-
"Yeah, we rocked the vote all right. Those little
bastards betrayed us again."
- Hunter S. Thompson on the 2004 election.
-- Forwarded message --
Date: Tue, 22 Mar 2005 19:16:09 -0800 (PST)
From: Matt Klein <[EMAIL PROTECTED]>
T
On Wed, 2005-03-23 at 17:50 -0800, Sean Kennedy wrote:
> I am trying to compile spandsp on my asterisk server, and it keeps
> failing out with the following
>
> t4.c:38:21: tiffiop.h: No such file or directory
> In file included from t4.c:41:
> spandsp/t4.h:62: error: syntax error before "TIFF"
>
Wired SIP.
Wired Analog.
Cordless analog.
and what not.
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What are these messages that * constantly sends to the client? And
why does it always say 'looking for [server ip] in default' ?
Sometimes I see 404's instead of 200's, and I know something is wrong
but I don't know what is being signalled here.
In this case 206.80.111.117 is the * server, and
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Scheda wrote:
Hey, I currently have the voicemail set up so I can dial out from a
voicemail box. I seem to not be able to place them. I have put the
number in with and without a 1, and then hit pound and it says "Please
wait while I connect your call" and then disconnects. Any ideas as to
why?
Hi Remco,
The IP Column should be the IP number that Asterisk has registered.
If names do not show up in the left box, it's because there's no callerid in
extensions.
Import from server will only import any "cidname" entries from Asterisk
database. If you want to refresh your extensions, you nee
Yes, the X100P clone cards will work.
--- Dan Morin <[EMAIL PROTECTED]> wrote:
> Does anyone know if the X100P clone cards provide
> the timer needed to
> run MOH and the Conferencing service? I have no
> need for a T1 card, but
> I'm running asterisk on a dual processor machine
> with the wrong
I love this message, just the perfect one to get some people on the
list pi**ed off. The subject is *very* descriptive. The disclaimer
just a beauty.
On Thu, 24 Mar 2005 10:40:58 +0530, [EMAIL PROTECTED]
<[EMAIL PROTECTED]> wrote:
> hi,
>
>I have
>
> 1.Registered Asterisk with Proxy a
On Tue, 22 Mar 2005, Matt wrote:
> Hi,
> The reason I didn't look into that is it says:
>
> You CAN NOT:
> Accounting:
>
> * generate Start or Alive records, which is doable easily for
> connected calls, but
>
> If I can't generate a start record... what good is it for CDR recording?
You d
hi,
I have
1.Registered Asterisk with Proxy as [EMAIL PROTECTED]/.
2.Sucessfully registered and got 200
OK.
3.It shows contact as sip:[EMAIL PROTECTED] IP
4.But if i send an INVITE request to Proxy it says
404 not found since that extension does not exist at the Proxy.
Thi
I heard about it as well on IRC. But I don't think it is out yet,
still testing last I heard.
On Wed, 23 Mar 2005 15:15:05 -0600, Matthew Boehm <[EMAIL PROTECTED]> wrote:
> Kris Boutilier wrote:
>
> > Interesting - is there a similar program available for the single
> > port T100P cards?
>
>
Is it possible to call an application (e.g. SayDigits) from within source file of another application ?.
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Hey, I currently have the voicemail set up so I can dial out from a
voicemail box. I seem to not be able to place them. I have put the
number in with and without a 1, and then hit pound and it says "Please
wait while I connect your call" and then disconnects. Any ideas as to
why?
__
I just got everything working the way I want except 2 things
#1 the timestamp on voicemail is not the local time zone - I am in US
Central (-6) timezone and the voice mail is timestamped 6 hours ahead of
local time.
#2 incoming faxes - I get a comm err message from several different fax
machine
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Jason Brown wrote:
| Anyone have experiece with polycom phones?
|
| I am experiencing a really weird problem. In an office where I have the
| following extensions:
| On the Polycom phones, when I want to dial from extension 100 to any
| extension 120 o
Hi
I am using TDM FXO (4) with one of my server , in middle east and there
internet not so good, every time its has some packet loss happend. but speed
is good. quite enough for 4 port with ILBC. my problem is i setup the same
thing with same config in several country like singapore, bangladesh a
that is asterisk-1.0.R2 you can download from digium or asterisk.org
- Original Message -
From: "Joseph" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Wednesday, March 23, 2005 5:15 PM
Subject: Re: [Asterisk-Users] *-1.0.7 DTFM => Not working
Using PRI, have you gotten past the Caller-ID Problem? I have an
Asterisk box connected to an Option 61C with a PRI Trunk, but the Option
61C sees the PRI Trunk as an outside line (configured with 5ESS
Switch-type), so it only passes its External CallerID to Asterisk, not
the actual CallerID or St
I have set 2 extensions. 820 and 821。
The default language is fr。
and I have created the following call
file:
Channel: SIP/820
MaxRetries: 2
RetryTime: 30
WaitTime: 30
Context: c820
Extension: 821
Priority: 2
The 820 hear the english greeting when
821 on thephone。Normal,It will
I want to connect 2 extension by AGI.
like auto dial out. How can i do?
Bill Chen
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I want to use auto monitor function
in version 1.0.3 .
I have put the options 'wW' to Dial
application. but it do nothing when pressing *1 in call.
How can auto monitor in 1.0.3?
Bill Chen
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I'm doing the same scenario for 2 customers right now. Works good, On the
opt11 I connected it to asterisk with PRI. Have a good day
Henry
- Original Message -
From: "Friend, George E." <[EMAIL PROTECTED]>
To:
Sent: Wednesday, March 23, 2005 8:55 PM
Subject: [Asterisk-Users] Nortel Op
Question...I'm fairly new to Asterisk, but one location I'm looking at
deploying Asterisk has an Option 11 in place already (it's actually in
someone's HOME - long story).
Does anyone know if it's feasible to interconnect the two and use Asterisk to
interface with the other offices and lines,
Thanks,
I got everything to load via ftp. The phone appears to correctly boot
from the config files. I also put the latest firmware there and the
phone sucessfully loaded it.
For some reason, the phone and * don't see each other. This is the part
that confuses me. Any clues as to why the
On March 23, 2005 08:53 pm, Josh Alberts wrote:
> Hello, I'd like to make it so that after 5 invalid attempts of entering
> an extension, the Hangup command will be issued. How would I go about
> doing this?
My guess would be a combination of
SetVar($[${VAR} + 1])
and GotoIf($[${VAR} < 5 })
No
Hello, I'd like to make it so that after 5 invalid attempts of entering
an extension, the Hangup command will be issued. How would I go about
doing this?
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I am trying to compile spandsp on my asterisk server, and it keeps
failing out with the following
t4.c:38:21: tiffiop.h: No such file or directory
In file included from t4.c:41:
spandsp/t4.h:62: error: syntax error before "TIFF"
spandsp/t4.h:62: warning: no semicolon at end of struct or union
spa
On Wed, 2005-03-23 at 22:54 +0100, David Hajek wrote:
> Hello,
>
> is it possible Asterisk's ChangeLog will contain a reference to appropriate
> bug number in
> bugzilla? This can be very handy.
It isn't bugzilla, it is mantis. There is a cvs list that would be more
appropriate for you to watch
FYI - new soekris boxes ..
Original Message
Subject:Re: [Soekris] net5801 & net7501
Date: Wed, 23 Mar 2005 15:34:20 -0800
From: Soren Kristensen <[EMAIL PROTECTED]>
Organization: Soekris Engineering
To: Jabbar Fagan <[EMAIL PROTECTED]>
CC: [EMAIL PROTECTE
i am using redhat 8 with Asterisk CVS-v1-0-12/27/04-11:34:42
I am having some problem with the Individual Caller Authentication
I am manage to use global authentication
Can anyone guide me using databases or examples of working one
I try to look at so many places but couldnt find any documentaion
(On top of which, they charged me a $40 termination fee to terminate
my account - just a parting shot I guess).
People need to read the fine print more. From Vonage's website:
"If you cancel after the first 14 days of service, you will be subject to
the $39.99 termination fee. If you return the d
If this is the case it would seem to me that chan_sip.c is buggy.
Where did you get R2 version? I'll try it.
I don't understand how such a major bug got into the CVS-Stable branch.
#Joseph
On Wed, 2005-03-23 at 17:00 -0800, Bashir Ullah - www.Lamsre.Com wrote:
> Hi
>
> I am not good at coding,
Are you at run level 3? X can cause this if you are at run level 5.
Paul
Paul Mahler
[EMAIL PROTECTED]
www.signate.com
Hello
We are running Asterisk CVS 22/12/04 and pwlib/oh323 pandora version to work
with our call agent.
Unfortunately **VERY** frequently, asterisk stops responding and goe
So this is what i've done so far...
my extconfig.conf looks like this
[settings]
;example => odbc,asterisk,alttable
;iaxusers => odbc,asterisk
;iaxpeers => odbc,asterisk
sipusers => mysql,voip,sip
sippeers => mysql,voip,sip
voicemail => mysql,voip,voicemail
extensions => mysql,voip,extensions
--
Hi,
We recently got a PRI installed at one of our local centres however, we appear
to be having interesting issues.
Everything seems to be installed in the correct manner.
asterisk*CLI> pri show span 1
Primary D-channel: 16
Status: Provisioned, Down, Active
Switchtype: EuroISDN
Type: CPE
Window L
OK I have a TDM400 with 3 incoming lines, with call
hunting. When u call any of the numbers it is supposed to goto our
greeting message. Now when you call the first number that is exactly what
happens. Problem is when you call either of the other 2 it just rings
forever and I see no mess
On Wed, 23 Mar 2005, GP wrote:
I've read that someone was able to do it by contacting vonage and getting
instructions for clearing the router of the vonage information. Does anyone
have the instructions for completing this or is this something that only the
vonage people can provide.
I've spen
You can also use * and linux to partialize the T1. You better plan on
spending a lot of time on making it work. You have to install the Linux
packages to split the line. NON trival. Works great,
though.
Paul
Paul Mahler
[EMAIL PROTECTED]
www.signate.com
On Mon, 2005-03-21 at 21:16 -0700, Tim C
There is nice script that you can use cvs2cl.pl
that creates changelog file from cvs entries which has the tags you are
talking about
example
http://asterisk.gnuinter.net/files/changelogs/asterisk-ng.ChangeLog
On Wed, 23 Mar 2005 22:54:35 +0100, David Hajek
<[EMAIL PROTECTED]> wrote:
> Hello,
Don,
Thanks.
I've got a box that is dedicated only for Asterisk. It has a new
Slackware installation and nothing else running.
I installed and build the Digium and Asterisk drivers from source.
I can get the phones to load their configuration via FTP. That works
perfectly.
My problem is that
Hi
I am not good at coding, what i did, i just replace chan_sip.c by version R2
and now my DTMF working , I also faced same problem too. I know this is lay
man solution but works.
Bashir
- Original Message -
From: "Joseph" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commer
My DTFM is not working in current CVS-stable *-1.0.6 and *-1.0.7 but it
works in version 1.0.5 (was working with 1.0.3).
I'm using SPA-3000 and dtmfmode=inband
--
#Joseph
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I am wondering is there any silence suppression on asterisk? because when i
try to call from X-lite to one of my FXS port, when FXS don't talk, x-lite
felt like totally get cut off.
thank you.
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Noah Silverman wrote:
Dean,
I appreciate the suggestion. Is it really necessary.
I've got slackware already installed on the box. (I consider myself a
bit of a Linux guru.), all the drivers, and Asterisk. Everything
seems to work fine, EXCEPT getting it to talk to the extension.
wouldn't it b
This happens here, it is due to the lack of jitter buffer in the sip
channel, and using ulaw as the codec. Switch to GSM if you can, or wait
for the sip jitter buffer to be completed...
On Wed, 2005-23-03 at 15:29 -0800, Sean Kennedy wrote:
> Pol wrote:
>
> > I'm using asterisk as a sip client wi
Anyone have experiece with polycom phones?
I am experiencing a really weird problem. In an
office where I have the following extensions:
100
101
102
103
104
110
111
120
130
140
141
150
200
On the Polycom phones, when I want to dial from
extension 100 to any extension 120 or above, or dial o
> > They probably don't want to deal with the snickering and laughter that
> > this code will ensue. Digium has a good rep. in the open source
> > community for Asterisk, they don't want to release this mockery on
> > anyone!
>
> I truly hope your not right. While I know the code might be bad, th
Pol wrote:
I'm using asterisk as a sip client with a sip proxy server... I've
made the pertinent extensions and I've configured the sip.conf
correctly or I think so..
I'm using x-lite as a client and when I ring to a public telephone
through proxy, the arriving sound it's perfect but the sound
David Hajek wrote:
I have no such issues with Sipura/Asterisk. I don't think this is
common to Sipura. Try Asterisk's SIP debug to see what happens when
registration fails.
I guess YMMV, as I am seeing this problem from Sipura SPA-2000s and
SPA-841s set up at various locations with various ISPs
Dean,
I appreciate the suggestion. Is it really necessary.
I've got slackware already installed on the box. (I consider myself a
bit of a Linux guru.), all the drivers, and Asterisk. Everything seems
to work fine, EXCEPT getting it to talk to the extension. wouldn't it
be easier/faster to jus
I'm using asterisk as a sip client with a sip proxy server... I've made
the pertinent extensions and I've configured the sip.conf correctly or I
think so..
I'm using x-lite as a client and when I ring to a public telephone
through proxy, the arriving sound it's perfect but the sound I send is
Hi,
I'm just setting up my first Asterisk box. So far everything is working
fine. I have the digium card in and connected to a regular telco line.
The Asterisk box answers the line and goes through the demo voicemail
functions. Sounds great!
I bought a Polylcom ip500 phone. I can't seem to
Hi,
I wanted to say a date (and time) in german, but did not found a way to
do this with sayunixtime, so I wrote this agi script.
(I could switch to german, but it was not correct)
be sure to use the sound files from:
http://www.stadt-pforzheim.de/asterisk/dateien/ast_prompts_de.tgz (which
are
Erik Espinoza wrote:
They probably don't want to deal with the snickering and laughter that
this code will ensue. Digium has a good rep. in the open source
community for Asterisk, they don't want to release this mockery on
anyone!
I truly hope your not right. While I know the code might be bad, th
How to change the date from:
Mon Mar 23 13:50:43 2005
to:
03-23-2005 09:50:43 Wed
???
1. The form as ASTCC stores the date / time does not allow to sort the
records (ASC/DESC) by date.
I would like to change it to a form that allows me to sort the records.
2. Is there a way to change existing r
post your mgcp.conf (mask out any ips if you need to)
duane cox
- Original Message -
From: <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Wednesday, March 16, 2005 3:00 PM
Subject: [Asterisk-Users] MGCP Channel Lockup and other probelms
Hi All,
GP wrote:
> I setup a vonage account last year, and cancelled it last night when I
> put my asterisk box together and signed up for a Broadvoice account to
> use with it.
>
> Now I would like to use my Linksys router as an MTA, but realize it is
> still programmed with all of vonage's proprietary i
Thanks.. definately will keep the elist posted. Today I got a cost
comparison from other PBX vendors and integrating the legacy phones.
Nortel and Asterisk with the RHINO channel banks are similar (about US$
36,000 in equipment costs).
C F wrote:
On Tue, 22 Mar 2005 19:36:26 +, cmould <[EMA
I have no such issues with Sipura/Asterisk. I don't think this is common to Sipura. Try
Asterisk's SIP debug to see what happens when registration fails.
-D
GliTcH wrote:
This is a common issue with all Sipura devices I've seen. I set the
registration interval to 5 minutes, so that NAT doesn't in
Hello,
is it possible Asterisk's ChangeLog will contain a reference to appropriate bug number in
bugzilla? This can be very handy.
Thanks,
David
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hi all, was wondering if someone could assist with a slight problem
i'm having. I have asterisk setup with extensions 101 to 109 and am
using xlite, grandstream budgetone, polycom ip500 and a couple of
other phones. the problem is:
1. only the xlite extension (107) can receive calls.
2. all extens
They probably don't want to deal with the snickering and laughter that
this code will ensue. Digium has a good rep. in the open source
community for Asterisk, they don't want to release this mockery on
anyone!
On Wed, 23 Mar 2005 15:38:06 -0600, Chris Wade <[EMAIL PROTECTED]> wrote:
> I've noti
is some one from digium reading this thread. !!
Looks like they have a ready and a big market for this device. And all
they need to do is invest say 6 man months of development effort :)
come on digium do it !!
How about making the firmware open source so we can hack on it ...
t
On Wed, 23 Ma
I setup a vonage account last year, and cancelled it last night when I
put my asterisk box together and signed up for a Broadvoice account to
use with it.
Now I would like to use my Linksys router as an MTA, but realize it is
still programmed with all of vonage's proprietary information and I d
looks like you are missing the trailing '/' in the
url
- Original Message -
From:
Dov Bigio
To: asterisk-users@lists.digium.com
Sent: Wednesday, March 23, 2005 12:38
PM
Subject: [Asterisk-Users] slim server for
moh
Hello,
I have installed SlimSer
Hi,
I was wondering if there was a way for Asterisk to direct dial into an
ISDN modem. Basically we want to do the following:
1. Initiate the ISDN call into the remote router.
2. Authenticate itself to the password in the router.
2a. Possibly negotiate a reasonable compression mode using STAC.
3.
Couldn't you do call files and have them drop the call into the meetme room?
- Original Message -
From: "Alex Pepper" <[EMAIL PROTECTED]>
To:
Sent: Wednesday, March 23, 2005 12:15 PM
Subject: [Asterisk-Users] Multicall
This is my question:
it is possible to call from one phone many othe
Now here's a thread I've been waiting to see! I have had issues with
what is considered to be a decent phone, the siemens DECT line.
Fortunately, the problem is just callerID which although annoying,
isn't mission critical, and we are in Europe. Still, the USA isn't the
world, the more adaptable th
Jared Watkins wrote:
>Adam Robins wrote:
>
>>So, I switched to IAX2. Now, everything works fine 95% of the time .
.
>>. but every once in a while, perhaps 5 seconds into a call or 20
minutes
>>into a call, the call will simply drop. This occurs several times per
>>week with no observable pattern
I've noticed a few recent messages regarding if/when a fixed firmware
for the IAXy will become available that deals with everyone's issues
regarding the little device.
After some thought on this subject, I wonder what Digium's stance on
open-sourcing the firmware for the device? Is there anyth
On Wed, 2005-03-23 at 20:58 +, Filipe Abrantes wrote:
> Hi all,
>
> I've just got asterisk and i was trying to compile it for my amdk6
> running 2.4.27 debian patched kernel. However I got some some undefined
> references when trying to load the zaptel module:
>
> # modprobe zaptel
> /lib/
On Wed, 23 Mar 2005 16:13:12 -0500, Time Bandit <[EMAIL PROTECTED]> wrote:
> > So how am I going to provision the device in the first place, to be
> > able to dial this extension, if I don't even know the IP?
> Oups, sorry, didn't think about this one.
>
> Check winiaxyprov, the version 1.01 can s
> Where do you put the module load and init commands on a RHEL 4 box and
> where to put it on a RHEL 3 box?
If you are talking about loading the Zaptel modules, here's the easy way :
- Go in your Zaptel src directory (usually /usr/src/zaptel)
- # make config (this wil copy the init script)
N.B.: if
On Wed, 23 Mar 2005 12:55:45 -0800, Sys Admin <[EMAIL PROTECTED]> wrote:
> so seems like the verdict is go IAXy with a IAX only network ? Most of
> the problems of the IAXy device seems like will be fixed with firmware
> updates and wont require a hardware update..
>
> this way we get the advantag
Hi,
Im testing asterisk for callback functionality and want to reject a call
after a few seconds for freeing the line for callback. But if I use a
"congestion", there is a connection for a (billing) short time. Is there
an ability to reject a call (like the red button on a mobile phone)
without
James,
After three months of echo, I finally got mine to go away.
I am using a Dell Optiplex GX280 with a P4 3.2GHz processor and 512MB RAM
and Asterisk version stable 1.07.
Here's what I came up with after reading many of the posts and much trial
and error.
1. Make sure that ther is no GUI load
> so seems like the verdict is go IAXy with a IAX only network ? Most of
> the problems of the IAXy device seems like will be fixed with firmware
> updates and wont require a hardware update..
The best part is how you update the firmware : each time an IAXy
connect to Asterisk, it check what firmwa
Kris Boutilier wrote:
> Interesting - is there a similar program available for the single
> port T100P cards?
No clue. But IIRC the T100P doesn't have any "expansion" slots on it.
-Matthew
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Jeb Campbell wrote:
> Matthew Boehm wrote:
>> For $2500 you can buy the TE411P which is a 4 port T1/PRI card
>> that has onboard echo cancellation. Or you can send in your TE410P
>> and $1000 and they will upgrade it for you.
>
> Do you know where we can read up on this? I couldn't find any i
> So how am I going to provision the device in the first place, to be
> able to dial this extension, if I don't even know the IP?
Oups, sorry, didn't think about this one.
Check winiaxyprov, the version 1.01 can scan your network to find
IAXy. Now the only thing we need is for Digium to write the
couldnt agree with u more !!
On Wed, 23 Mar 2005 11:15:55 -0800, Robert Goodyear <[EMAIL PROTECTED]> wrote:
> >>
> >> Confidentiality Notice
> >>
> >> The information contained in this electronic message and any
> >> attachments to
> >> this message are intended
> >> for the exclusive use of t
Hi all,
I've just got asterisk and i was trying to compile it for my amdk6
running 2.4.27 debian patched kernel. However I got some some undefined
references when trying to load the zaptel module:
# modprobe zaptel
/lib/modules/2.4.27-1-386/misc/zaptel.o:
/lib/modules/2.4.27-1-386/misc/zaptel.o
so seems like the verdict is go IAXy with a IAX only network ? Most of
the problems of the IAXy device seems like will be fixed with firmware
updates and wont require a hardware update..
this way we get the advantage of a Hardphone (human factor, just feel
good to talk on a real phone) with all th
Michael Sanders wrote:
Hi,
How do I connect two Analog PBX together with Asterisk.I want two
simultaneous voice channels between sites.
My (simple) solution was to put an * server at each site with an
extension from the PBX connected to an * FXO port. Users dial the
regular pbx ext to access
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Matthew Boehm
> Sent: Wednesday, March 23, 2005 12:12 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Problem compiling asterisk-addons
>
>
> Eric
Matthew Boehm wrote:
For $2500 you can buy the TE411P which is a 4 port T1/PRI card that has
onboard echo cancellation. Or you can send in your TE410P and $1000 and they
will upgrade it for you.
Do you know where we can read up on this? I couldn't find any info on
digium's site. I'm very int
[EMAIL PROTECTED] (Chuck Ramirez) writes:
> Looking at the source code I noticed that rand() is
> used four times to get a callid. Is that safe enough?
RAND(3) OpenBSD Programmer's Manual RAND(3)
NAME
rand, srand - bad random number generator
...
Is there
On Wed, 23 Mar 2005, McQuiggan, Mark xt46480 wrote:
> I have noticed that any of the zapata.conf echo cancel parameters seem to
> have no effect on an ISDN-PRI line, using pri_net signalling (I used the
> voip-info.org wiki for the configuration). If this is true, and I am not
> making some dumb
> -Original Message-
> From: McQuiggan, Mark xt46480 [mailto:[EMAIL PROTECTED]
> Sent: Wednesday, March 23, 2005 12:28 PM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] TE405P and echo
>
>Peter Svensson wrote:
>
>>On Tue, 22 Mar 2005, McQuiggan, Mark xt46480 wrote:
>>
>
Hi Shaun,
Your suggestion was right on the money. I make a symlink, rather then
renaming my asterisk source directory (I keep backup copies when upgrading),
and the addons compile now.
Thanks.
On Wed, 23 Mar 2005 14:00:36 -0600
"Shaun Tierney" <[EMAIL PROTECTED]> wrote:
> I believe that it
>Peter Svensson wrote:>>>On Tue, 22 Mar 2005, McQuiggan, Mark xt46480 wrote: >I am using a SIP softphone (X-lite, SIPPS or Firefly) connected to
an>>>Asterisk v 1.0.3 PBX. The PBX is also connected
via a ISDN-PRI crossover>>>cable to a Avaya Definity Generic 3 PBX via a TE405P card.
I have asterisk running on my personal computer and am using Kphone to
connect to it. My provider is broadvoice which is Ulaw and I had kphone
connected as GSM. The lag was terrible coming from
Pots-->--Broadvoice-->Kphone. About 2.5 seconds! Going the other
direction seemed fine. I did a:
show
I was having the same problem with simpletelecom, but FWD was working
fine. I just upgraded to the latest CVS and it fixed the problem for
me also.
Chris
On Wed, 23 Mar 2005 03:49:08 -0500, Kris Edwards <[EMAIL PROTECTED]> wrote:
> Compiled from CVS today and no more dropping outbound calls aft
> -Original Message-
> From: Matthew Boehm [mailto:[EMAIL PROTECTED]
> Sent: Wednesday, March 23, 2005 11:58 AM
> To: C F; Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Any Software Echo Cancellation in
> Asterisk?
>
>
> C F wrote:
>
> >>
> >> sh
Eric wrote:
> Hi,
>
> I am getting an error trying to compile the asterisk addons:
>
> cdr_addon_mysql.c:24:22: asterisk.h: No such file or directory
> make: *** [cdr_addon_mysql.o] Error 1
>
> Can anyone suggest something I could try?
>
Are you actually installed asterisk? Do you have
/usr/includ
On Wed, 2005-03-23 at 14:30 -0500, Dan Morin wrote:
> Does anyone know if the X100P clone cards provide the timer needed to
> run MOH and the Conferencing service?
Yes, it will work fine.
-Seth
--
Seth Remington
SaberLogic, LLC
661-B Weber Drive
Wadsworth, Ohio 44281
Phone: (330)335-6442
Fax: (
Every now and then, I notice that one of my queues suddenly
reports that it has anywhere from the millions to billions when reporting how
many calls are in the queue, and the “max holders” reports an
insanely high number as well, even when max holders is set to be “unlimited”.
Has anyone
Thanks for your reply. I already created a new extension, I will tag it to
my deskphone. Your app looks really neat :)
The field IP address in the extensions TAB is showing all kinds of weird
data, but no valid ip addresses (some fields do have something that looks
like an ip addres but the num
I believe that it looks in ../asterisk/ for the asterisk.h file. If your
source directories for both Asterisk and the Asterisk Addons are within the
same directory, try renaming the asterisk-x.x.x directory to just asterisk
then recompile. It should work for you after that.
Regards,
Shaun
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