> I want to use this to call on to a Telecom line(PSTN) and vice versa. I read
> somewhere that we need to use some provider for it like FWD or iconnect, do
> we need to use them to make outgoing and incoming calls to PSTN lines or we
> can do it without them.
Try reading these articles:
http://ww
> How can I have asterisk ignore incoming rings so it doesn't answer a
> specific line. I tried setting up an empty context section but that didn't
> work.
Make a long delay the first line of the phone's context. This can even
be turned on and off using a few more lines.
> using Allison for the English prompts and are looking for
> recommendations for Spanish.
You could check here: http://declic.com/voices/
There are 10 Spanish-speakers listed
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> My message to the list was definitely flame bait for my
> ignorance, thanks for just giving me links and a point in the right
> direction.
In an ideal world, where insecure people wouldn't need constant
validation on mailing lists to bolster their self worth, what just
happened here would be mo
> I want to set up a * box that connects to my telephone line so I can
> make calls using VOIP. Do I have to sign up for a service to do this
Yes, read the first doc below.
> can I buy a card that goes into my * box that allows me to use VOIP over
> my telephone line. If so, what card do I need?
> I'm trying to use some VoIP phones behind a Linksys WRT54G router but can't
> get them to register.
I am doing this now.
> Am I missing some trick to get Linksys to cooperate with my asterisk setup?
Are you forwarding ports on the WRT54g to the phone?
> On the phone I'm using BT100 .. and
> When at the CLI, "show channels" shows nothing.
Look for ztcfg
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I'm posting this here because I'm betting many of you are qualified
and someone may be interested. Please, no flames, just act if this is
something that interests you, it may be worthwhile. If not move on.
Saw this on comp.dcom.voice-over-ip. I want and looked at the site and
they do have several
Today I ran a different box with Windows XP on a different router and
disconnected the phones from the filter. I ran pingplotter, which is
like mtr for windows and was able to determine that there is nearly
always about 8 minutes between these glitches.
I pinged a server of mine in the US and the
Slightly OT but if you know much about networks, please read and maybe you can suggest some things.
I have been running asterisk for about 1 year. Currently running 1.0.6
with 2 X100P and 1 TDM400P with 3 FXS modules on a Pentium III-800 with
512M RAM. The system performs very well for both POTS a
> the recommended documentation source to study and understand dial plans.
Here's a few
http://www.voip-info.org/wiki-Asterisk+Dialplan+Introduction
http://www.voip-info.org/wiki-Asterisk+config+extensions.conf
http://www.voip-info.org/wiki-Asterisk+Extension+Matching
http://www.voip-info.org/wik
> > Are you using nat=yes in sip.conf entries and giving the externip and
> > localnet parameters?
> Also set the canreinvite=no for the external phones. that
I keep forgetting that, thanks!
So to recap in a single post:
In sip.conf
[general]
port=5060
; other stuff
externip=your.ip.addr.here
Sign it: http://www.petitiononline.com/mod_perl/signed.cgi?IAXPhone
> Now it would be even more interesting to see if Cisco or maybe
> Siemens/Polycom would bring out a firmware for IAX, now that would be a
> revolution.. :)
Cisco et al won't exactly be blown away by the not even 200 sigs :)
I
> However, and I know this is a running issues, I cannot get external sip
> users behind a NAT to be able to successfully connect to asterisk when it's
> behind a NAT as well.
> I have done port forwarding at both ends dealing with the usual ports of
> 5060, 4569 and 5036 as well as opening up the
http://www.petitiononline.com/mod_perl/signed.cgi?IAXPhone
Sorry for the 170 or so who have already signed. This list supposedly
has 10,000 or more subscribers. 170 isn't very impressive. Please
sign!
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Ancillary question, I wonder if anyone knows a source for high-end
(but cheap because they're a generation behind) PIII motherboards and
CPU that would be a good match for * ? That should keep the price
down? Rather than look for those small specialized but expensive
boxes.
Hi,
I am running 1.0.6 STABLE and I use SMS occasionally, but not often. I
wanted to send one today and discovered that the SMS app would still
receive and process them, but smsq doesn't seem to be formatting the
call files properly.
What is missing in the .call file is the destination number and
> My first szenario connects two servers via IAX2. One is static IP the second
> is a nated dnyamic host. I could register the dynamic host succesfully at
> the static one. Routing calls to it with my dialplan gets denied/rejected
> due to missing authority on the remote side. I REALLY put this up
> [marco-voicerec]
> exten => s,1,noop(${ARG1})
> exten => s,2,Background(custom/recordwarn)
A nice thought, to name macros for Mark, "marco". Won't work in the
dialplan though.
Also, *8 is usually used for picking up a ringing phone. See features.conf.
___
> After dialing the extension it takes about 10 sec
> for the other phone to ring. A little more the
> Cisco to ATCOM, a little less in the other direction.
Look at the CLI when you dial a number. How does the dialplan look? It
sounds like asterisk is waitng for more digits.
_
> Has anyone had any experience with the IAX2 phones being marketed by
> Netweb?
I have received one and am waiting for a second one. There is an
extensive wiki page discussing the first phone which is now obsolete,
the 302. I'd agree with most of what is said there. However, for the
price these a
> deploy VoIP, when a call arrives it comes on operator
> phone; but sometimes operator is away (coffee break
> and so), the other workers in the room have
> possibility to intercept an incoming call by pressing
> *02 (note that they can
> intercept not only operator phone but any number
> ringing
> Good point! I was actually meaning more the commands to put a call on
> hold, forward to other extensions etc.
That's called documentation and is given almost daily here
http://asteriskdocs.org - a large downloadable PDF is available
Starter tutorial:
http://www.onlamp.com/pub/a/onlamp/2003/0
> 0: 184892 XT-PIC timer
> 1: 5 XT-PIC keyboard
> 2: 0 XT-PIC cascade
> 8: 1 XT-PIC rtc
> 11:3585589 XT-PIC wcfxo, ztdummy, usb-uhci, eth0
> 14: 5360 XT-PIC ide0
> 15: 0
> Just wondering if there's an operators guide around somewhere for
> Asterisk that I can give to my users (like the little guides you get
> with a new PABX system).
Wouldn't it need to be written by the person who did the dialplan? I
finally got around to doing mine 1 year after install :)
__
> The problem is - and i was wandering if anyone knows the solution - is
> that When I dial from my windows machine,
> to an external phone line through Zap, then the receiving party does not
> hear my voice - but when the receiving party
Do you have Transmit Silence=YES on the X-Lite? It's under
> Where can I find a good how-to to do this job. A small starting
> how-to that let me understand the principles of setting a PBX with
> asterisk. The handbook does not like starting guide.
Try this:
http://automated.it/guidetoasterisk.htm
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> = An incoming call is received on any of the incoming POTS ZAP channels
> = Call is immediately picked up by Asterisk and the caller is greeted with a
> message like, "Thank you for calling , please hold the line your call
> will be attended to shortly"MUSIC
> = What I would like to
> How do I dialout using an extensions.conf and connect to an outside number?
> For example, I would like for a person in the IVR to be able to press a
> number and it dial out using another FXO card and another POTS line and
> then bridge the two calls together.
So you've given up on trying to u
> Is there a list of these anywhere? This is now the third one I've heard
> of, with no documentation: lowpower (IIRC), robust and now boostringer.
> Do I have to go diving in the source, or is there a Wiki I can't find?
Good point! There is the bugtracker to search but this is one of those
subj
> > Is it possible to have 2 (working) iax2 phones behind port restriced nat?
Interesting you ask, since I just had an incident concerning this. I
have an IAXy and got an IAX hardphone which I tested at home behind
the same NAT. Using IAX soft clients before in this situation, they
would work, but
> connected to one of them. Basically my goal is to have someone call into the
> incoming POTS line and be presented with a menu where they would select an
> exten => 1,2,Goto,cellphone|s|1
Nice try, but take a look here:
http://www.voip-info.org/wiki-Asterisk+tips+ivr+menu
or here
http://users.p
> I'm dialling 17094009, as instructed in the BT documentation. Where
> does the extra digit go?
Quoting the wiki at http://www.voip-info.org/wiki-Asterisk+cmd+Sms
"The specification for landline SMS allows for the possibility of more
than one device on a single landline. These can be configured
> Incoming however just isn't working. I've got a nice list of numbers
> from which SMS messages come:
You are sending the extra digit to say which "mailbox" the message is
for, right? In this country, if you do not send that digit, it will
try to vocalize the message during the calls.
_
> I am a beginer trying to install my first TDM04B.
Hi RR,
> [outgoing]
> exten => _0X.,1,Zap/1/${EXTEN}
> I cant send them out.
The error is telling you that ZAP is not an application. To dial out
you need the dial application exactly as you have in the incoming
section. Something like this:
Hi Andrea,
> register => user:secret:[EMAIL PROTECTED]:port/extension
> well, it's only local, or is important for authentication on external
> sip server?
As far as I understand it, that extension will be the one the external
server requests. Consider FWD example:
register = 123456:[EMAIL PROTE
> If this handles the case where 10 digits are required:
>
> exten => _9NX,1,StripMSD,1
>
> exten => _NX,2,Dial,Zap/4/BYEXTENSION
>
> How do you create a handler which works for either this or
> the case with a leading '1' plus 10 digits?
I'd start by looking at this material:
> did was throw away the original brick and connected a generic one where
> you can select voltage and polarity.
You have to be careful about those. Some of them have lousy regulation
and just send otu 15vdc when set to 12 or lower. In fact, my first
supply was a 1A adjustable but I had lots of p
> for the same price, siemens dect phones with sms feature
> display caller name
The C200 with 2 C2 sets does SMS and now displays CallerID name
(thanks to your post). They just don't have color screen.
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> I have the same problems with siemens dect phones S100
> The caller id don't work on tdm...
> In France, the CID is differant than other country.
> Then standard ring cadence is: 1500 3000 1500 3 and so...
>
> The etsi standard (used in France) say:
>
> The short ring is called Ring Pulse A
Got posted unfinished:
>Dial("SIP/casa1-7552", "IAX2/fwd-gw/*18005551212|60|rwt")
Take a *close* look at that line and compare it to the one on the FWD page:
Dial(IAX2/${FWDNUMBER}:
${FWDPASSWORD}@
iax2.fwdnet.net/${EXTEN:3},60,r)
or to simplify
Dial(IAX2/[EMAIL PROTECTED]/*18005551212,45)
___
> Dial("SIP/casa1-7552","IAX2/fwd-gw/*18005551212|60|rwt")
Take a *close* look at that line
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> Besides that, how would you make * work for sip if for example this:
>
> Softphone - nat - internet - nat - *
My grandson did it this way:
http://willypick.mindsay.com/?entry=10
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> I am looking for a solution that will make a call based on SMS request. Can
> you solve this problem with Asterisk?
If you are in a place where SMS is received on regular phone lines,
there is the SMS application that comes standard in asterisk to
receive and store the messages. You can then use
> So I went a step ahead and configured eyebeam to connect not to astersis
> kinternal IP but rather the external firewalls IP, much like any other
> external computer would do.
Why? What you did doesn't accurately simulate someone on the outside
trying to get through your firewall because you are
> Can you dial 800 and 888 toll free numbers using FWD? how do you dial them
> cause I tried using 1800x and 1888x and I simply get a "nobody can
> asnwer the call" signal on asterisk.
>
> Can you dial 800 toll free from FWD?
The best answer to that question is on the fwd site. Last time
> IAXy users on this list -- who of you would trust their home/business
> security to an IAXy? Is there a silent majority who loves this product,
> or are the over-heating, looking-up issues reported as commonplace as
> the frequency of the posts would indicate?
I would not trust the above to voI
> Sorry for my bad english. I'm a french guy.
Absolument rien à critiquer de ton anglais
> I have the same problems with siemens dect phones S100
> The caller id don't work on tdm...
> Try adding
> cadence=250,1500,1500,3000,1500,3000
> In zapata.conf
> And use in extension.conf
> exten => 200,
> What are the CID issues? I am using Siemens Gigaset DECT phones and CID
> (albeit internal - not from PSTN) is working.
The phones are C200, one base two phones. Connected to the PSTN, CID
works perfectly. From the TDM400 they do not, no matter what I've
tried.
When Digium ssh'ed in, he changed
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
> I have searched the list and the wiki and have seen references to
> changing this in the wcfxs.c file but I am not using that. Likewise, I
> have not founf anything in by looking into the wctdm.c file. I am no
> programmer but can somewhat follow the code.
This is one of the best kept secrets
> forum-like tool to search thru the posts by keyworks for example.
You can use google by specifying site:lists.digium.com before or after the words
Most if not all of your questions are answered on the wiki (which does
not seem to be responding as I write this) and at sites like
http://www.ast
> I need to be able to send an sms alert to one's mobile/cell phone. For
> instance, when I receive a voicemail message in my inbox, I also want to
> be able to get a message on my cell phone alerting me of this e-mail. How
> possible is this?
This is probably cheating: I have a free email accoun
> Where is the mistake?
Right here
> [local]
> exten => _XX,1,Dial(IAX2/${EXTEN})
You are telling it to dial ALL 10 digit numbers using IAX. What you
need to do is have an extension that determines if a call is meany for
the X100P and if so the dial would be
> exten => _XX,1,Dia
> This post is a FWDout specific one, but may be of interest for many
> Asterisk users and may even make more of them use FWDout for the good
> of everyone :)
Sam, some people have come to the conclusion that while FWDOut is a
ince idea, it isn't a good idea. The first two things that come to
mind
Say Jim, that bridge in Manhattan I bough from you last yera? It's
still rusty! :)
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> My problem is as well described on
> http://www.voip-info.org/wiki-Asterisk+i+extension
Yup. I believe the 'i' reacts to dialed digits input during that
particular extension. There aren't any. That wiki page concludes that
you need a "fallthrough" extension like
_.,1,Playback(Idiot)
_.,2,Hangup
> This works exactly as i want so users basically can dial 0, wait for the
> dialtone and then dial the requested number.
>
> The only problem that i have is that from when a user dial 0 to when i get
> the dialtone from the telephone line, something like 5 seconds pass... is it
> possible to pull
> Does anyone know of a Print-to-Fax client that works with asterisk &
> spandsp? Astfax is a partial solution but that only lets us email the fax
> in, we'ld like to set it up so the user can hit the print button and send
> the fax (even if all it does is email - transparently to the user - the fa
> I would like to deploy a (very) small PBX at my place, so that I can
> stop answering phones for my kids or my wife, using distinctive
> ringings.
Why not just buy a phone capable of distinctive ringing? I think
Siemens makes a few for example?
___
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> http://www.farfon.com/
> http://ipphone.eezeephone.com/
>
> Looks like all URLs on IAX-capable phones, http://www.iaxtalk.com/
> included, point to China. Interesting...
Farfon is in Pakistan, not China
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This is a heads up, I heard nothing about this I suppose in part
because few of you use SMS, but note that the SMS application has
changed the directory where it stores messages and the format of the
file they are stored in.
If ythe author or someone could chime in with a valid URL that
mentions t
> This setup can cause any problems to the comunication process? I'm aware
> that the IAX2 protocol is NAT friendly so I think this will work, but to
> be sure I want to hear some oppinions.
Should be no problems with this
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> The rtp ports used for voice (1:2 in your example) vary by phone type.
> Cisco uses a different range of ports, Xten another range, Grandsteam yet
> another. The ports you have listed are what asterisk uses and are probably
> not the same ports as what your remote phones use. Therefore, t
> But this doesn't work when I press any non-existent extension I get
> congestion.
Can you put your 'i' extension directly in [internal] ?
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> HP's lawyers have contacted me about the conflict.
I suppose lawyers suffered as much as other businesses when the bubble
burst, but now they're coming back strong. Funny it took them a few
years to find your domain name :)
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> You would need to somehow have an external fixed address which would
> redirect all of your traffic to the dynamic address, I have found no way to
> do this, your best bet is to pony up the extra cash for a fixed address
> (usually 3-4x the cost)
>
> I would love to hear if anyone has figured
> For anyone playing around with IAXy(S100i) devices, I am making the
> following available:
>
> Windows IAXy Provision v1.00
Tony, thanks for this, it was sorely needed! Especially useful when
travelling to an office that has MS only boxes for example.
___
I have two phones which are callerid num and name capable connected to
asterisk 1.0.3. Both of these phones will display number and name of
caller when available and when connected to the French phone company
(France Télécom). However, one of these phones will not show it on
asterisk connected via
> I just want one of my incoming numbers to go to an IVR service that will
> allow me to select what I want.
IVR is a key word. Try this:
http://asteriskdocs.org
There is a movie theatre example that will give you this.
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> Just need to learn how to configure services now so that I can put a menu on
> one of my numbers!
Elaborate please, I'm not clear on "put a menu on one of my numbers".
Give an example of what you want to accomplish and I'm sure many
people here will help you.
The answers to the questions you've been asking are probably here:
Starter articles:
http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html
http://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html
Full install etc.
http://automated.it/guidetoasterisk.htm
And of course:
http://www.asteri
Yousri,
You may want to look at these two articles which will give you a good
idea of what hardware you need for a simple system with one or two
phones and phone lines:
http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html
http://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html
__
> > They can always check the archives to read up on missed posts, and it
> > would save us all the trouble in the mean time ;-)
Isn't it obvious that with a choice of hundreds of free email
providers, anyone who wants to avois this problem need only use a
throwaway account like gmail with mailing
Hello all,
In changing the phone connected to my IAXy I remembered that this
phone will not ring on the standard frquency (20hz?) when connected to
the FXS. There is a patch, one line to change in wcfxs.c for use with
a TDM400P, but obviously this doesn't change the FXS on the IAXy.
Is there any
> However, outbound calls use our secondary number rather than our primary
> number first. This undesirable because of Caller ID; we'd like the primary
> number to appear instead.
> Yes, I can correct this by swapping the PSTN lines. But which config file
> assigns the physical FXO ports to PSTN li
> Have you tested it directly on a PSTN line?
Repeating the third sentence in my original message:
"CallerID is correctly shown when the phone is hooked up directly to
the PSTN. "
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I just bought one of these for the office. It has all the features we
wanted including Caller name and number display and SMS capability.
CallerID is correctly shown when the phone is hooked up directly to
the PSTN. The terms look the same in the book (CLIP, CLID).
When Hooked to an FXS in asteris
> I have a problem with ZAP interface bridging in France (FXO
> interface): hangup is detected through a busy tone (no polarity
> inversion or whatever). When I dial out from a zap line when I receive
> an incoming call on another zap line (for example to redirect calls to
> my office when I'm not
> We have an open support incident with Digium but have not yet heard back.
> FWIW we have stopped selling & deploying the IAXys until we have a
> resolution to the problem.
Bryan what kind of power supplies do you use?
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A
> How can I make it so
> Extension 500 always dials out on line 1 and extension 600 always dials out
> on line 2. Below is the relevant snippets of the extensions.conf, sip.conf,
> and Zapata.conf.
>
> extensions.conf
Remove the groups an dput "hard wired" channels 1 and 2
> [sales1]
> exten
> still in the mess. So I want to know where I can find a
> detailed explanation of the Asterisk which including the
> Architecture, Install, Configure, usage example document.
The answers to the questions you've been asking are probably here:
Starter articles:
http://www.onlamp.com/pub/a/onlamp/2
> When I place a call with asterisk, asterisk will try to dial
> out on the first line even if the first line is already being
> used by someone else. Any ideas on what I'm doing
> wrong?
My question would be, how would asterisk know the line is in use if it
isn't controlling it?
> L( ) option is
> applicable here? And, if your version of Asterisk doesn't have a Dial app
> with the L( ) option,
> will it be worth your while to upgrade to have the L( ) option?
A third question might be in what version was this introduced.
___
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> At a couple of locations,
> the IAXy's have just stopped working after 1 or 2 days use. No lights go on,
> no DHCP lease is renewed as far as we can tell, and of course no dialtone
> and no registration with the server.
I had that problem intermittently while using a 1A power supply. The
behavio
> I would also suggest that while it is possible to do something, it is
> not always wise :) See the significant volumes of reports in the
> archives regarding multiple zaptel cards in one system.
I must be lucky: I have 2 X100P and a TDM400 with zero IRQ or other
issues. And double NAT for the vo
> about buggy firmware I want to know what I'm getting into before upping past
> my current 1.0.5.11.It's relatively stable, and the last thing I want to
> do is update to a flaky firmware
I don't know why, and Grandstream *swears* I'm the only one having
this problem, but I am "stuck" wi
> I've had ordered some items (VoIP starter kit, TDM400P, a couple of ATAs
> and an IAXy) from Digitnetworks to Reunion Island (which *technically*
> is in the EU despite its remote location) and with UPS shipping it came
> fast (about 1 week - but expensive though). No major troubles except
We've
You have already had several good answers but none included these
intro articles by John Todd which IMO are an excellent intro to the
whole thing:
Starting from zero:
http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html
The second article adds a bit of hardware to interface with PSTN
htt
> Is there a mistake in my asterisk config or is the French number only
> accessable from France?
No that happens when your server is not correctly logged in or
registered. I had that for a couple of days.
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> The problem is that I am a little confused as to what I all I need. I have
> a lot of Linux/Windows development and networking experience but almost
> zero VoIP experience.
http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html
Will give you a good background in voIP and asterisk capabilit
> /hax0r n00b mode on
> Which command and parameters do I need to use to get some legible (usable)
> output to do the packet sniffing? I tried ethereal but it only gives me
> loads of garbage?
> /hax0r n00b mode off :)
Go to the Wengo forum, there is a thread in the technique section that
gives t
> Any one have any solution for this?
> We need to have the caller id information announced when the phone is
> answered.
> for example
> I am sitting at my desk, my phone rings.
> I pick it up and hear call from 55 to except press 1 to decline
The Grandstream BT100 series phones will
> -- Call accepted by 66.225.202.72 (format gsm)
> -- Format for call is gsm
> -- Hungup 'IAX2/NuFone/1'
>== No one is available to answer at this time
Is the callerid a number like 7073131 ?
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> so you mean that the even the same network, the firewall will affect the
> channeling of data accross the asterisk?
Not likely. Look for the transmit silence setting on X-Lite which is
another gotcha that causes no audio and or try other SIP (and why not
IAX) clients.
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> Can some one advise me an PC to Phone client software to be used under
> Windows OS at the client side, to be communicated with Asterisk PBX?
Phones hard and soft
http://www.voip-info.org/wiki-VOIP+Phones
A few clients I've tried
for SIP X-Lite, SJPhone
for IAX2 IAXPhone, Firefly, IAXComm, D
> i cannot
> hear any voice from client's end
This is a common problem if you are using NAT (behind router)
Google for "asterisk one way audio"
and take a loog here
http://www.voip-info.org/wiki-Asterisk+SIP+NAT
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> I am starting to know about Asterisk and trying to use it and knowing its
> functionalities, I hope that here I can find answers for my following
> questions:
Welcome to the community! Most of the answers are here
http://voip-info.org/tiki-index.php?page=Asterisk
http://asteriskdocs.org
_
> Hahawell the MWI is the blinking blue LCD. The message button
> is "reserved for future use" Hang in there. There will soon to be some
No the message button call the number you configure in the web
interface. Presumably voicemail, but could be your mistress.
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