hi guys:
my asterisk 1.2.1 just won't play music on hold, it
will play a tiny bit of music at the beginning, then go silent, doing:
set debug 1,
found error msg:
monmp3thread: Only wrote -1 of 1600 bytes to pipe:
(11)Resource temporarily unavailable
is this a bug?
anyone know what's w
I agree. Perhaps put the time into making new cool
features so that Skype folks can look at some SIP
client and say 'wow - I want that too, let's
switch from Skype' ;)
The world would be a better place without Skype,
without proprietary standards..
Let me disagree with that. The advantage of
On Tue, Dec 20, 2005 at 09:54:50AM +0900, [EMAIL PROTECTED] wrote:
>
> I sincerely believe that it's completely non-sense to make a channel for
> Skype.
> Skype is a *proprietary* protocol. If they(ebay) don't like the idea of
> someone messing around their network,
> they will change the protoc
Has anyone an example to pass variables to a fagi script?
I have succesfull made some examples with traditional AGIs, but i could
not find a way to do with FastAGI.
Regards
--
Alberto Sagredo
Departamento Técnico
Peoplecall
Email : [EMAIL PROTECTED]
Blog: http://www.voip-novatos.es
Tel./Ph
Sorry I can't help you further. Maybe someone else can chip in.
- Waldo
On Dec 20, 2005, at 1:27 AM, Steven Job wrote:
I have:
1) nat=route
2) dtmfmode=inband
Tried that and no luck. :-(
Yes, I have local and remote (behind NAT) UIP200.
You also need to make sure to specify in the [ge
I am trying to compile and document and list of server makes and models
that are physically compatible with the new Digium TDM2400 series cards,
and which meet both the unique power requirements as well as physical
chassis "space requirements" necessary to successfully integrate these
boards.
While his if statement will never do what he expects (check for the
value "ANSWER" in $dialstatus) it should work when the call is answered
as an assignment in an if statement will pretty much always return
true... and he isn't overwriting a variable he uses in his execute
so the thing shou
Hi list,
After search thru goggle and voip-info, i discover sphinx is unpopular
among asterisk community. Simply because there is no much information
contributed :( I imply there wont be much respond for this mail :sad:
I tried to use Sphinx2 for my testbed project, but i hv some
segmentatio
Hi Christian,
Yes, it is possible. You will have to make your agi file for this.For
example abc.pl.
Use the following command in your agi file:
my $DialedNo= $AGI->get_data("sound_filename", 3);
where $DialedNo is any varaible, you can use any name.
"sound_filename" is the name of the file y
Hi Christian,
Yes, it is possible. You will have to make your agi file for this.For
example abc.pl. Use the following command in your agi file:
my $DialedNo= $AGI->get_data("sound_filename", 3);
where $DialedNo is any varaible, you can use any name.
"sound_filename" is the name of the file you
I have:
1) nat=route
2) dtmfmode=inband
Tried that and no luck. :-(
Yes, I have local and remote (behind NAT) UIP200.
You also need to make sure to specify in the [general] section:
externip
localnet
That really wouldn't matter for me since my Asterisk box is not behind NAT.
Only the
hello all,
i wanted to know when will an fxs not
send/receive audio if called to. it sends/receives when dialled from to other
fxo's, sip, conference and playback of voice files but not to other fxs's.
basically the situation is:
sip->fxs failfxo->fxs fail
fxs->sip passfxs->fxo pass
t
You can configure Asterisk to log an agent in/out of an ACD Queue with
AgentCallBacklogin, AddAgent etc, just like with any other phone. The Polycom
themselves don't seem to display any sort of visual indicater that an
appearance is logged in/out however, which is dissapointing. I'm sure you cou
Hi Code Lover,
Please try this code :
if ($dialstatus eq
"ANSWER"){$Accounting_update->execute($fdatetime,$Cuniq,$UserName,$CalledN);}
I hope it will solve your problem. We use "eq" instead of "=" in perl or cgi
for strings.
Thanks
Furqan Ahmed
Software Engineer
B.E. (Computer System), DB
In this connection there are min of 2 hybrid circuits on my end
Vonage ATA box SIPURA ATA Asterisk - SIPURA ATA (phone)
hybird 1 - 2 wire -- hybird 2 --hybird 3 (if not 841 phone)
I also think vonage has one more hybird on there end.
Since the connection
I compared your sip.conf entry against mine and the only differences
are:
I have:
1) nat=route
2) dtmfmode=inband
Yes, I have local and remote (behind NAT) UIP200.
You also need to make sure to specify in the [general] section:
externip
localnet
- Waldo
On Dec 19, 2005, at 11:19 PM, Steven
Which version of Asterisk are you using?
Version 1.2.1
Did you use a STUN server? Are you using NAT?
I meant "qualify" not "quality" :)
I just set it to qualify=no.
It doesn't give me the error anymore of UNREACHABLE (which makes sense).
But I am also still not able to dial the extension of
Asterisk 1.2.1 installation. It seems that calls are being dropped for no valid reason, completely random, in the middle of the call. I first thought that it was either the network or the VoIP provider dropping packets and confusing Asterisk into hanging up the call.
However I happened to be ru
I meant "qualify" not "quality" :)
- Waldo
On Dec 19, 2005, at 11:02 PM, Waldo Rubinstein wrote:
I've found that I have to disable quality on the UIP200 when I
switched to Asterisk 1.2.X. It worked find with 1.0.9 and under.
Which version of Asterisk are you using?
- Waldo
On Dec 19, 200
I've found that I have to disable quality on the UIP200 when I
switched to Asterisk 1.2.X. It worked find with 1.0.9 and under.
Which version of Asterisk are you using?
- Waldo
On Dec 19, 2005, at 10:34 PM, Steven Job wrote:
Having the strangest time getting the uip200 to work with Asteris
Having the strangest time getting the uip200 to work with Asterisk.
We can send outgoing calls, however we can not receive phone calls.
I have tried listening to all of the recommendations in this list such as
setting the nat=never in the sip.conf and that didn't work at all (phone
stopped regist
In sip.conf make sure that the account for the cisco phone has
[EMAIL PROTECTED] set
On 12/18/05, Tim Connolly <[EMAIL PROTECTED]> wrote:
> I'm looking for ideas on how to implement voicemail notification on
> Cisco 7960 SIP phones. Something like a light on the legacy pbx-phone
> would be per
I have nufone.net for 800 and all seems fine... although my useage is very low. 2 cents a min.
JR
On 12/17/05, Tom Vile <[EMAIL PROTECTED]> wrote:
Looking for a good toll free DID provider. Any suggestions?All ready tried Sellvoip and Gafachi and the experience was not desirable.Thanks,Tom Vile
On 12/18/05, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
> Hello,
>
> Polycom ip soundpoint support ACD login/logout .
> Can we configure asterisk with polycom ACD support?
>
Like the Park button functionality that was recently "figured out"
here with Polycom phones, I'd bet ACD login/logout fun
On 12/19/05, Kevin P. Fleming <[EMAIL PROTECTED]> wrote:
> BJ Weschke wrote:
>
> > You should be fine with the 411P because I believe the latest version
> > of the card has the software updateable PROM to deal with the
> > different PCI bridge chips that were previously causing problems. The
> > s
On 12/19/05, Nikhil Yogesh Jogia <[EMAIL PROTECTED]> wrote:
>
> It turns out the you were right the first time - it needed to be _ALERT_INFO
> not ALERT_INFO. Strange, considering I am using asterisk 1.2.1 downloaded
> from the asterisk home page.
> ___
Tom,
I've been very pleased with Clearpath (www.clearpath1.com). I've used
them for two years a never had any downtime. Their web site is lame,
but call or email and they will respond.
Michael Graves
On Sat, 17 Dec 2005 20:28:56 -0500, Tom Vile wrote:
>Looking for a good toll free DID provider
On Tuesday 20 December 2005 08:01 am, BJ Weschke wrote:
> On 12/19/05, Mojo with Horan & Company, LLC <[EMAIL PROTECTED]>
wrote:
> > I'm pretty sure ALERT_INFO became _ALERT_INFO, so change your line to
> > read: SetVar(_ALERT_INFO=bellcore-r4)
> > and it should work again.
>
> Actually, having j
BJ Weschke wrote:
You should be fine with the 411P because I believe the latest version
of the card has the software updateable PROM to deal with the
different PCI bridge chips that were previously causing problems. The
single and dual span cards do not have this and will have problems
with cer
Dakota wrote:
Are there any IP Phones that can work with Asterisk, that cost less
than $60 ?
if so, what's the model/manufacturer?
Dakota
___
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Asterisk-Users mailing list
To UNSUBSCRIBE or updat
Are there any IP Phones that can work with Asterisk, that cost less than $60
?
if so, what's the model/manufacturer?
Dakota
___
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Asterisk-Users mailing list
To UNSUBSCRIBE or update options vis
I'm looking for ideas on how to implement voicemail notification on
Cisco 7960 SIP phones. Something like a light on the legacy pbx-phone
would be perfect. Even maybe go so far as a quick ring to the extension
every 15 minutes or so, but then that would increment the on-screen
missed call co
I sincerely believe that it's completely non-sense to make a channel for
Skype.
Skype is a *proprietary* protocol. If they(ebay) don't like the idea of
someone messing around their network,
they will change the protocol specification, launching a new version, for
example, and *all* the work a
When a channel is released only a part of memory is released, not all,
so in few days I have no more memory, only swap...
This system is a Slackware 10.1 with kernel 2.6.12
Asterisk CVS, wct2xxp with unicall.
When I stop the asterisk no memory is released, it 's only whith reboot command.
I have
Harry,
We have a ibm x346 with a one port Digium E1 card in it.
Ilan
On 12/19/05, Harry McGregor <[EMAIL PROTECTED]> wrote:
Hi,Has anyone used a Digium PRI card in an IBM eServer x346? I know thatDigium's website lists the x345 as having problems, but I am restrictedto buying only IBM eServer
Greetings,
We have just completed a successful test of DNS SRV failover with
Polycom phones, but feel that the failover was a little sluggish (up to
15 minutes for some phones).
We are using the default Polycom settings for Retry Interval and other
related settings and I'm wondering if anyon
On 12/19/05, Jolly M. Recto <[EMAIL PROTECTED]> wrote:
> Harry McGregor wrote:
> > Hi,
> >
> > Has anyone used a Digium PRI card in an IBM eServer x346? I know that
> > Digium's website lists the x345 as having problems, but I am restricted
> > to buying only IBM eServers for this possible project
On Mon, Dec 19, 2005 at 03:18:18PM -0800, Kerry Garrison wrote:
> I don't know exactly how it works, but since it appears to just be SIP, I
> would have to assume a STUN setup. I haven't bothered to sit there and watch
> the packets go by to see what its doing under the hood.
thanks - luigi
__
Harry McGregor wrote:
Hi,
Has anyone used a Digium PRI card in an IBM eServer x346? I know that
Digium's website lists the x345 as having problems, but I am restricted
to buying only IBM eServers for this possible project.
I would like to use the TE411P
Harry i am using 3 deferent IBM eServ
Greetings all,
A couple of clients have recently decided they'd like extensions to their
office PBXs at their homes, so they've duly been provided with preconfigured
phones which register with the Asterisk server at their offices (public IPs)
quite happily.
Every 3-5 days it seems that these remo
On 12/19/05, BJ Weschke <[EMAIL PROTECTED]> wrote:
> On 12/19/05, Mojo with Horan & Company, LLC <[EMAIL PROTECTED]> wrote:
> > I'm pretty sure ALERT_INFO became _ALERT_INFO, so change your line to read:
> > SetVar(_ALERT_INFO=bellcore-r4)
> > and it should work again.
> >
>
> Actually, having jus
Yes call progress tones (busy) can be generated by ATA devices. Not
sure if the 486 has a digitmap or not, may wish to checkit out if so.
I also believe it has an internal status webpage and syslog info you
could look at for clues.
On Dec 19, 2005, at 5:34 PM, Craig Bruenderman wrote:
I'v
> Asterisk -r makes no mention of any activity when this occurs
> so it seems that Asterisk is not even generating the busy
> signal. Is the Handytone capable of doing this and if so, why
> would it be?
Make sure "early dial" is disabled in your HT486 config. I've never been
able to get it work
On 12/19/05, Mojo with Horan & Company, LLC <[EMAIL PROTECTED]> wrote:
> I'm pretty sure ALERT_INFO became _ALERT_INFO, so change your line to read:
> SetVar(_ALERT_INFO=bellcore-r4)
> and it should work again.
>
Actually, having just checked then chan_sip.c source, ALERT_INFO
should be working.
I'm pretty sure ALERT_INFO became _ALERT_INFO, so change your line to read:
SetVar(_ALERT_INFO=bellcore-r4)
and it should work again.
Moj
Nikhil Yogesh Jogia wrote:
Hi All,
The setup I have is an asterisk box with a sipura spa 2000 for the fxs ports
and a TDM400P with 2 fxo ports.
Anyway, I
I've got my Handytone 486 registered fine with SIP. Using an analog phone attached to it, I can dial 2 digit extensions in the main context just fine. I have a DID mapped to it from the outside which I can also dial and ring through to fine. For some reason thoug, a 7 digit or 10 digit dial string
On 14:56, Mon 19 Dec 05, Kerry Garrison wrote:
> Yes you can send and receive calls via Asterisk.
>
> http://voipspeak.net/index.php?/content/view/19/28/
Is there any change you can provide the sip.conf and
extensions.conf stuff this generates?
I'm not an amp user, nor do I want to use it just to
I don't know exactly how it works, but since it appears to just be SIP, I
would have to assume a STUN setup. I haven't bothered to sit there and watch
the packets go by to see what its doing under the hood.
-Kerry
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Be
You may want to check this out:
http://www.digium.com/asterisk_handbook/agentlogin_queues.html
A number of our clients use it to analyze outgoing calls from within
QueueMetrics.
Yours
l.
In data Mon, 19 Dec 2005 20:57:54 +0100, Jim Miller
<[EMAIL PROTECTED]> ha scritto:
Is there a w
On Mon, Dec 19, 2005 at 02:56:30PM -0800, Kerry Garrison wrote:
> Yes you can send and receive calls via Asterisk.
>
> http://voipspeak.net/index.php?/content/view/19/28/
so let me understand.
One nice feature of skype is the excellent (for the user; i
understand the sysadmin may see this as a ni
Hi All,
The setup I have is an asterisk box with a sipura spa 2000 for the fxs ports
and a TDM400P with 2 fxo ports.
Anyway, I recently upgraded from 1.0.9 to 1.2.1 and have found a very annoying
problem.
One of my dialplans includes this statement in order to distinguish between
the 2 incomi
Yes you can send and receive calls via Asterisk.
http://voipspeak.net/index.php?/content/view/19/28/
Kerry Garrison
Publisher - http://GeekGazette.com - http://VOIPSpeak.net
(949) 502-7819 x200 - [EMAIL PROTECTED]
http://www.techdatapros.com
-Original Message-
From: [EMAIL PROTECTED]
could you please tell how it interfaces with Asterisk? Could I receive calls
into Asterisk? send calls out?
I've just downloaded it and am searching (unsuccessfully) for these on
Gizmo's site/software.
- Original Message -
From: "Kerry Garrison" <[EMAIL PROTECTED]>
To: "'Asterisk Users
Hi everybody!
Jonathan wrote:
>
> Hi,
>
> I'm using a td400p card with an FXO port and asterisk 1.2.1 in South
> Korea and asterisk isn't detecting when PSTN callers hangup.
> I've gone through all the settings related to hangup detection and none
> work. I've tried:
> hanguponpolarityswit
Thanks for this information!
Lan
- Original Message -
From:
ram
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Friday, December 16, 2005 8:50
PM
Subject: Re: [Asterisk-Users] A2billing
Trunk
Hi
check this URL will help you for
Well this seems doable although can extensions be contexted? Meaning
that can you have an extension of 100 for company A and the same for
company B?
Scott.
On 6/30/05, Iqbal <[EMAIL PROTECTED]> wrote:
> Hi
>
> If I have ser sending calls to asterisk, is there a way to get a
> different block cal
Hi Chuck,2005/12/17, Chuck Bunn <[EMAIL PROTECTED]>:
If you do not have QOS assigned to the SIP protocol it is quite possiblethat there are packets time outs and the packets are discarded. Is itpossible to test the network during the evening or at a time whentraffic is at it lowest?
It took me some
Hello,
Does it make a difference for the NAT traversal capabilities of Asterisk
if the users are registered directly to Asterisk or if they are
registered to a SIP proxy which just relays the calls to Asterisk?
Are there any cases where Asterisk would be able to traverse the NAT if
the user is re
We can provide this with IPSEC DES - 3DES or AES encryption
We are based in Strasbourg ( near Basel - Switzerland )
Best regards
Thierry
tél: +33 (0)3 90 40 06 75
fax: +33 (0)3 90 40 06 75
email: [EMAIL PROTECTED]
web: http://www.widevoip.com
> -Message d'origine-
> De : [EMAIL PROTECT
C F wrote:
I geuss this explains why you should test :)
It should work, you miscofigured something.
I am testing, it is a test setup. The ATA rings a phone connected to it,
and the Channel bank answers a call on an incoming CO line. Any ideas?
--
Chris Mason
NetConcepts
(264) 497-5670 Fax:
Hi,
I'm using a td400p
card with an FXO port and asterisk 1.2.1 in South Korea and asterisk isn't
detecting when PSTN callers hangup.
I've gone through
all the settings related to hangup detection and none work. I've
tried:
hanguponpolarityswitch=yes
callprogress=yes
busydetec
Hello,
I have two questions.
First, Can I redirect calls with a regular sip-phone like ZyXEL Prestige 2000W,
typing something like #0#SIPNUM# or another key-combination?
I have a queue configured. How can I configure it to jump to voicemail when
pressing like, # when in the queue? and is there
Have you considered the Sangoma cards? I have an a102 running in 2x
X306's and they're running fantastic.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Harry
McGregor
Sent: December 19, 2005 4:20 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk
The 8 byte chunk size is used in zaptel because:
1. For lowest "cost" echo cancellation in software you need the lowest
possible delay between outbound and inbound data streams, which demands the
minimum possible buffering.
2. The 8 byte chunk per channel is the source of the millisecond timer t
I geuss this explains why you should test :)
It should work, you miscofigured something.
On 12/19/05, Chris Mason (Lists) <[EMAIL PROTECTED]> wrote:
> For testing purposed I want to simulate a CO line into my channel bank
> using an FXS ATA. I thought I could use a Sipura 1001 ATA to feed to the
>
Have it send an email to everyone that there is a voicemail in the box, then
anyone can log into the voicemail system and retreive it
Kerry Garrison
Director of Technical Services
Tech Data Pros - Orange County's Mobile IT Service Provider
(949) 502-7819 x200 - [EMAIL PROTECTED]
http://www.techdat
Hi,
Has anyone used a Digium PRI card in an IBM eServer x346? I know that
Digium's website lists the x345 as having problems, but I am restricted
to buying only IBM eServers for this possible project.
I would like to use the TE411P
Harry
I have four DIDs. 2400,2401,2402, and 2403
There is no phone attached to 2400 but the other three DIDs do have
phones attached
All the four DIDs have their own voicemail and voicemail works on all
the DIDs. When you dial 2400 it rings the other three numbers. If no
one picks up, it goes to the
note:
What I have found is that when many people say bad, they really mean
different.
(from training staff on Asterisk systems)
PaulH
- Original Message -
From: "James Sturges" <[EMAIL PROTECTED]>
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
Sent: Tuesday, December 2
For testing purposed I want to simulate a CO line into my channel bank
using an FXS ATA. I thought I could use a Sipura 1001 ATA to feed to the
FXO module of the Adtran 750 channel bank, FXS feeds so why does this
not work? Does the channel bank rely on seeing something different that
an analog
Does anybody know of any VoIP provider in Switzerland (or other Euro country
not far from it) that could give me a DID with VPN termination. What I need
is to have a SIP or IAX connection encrypted inside a VPN (ipsec preferably)
to make and receive calls. Fax support would be a huge plus.
Than
Everyone should simply uninstall Skype and switch to the Gizmo project
because it interfaces quite nicely with Asterisk.
Kerry Garrison
Publisher - http://GeekGazette.com - http://VOIPSpeak.net
(949) 502-7819 x200 - [EMAIL PROTECTED]
http://www.techdatapros.com
-Original Message-
From:
Mark Hulber wrote:
The paper is definitely interesting and I commend them for their effort
but it doesn't represent a complete understanding of the Skype protocol
to the extent that an Asterisk server could speak the Skype protocol.
They say that much of the Skype protocol is encrypted and nee
yikes,
25 and 110 will allow mail - but please without the whole digest attached.
And wouldn't your question be more useful with a better subject field? now
no one will see me addressing a question I know an answer to.
___
--Bandwidth and Colocation
Is there a way to use the manager interface to originate a call between
an outside phone number and an inside queue?
I've thought of maybe just doing it by originating the call between the
outside phone number and an outside DiD that gets routed to a queue, but
that seems messy.
Thanks for your h
The paper is definitely interesting and I commend them for their effort
but it doesn't represent a complete understanding of the Skype protocol
to the extent that an Asterisk server could speak the Skype protocol.
They say that much of the Skype protocol is encrypted and needs to be
inferred t
On Mon, Dec 19, 2005 at 06:36:16AM -0600, Rich Adamson wrote:
>
ok rick i will check this directives and write you again.. thanks
> > ok rick all of my conf...
> > asterisk 1.2.1
> > zaptel 1.2.1
> >
> > i have a pbx simple with digital phones in one side. and the other side
> > are xten
I think the broken pipe issue is related with the mpg123 player,
try disabling moh and see if it behaves the same way
On 12/19/05, Maximiliano J. Goldsmid <[EMAIL PROTECTED]> wrote:
> I have the same problem !!
> :-(
>
>
> 2005/12/18, Mohammad Shokuie <[EMAIL PROTECTED]>:
> > Hi there,
> >
> > Any
I am trying to configure zapata.conf to handle distinctive ring.
Everytime someone calls my main number, I get a ring pattern of 0,0,0
which works consistently. The problem is that every time someone calls
one of the other phone numbers (same number each time), I get a
different ring pattern
I have the same problem !!
:-(
2005/12/18, Mohammad Shokuie <[EMAIL PROTECTED]>:
> Hi there,
>
> Any one confronted a crash in asterisk when using mixmonitor app. When i'm
> using the mixmonitor app on a briged call as soon as the called party hangs
> up the call asterisk crashes and the process
> I've got an * sitting behind a linux iptables firewall. I have an
> account with teliax. After entering the registration information
> accurately and restarting *, iax2 show registry shows a registered
> status on that connection.
>
> However, whenever I try to place a call, I get a "No on
What hardware/?
Sorry I have missed part of the message?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Evert Meulie
Sent: Monday, December 19, 2005 4:03 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Re: Does hardware like this exist...
No, I don't know why. But, the dtmf mode used between the phone and asterisk
stays the same regardless of where you call. That part of the call doesn't
change. Once asterisk has the dtmf digit(s), the next channel (eg, zap, iax)
will use whatever "it" deems to be the correct dtmf mode (unless you h
On Mon, 2005-12-19 at 10:13 -0800, Wolfgang S. Rupprecht wrote:
> >> from: http://www.trac.org/news/2005/tracnotes-vol-3-22.html
> >
> > The scam isn't new, and its certainly not limited to home 800 numbers.
> > The same basic principles were used by some of the 900 number folks
> > a few years
I have a question about how to set up some variables. I have
an extension that my teachers are going to dial to record their information
messages for the public. I need to know what I need to put in step two for my
teachers to enter 6 digits on their phone and have that saved as a variable.
We can call out almost every number. But when calling to numbers with automatic
attenders the
asterisk returns a NO ANSWER as dial status, like the number doesn't exist. Can
any one help as?
We have no idea about was it's happening.
We are runnig an Asterisk 1.2 with a TE210p digium card.
thx
>> from: http://www.trac.org/news/2005/tracnotes-vol-3-22.html
>
> The scam isn't new, and its certainly not limited to home 800 numbers.
> The same basic principles were used by some of the 900 number folks
> a few years ago as well.
My fear wasn't that someone would stuff phony charges on m
As it turns out I can dial from the Infinity PBX into the Asterisk
box. So it must be something to do with contexts or configs I guess.
So when I set up the ZAP trunk in AMP it automatically did it as
ZAP/g0. Well I just assumed that was the first one. Most spans
numbering that I deal with st
do i need any ports open inorder to use send mail from behind a router
[EMAIL PROTECTED] wrote:
Send Asterisk-Users mailing list submissions to
asterisk-users@lists.digium.com
To subscribe or unsubscribe via the World Wide Web, visit
http://lists.digium.com/mailman/listinfo/aste
($dialstatus="ANSWER") replace with $dialstatus eq "ANSWER"
perl treats = as an assignment operator. For comparison, you need eq or == .
-apu
On 12/18/05, Code Lover <[EMAIL PROTECTED]> wrote:
Hi all,I wanted to execute one of mySQL query when the call is answered itried with the following code
Lists wrote:
I have gotten the tftp server working and the 9133i is doing a firmware
update and finds the aastra.cfg file as well as the 00XXX.mac file. The
issue is that I can't figure out what is wrong in the configuration files
that it is not loading the extension, proxy, etc. info.
Could s
nr k wrote:
HI all
How to configure voice mail in asterisk . pls do the needful.
regards
ramakrishnan.n
here is the wiki page on voicemail
http://www.voip-info.org/wiki-Asterisk+config+voicemail.conf
Tom
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Hi all,
I've got an * sitting behind a linux iptables firewall. I have an
account with teliax. After entering the registration information
accurately and restarting *, iax2 show registry shows a registered
status on that connection.
However, whenever I try to place a call, I get a "No one
When I used MixMonitor, and I hangup the channel, Asterisk exit whit error in segment. Why? Thanks
Yahoo! Messenger: chiamate gratuite in tutto il mondo ___
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Asterisk-Users mailing list
To UNSUBS
The only other thing I can think of is that your
contexts etc... need checked.
It would be very helpful to know if calls can come into
the system from the PBX, that would be the only way to know the span is alive
and well truely. Once you know that then its down to the contexts and
conf
No matter what I try, I can't seem to pass variables to the local channel
when using the Originate command through PHPAGI. Here is a small snippet of
the PHP code:
/***
**/
$channel = 'Local/[EMAIL PROTECTED]/1';
$exten = '';
My other pbx vendor told me they supported pretty much all of the
switchtypes and that the system would automatically detect the correct
one. I've tried Qsig and National and both seem to bring the span up
fine.
I just switched to span=1,0,0,esf,b8zs
to have asterisk provide the timing. That
The parameter in zaptel.conf that sets up timing etc
is:
span=1,1,0,esf,b8zs
The first 1 means this is span 1.
The second one defines the timing of the link. For asterisk to provide the
timing use 0 instead. For instance my Asterisk box,
hooked directly to my Avaya G3 uses:
span=1,0,0
I unsubscribed but I am still getting
emails to this account.
Please remove [EMAIL PROTECTED]
I think that the moderator added my reply
to address to the list.
Thanks for your help.
-Jason
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael Sampson
Sent: M
Well, I formated the system installed slackware 10.2 compiled a new
kernel with 2.4.31, modprobe usb-uhci, install zaptel, no problem,
modprobe ztdummy, modprobe zaptel. Got this in lsmod
ztdummy 1656 0 (unused)
zaptel219520 14 [ztdummy]
usb-uhci
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