Hi,
I have just bought another TDM400P card from Digium directly, purchased
last Thursday, received it today:
Span 1: WCTDM/0 "Wildcard TDM400P REV E/F Board 1"
1 WCTDM/0/0 FXOKS (In use)
2 WCTDM/0/1 FXOKS (In use)
3 WCTDM/0/2 FXSKS (In use)
4 WCTDM/0/3
Any why would that make it work with cvs-head but not cvs-stable?
By the way, I no_load the module so I can load it manually later and see the
console output. Either way, it still kicks out the error and crashes, or
just kicks out the error if I no_load it first...
-Original Message-
From
Your right on the overpriced junk! But yes now it works great.
Rod Bacon wrote:
I found new firmware at:
ftp://ftp.us.zyxel.com/P2000W/firmware/P2000W_WJ.00.10_Standard.zip
The phone is now finally (almost) useful. Still a cheap piece of crap,
with new bugs to replace the old, but at least it sor
On Mon, 11 Apr 2005, Michael Loftis wrote:
> --On Tuesday, April 12, 2005 1:28 PM +1000 Ben Ryan
> <[EMAIL PROTECTED]> wrote:
>
> > I have a question probably for those in the know in business Asterisk
> > solutions. I have searched high and low but have not been able to get
> > any answers. I h
Greetings,
I have a question regarding setting the CallerID, more specifically the
Caller Name. In all of my menus I set the current Caller Name so it
displays what menu they are in when the phone rings for my users. We run
seperate companies so it's easy for us to distinguish how to answer the
Tim Connolly wrote:
Well crapola... cvs-head works with Digium's te110xp, but not cvs stable.
Looks like there's a huge difference:
Stable=-rw--- 1 root root 248572 Jun 9 2004 chan_zap.c
Head =-rw--- 1 root root 326585 Apr 6 14:17 chan_zap.c
I run a te110p with 1.0.x CVS stable all th
Hello,
I need a provider that will allow me do create more than two channels at
the same time. Calls will be place within illinois. Does anyone know any
good provider that work (was tested for a long period) and is not
expensive? I want something for 15-20 dollars for unlimited calls within
one
Ok here are my config files.
Are you behind any firewall? Are you using NAT?
//cat sip.conf
[general]
externip=your domain name
bindaddr = 0.0.0.0
port=5060
localnet=192.168.1.0/255.255.255.0
disallow=all
allow=ulaw
register => phonenum:[EMAIL PROTECTED]
tos=0x18
srvlookup=yes
nat=never
insecure=y
More on this...
While the new features are a welcome change, the most annoying of the
original problems remain.
If anyone knows why I can make one call, then not make or receive another
(until I reboot it), I'd like to know. I think it has something to do with a
lack of or a malformed BYE.
--
Hi
Thanks!
The zapata.conf is as followings,
I also check the callerid used here, it's ETSI Standard ETS 300-659-1(both FSK and DTMF Caller ID transmission formats )
FSK use Dual Tone Alerting Signal (DTAS) as the start signal, DTMF seems no start signal("start" digit only)
http://www.adv
I found new firmware at:
ftp://ftp.us.zyxel.com/P2000W/firmware/P2000W_WJ.00.10_Standard.zip
The phone is now finally (almost) useful. Still a cheap piece of crap,
with new bugs to replace the old, but at least it sort of works now.
___
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Well crapola... cvs-head works with Digium's te110xp, but not cvs stable.
Looks like there's a huge difference:
Stable=-rw--- 1 root root 248572 Jun 9 2004 chan_zap.c
Head =-rw--- 1 root root 326585 Apr 6 14:17 chan_zap.c
Arrrg...: (results of the first test call)
-- Executing Dial("
> It seems that sometimes broadvoice honors my g.729 request
Be careful with this. I tried setting G726-32 as a prefered codec and
SOME calls would accept it (depending on call destination) but usually
the caller did NOT hear me, although I could hear the caller just
fine.
So there's truth to it:
On Mon, 11 Apr 2005, Noah Miller wrote:
> > This this may sound ridiculous, but we've had problems with this when
> > the
> > users did not plug the handset cord in completely. 8 out of our 12
> > employees
> > made the mistake, as the plug on the IPX00's appears to be all the way
> > in
> > w
Guys.
Im playing with agent autologoff but I have a question. My agents are
configured to timeout after 10 seconds of ringing but if you configure
autologoff to 15 seconds for example, then they never autologoff cause it
never rings for 15 seconds.
Is there a way to make the autologoff timeout be
--On Tuesday, April 12, 2005 1:28 PM +1000 Ben Ryan
<[EMAIL PROTECTED]> wrote:
I have a question probably for those in the know in business Asterisk
solutions. I have searched high and low but have not been able to get
any answers. I hope there is someone on the list that can answer my
question.
I have a question probably for those in the know in business Asterisk
solutions. I have searched high and low but have not been able to get
any answers. I hope there is someone on the list that can answer my
question.
How do you implement "trunk seize"? This is a feature that is almost
universal i
Hello, I did that using SIP. I setup an extension on both servers.
Register both servers using the new extensions. Then just route the
calls.
David
On Sat, 2005-04-09 at 15:01 -0300, Juan Luis Moyano wrote:
> Hi all, I am trying to set up two asterisk servers (SrvA and SrvB), and
> what I want
Try wwwadmin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael D
Schelin
Sent: Tuesday, 12 April 2005 1:03 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Asterisk management portal
Hi everyone, Why doesn'
Try
wwwadmin
Password
BRW
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael D
Schelin
Sent: Monday, April 11, 2005 8:03 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Asterisk management portal
Michael D Schelin wrote:
Hi everyone, Why doesn't this work? I can't get in. Is it because I
changed the root?
User: admin Pass: password
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My experience with VOIP to date has surounded SER and Asterisk, SIP and
IAX. It would appear as though I am about to be inducted into the world
of H.323 and as such I am interested in hearing from anyone who is using
Asterisk extensively in a mixed protocol environment, especially in
using H.32
Hi everyone, Why doesn't this work? I can't get in. Is it because I
changed the root?
User: admin
Pass: password
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-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
I've been playing around with ChanSpy(), using it to monitor a
SIP<->SIP call.all seems to work fine, except the audio is very
slow (sounds like a record player on the slow speed) and I get flooded
on the console with this error:
ERROR[9515]: ch
On Mon, 2005-04-11 at 17:06 -0400, Andre Normandin wrote:
> Hi,
>
> I'm having very similiar problems.. However, I'm running a development
> version, and it happens on both SIP phones, and on Analog phones
> connected via Sipura SPA-2000's (I have 2 different SPA2000's, and 4
> analog lines.. See
Craig Simon wrote:
Thank you, thank you, thank you! That was it! Thanks alot for the
description, it made the call flow much easier to understand.
You're welcome. I only discovered Asterisk about a month ago myself,
and understand first hand how difficult it can be for the uninitiated.
Aster
Is there a possible settings for a remote SIP phone, so that a router
will not close the connection due to long time inactivity?
bye
Ronald
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I have configured realtime according to the wiki.
However Asterisk can't see the new sip friend that is created in the table. I'm
sure I have failed to do something very basic which allows Asterisk to see
the records in mysql
realtime load sippeers name 6000
No rows found
/var/log/asteris
> I see where you are coming from. However, when I read or hear someone say
"We
> have built in" something into a system, to me that implies they designed
it and
> are taking the credit for it. Yes, that have given a bulleted list of
items
> included in the package. But no where in their adverti
With all the press regarding the lawsuits around 911 services and
Vonage... I wonder how well Asterisk supports E911?
Comments?
- matthewk (MSK2)
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Hello,
On my setup, I can't seem to get asterisk to play music on hold when i press
the hold button on sjphone (does not work on x-lite as well). I have already
set
the musicclass=default in sip.conf and default =>
mp3:/var/lib/asterisk/mohmp3 in musiconhold.conf.
the music play fine when pressin
I have a writeup on configuring BroadVoice with Asterisk that many people
have used successfully at http://www.geekgazette.com
-Kerry
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Craig Simon
Sent: Monday, April 11, 2005 3:00 PM
To: Asterisk Users Mail
> I'm not quite sure if this can be done, but..
>
> I use BROADVOICE as my outbound primary. I have both g.729 and ulaw as my
> outbound preference with BROADVOICE. It seems that sometimes broadvoice
> honors my g.729 request, and that is the codec chosen for the outbound call
> via broadvoice..
Johnathan,
Thank you, thank you, thank you! That was it! Thanks alot for the
description, it made the call flow much easier to understand. Now I am
off to integrate my 7920 and 7960.
Craig
Johnathan Corgan wrote:
Craig Simon wrote:
The 100 is an extension I created for my softphone to log i
This is an off-topic message so please reply to me by
e-mail if you can help:
I've installed and started using Thunderbird at the suggestion of many
members. I'm only using it for the Asterisk list and as it's
apparently very popular here I thought someone might have a quick
answer which I've
> > On Apr 11, 2005 3:58 PM, Craig Simon <[EMAIL PROTECTED]> wrote:
> > > This is the pieces of my extensions.conf. All this has been sent to me
> > > from broadvoice, so I can't tell you if it's correct or not. I do have
> > > an extension 100 created, and that is what I am logging on with my
>
Hello, again.
Here is some more information. I got this by starting 'asterisk -cvvvr'
and typing 'pri debug span 1'.
I'm not a PRI person, but the calls look sufficiently different to be
interesting. Any ideas, or better place to post?
Thanks!
-Jesse
A normal call to 800-950-5114:
-- Exec
> what sort of processing power etc should I be aiming for to support 60
> SIP extensions and 60 SIP based lines?
If the 60 phones never make a call, a 150 mhz machine should be
just fine. If all 60 are making calls at exactly the same time,
get the fastest machine you can aford.
The machine siz
> On Apr 11, 2005 3:58 PM, Craig Simon <[EMAIL PROTECTED]> wrote:
> > This is the pieces of my extensions.conf. All this has been sent to me
> > from broadvoice, so I can't tell you if it's correct or not. I do have
> > an extension 100 created, and that is what I am logging on with my
> > softph
You have to put the monitor after the person presses their selection.
This is how ours is:
exten => s,1,answer
exten => s,2,SetCIDName('PMG')
exten => s,3,SetVar(company=PMG)
exten => s,4,Wait(1)
exten => s,5,DigitTimeout,5
exten => s,6,ResponseTimeout,40
exten => s,7,Background(/var/lib/asterisk/s
I upgraded to .8 from .4 and I'm having problems with SPA-2000 ATA. Its
unable to re-register. I'm using the same hardware. I see the spa-2000
try using tcpdump. is there anything I should check???
Thanks, David
On Wed, 2005-03-30 at 07:50 -0800, [EMAIL PROTECTED] wrote:
> [EMAIL PROTECTED] 0.7
> > > I was following a discussion on this list about the TDM400P
> > revisions.
> > > It is my understanding that the current revision that one
> > should have
> > > is the Rev. H and not the E/F. I have not yet been able to
> > verify the
> > > rev stamped on the board, but zaptel is repor
Matt Fredrickson wrote:
Display-style name should work already. If it's facility-style name,
you'll probably have to pay more than $15 for someone to code that :-)
I did the facility-name receive and a good chunk of the newer features
that have been going into libpri and anything involving ASN.1 i
Hello, all!
I have an asterisk setup that I am just starting to deploy, but I've run
into a snag. I have a problem that manifests itself as the user being
unable to use the 1-9 keys on his phone to interact with a voice menu, but
only on some calls.
Here are the system details:
I am running CVS
I am sure that this was answered somewhere but my lack of
being able to find an answer using google I turn to the pros.
What would be the easist way to record all conversations
using Monitor command with the latest [EMAIL PROTECTED] ?
Using a FXO card with SIP extensions
I have tri
> Im facing a strange problem using a linksys-pap2 (two ports) ATA: I cant
> have two simultaneous incoming calls when i use g729 codec, if i use g711
> (alaw) there is no problem, is this a know issue or am i missing something?
The PAP2 only supports one G.729 call at a time.
Same as the Sipura
On Mon, Apr 11, 2005 at 10:47:19AM -0700, Trevor Peirce wrote:
> Hello,
>
> We are prepared to offer a small bounty to the individual who can
> provide us with a debug of an outgoing call from a PBX that supports
> outgoing Display Name to the upstream PRI provider.
Display-style name should wo
> Robert Webb wrote:
>
> >
> >
> >
> >
> >>-Original Message-
> >>From: [EMAIL PROTECTED]
> >>[mailto:[EMAIL PROTECTED] On Behalf Of dean
> >>collins
> >>Sent: Monday, April 11, 2005 5:35 PM
> >>To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
> >>Non-Commercial Discussion
> >>Subject:
Sorry to disappoint you, but questions only appear if the SELLER
wants them to. So if the seller doesn't like your question, he doesn't
have to make it show up on the listing.
Jon.
On Monday 11 April 2005 04:35 pm, dean collins wrote:
> Lol, just posted a question to the list that should kee
nat=no
disallow=all
allow=g729
allow=g726
auth=plain
context=default
canreinvite=yes
username=USERNAME
secret=PASSWORD
dtmfmode=info
fromdomain=REALM
fromuser=USERNAME
qualify=1000
insecure=very
I am using Asterisk 1.0.7 compiled into RPMs from the tarball running on
CentOS Enterprise 4...
Can so
On Apr 11, 2005 3:58 PM, Craig Simon <[EMAIL PROTECTED]> wrote:
> This is the pieces of my extensions.conf. All this has been sent to me
> from broadvoice, so I can't tell you if it's correct or not. I do have
> an extension 100 created, and that is what I am logging on with my
> softphone as. A
--On Monday, April 11, 2005 7:08 AM -0700 Sean Kennedy
<[EMAIL PROTECTED]> wrote:
Does a great job. My only fear is it doesn't specifically target IAX2
traffic as high priority, but I can modify it later to do so if needed.
Just add some iptables rules to prioritize UDP traffic by marking them w
Robert Webb wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
dean collins
Sent: Monday, April 11, 2005 5:35 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: RE: [Asterisk-Users] RE: Ebay listing selling
>
> > I was following a discussion on this list about the TDM400P
> revisions.
> > It is my understanding that the current revision that one
> should have
> > is the Rev. H and not the E/F. I have not yet been able to
> verify the
> > rev stamped on the board, but zaptel is reporting that I
> have
Craig Simon wrote:
The 100 is an extension I created for my softphone to log into.
* is tricky with terminology.
You didn't create an "extension" 100, you created a SIP peer/user named
"100", which the softphone connects as.
"Extensions" (that are within "contexts") are lists of commands that *
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> dean collins
> Sent: Monday, April 11, 2005 5:35 PM
> To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
> Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] RE: Ebay listing selling
> Aster
On April 11, 2005 06:43 pm, Bicom Systems wrote:
> Come June/July an USB/PCI DSP cost effective solution should be available
> to address this issues. It will transcode nearly all codec's.
> I am not in position to reveal the company name
> at this stage unless "MN" wants to "speak up" :)
seconda
Hello
Is there a set of ivr speech prompts availible for the ASTCC card system?
I can't find them in the CVS or any reference in the WIKI?
Thanks
David
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There are many analogue gateways to choose from:
http://www.voip-info.org/wiki-VoIP+Gateways
Does anyone have experience with several that could point me in the
right direction? I need 5-8 ports. At some point I see us going digital
but I'm not sure when TCO will make sense.
Advice based on real
Another option is to use one of the 'ready to go' asterisk cd's.
I haven't used them myself, but they might be good for someone starting out.
PaulH
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Race Vanderdecken
Sent: Tuesday, 12 April 2005 2:25 AM
T
what sort of processing power etc should I be aiming for to support 60
SIP extensions and 60 SIP based lines?
Thanks
David
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To UNSUBSC
Nathan, that’s for express I have full Callmanager. Heaps of good info in it and this is what we used to get the bulk of voicemail working initially (minus MWI). Have been supplied config for using the 2 dial out numbers into Callmanager and have it all working (thanks Dunc) Cheers D Kemp >
Come June/July an USB/PCI DSP cost effective solution should be
available
to address this issues. It will transcode nearly all codec's.
I am not in position to reveal the company name
at this stage unless "MN" wants to "speak up" :)
Put me on the mailing list for the PCI DSP card, I'll beta test
The 100 is an extension I created for my softphone to log into.
Here is my extensions.conf:
[default]
exten => _1NXXNXX, 1, dial(SIP/[EMAIL PROTECTED],30)
exten => _1NXXNXX, 2, congestion()
exten => _1NXXNXX, 102, busy()
[from-broadvoice]
exten => s,1,Answer
exten => s,2,Wait(1)
exten
Hello all,
Here is the issue ..
Latest Asterisk install
SIP trunk installed via Broad Voice
Outbound calls no issues other than I get no sound of ringing, Inbound no
calls at all .. Anyone have any ideas of where or what?
BRW
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> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> [EMAIL PROTECTED]
> Sent: Monday, April 11, 2005 3:37 PM
> To: Asterisk-Users@lists.digium.com
> Subject: [Asterisk-Users] Linksys PAP2 Dual Incoming Calls
>
>
> Hi List,
> Im facing a strange pro
Thanks for that Rich. Etheral trace is going to be almost impossible
for various reasons, but will try the other two options.
Can't find much online re. debugging - any chance of killing the box by
turning this on?
SIP show channels and the various CAPI show commands do not show
anything out
This is the pieces of my extensions.conf. All this has been sent to me
from broadvoice, so I can't tell you if it's correct or not. I do have
an extension 100 created, and that is what I am logging on with my
softphone as. And testing my outgoing calling.
Thanks for the help!
Craig
[default]
Is anyone using an 8-port analog trunk card and 24 port
station card with Asterisk and Televantage?
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Craig Simon wrote:
Looking for 100 in from-broadvoice
It looks like * is searching for extension 100 in the 'from-broadvoice'
context, not finding it, and sending a 404 back.
First, you can create a extension 100 in that context in your dialplan,
then see if that allows the call to come through.
Hi. I am considering building an asterisk system for
home use, but I am having some trouble understanding a
few things after reading the wiki and the various
mailing lists. Let me apologize in advance if I have
missed something...
1) What are the hardware needs to run asterisk
reliably? It appe
[EMAIL PROTECTED] wrote:
> On April 11, 2005 03:17 pm, Andres wrote:
>> Can you confirm if there will be some sort of DSP daughther card add
>> on of some sort for the DS3000 so that we can run G729 transcoding?
>> I don't see how the DS3 interface would be usefull unless we could
>> offload trans
Joseph Gutowski wrote:
Are you trying to setup a seperate extension just for IP Switchboard?
That's what it sounds like you're trying to do.
You don't have to do anything to your Asterisk to use the program,
except enable the manager interface and add the 77 and 88 stuff to
your extensions.conf to
Hi List,
Im facing a strange problem using a linksys-pap2 (two ports) ATA: I cant
have two simultaneous incoming calls when i use g729 codec, if i use g711
(alaw) there is no problem, is this a know issue or am i missing something?
These are the relevant config lines
sip.conf
[general]
port =
> I have been fighting with * for a couple of days now. I have recieved
> some help from the list but have not been successful in receiving calls
> from broadvoice to my asterisk box yet. I can place calls however, just
> not receive them. I enabled sip debugging today and here is the output
Thorben Jensen wrote:
| It is there!
| However, if I try to call 650, than I get 650 is unavailable
| If I try to make a call on the IPswitch, e.g., to 601, and click on
| call, it just closes the window.
|
| sip show usersshows the extension 650
| sip show peers shows the extension 650 with
Hello Everyone,
I am working with a J4BRI and it works ok
with our national provider ( TelecomItalia ) and when
connected
to ISDN pabx ( Ericsson and Tenovis ). We now have
switch to a new telco provider ( Fastweb ) which brings
you S0 via Cisco 1760 VIC-2BRI interface. They
say ( and Cisco
Angel Diaz wrote:
I want to use the Voicemail app and before that, I would like to play
an audio file but not billable in the Switch side. Than, to do so, I have to
be able to no send the Answer message during the play of the file. Then
after finish the file, I'w xecute the Voicemail app.
Tha
> Can't help but wonder if this isn't a bug in Asterisk or one of it's
> modules, as there seems to be a lot of people experiencing the same
> problem, seemingly with different hardware and software configurations.
>
> Anyone know how (or if it's possible) to submit a bug report to Digium
> re
How many people (or remote sip clients) have people actually
seen/gotten to work in a real world environment?
Say a 2.8Ghz machine with a GIG of ram. How many G711 or G729
calls could you handle?
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d with no avail!
>
> You could always download the Vanilla kernel source from
> http://www.kernel.org and compile a kernel from source. I tend to always
> use the Vanilla source, it's what everything has been tested against and
> it taste
> What it SHOULD do is, check the DC voltage on the line, and if less than
> 8-10 volts, consider it BUSY/ unavailable/not connected, THEN check for
> dialtone before dialing. Also optionally listen for stutter dialtone before
> dialing, making the detection of stutter dialtone available for som
Hello list
I have been fighting with * for a couple of days now. I have recieved
some help from the list but have not been successful in receiving calls
from broadvoice to my asterisk box yet. I can place calls however, just
not receive them. I enabled sip debugging today and here is the outp
I have a woking system now with 3 fritz cards with DID running chan_misdn..
Take Capi out of the Picture all together and it works fine.
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Craig Guy
Sent: Saturday, 9 April 2005 7:31 PM
To: Asterisk Use
The ONLY way to MAYBE play an announcement DURING a call is by using the
stuff put in for calling cards. See "show application dial"
Chris wrote:
That won't work on outgoing calls, will it?
Regard,
Chris
- Original Message -
From: "Eric Wieling aka ManxPower" <[EMAIL PROTECTED]>
To:
Bart,
Greetings. I saw your posting on the asterisk users list regarding
broadvoice. I am trying to fix my own broadvoice issue with asterisk as
well, but I have not even been able to get to your issue yet. I can
make outgoing calls, however no incoming calls work. If I may ask, how
did you
Mac Mini? Can't get much smalelr than that. BSD Core.
On Apr 11, 2005 3:55 PM, Chuck Bunn <[EMAIL PROTECTED]> wrote:
> Hi,
>
> Yes cheaper than that - do not get me wrong I love Dell hardware but I
> do not need an installed OS, CDROM, Keyboard/mouse, and floppy. Minus
> all those I can get in d
Hi all,
I am using AstCC and I have found a problem that I hope someone else has found
and fixed. When going thru this AGI script and your 'card' runs down to a
minute, it plays a series of sounds telling you have a minute left. After this,
it breaks the RTP stream (I am guessing) and either si
Might you have a serious typo in zapata.conf?
> I'm assuming I would see an error if this was bad:
> ldd /usr/lib/asterisk/modules/chan_zap.so
> linux-gate.so.1 => (0xe000)
> libpri.so.1 => /usr/lib/libpri.so.1 (0xb7f89000)
> libtonezone.so.1.0
That won't work on outgoing calls, will it?
Regard,
Chris
- Original Message -
From: "Eric Wieling aka ManxPower" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Monday, April 11, 2005 2:46 PM
Subject: Re: [Asterisk-Users] timed Loop
>
I want to use the Voicemail app and before that, I would like to play
an audio file but not billable in the Switch side. Than, to do so, I have to
be able to no send the Answer message during the play of the file. Then
after finish the file, I'w xecute the Voicemail app.
That's why I need t
I think I know whats causing this. The Sipura's/Linksys's don't like
the SIP Poke thing, the qualify=yes option. They'll normally respond,
but every once in a while they will respond in such a way that
Asterisk does not like, and Asterisk will consider them unreachable.
Maybe a patch where qualify
And, what asterisk version are you running?
> Ooops, sorry folks.. A correction..
>
> I don't have digium X100 cards, I have Digit Networks X100P clone cards..
> Don't know if it
matters, but wanted to get the facts straight :-)
>
> -Original Message-
>
>[top-posting to continue the tradition here, and because there's so
>much context below - sorry]
Sorry. I'm stuck with Outlook, and far too often forget
to fix the posting.
>How about encoding the conference number in the extension dialled?
>You can use whatever prefix you like.
Can't help but wonder if this isn't a bug in Asterisk or one of it's
modules, as there seems to be a lot of people experiencing the same
problem, seemingly with different hardware and software configurations.
Anyone know how (or if it's possible) to submit a bug report to Digium
regarding this
rm /var/lib/asterisk/astdb always worked for me.
On Apr 11, 2005 2:34 PM, Matt <[EMAIL PROTECTED]> wrote:
> If the asterisk internal database becomes corrupt... how does one dump
> it and start the database over?
> ___
> Asterisk-Users mailing list
> Ast
What it SHOULD do is, check the DC voltage on the line, and if less than
8-10 volts, consider it BUSY/ unavailable/not connected, THEN check for
dialtone before dialing. Also optionally listen for stutter dialtone before
dialing, making the detection of stutter dialtone available for some other
I'm assuming I would see an error if this was bad:
ldd /usr/lib/asterisk/modules/chan_zap.so
linux-gate.so.1 => (0xe000)
libpri.so.1 => /usr/lib/libpri.so.1 (0xb7f89000)
libtonezone.so.1.0 => /usr/lib/libtonezone.so.1.0 (0xb7f68000)
libc.so.6 => /lib/tls/libc.so
- Original Message -
From: "Eric Wieling aka ManxPower" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>; "Asterisk Users Mailing List -
Non-Commercial Discussion"
Sent: Monday, April 11, 2005 9:43 PM
Subject: Re: [Asterisk-Users] Why 's' doesn't take over unknown
extensionincontext ?
Stev
Here's a complete system for $91 US. I use this box at co located office
with * and 10 SIP-841's and works great
http://www.hcditrading.com/Shop/Control/Product/fp/vpid/1377166/vpcsid/0/SFV/29664/rid/117517
- Original Message -
From: "Ken Godee" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED
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