16, 2005, at 7:19 PM, Eric Wieling wrote:
Jerry wrote:
On Feb 16, 2005, at 3:07 PM, Eric Wieling wrote:
I have the following configuration:
CLEC - T-1 - Asterisk - Adtran Channel Bank - (analog) - Nortel
Don't complain that it's ugly. I've already done plenty of that.
The CLEC manages
Hi!
How do I test if the card is working or not ? Is there something that I
can
do
to get a response from the card ?
Ive put the card in, installed drivers ect but can't dial out and can't
see
a
response when I try dial in from external number.
Did you configure groups in sirrix.conf? See
Hi Denis,
- Original Message -
From: Denis Galvão - iSolve [EMAIL PROTECTED]
' - audio delay when IAX bridging inside Asterisk
Will it cover that problem of long delays that we talked before!?
Yes, with a small remark.
In some situations is possible to loose the audio for the first 2-3s
Pablo Fernandes wrote:
Can i use the Asterisk functions (call recognition for example), using
conventional telephony (in Brazil) ?
Generally, yes. (VOIP is just a cool thing to be into these days.)
Can you define call recognition for me? Do you mean
CallerID(determining the phone number that is
On Thu, 17 Feb 2005, Deti Fliegl wrote:
Peter Svensson wrote:
Asterisk only expects INFORMATION elements when expecting overlap digits
(i.e. before CONNECT, PROCEEDING etc). After that it expects digits as
inline dtmf.
Yep - but ISDN phones normally do not encode inline DTMF. Therefor
Hi,
I have two HFC-s boards I configured in NT and TE mode respectively.
When I connect the two boards together, I can dial extensions and I
see the correct called and caller ID numbers:
-- Executing SetCallerID(Zap/2-1, 7516862) in new stack
== CDR updated on Zap/2-1
-- Executing
We have a brand new T100P that has never been used for sale.
We purchased this card from NETXUSA and then decided to use an external VoIP
gateway. So I have this unit for sale.
Price: $450.00 plus shipping.
If interested, please reply off list.
Ty Carter, President
Strategic Network
Kim Daeyong wrote:
I downloaded asterisk to use cvs to checkout the release version.
After installing, I would like to load module chan_h323.so but there is some
error :
*CLI load chan_h323.so
Feb 17 15:22:38 WARNING[2865]: loader.c:258 ast_load_resource:
/usr/lib/asterisk/m
odules/chan_h323.so:
Hi,
I have compiled asterisk-addons successfully, but when I put
res_config_mysql.so in modules directory asterisk fails to load, here
is the error:
7:29 WARNING[19097]: loader.c:301 __load_resource: libmysqlclient.so.14: cannot
open shared object file: No such file or directory
Feb 17
Title: Cyclades-PC300/TE 1 Compatibility?
Hello,
Has anyone on this list tried the Cyclades PC300 card with asterisk?
Thanks,
Brad.
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beonice wrote:
The resulting extensions_from_mysql.conf file looks
something like this:
[vp_context]
exten = 1000,1,Record(/tmp/rec:gsm);
exten = 1000,2,Playback(/tmp/rec) ;
exten = 1000,3,Background(goodbye) ;
exten = 1000,4,Hangup();
I decided to #include this in my main
Hi,
If digital voice circuits(in any form) are available in your area,
you'll likely be happier using them than POTS lines.
yes, here is available Digital voice circuits.
You will need hardware that is compatible with your areas telephone
network. (Stating that as it is likely different from
Pablo,
Brazil uses normal PRI (primary rate ISDN) over E1, so the Digium TE
cards will definitely work. As always with PRI, you will need to get the
correct settings for framing, line coding, and so on.
I would imagine that BRI (basic rate ISDN) would also be normal in
Brazil, but have not
Hello,
I have following problem with Sangoma A104 card:
CLI pri show span 1
Primary D-channel: 16
Status: Provisioned, Down, Active
Switchtype: EuroISDN
Type: CPE
Window Length: 0/7
Sentrej: 0
SolicitFbit: 0
Retrans: 0
Busy: 0
Overlap Dial: 0
T200 Timer: 1000
T203 Timer: 1
T305 Timer: 3
Run this from inside asterisk-addons:
make clean; cvs update; make; make install
then try again. be sure you have v1.7 of res_config_mysql
The Makefile seems to check most places for mysql libraries but check it
again to make sure. Also make sure your mysql lib path is in ld.so.config
then
On Thu, 17 Feb 2005 15:04:50 +0100
Stefan Gofferje [EMAIL PROTECTED] wrote:
Hi folks,
I'm registered with sipgate, a German SIP provider.
Configs works fine so far. Trouble is, after a while, it
seems, my registration is dropped by sipgate. How do I
tell * the interval for * registering with a
Hi folks,
I'm registered with sipgate, a German SIP provider. Configs works fine
so far. Trouble is, after a while, it seems, my registration is
dropped
by sipgate. How do I tell * the interval for * registering with a
provider? I suppose, the re-registration interval is to long...
Howard Lowndes wrote:
On Thu, 2005-02-17 at 15:51, [EMAIL PROTECTED] wrote:
I've installed a TDM400. Having a go with AMP.
I would like incoming calls to be put throuhg to an extension (at my desk)
and a mobile (cell phone) at the same time. Whichever picks up, gets the
call..
This should be
Hi all
I want to connect Asterisk with my Siemens HiPath PBX, to use it as a
Gateway to the PSTN.
I already have a RDIS entry in the Siemens HiPath, but the PC with Asterisk
doesnt have any RDIS board, can someone tell me about good and cheap PCI
RDIS boards that supports QSIG?
The Eicon boards
After updating to the latest CVS
Feb 17 15:20:03 WARNING[15317]: config.c:819 read_config_maps: The
'sipfriends' table is obsolete, update your config to use sipusers and
sippeers, though they can point to the same table.
== Binding sipusers to mysql/asterisk/sip
== Binding sippeers to
Mark Benson wrote:
Yesterday I asked about a user manual - ie a user guide to actually
using asterisk (now on how to set it up) the doc project (v2) isn't
anywhere near complete and is the closest thing I could find.
Does anyone know of such a doc? The reason I ask is that while a lot of
this
Muhammad Muzzamil Luqman wrote:
I have successfully installed and configured the asterisk, the incoming and the
outgoing calls are working fine, its a tremendous solution :)
Now i want to enable trunking between two asterisk boxes, in the iax.conf i
have put:
[karachi]
...
...
...
trunk=yes
...
If your X-ten phones are on the same lan as asterisk then try nat=no.
David
On Thu, 2005-02-17 at 07:28 +0300, Julius Kidubuka wrote:
My sip.conf file;
[luke]
type=friend
host=dynamic
username=luke
secret=luke
;dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info
dtmfmode=rfc2833
Thomas RULMONT wrote:
Hi everybody,
I have a problem with voicemail:
I have two TDM boards for a total of 5 fxs and 3 fxo. One of the fxo is connected to the local tel provider and is redirected to a voicemail box.
When I call asterisk from outside, I leave my message, but, after hanging on,
--- Andrew Thompson [EMAIL PROTECTED] wrote:
--- snip ---
The only thing that seems out of place to me is your
#include in
[main_vp_context]. It looks to me like you intend
for the s, #, t, and i
extensions to be in [main_vp_context]. The way you
layed out this
example, that's not
When I do apply nat=no, the X-ten phones don't login at all!
If your X-ten phones are on the same lan as asterisk then try nat=no.
David
On Thu, 2005-02-17 at 07:28 +0300, Julius Kidubuka wrote:
My sip.conf file;
[luke]
type=friend
host=dynamic
username=luke
secret=luke
MB The Makefile seems to check most places for mysql libraries but check it
MB again to make sure. Also make sure your mysql lib path is in ld.so.config
MB then rerun ldconfig. (Oh..do that before you do the above commands)
That was the problem, tnx !
P.S.
Any skill in realtime ?
I'm
Peter Svensson wrote:
Ok, then INFORMATION with keypad IE needs to be handled differently from
IE called number.
This is what it looks like with pri intense debug enabled:
Informational frame:
SAPI: 00 C/R: 1 EA: 0
TEI: 000EA: 1
N(S): 116 0: 0
N(R): 126 P: 0
8 bytes of data
--
Has anyone figured out how to make a Sipura to dial an extension
automatically as soon as you pick the the handset?
I need to make all my users go thorugh a menu to place a call. Users should
not be able to dial directly, only through the menu.
Any ideas?
O.A.
As far as I am aware there isnt a way for * to
receive/send audio to a multicast group. There needs to be a way
for Asterisk to tell the phone which ip multicast group to join in order to
receive the page. This method varies by vendor. I know that
with Cisco ip phone multicast paging
Hi all
I want to connect Asterisk with my Siemens HiPath PBX, to use it as a
Gateway to the PSTN.
I already have a ISDN entry in the Siemens HiPath, but the PC with Asterisk
doesnt have any ISDN board, can someone tell me about good and cheap PCI
ISDN boards that supports QSIG?
The Eicon
[EMAIL PROTECTED] wrote:
IS Anything changed?? Missed something?
You're running head and not watching -dev?
How should the iaxpeers and sippeers tables look like then?
This message was posted to asterisk-dev recently:
http://lists.digium.com/pipermail/asterisk-dev/2005-February/009445.html
--
I'm not using any Digium cards. I'm actually using SpanDSP and
app_rxfax to process incoming faxes.
After drilling into it for about 8 hours yesterday I come to realize
that there is a lot more to it than the asterisk upgrade.
I patched my FC3 box, which means libtiff is now 3.6.1 which
On Wed, Feb 16, 2005 at 08:58:41PM +0100, Robert Rozman wrote:
Hi,
I'm trying to get capiECT working. I'd like to transfer call to another ISDN
router connected extension and free channel from router to Asterisk.
I get incoming call on CAPI and would liek to transfer it to dialed local
On Thu, Feb 17, 2005 at 03:02:07PM +0100, Marc SCHAEFER wrote:
Hi,
I have two HFC-s boards I configured in NT and TE mode respectively.
When I connect the two boards together, I can dial extensions and I
see the correct called and caller ID numbers:
-- Executing SetCallerID(Zap/2-1,
beonice wrote:
Yes, I see what you are saying. This sounds backwards,
but it's actually doing what I _want_ it to do. :)
From what I see in the dialplan, what asterisk does
is, it loads the handlers for '#', 't' and 'i' as part
of vp_context, not as part of main_vp_context. That
actually happens
No kidding, every time.
I know I have the config via tftp working. Funny
story - I was getting nowhere with it and then decided
to tcpdump on the tftpd box, and wow! The UIP-200
tftp client was looking for the unidenmac.txt in
lower-case! Hah!
That was easy to fix. Now the config is
Look at an EXTENSIONS RELOAD and make sure the include is being parsed
-- and not throwing file not found errors. I broke my include
functionality last week by reMAKEing and not paying attention to a
known bug in the #INCLUDE function that existed in non-HEAD versions.
/rg
On Feb 16, 2005, at
Hi all
I want to connect Asterisk with my Siemens HiPath PBX, to use it as a
Gateway to the PSTN.
I already have a ISDN entry in the Siemens HiPath, but the PC with Asterisk
doesnt have any ISDN board, can someone tell me about good and cheap PCI
ISDN boards that supports QSIG?
The Eicon
On Thu, 17 Feb 2005, Deti Fliegl wrote:
Protocol Discriminator: Q.931 (8) len=8
Call Ref: len= 2 (reference 7/0x7) (Originator)
Message type: INFORMATION (123)
[2c 01 31]
Keypad Facility (len= 3) [ 1 ]
Feb 16 11:42:25 VERBOSE[2975]:
[ 02 01 e8 fc 08 02 00 07 7b 2c 01 31 ]
see
I remember reading some people were talking about being able
to use packet 8 without the ATA (I currently connect via an X100P card).
Did this ever get anywhere?
The wiki doesnt have any information on this
lots of referrals but thats it.
On Thu, 17 Feb 2005, Oswaldo Arratia wrote:
Has anyone figured out how to make a Sipura to dial an extension
automatically as soon as you pick the the handset?
I need to make all my users go thorugh a menu to place a call. Users should
not be able to dial directly, only through the menu.
Oswaldo Arratia wrote:
Has anyone figured out how to make a Sipura to dial an extension
automatically as soon as you pick the the handset?
Go to google and type: sipura hotline
Read the first three links.
Test.
Send us a note telling what worked for you.
--
Andrew Thompson
http://aktzero.com/
dean collins wrote:
I remember reading some people were talking about being able to use
packet 8 without the ATA (I currently connect via an X100P card).
Did this ever get anywhere?
Packet8 made changes at least a year ago that prevents this. Just
like Vonage did.
Jerry wrote:
On Feb 16, 2005, at 7:19 PM, Eric Wieling wrote:
Jerry wrote:
On Feb 16, 2005, at 3:07 PM, Eric Wieling wrote:
I have the following configuration:
CLEC - T-1 - Asterisk - Adtran Channel Bank - (analog) - Nortel
Don't complain that it's ugly. I've already done plenty of that.
The CLEC
Thanks, I will begin my testing
Erick
- Original Message -
From: Race Vanderdecken [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Wednesday, February 16, 2005 8:18 PM
Subject: RE: [Asterisk-Users] Help Please
Hi Keith,
I have a TFTP server set up with the proper files on it, but after a
factory reset, how does the phone know where to find the TFTP server..?
I cannot get into it to set the TFTP server IP address.
Thanks
On Wed, 16 Feb 2005 20:02:22 -0500
Keith O'Brien [EMAIL PROTECTED] wrote:
It
Thanks for the headsup and saving my time.
It's a great service, still highly recommended I use 2 of them here
Guess I'll just have to stick with running connections to the ATA's via
X100P
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
how does the phone know where to find the TFTP server..?
Dude, option 150 in your DHCP server:
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note09186
a00800942f4.shtml
We use the same option for our Mitel phones. HTH.
___
Hello,
I have a crossover PRI(Asterisk server to PBX) and a regular telco PRI T1
line and currently have echocancel=yes and echocancelwhenbridged=yes on
those spans in zapata.conf. I was discussing CPU load with another Asterisk
user and he mentioned that PRIs don't need echo cancelation and that
Has anyone figured out how to power a Digium TDM 400P card
in a Dell 1750 server? I opened the server and noticed that there is no
access to 4 pin power to power the card. Is there some sort of adapter that I
need to power the Digium card in a Dell Server? I see that the 1750 is listed
on
For anyone playing around with IAXy(S100i) devices, I am making the
following available:
Windows IAXy Provision v1.00
This is a from-the-ground-up development of a means of provisioning IAXy
devices using a Windows environment. For some users, being bound to Linux
for IAXy provisioning is not
On Thu, 17 Feb 2005 09:42:54 -0600, Eric Wieling [EMAIL PROTECTED] wrote:
Asterisk lacks good documentation. The documentation that is
available is fragmented. This is bad. Fortunately, we are seeing a
slow consolidation of documentation. SineApps is now syndicating the
updates information
Keith O'Brien wrote:
Has anyone figured out
how to power a Digium TDM 400P card
in a Dell 1750 server? I opened the server and noticed that there is
no
access to 4 pin power to power the card. Is there some sort of
adapter that I
need to power the Digium card in a Dell Server?
Wow. This list is high traffic Just to add to the noise, here's
some of my extensions.conf that implements what you are talking about.
In particular, the macro featureexten takes an argument that is the
same as the context the user uses for outbound dialing. The result
being that whatever
On Thu, 17 Feb 2005 16:08:26 +0500, Muhammad Muzzamil Luqman
[EMAIL PROTECTED] wrote:
It was missing the kernel-source rpm. I installed the version that i found
but now the first error is still there and when i modprobe ztdummy it gives
the following response.
Hm, do you have the right settings in zapata.conf? (switchtype,
pridialplan...)
So, in Switzerland, I assume
switchtype = euroisdn
now, for the pridialplan, am I right that the pridialplan configures the
way the phone number to be dialed (called ID) is sent, and that the
prilocaldialplan
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