Chris A. Icide wrote:
On 07:12 PM 8/16/2004, John Todd wrote:
g stupid top-posting confuses the hell out of these threads,
but I'll continue the insanity.
I _swear_ I already brought this problem up and it got resolved, like
1.5 years ago. I explicitly remember talking with kram about
However, I tried the patch ## transfer into CVS when? Plus Suggestion.
Attendant Transfer possible..
But they have the same problem to keep the code up to date.
To my best knowledge: currently not. If the phone can't do it, you're
lost.
Asterisk currently has blind transfer built in, this
On Tue, 17 Aug 2004, Roland Zagler wrote:
Hello! has anyone already successfully installed Digium TE410P card on
RedHat Enterprise Server 3.0?
Yes, it works very well. No problems at all.
Peter
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hello,
if anyone is using asterisk as a voicemail system for SER I would be
grateful if i could see a working ser.cfg and extensions.conf of such a
setup. I am having some issues with rollover to voicemail when busy, and in
setting up a VM extension for users to retrieve their mail without having
(sorry, posted without subject)
hello,
if anyone is using asterisk as a voicemail system for SER I would be
grateful if i could see a working ser.cfg and extensions.conf of such a
setup. I am having some issues with rollover to voicemail when busy, and in
setting up a VM extension for users to
I'm reading a lot about Call Parking and Conference Room features of
Asterisk, but don't know how to set up in RC2. For a start,
parking.conf is missing, and also MeetMe application doesn't exist...
parking.conf has been renamed into features.conf
MeetMe is there, it's in in
Hi All,
Just a heads up - I was looking around the Cisco FTP a little while
ago and noticed that the SIP 7.2 images for Cisco IP Phone 7940/7960
were released yesterday (16th August).
No new features - all bug fixes according to the release notes. I've
already started using it.
I thought those
On Mon, Aug 16, 2004 at 04:46:56PM -0500, Lex Lethol arranged a set of bits into the
following:
I recently bought a 7910. I found out too late that it would not do
SIP as I initially thought. Anyway before ditchingit for a 7960 I
wanted to try it out, I read that the guys at
Hi,
We try to start TDMoE but the result is that the Asterisk and the
Network are crashed.
Are there some successful stories with TDMoE? Any help will be very
useful.
Best Regards,
Miroslav Nachev
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They seem to be very good ! but where the hell could we buy them in Europe !?
Michael
-Oorspronkelijk bericht-
Van: [EMAIL PROTECTED] namens John Todd
Verzonden: di 17/08/2004 4:56
Aan: [EMAIL PROTECTED]
CC:
Onderwerp: RE:
Surely, 802.1Q wasn't designed with security in mind...change tagging,
change vlan...
Cheers,
Simone.
Steve Szmidt ha scritto:
Thus the Virtual part of VLAN. Though it's still a very good idea, from a
security standpoint, to keep them apart. You do not want to have your LAN
owned because of
On Tue, 17 Aug 2004, Simone Ricci wrote:
Surely, 802.1Q wasn't designed with security in mind...change tagging,
change vlan...
I have not used any ip-phones with vlan support but several switches. They
all allow you to confgiure which tags are accepted on which ports. Can the
phones be
hi all
I keep getting 'bzzt'-sounding dropouts, but I've only heard them when
from behind NAT. I don't think it's related to bandwidth, as I'm on a
2400/640 ADSL line. I'm currently using ALAW. Any ideas how to debug
this? SIP DEBUG doesn't show me anything when dropouts happen, although
I'd
Hi Julien,
Thanks for the feedback. I am currently trying to compile with gcc-2.96-113
I've been trying all day to get CVS from sf.net to try to compile from
latest version. Hope they fix it soon :S.
Lethol
On Tue, 17 Aug 2004 16:35:11 +1000, Julien Goodwin
[EMAIL PROTECTED] wrote:
On Mon,
I don't really think so. They only can tag traffic they generate, AFAIK.
My ATA-186 from cisco has that functionality, but never tried myself.
Peter Svensson ha scritto:
I have not used any ip-phones with vlan support but several switches. They
all allow you to confgiure which tags are accepted
I've got a problem with DTMF, again.
My asterisk box is connected with the outside world (PSTN) via a sip
proxy. The problem is that for some reason, I need to use rfc2833 for
signaling digits to the gateway and inband to accept digits from outside
(eg. when someone dials one of our DIDs). It's
On Tue, 2004-08-17 at 08:54, Michael Devenijn wrote:
They seem to be very good ! but where the hell could we buy them in Europe !?
Michael
[snip]
If I understood Polycom's EMEA Product Manager correctly they have a
number of technology partners and you can only buy their phones when you
On Tue, 2004-08-17 at 02:17, Tom Masterson wrote:
What we are finding is that things work quite well with a small number of
users/agents and callers i.e 10 or less. However when we put the stuff in
production (normally changes to the configurations) where they can and are
hit with hundreds
Hi,
I installed TE410P card on RedHat9.0 successfully. If u want any help contact me.
On Tue, 17 Aug 2004 Roland Zagler wrote :
Hello! has anyone already successfully installed Digium TE410P card on
RedHat Enterprise Server 3.0?
Roland Zagler
mailto:[EMAIL PROTECTED]
@fog smart
Julien,
Just to let you know that I manually included your patch and
everything compiled OK.
I'll begin testing now.
Thanks!
Lethol
On Tue, 17 Aug 2004 02:02:58 -0600, Lex Lethol [EMAIL PROTECTED] wrote:
Hi Julien,
Thanks for the feedback. I am currently trying to compile with
Hi Tom-
I wrote that (rudimentary) Perl script last year to simulate traffic from
one system to another, although it can also be used between spans on the
same machine. It's much better to have the load generated on a separate
system , for obvious reasons. A couple of things: the traffic
Hi all,
Can anyone please mail me the files from ftp.opencall.org (app_rxfax.c
app_txfax.c, Makfile.patch, ...), because I need them desperately and
the server seems to be down since yesterday. :-/
Thank you and sorry for the inconvenience.
Manfred
___
Hi,
The company i work for has been active with asterisk for more than a year
and have done plenty of * installations .
Their VOIP team is based in delhi.
Navnit
- Original Message -
From: Kannaiyan Natesan [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, August 14, 2004 2:07 AM
Hi,
thanks for response. I don't have Zaptel HW, but have read that some kind of
SW timer is available (zt_dummy or similar).
I wonder what is its performance and how to compile it...
Asterisk Conference room is feature that is too nice not to have it ...
Regards,
Robert.
- Original
I get this warnings while compiling zaptel
Any suggestions would be very helpfull.
make -C /usr/src/linux-2.6 SUBDIRS=/usr/src/bri-stuff.0.1.0-RC4/zaptel
modules
make[1]: Entering directory `/usr/src/linux-2.6.8.1'
CC [M] /usr/src/bri-stuff.0.1.0-RC4/zaptel/zaptel.o
Greg Hill schrieb:
I can makes calls with my asterisk using X-lite softphone, I can even
call the 877 number and it works perfectly! But when I call the 877
number over the PSTN, it does nothing. A busy signal.
Here in Germany when you order a line with 3 digit extensions you often
only get
I think most people just tag the frames for priority by using VID 0
- Original Message -
From: Simone Ricci [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, August 16, 2004 12:15 PM
Subject: Re: [Asterisk-Users] New $89 VOIP phone
That's not true: with some equipment you can use
Hi Shaun,
Saw you post, and rushed to their ftp-server and downloaded it :-)
But, I can't make my phone (7940) upgrade, so maybe you can give me a hint.
I added the files to my tftpd folder, changed the version-number in the file
OS79XX.TXT - from P003-07-1-00 to P003-07-2-00
In my
Typo in your OS79XX.TXT P00 ? instead of P0S !?
-Original Message-
From: Michael Løjtnant [mailto:[EMAIL PROTECTED]
Sent: 17 August 2004 13:31
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Cisco 7.2 firmware for SIP 7940/7960
released
Hi Shaun,
Saw you post, and rushed to their
Ok please ignore me, I just tried 7.2 myself and worked fine with the same mods you
made !?
-Original Message-
From: Michael Løjtnant [mailto:[EMAIL PROTECTED]
Sent: 17 August 2004 13:31
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Cisco 7.2 firmware for SIP 7940/7960
released
After i removed my SEPMAC.cnf.xml file it successfully upgraded :-)
On Tue, 17 Aug 2004 13:30:38 +0200
Michael Løjtnant [EMAIL PROTECTED] wrote:
Hi Shaun,
Saw you post, and rushed to their ftp-server and downloaded it :-)
But, I can't make my phone (7940) upgrade, so maybe you can
Title: Message
Hi,
I have some queries
regarding Asterisk. They are as follows, any help would be greatly
appreciated.
1.We would
like to download modify Asterisk's sources distribute the binaries.
Is there any constraint on commercial use of Asterisk. Where can I get
more
Title: Message
Thanks to Greg Hill
for pointing me to the 'sip debug on' cmd that helped me resolve the sip
connection problem!
The other issue I'm
trying to resolve is configuring outgoing calls. I need to configure outgoing
calls to use the FXO card in the PBX (zaptel device) via sip
Hello,
I am trying to compile the Zaptel drivers for an embedded project using
the Bering 1.2 kernel(2.4.20). All compiles well however receive a
number of unresolved symbols when I try and load the modules. They look
like kernel things, anyone clue me in on what the dependency may be or
have I
Are there any considerations to take in account when faxing from analog
to SIP using ULAW? The problem we're having is faxes are only making it
halfway, getting cut off. Neither fax machine seems to report an error.
Pretty diagram:
FXS -- SIP -- PSTN Provider -- FAX
^ULAW
With regards to the XML documentation, I should be getting that today.
If you're looking for a vendor, I can recommend ReviewVideo. They were
very helpful with my purchase and also with support. Here's my contact:
Michael F. Sweetman
P: 630-388-0491
email: [EMAIL PROTECTED]
I've done quite a
On Tuesday 17 August 2004 09:33, Matt Schulte wrote:
Are there any considerations to take in account when faxing from analog
to SIP using ULAW? The problem we're having is faxes are only making it
halfway, getting cut off. Neither fax machine seems to report an error.
Pretty diagram:
FXS --
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hmm,
My music on hold has always worked fine. But I discovered that under incoming
IAX2 calls they don't get any MOH! All I could find was a comment saying let
me know if you find a solution... Nor does the debugger does say:
Started music
Title: Message
Nevermind. Figured this out. I needed the following in
extensions.conf to enable outbound dial.
exten
= _9.,1,Dial(Zap/2/${EXTEN:1},70,Tt)
Thanks
-Original Message-From: Info
[mailto:[EMAIL PROTECTED] Sent: Tuesday, August 17, 2004 9:27
AMTo: '[EMAIL
On Mon, 2004-08-16 at 16:39, Derek Listmail Acct wrote:
Has anyone been able to get the minibrowser on the Polycom SoundPoint IP
500/600 phones working? If so could you share the relevant sections of
your config with me?
According to the docs the microbrowser ONLY works on the 600. I only
Dear all,
I have an Asterisk box serving as gateway to a set of POTS phones that
all share the same telephone number, and I register this box as gateway
to a H.323 gatekeeper with the telephone number as the gateway prefix.
I am wondering if there is a way to have Asterisk-wide (perhaps to be
A newbie on Zaptel asks:
Is there any way I can force CPC on a Zap channel - Digium TDM four port FXS card?
/O
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Whenever I do a Show version, I get CVS-02/10/04. I have updated and
recompiled Asterisk a number of times and it comes up clean. Should'nt this
show the lastest version number?
I have been using CVS download, then make clean;make install. I have also
tried make clean; cvs update; make; make
Aaron Johnson wrote:
I attempted to update an Avaya 4602 phone to the latest SIP firmware
and now the phone stops at the bootloader. It keeps requesting an
appsip.ebin file from my HTTP server and is no longer checking my TFTP
server for update files. Since no appsip.ebin file was included in
Martin Keding a écrit :
Whenever I do a Show version, I get CVS-02/10/04. I have updated and
recompiled Asterisk a number of times and it comes up clean. Should'nt this
show the lastest version number?
Yes. You have to stop * before building the new version.
I have been using CVS download, then
Martin Keding wrote:
Whenever I do a Show version, I get CVS-02/10/04. I have updated and
recompiled Asterisk a number of times and it comes up clean. Should'nt this
show the lastest version number?
I have been using CVS download, then make clean;make install. I have also
tried make clean; cvs
If you're looking for a vendor, I can recommend ReviewVideo. They were
very helpful with my purchase and also with support. Here's my contact:
Michael F. Sweetman
P: 630-388-0491
email: [EMAIL PROTECTED]
I would not recommend this vendor, they shipped us a bunch 500's with the
wrong
Guys,
For what it's worth, after months of trying to troubleshoot issues with
them, and after paying them around $2500 for setup and a down payment
(it's unclear what of that will be refunded, if any) BroadVox --
http://www.broadvox.net/ -- decided to terminate our contract without any
valid
Trevor thanks,
I think this solved the problem.
The only think where I am not happy with
is that I got one internal segmentation error when I compiled Asterisk.
At this moment everything is working and I
am now testing the stability.
And I made an additional softlink
+++ Vikram Rangnekar [13/08/04 13:35 +0200]:
Does anyone know if the E1 cards that digium sells work in India. Also are
there any distributers for those cards in India. By E1 cards I mean E100P,
TE410P or TE405P
--
regards
Vikram (http://www.vicramresearch.com)
I dont agree with this view of yours below about the usefulness of ethernet ports.
but they're 10mbit, so they're essentially useless,
unless you plug in a printer or something slow
All that you need is 64Kbps at the highest on these ports to
establish a Voice call. Even if you are running a
Are there any known issues with using the Digium licensed G729 codec with
the OH323 channel?
-Chris
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According to the docs the microbrowser ONLY works on the 600. I only
have 500s so have not messed with it.
That was part of my problem.
I can now get the 600 to download XML, I tried using
http://phone-xml.berbee.com/menu.xml and the phone displays XML Error
(1,0) syntax error. I'm guessing
Suse 9.1 keeps being unstable.
When I compile asterisk I get on random
positions segmentation fault.
gcc -shared -Xlinker -x -o app_playback.so app_playback.ogcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT
quote who=Kanuri, Seshu
All that you need is 64Kbps at the highest on these ports to
establish a Voice call. Even if you are running a Video Phone,
all you need may be 512Kbps at 32Fips.
Why do you need any thing that is capable of a larger bandwidth?
He wasn't talking about it being useless
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Tuesday 17 August 2004 01:13 pm, Kanuri, Seshu wrote:
I dont agree with this view of yours below about the usefulness of ethernet
ports.
but they're 10mbit, so they're essentially useless,
unless you plug in a printer or something slow
All
I have the same phone sitting here... Not worth $89, IMHO. If does have
two eth ports, but they're 10mbit, so they're essentially useless,
unless you plug in a printer or something slow. Only one call
appearance, and (with my software) the caller-name didn't work, only
calling number.
Do what it says! Replace instances of pthread_create() with
ast_pthread_create instead! :)
Go into chan_oh323.c and replace all instances of pthread_create() with
ast_pthread_create(), then recompile and load... it should work...
-Chris
- Original Message -
From: Krystian.Filiks
Hello all-
So I have * up and running and connected to a legacy system via em_w lines
and have no trouble dialing from * through the tie line but from the PBX
across the tie line I am having intermittant receipt of the DTMF. T-Berd
testing is showing that the digits are coming across but * is
Johannes:
I've compiled Asterisk and drivers on SuSe 9.1 several times without
problem, use several different machines.
If you are getting segmentation faults at random points each time, I
would STRONGLY suspect the problem is bad hardware; most likely faulty RAM.
Robert
Johannes van Hulst
Sorry for the OT but where do you all go to buy this stuff?
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My desire to run Asterisk is finally giving me the reason to install a Linux box at home.
Is RH9 the onlydistro that Asterisk will run on, and can anyone recommend a good source for a cheap Linux (RH9) box?
For example, walmart.com has microtel boxes with no OS. Will RH9 and Asterisk run on
On Tue, 2004-08-17 at 14:45 -0300, Johannes van Hulst wrote:
Who has the same experience with suse linux or who can help me with
resolving this problem
I don't use Suse but last year I had a Mandrake system seg faulting on
every compile. If I just carried on with make everything did eventually
Linux will run on a variety of boxes with many different processors... Linux
will even run on PocketPC and PALM! If you have an old machine lying around
it will work perfectly for running Linux...
However... I hate to tell you this, but Asterisk is not well suited to
someone who doesn't know
I have a question about how Asterisk Parses the Dial Plan.
To create a hunt-group which would be the appropriate dial plan:
[CompanyABC]
exten = 722,1,Dial(SIP/801722,60,r)
exten = 722,102,Dial(SIP/8014361234,60,r)
exten = 722,103,Dial(SIP/8014362345,60,r)
exten =
Ebay is your friend. Dell / HP
/ IBM / Compaq workstations and servers are available for essentially 1
handful of dirt. I have run RedHat on all of the above.
Tom Meeks
Technology Group of Dixon Hughes PLLC
[EMAIL PROTECTED]
John Williams [EMAIL PROTECTED]
Sent by: [EMAIL PROTECTED]
On Tue, 17 Aug 2004, John Williams wrote:
My desire to run Asterisk is finally giving me the reason to install a
Linux box at home.
Is RH9 the only distro that Asterisk will run on, and can anyone
recommend a good source for a cheap Linux (RH9) box?
For example, walmart.com has microtel
Why is everyone stuck on the ethernet port (or lack thereof)? That's like
the least important feature of the phone! Besides the switches built into
many of these multiport phones are probably RealTek chipsets or worse
anyway... You couldn't pay me to plug my ethernet card into one of
those...
What's the URL. I have 7960 with the old firmware, it works fine..but I
wouldn't mind to update to the latest/
Wojtek
- Original Message -
From: Shaun Ewing [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, August 17, 2004 2:28 AM
Subject: [Asterisk-Users] Cisco 7.2 firmware for SIP
I have just started using * and have been trying to set up MeetMe. So far I
have not been able to start a conference. When I dial the conference
extension (I am using X-Lite softphone), the call hangs up. I am not using
Zaptel cards so I uncommented the ztdummy in the Zaptel Makefile. My
Chaps,
I recently added an incoming VOIP account to my asterisk box. When the
PSTN number associated with this account is dialled, the call rings once
and then asterisk starts playing music on hold, even though all the
extensions continue to ring. Variations of answer() and ringing() don't
seem to
I'm trying to install the RxFax in Asterisk.
After much trouble getting app_rxfax.c(and friends) compiles and installed
into Asterisk (thanks to the many asterisk-users messages) it did get it
installed. I have been unable to receive a legible fax. app_rxfax
provides lots of debugging
Hi Guys,
I work with Dan and yes this whole Broadvox problem is a huge piece of
[EMAIL PROTECTED]
However, I've noticed that there is a useragent= cmd that can be used in the
sip.conf. Right now it's set to useragent=Asterisk PBX. I was wondering
if we change that to reflect a Cisco Gateway or
A solution to this very problem has already been discussed... in fact... in
this very thread!
There are several options (much better than plugging your PC into the phone
I might add) one being to buy a small desktop switch... Another would be to
install a switching wall plate like those from
Hi,
You need ztdummy. Follow this link, it worked for me,
http://www.voip-info.org/wiki-Asterisk+timer+ztdummy
Thanks,
Guru.
I have just started using * and have been trying to set up MeetMe. So
far I
have not been able to start a conference. When I dial the conference
extension (I am using
Hi,
You need ztdummy. Follow this link, it worked for me,
http://www.voip-info.org/wiki-Asterisk+timer+ztdummy
Thanks,
Guru.
I have just started using * and have been trying to set up MeetMe. So
far I
have not been able to start a conference. When I dial the conference
extension (I am using
Hi,
You need ztdummy. Follow this link, it worked for me,
http://www.voip-info.org/wiki-Asterisk+timer+ztdummy
Thanks,
Guru.
I have just started using * and have been trying to set up MeetMe. So
far I
have not been able to start a conference. When I dial the conference
extension (I am using
On Tue, 2004-08-17 at 15:44, Patrick Lidstone (Personal e-mail) wrote:
Chaps,
I recently added an incoming VOIP account to my asterisk box. When the
PSTN number associated with this account is dialled, the call rings once
and then asterisk starts playing music on hold, even though all the
There are several options (much better than plugging your PC into the
phone
I might add) one being to buy a small desktop switch... Another would be
to
install a switching wall plate like those from Cisco or 3com... Another
would be to think ahead and install another wall jack if your
On Tue, 2004-08-17 at 15:28, Gurdeep Singh Bagga Guru wrote:
Hi,
You need ztdummy. Follow this link, it worked for me,
http://www.voip-info.org/wiki-Asterisk+timer+ztdummy
While he may need ztdummy, this is the 4th time I received the same
exact message from you with differing timestamps
On Tuesday 17 August 2004 16:54, Derek Listmail Acct wrote:
Pulling another cable is fairly expensive, requires another port on the
switch, and having people on ladders stringing cable is usually a bit
annoying to the employees of the business.
That's why you use the existing telco wiring and
A matter of preference I suppose... I wouldn't count the number of ports or
the type of ports in the phone as a reason for buying... The phone could
come with a 4-port switch for all I care...
Small desktop switches generally don't have any QOS, tend to look bad
under a desk...
No phone that I
Speakers, pencil sharpeners, cocks, cellphone chargers, etc... All use
another power outlet also that's what a power strip is for...
*clocks* Oops... well that threw my whole argument...
-Chris
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Either way a decision needs to be made. There's no magic fairy gonna come
down and wiggle her pretty lil' ass over the walls and you magically have
dual Cat5e to every desk and some great POE-injecting switches upstairs.
:-)
LOL I like your answer better...
On Tuesday 17 August 2004 17:21, Chris Shaw wrote:
Speakers, pencil sharpeners, cocks, cellphone chargers, etc... All use
another power outlet also that's what a power strip is for...
*clocks* Oops... well that threw my whole argument...
HAHAHAHA
What kind of company are you involved
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Chris Shaw
Sent: 17 August 2004 21:12
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] New $89 VOIP phone
A solution to this very problem has already been discussed... in
fact...
Hi all,
Does anyone know if the fax detect will work over analog? I have a
Wildcard TDM400P with 4 lines in it and would like to use the fax detect
to forward fax calls to a fax machine on the other end of a channel bank
that is connected through a T1 on a Wildcard T100P.
Thanks,
I am making some changes to the dial plan at the request of the company
president and have run into some problems. I have a couple of layers of
menu's and I am not sure how to handle them.
Here is how it should work (sorry for the crappy diagram)
main menu
Dial 1 for support
|
Title: Nachricht
you
looking for:
http://www.voip-info.org/wiki-Asterisk+OS+Platforms
Is RH9 the onlydistro that Asterisk
will run on, and can anyone recommend a good source for a cheap Linux (RH9)
box
For example, walmart.com has microtel boxes
with no OS. Will RH9 and Asterisk run on
I'm not sure if this is your issue or not, but it looks like ext= 1,
starts over at the bottom of the 1's. You have 1,1-10 and then 1,1 and
2 after it. I can see how asterisk might get confused if you sent your
call back to ext 1 at starting point 1 or 2.
On Tue, 2004-08-17 at 15:03, defiance
On Tue, 2004-08-17 at 17:03, defiance wrote:
I am making some changes to the dial plan at the request of the company
president and have run into some problems. I have a couple of layers of
menu's and I am not sure how to handle them.
Here is how it should work (sorry for the crappy diagram)
On Tuesday 17 August 2004 18:03, defiance wrote:
exten = 1,1,SetCallerID(Toll Free No Cpub)
...
exten = 1,1,Playback(cpub-support)
Do you see a problem? 'cos I sure do...
You can use the numbers over again if you use Goto and jump to a different
context.
-A.
That makes sense, but how do I send it to each context?
chris
On Tue, 2004-08-17 at 17:07, Steven Critchfield wrote:
On Tue, 2004-08-17 at 17:03, defiance wrote:
I am making some changes to the dial plan at the request of the company
president and have run into some problems. I have a
CONTEXTS
[main context]
Dial 1 for support
| Dial 2 for special
| Dial 3 sales
| Dial 5 For sales
[support context]
; don't include main context
However... I hate to tell you this, but Asterisk is not well suited to
someone who doesn't know Linux... By your question I can tell that you DO
NOT know Linux...
Points well taken. People who ask will * work with whatever distro don't
seem to
yet grasp the difference between kern and distro;
On Tue, 2004-08-17 at 17:33, defiance wrote:
That makes sense, but how do I send it to each context?
use goto(newcontext,s,1)
On Tue, 2004-08-17 at 17:07, Steven Critchfield wrote:
On Tue, 2004-08-17 at 17:03, defiance wrote:
I am making some changes to the dial plan at the request of the
Hi,
I'd love to have multiple sound consoles on Asterisk (like to have
speaker+mic on each sound card as simple IP phone).
Is this possible?
Should I better try some of command line softphones ? Which one ?
Regards,
Robert.
___
Asterisk-Users
Thanks a million man that works beautifully, and thanks for giving me an
example, I am still pretty new at this so that helped alot.
chris
On Tue, 2004-08-17 at 17:31, William Glynn wrote:
CONTEXTS
[main context]
Dial 1 for support
| Dial 2 for special
Hello list,
I have started writing a little log_queue parser that will display stats
in a graphical way based on the involved queue(s) and a start/end date.
You can see a sample analysis here: http://demo.xcept.it/xc-ast/XC-AST.htm
Strings are in italian, but I guess the uploaded page it's easy
Answers of this kind are what turns people off to linux. It would appear
that some have the attitude of if you don't know the basics then stay
out. Rmember you once asked stupid questions as well and answer with
either the appropriate answer along with some education and pointers or
don't
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