Been trying to get this to go but nongo :-).
I'm asking for some guidance especially if I should not be doing this on
an RT kernel.
I've installed what is supposed to be all of the requred deps.
Some factors that may be adding to my problem are:
1. this is only a test.. it's a 32bit guest OS
much.
-Steve
Steve Gladden wrote:
2. This is ubuntu Studio which uses an RT (realtime kernel)..
There seems to be very little aout there regarding running asterisk on
RT linux... one woudl think this would have some benefits..
Big benefits.. I've always wondered.
But moreso in a nn
I REALLY like the Snom M3 DECT SIP base.
You can have up to 3 simultaneous calls through the base
and you can have up to 8 phones registered with it.
It's all web managed as well as http/s provisionable and has
this nice phone to line matrix where you can set which phones
ring on inbound calls and
Ah.. so Debian's asterisk build didn't include zaptel.
OK will give that try!
Thanks for the pointer.
Steve
On Sun, Mar 22, 2009 at 12:09:09AM -0400, Steve Gladden wrote:
I've recently installed the latest Debian Linux for powerpc onto
and old iMac (version A) the original iMac with a 233Mhz
I've recently installed the latest Debian Linux for powerpc onto
and old iMac (version A) the original iMac with a 233Mhz G3 processor
and 160MB of sdram.
The debian install went smooth and so the the apt-get install of
Asterisk 1.4.21
It appears to have no functioning zaptel or ztdummy module.
Just watned to kick this back out here one more time.
Anyone done this or ever looked into a work-around?
Thanks!
Steve
I have meetme working with BLF on polycom phones however when
meetme is not actually being used by anyone the 'status' of meetme
becomes idle.
Which the Polycom phone
I have meetme working with BLF on polycom phones however when
meetme is not actually being used by anyone the 'status' of meetme
becomes idle.
Which the Polycom phone sees and produces a clock symbol and FLASHING red
LED.
Are there any 'tricks' or work-arounds to change this status to something
If you wanna go low tech. down dirty you could also go with a conventional
POTS phone line 'loud ringer' device and simply hook it to an ata such as
a PAP2, and add the PAP2 to the ring group.
Why don't you put a PC in the storeroom with a softphone to be the loud
ringer? You could make
Thanks all very much for the help pointers.
I've found all of the documentation on asterisk (especially 1.2-1.4) to
be more than adequate, and the voip-info wiki to be almost complete for many
things I've had to do in the past.
I also back in 2004 was able to bring up several high end large
Of fly me out there and put me to work.
I'll hook the PAP2 to a relay/air valve, air compressor/tank and dual Amtrak
Train Horn!
I am looking for work!
How'd that be for a loud ringer?
Danny,
Thanks for the idea, I thought of it but I was looking for a more elegant
solution, and one
IF you are going to go the electronics/expert router and use audio from
the speaker of an IP phone such as a Polycol 650 (I love mine)
I'd take it even a setp further and use some kind of VOX circuit or LED
triggered relay signal and actually put either the low level audio or
the amplified speaker
New to Aserisk 1.6 and find the 'installation tutorials' seem low to non
existent.
You go to the main Asterisk page (digium.org) and really just old install
instructions for 1.2 are in the examples.
Download links only give you asterisk itself and not dahdi or libpri
which also are needed to run
I meant digium.com.
Yay for messups!
It's been one of those weeks.
Really.
New to Aserisk 1.6 and find the 'installation tutorials' seem low to non
existent.
You go to the main Asterisk page (digium.org) and really just old install
instructions for 1.2 are in the examples.
Download
?
If not is there another product PAID or FREE software or hardware that can
do this easily and reliably?
Thanks very much!
Steve Gladden
--
This message has been scanned for viruses and
dangerous content by MailScanner, and is
believed to be clean
Scratching my head and trying this.
Asterisk Version: Asterisk 1.4.21.2
Tried:
exten = 4771,1,ExtenSpy([EMAIL PROTECTED])
exten = 4771,2,Hangup
Also tried:
exten = 4771,1,Answer
exten = 4771,2,ExtenSpy([EMAIL PROTECTED])
exten = 4771,3,Hangup
Also tried many variations including option ,b
I
ChanSpy works fine..
Just have not been able to get any audio out of ExtenSpy
Not sure if that's useful info..
Maybe.
Scratching my head and trying this.
Asterisk Version: Asterisk 1.4.21.2
Tried:
exten = 4771,1,ExtenSpy([EMAIL PROTECTED])
exten = 4771,2,Hangup
Also tried:
exten =
to get these calls into 3 different contexts
as the registration lines are identical EXCEPT for the DID or
username.
On Fri, Jun 27, 2008 at 2:20 AM, Steve Gladden
[EMAIL PROTECTED] wrote:
In other words how to match a registration to a peer or inbound context
other
The scenario:
This is all done SIP with a VOIP provider (have to register to single IP)
We have two inbound DID numbers / Accounts.
We have to register each individually with the VOIP provider.
I'd like inbound from each registered account (DID)
to be able to come into a unique PEER or dialplan
The scenario:
This is all done SIP with a VOIP provider (have to register to single IP)
We have two inbound DID numbers / Accounts.
We have to register each individually with the VOIP provider.
I'd like inbound from each registered account (DID)
to be able to come into a unique PEER or dialplan
Thanks for the good pointers!
I tried the manual fix first which got me a step closer only to find
something else expecting config.h to be there.
I gave up and grabbed the latest SVN.
Been awhile since I've used that.
My first 1.4 box WOW! different!
Works well now ;-)
Thanks!
Steve
On
I keep running into the dead end that it can't find config.h in the source
tree.
It looks like newer kernels don't use it anymore.
Someone ran into this awhile back when compiling 1.2 and it looks as
though the issue was never resolved.
Any ideas?
Last time I tried this, I was on fedora core 5
Hey!!!
I spent an ENTIRE Day googling for this issue only to find your single
solitary response ( solution) here on the mailing list.
How in the heck did you know to do that? and as popular as IIS (is)
I wonder why the answer was so hard to find on the web.
If you have time, please let me
Is it a bug or is it me?
For the longest time I have been using the feature within voicemail to
call back a number by caller ID.
Never had a problem with it at all.
I just updated to the latest (stable) asterisk from asterisk.org
Option 3 (advanced) then 2 then 1
caller number 7347292615
and
Is it a bug or is it me?
For the longest time I have been using the feature within voicemail to
call back a number by caller ID.
Never had a problem with it at all.
I just updated to the latest (stable) asterisk from asterisk.org
Option 3 (advanced) then 2 then 1
caller number 7347292615
and
Is it a bug or is it me?
For the longest time I have been using the feature within voicemail
to call back a number by caller ID.
Never had a problem with it at all.
I just updated to the latest (stable) asterisk from asterisk.org
Option 3 (advanced) then 2 then 1
caller number 7347292615
and
Is it a bug or is it me?
For the longest time I have been using the feature within voicemail
to call back a number by caller ID.
Never had a problem with it at all.
I just updated to the latest (stable) asterisk from asterisk.org
Option 3 (advanced) then 2 then 1
caller number 7347292615
and
,Hangup()
I didn't test this code, but this is my tip the main idea is that you
need to catch de DID and make a GoTo for the context you want.
Best regards,
Marco Mouta
On 4/2/06, Rich Adamson [EMAIL PROTECTED] wrote:
Steve Gladden wrote:
What version of asterisk? (been lots of changes
()
exten = 1234,2,Playback(vm-goodbye)
exten = 1234,3,Hangup()
I didn't test this code, but this is my tip the main idea is that you
need to catch de DID and make a GoTo for the context you want.
Best regards,
Marco Mouta
On 4/2/06, Rich Adamson [EMAIL PROTECTED] wrote:
Steve Gladden wrote
Hello!
I've been struggling with the documentation for months on this simple
subject...
I still have not been able to get this concept down...
I have 3 sip accounts (PSTN DID's) that come into my asterisk box
and give me phone service from my itsp via SIP.
I for the life of me have not been
What version of asterisk? (been lots of changes happening to the sip
code over the last year)
SVN-branch-1.2-r9156
I think what I am trying to do is pretty basic and should not have changed
much in the past year.
I got started in July of 2005 and I upgrade about once per month.
In all this
Sorry for being a bit of a newbie here but I find the
docs or README for downloading the G.729 codec from Digium
are not as detailed as I would like or just don't really
break down the different versions to a point that I am clear
on which one to grab.
The choices for 32bit are:
drwxr-xr-x3
And also curious why a K6-3? but not a K6-2?
And then of course why do we get a K6-3 but not an Athlon?
:-)
Steve
Sorry for being a bit of a newbie here but I find the
docs or README for downloading the G.729 codec from Digium
are not as detailed as I would like or just don't really
break
Not tried 1.2.4 yet I'm using 1.2.3 and an old version of
mpg123
You should be able to use any streaming mp3 that you can find on shoutcast
for test.
http://www.shoutcast.com
Click one of the 'tune in buttons' to download a playlist (pls) file
and open in your favorite text editor.
Or let it
!
Steve
Steve:
I'm picking up the tail end of a thread, so apologies if this is
offtrack...
Have you perhaps got an old set of EXECUTABLES in your path, that are
being picked up before your newly compiled ones?
Roger
Steve Gladden wrote:
Yes I have.
I have been battling this issue
Trying to compile asterisk (again) from scratch.
I seem to be still experiencing the effects fro Jan 25 where I get no sip
to sip audio.
I have tried upgrading to 1.2.3 which has made no change in the
problem.
I am starting over and now trying to compile/install /trunk
zaptel
libpri
asterisk
Thanks Tony!
You are (of course) absolutely correct.
I feel like an idiot for doing that when I know better.
Take care
Steve
In article
[EMAIL PROTECTED],
Steve Gladden [EMAIL PROTECTED] wrote:
I am starting over and now trying to compile/install /trunk
zaptel
libpri
asterisk
Sure enough we lost ALL sip-sip audio on 1-25
Pulled my hair out for hours before looking here or at the website
to find this problem reported...
Very greatful to find this I have upgraded to 1.2.3 but
still have no sip-sip audio!
what?!
Now I'm back to contnued hair pulling what culd I
Yep, tried that.
blew away all my source code, re-downloaded re compiled and re installed.
it's behaving exactly the same, calls go through but no audio in either
direction for sip-sip calls on the LAN or to-from the Internet SIP
providers tested.
I'm at a loss I feel like I have tried
No Firewalls involved, the test has been simplified down to two sip phones
on a LAN and still no audio.
For waht it's worth IAX2 still works fine.
Steve
-
Yep, tried that.
blew away all my source code, re-downloaded re compiled and re
installed.
it's behaving exactly the same,
Yes I have.
I have been battling this issue since wednesday 1-25
And so far have tried many things.
Have also tried RTP debug and do not see ANY RTP when the call is made.
I will keep working at this until I figure it out but right now am very
stumped and frusterated.
The software update SHOULD
and
am trying not to leave anything out.
Thanks for your time!
Steve
Steve:
I'm picking up the tail end of a thread, so apologies if this is
offtrack...
Have you perhaps got an old set of EXECUTABLES in your path, that are
being picked up before your newly compiled ones?
Roger
Steve
Hello and thanks for replying!
Steve,
The mission is to actually get a reinvite to work on the lan.
There isn't anything special to get this working... normally. I trust
you verified the traffic flow with a network monitor tool (tcpdump?),
Actully ethereal,
It is encouraging to hear that
It's been about 2 months since I have updated my asterisk box.
I was running CVS HEAD and I notice a whole lot has changed since
my last update!
I'm running Debian Sarge up to date on a 2.4 Kernel.
I was updating about every 2 or 3 weeks and never had any problems
compiling
Bummer - Possibly a bug
The stable stuff compiles and runs fine :(
Steve
-
It's been about 2 months since I have updated my asterisk box.
I was running CVS HEAD and I notice a whole lot has changed since
my last update!
I'm running Debian Sarge up to date on a 2.4 Kernel.
I was
been testing with a rather simple setup.
The mission is to actually get a reinvite to work on the lan.
I am trying with two sipura phones G.711 codec forced on both
both on the lan no nat no fancy options suchs as tT or H
No matter what we do asterisk hangs on to the media path, how
in the
please turn on all the debug, warning, error etc messages in the
console, see logger.conf, then type sip peer peer1 debug and sip
peer peer2 debug to see the SIP messages.
How are you testing if asterisk is in the media path?
Regards
On 1/23/06, Steve Gladden [EMAIL PROTECTED] wrote:
been
Anyone know if this can be made to work?
I've only been able to get SIP-SIP call pickup to work.
Steve
---
as far as I know, no.
Il lun, 2004-07-05 alle 18:56, Adolfo R. Brandes ha scritto:
I've looked in the obvious places but haven't found a definitive
answer to the following:
Anyone know if this can be made to work?
I've only been able to get SIP-SIP call pickup to work.
Steve
---
as far as I know, no.
Il lun, 2004-07-05 alle 18:56, Adolfo R. Brandes ha scritto:
I've looked in the obvious places but haven't found a definitive
answer to the following:
://edvina.net/broadvoice/
/Olle
Steve Gladden wrote:
Sorry this is a bit of a newbie question, I've been at this for a few
months and still have not quite figured this one out.
I've been able to setup one itsp (incoming calls) (sip account) with a
register line like this:
register
Hello,
I'd like to know if it is possible to get * to listen and respond on more
than just one single udp port.
I've run into several situations where I'd like IAX to work on an alternate
port as well as be able to work on the standard port.
I'm wondering if there is a way to do this?
Thanks!!
Yep thanks for the reply,
I figured out pretty quickly after one test that the /s did not work.
The issue remains that I have been unsuccessful in getting an incoming
call to come into any other context other than the one specified in
sip.conf [general] section
Anything I'm missing here?
I
Sorry this is a bit of a newbie question, I've been at this for a few
months and still have not quite figured this one out.
I've been able to setup one itsp (incoming calls) (sip account) with a
register line like this:
register = nnn:[EMAIL PROTECTED]
-or-
register = nnn:[EMAIL
Thank you for your reply and your help.
I am still confused here and apologize.
To some degree I still do not know what I am doing.
We use 2 ITSP's and one of them we have multiple SIP accounts
on so I will not be able to do this by IP address.
For incoming calls we use a register line in the
secret=password
context=my-incoming-context-3
;Disable canreinvite if you are behind a NAT
canreinvite=no
;Don't try to authenticate on incoming calls
insecure=very
Steve Gladden wrote:
Sorry this is a bit of a newbie question, I've been
on incoming calls
insecure=very
Steve Gladden wrote:
Sorry this is a bit of a newbie question, I've been at this for a few
months and still have not quite figured this one out.
I've been able to setup one itsp (incoming calls) (sip account
This was just a recent personal experience
Maybe I missed a thread on this:
We recently installed asterisk (CVS-HEAD) on a debian system using 2.6
kernel and the enhanced RTC for all timing.
Also a custom compiled kernel for the CPU on the box (P4).
We had a strange thing happen in that
% 99.975586%
99.975586% 99.975586%
--- Results after 49 passes ---
Best: 99.987793 -- Worst: 99.975586
--
On Sat, Sep 24, 2005 at 10:23:42PM -0400, Steve Gladden wrote:
On the same P2 450Mhz box.
I have tried both UHCI usb on a 2.4
On the same P2 450Mhz box.
I have tried both UHCI usb on a 2.4 kernel
and enhanced RTC on a 2.6 kernel.
Have not tried UHCI USB on a 2.6 kernel as of yet.
Both seem to work GREAT.
I have read in many places to be sure to use a digium card as a time source
and not to reply on the cheap
Tim,
We provide Linux/Asterisk installation setup, configuration and
maintenance services locally,
If you are still looking or simply wouldn't mind a good chat please
give us a call and ask for Steve at the phone number on our website
http://www.michiganbroadband.com
Steve
I'm in
of those two settings (registertimeout or registerattempts)
As I had not been able to find those on my own or in the wiki.
Thanks!
Steve
)
On Wed, 24 Aug 2005, Steve Gladden wrote:
I'm looking for some help in how to keep asterisk from doing this.
If we loose Internet or routing to our
I'm looking for some help in how to keep asterisk from doing this.
If we loose Internet or routing to our upstream provider even for only a
few short minutes asterisk quickly gives up never tries again.
I have to do a manual reload to get it to register with my
sip provider(s) again before
You also want to look at the registertimeout and registerattempts
Yes!!!, thank you VERY much this is what I needed.
Where are these options documented at?
I'm guessing the source code?
Or is there a better place to find this stuff?
A search on the wiki for registertimeout or registerattempts
I'm having difficulty understanding this 'hint' feature of asterisk.
My limited understanding is that it is somehow needed for 'informing'
some kinds of phones that can do shared line appearance to show the
state of the channel/user...
Is this true?
the wiki has this:
there.
On 8/19/05, trixter http://www.0xdecafbad.com/ [EMAIL PROTECTED]
wrote:
Steve Gladden wrote:
Well hey! let me know!!! :-)
I got my max232 chip sitting out and am building a converter board
right
now...
Gonna give it a shot soon as I get yer info!!! :-)
Have you done successful re-blast
Very Highly Internested
Any chance you could zip or tar your content up and email it to me or give
me a link to grab it?
Maybe I could help you get it hosted again too ifyou need that.
Thanks!!!
Steve
Steve Gladden wrote:
I have a small pile of them from customers who were too lazy
!
Steve
Steve Gladden wrote:
Very Highly Internested
Any chance you could zip or tar your content up and email it to me or
give
me a link to grab it?
Maybe I could help you get it hosted again too ifyou need that.
Thanks!!!
Steve
I would love to have a tarball of my web stuff. I
Greetings,
Running CVS HEAD about 3 weeks old,
I have been beating my head trying to get this to work properly..
Or at least figure out what's going on.
Maybe I have done things wrong...
I have created a 'catch all' extension at the end of our last context
where all phones voicemail extension
I have a small pile of them from customers who were too lazy to send them
back after switching to our local voice service...
Is there any hope of ever using these things with Asterisk?
Vonage does not want them back and they won't unlock them either.
A terrible shame!
Should I just toss them?
We had 2 fail at the office...
Red light and no talk battery on line #1 or line #2 port.
Unit completly responsive...
Not able to factory default due to unit being completly unresponsive.
It's some kind of hardware or total firmware fail.
They failed here before my very face and there were no
I am using asterisk (2 week old CVS) am for the first time have
been starting to experiment with busy and congestion.
At this point I am only using sip endpoints PAP2-NA devices.
All testing of this is being done on a local network.
my test extension looks like this:
exten = ,1,Answer
Oh GOSH
That was awesome!
I have an even better one in store but gotta capture it from it's
rather obscure source..
Stay tuned!!!
Steve (N8LBV)
Does anyone have a collection of stupid hold music? Y'know, the sort of
thing that would drive a person mad? Silly songs,
OK was actually able to pull it out of the archives!
It's now at http://stuff.michiganbroadband.com/asterisk
I'll leave it there for about a week or two then remove it.
T othe best of my knowledge it's public domain, if anyone needs more info
please contact me offlist.
This is right up there in
Generally speaking one works against one's own best interests when he
reminds the group that he has been posting on a topic repeatedly without
anyone answering.
Yes agreed,
In this case my only intention was to acknowledge the fact that I
realized I was asking a 3rd time and hopefully not
...
Is that Asterisk would have a way of doing this.
Seems that it it really does not (yet) though.
Thanks for all the help! I do appreciate it.
Take care!
Steve
Chris Mason (Lists) wrote:
Steve Gladden wrote:
Still looking for some direction with this subject:
I think the term is called multi
Still looking for some direction with this subject:
I think the term is called multi-line appearance
Is this something that is directly supported in Asterisk?
I can't seem to find any information on it or how to actually use it
This is where you have several sipura-841 SIP phones for
Still looking for some direction with this subject:
I think the term is called multi-line appearance
Is this something that is directly supported in Asterisk?
I can't seem to find any information on it or how to actually use it
This is where you have several sipura-841 SIP phones for
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