Dan,
Hence forth you will be called Dan the man!
Peter
At 23:20 3/11/03 +0200, you wrote:
Hi Dave,
- Original Message -
From: "David J Carter" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, November 03, 2003 10:28 PM
Subject: RE: [Asterisk-Users] DIAX Soft phone v0.9.1 is ava
I don't use it... Its an option for * to * communications. If I can get
the info on how to turn the 8kbps speex stuff on we might just see about
getting Mark to default speeks to 8k instead of what it uses now.
bkw
On Mon, 3 Nov 2003, Andrew Gillham wrote:
> Brian West wrote:
>
> >>Asterisk doe
On Mon, 2003-11-03 at 16:27, Alastair Maw wrote:
> On 03/11/03 20:03, Steven Critchfield wrote:
>
> > Sounds like you really need a C programmer and get into the guts
> > of asterisk. Can't get more flexible than having the source code
> > yourself to do anything you want. You could add your DSP r
(Email reordered from the braindead top-posting. http://learn.to/quote)
On Monday 03 November 2003 22:34, Peter Brown wrote:
> At 07:42 3/11/03 -0600, you wrote:
> >On Mon, 2003-11-03 at 07:07, Roy Sigurd Karlsbakk wrote:
> > > hi all
> > >
> > > anyone that've heard of any working IAX hardphones
On Mon, 3 Nov 2003 19:33:36 -0800 (PST), Chris Albertson wrote
>
> Did your patch make it to CVS? Sorry for being lazy and not looking.
> >From the sounds of thing maybe only half the patch made it
> But I'm not at the right machine to look at present.
No, but it may have something to do with th
Brian West wrote:
Asterisk doesn't seem to support SPEEX all that well. Has anyone had any
luck getting it to work with X-lite?
Speex works perfect with IAX but not that crack headed x-lite stuff.
bkw
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Markster,
Do you know where the product referred to by Steven is now?
Can it be purchased?
If so from whom?
Peter
At 07:42 3/11/03 -0600, you wrote:
On Mon, 2003-11-03 at 07:07, Roy Sigurd Karlsbakk wrote:
> hi all
>
> anyone that've heard of any working IAX hardphones yet?
During Phreaknic, M
> Asterisk doesn't seem to support SPEEX all that well. Has anyone had any
> luck getting it to work with X-lite?
Speex works perfect with IAX but not that crack headed x-lite stuff.
bkw
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> I'm going to use Cisco 7960's for the phones; is there a better phone I
> should be using?
excellent choice.
> I need to know if this is possible...
>
> On each phone program the appearances of 4 "Extensions" that are really
> the 4 phone lines?
yes for inbound. 1 rings 1... 2 rings 2.. and s
TDM400P
-gcc
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Joakimsen
Posted At: Monday, November 03, 2003 10:14 PM
Posted To: Asterisk User Group
Conversation: [Asterisk-Users] ADSI - PowerTouch 350
Subject: RE: [Asterisk-Users] ADSI - PowerTouch
I have a friend I am building a phone system for, I would
like to use *
He owns a “Take Out” restaurant with 4 Phone
lines and 4 Phones.
I’m going to use Cisco 7960’s for the phones; is
there a better phone I should be using?
I need to know if this is possible…
On each phone pr
> How much arrogance...
> I'm very sad that they are such people on this otherwise great distribution
> list.
> Maybe you are a big *nix expert , but this does not give you the right to
> talk like that with the other ones.
Ease up Dan, the comments are simple signs of immaturity, arrogance,
bias
--- William Waites <[EMAIL PROTECTED]> wrote:
>
> That only true for some NAT implementations and configurations.
> It is not robust in general. It would require double the ipfilter
> configuration and double the traffic on my NetBSD gateway, for
> example. The patch I submitted last week addr
I have 1.0.3.81. How do you execute the transfer?
Thanks,
Steven
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Brown (CV)
Sent: Monday, November 03, 2003 8:58 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Transfer from Grandstream BT100?
On Tue, Nov 04, 2003 at 02:49:39AM +0100, Christopher Arnold wrote:
> On Tue, 4 Nov 2003, Shoval Tom wrote:
>
> > Olle, www.voip-info.org still resolve to 192.168.168.3 from here, and many
> > other places (like our branch office, my home dial-up account, my parents
> > dial-up account)
> >
> Do y
I'll have a play with my 390 a bit later on and see if I can reproduce the
problems I had previously. Meanwhile, try calling 604 310 8350 and see if
you can download the Telus scripts to your phone.
I'm pretty sure the problem I had was during interactive ADSI sessions in
the voicemail app, leadin
What interface is the phone connected to?
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of PBX
> Sent: Monday, November 03, 2003 8:01 PM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] ADSI - PowerTouch 350
>
> I was wondering if
> In Bug ID 104, a patch was suggested that takes the netmask into
> effect and makes the right decision for phones on either side of the
> NAT.
> However, the code that was added for "externip" in the current CVS
> isn't
> that patch; it's just a way of giving me a choice of having SIP
> p
what version of GS firmware are you running ?
I call from PSTN to GS, GS does xfer to XTEN, hang up GS
call continues
if you aren't running 1.0.3.81 or newer, then upgrade :)
john brown
chagres
On Mon, Nov 03, 2003 at 06:15:02PM -0600, Steven Sokol wrote:
> Hi,
>
> Does anybody know how to
Hi,
I am currently working on an application that communicates with Asterisk
via the Manager interface. Two of the features that I have been working
on involve transfering a call and initiating a 3-way call.
I have been able to get the blind transfer working in most situations
with the manager c
Steve Underwood wrote:
Hi Thomas,
Unless you have a *very* specific need to use G.723.1 for
compatibility with someone else, forget it. It is pretty much an
obsolete product. Licencing is also a pain, as there is not patent
pool for it. G.729 is expensive to licence, but at least it is
relati
Gavin Hamill wrote:
On Mon, 2003-11-03 at 15:14, Eric Wieling wrote:
Licensing info for the G723.1 codec, direct from the holding company
that licenses the codec.
http://www.dspg.com/technology/LicensePricing.html
From what I remember when I looked into this about a year ago, this
isn't e
--- John Todd <[EMAIL PROTECTED]> wrote:
> >I'm replying to my own post because it is not clear.
> >
> >1) STUN can find all the firewalls between Aterisk
> >and whatever else. It can also find what your public
> >IP address it.
>
> Correct.
>
> >2) THere is an easy to use opensource STUN libra
Is there going to be a fix any time soon?
-gcc
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wade J.
Weppler
Posted At: Monday, November 03, 2003 3:31 PM
Posted To: Asterisk User Group
Conversation: [Asterisk-Users] MWI - I know this has been discussed
On Tue, Nov 04, 2003 at 02:43:36AM +0300, Shoval Tomer wrote:
> I see many posts about using ISDN cards and modem cards.
In my list lurking, I have gathered:
ISDN comes in two flavours, Euro and US, and two sizes BRI(2
lines)/PRI (24-30 lines) hardware generally only supports one size and
flavour
You have me convinced. It's a forwarding issue not a threeway calling issue.
Also, if the outgoing lines are configured for kewlstart, as long as the
called parties hang up at the same time, the conference will be torn down.
Steve.
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On Tue, 4 Nov 2003 13:00:44 +1100, Anthony Wood wrote
>
> Internals can use the IP address of the NAT box as the Asterisk
> Server IP and then it should work.
>
> i.e. don't set your internal SIP UAs to connect to the internal IP
> address of the Asterisk Server.
>
> The fix allows asterisk to w
At 11/3/03 2:41 PM, Martin Pycko <[EMAIL PROTECTED]> wrote:
>It's not for phones, it's for asterisk behind a NAT.
My apologies; I'm not making my question clear.
I realize this option is for Asterisk behind a NAT, but of course
Asterisk uses this parameter to talk to SIP clients (which I referr
None of these accounts use the same ISP or the same DNS.
Please post the ip address of a DNS server that does get the correct
address.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Christopher
Arnold
Sent: Tuesday, November 04, 2003 4:50 AM
To: [EMAIL P
Martin Pycko wrote:
It's new. It prevents asterisk from putting the private IP in the messages
that asterisk sends with SIP.
Hmmm. According to the sip.conf example:
[general]
externip = 200.201.202.203 :Address that we're going to put in SIP
messages if we're behind a NAT
Does this apply o
On Mon, Nov 03, 2003 at 09:46:40PM -0400, William Waites wrote:
> On Mon, 3 Nov 2003 17:10:10 -0600 (CST), Martin Pycko wrote
> > It doesn't care about the phones. If you phones are behind nat use nat=yes
> > for each defined account.
>
> The fix is incorrect. Asterisk chan_sip.c must distinguish
> > I don't think that is what keeping the original poster's system from
> > working. The issue is "one" extension is configured for canreinvite=no
> > and the other is canreinvite=yes. One extension believes all RTP must
> > be passed through * while the other is attempting to negotiate a
> > phon
I'm replying to my own post because it is not clear.
1) STUN can find all the firewalls between Aterisk
and whatever else. It can also find what your public
IP address it.
Correct.
2) THere is an easy to use opensource STUN library
available but I'm not yet sure how acivly developed it is
curren
i am using t/e410 and my config is just like yours, works great.
azher Eduardo Goncalves <[EMAIL PROTECTED]> wrote:
On Mon, 3 Nov 2003 17:15:21 -0600Don Pobanz <[EMAIL PROTECTED]>wrote:> You also need to verify that you are using loop timing and not> internal timing. (Your telco will provide timi
On Tue, 4 Nov 2003, Shoval Tom wrote:
> Olle, www.voip-info.org still resolve to 192.168.168.3 from here, and many
> other places (like our branch office, my home dial-up account, my parents
> dial-up account)
>
Do you by any chance use the same ISP?
Or migth it be so that all accounts use the s
On Mon, 3 Nov 2003 17:10:10 -0600 (CST), Martin Pycko wrote
> It doesn't care about the phones. If you phones are behind nat use nat=yes
> for each defined account.
The fix is incorrect. Asterisk chan_sip.c must distinguish between
SIP peers that are behind the firewall (together with the *) and t
I was wondering if anyone has had any experience with PowerTouch 350 &
ADSI.
I am able to configure the phone with ADSI.. Which is pretty cool in it
self. But I have a question regarding the Service feature. I have to
choose the service button and choose the Asterisk PBX ADSI. But let's
say I m
Gary,
What does your clocking setup look like? In a general telco sense I
would expect it should look something like this:
LEC PRI I/Fto *: Loop
* PRI I/F to Modem Bank: Internal
Modem bank PRI I/F to *: Loop
Got a smart CSU (or two) you can put inline on the PRI lines and take a
look a
I
see many posts about using ISDN cards and modem cards.
Does
this mean I can use a regular modem (that supports voice) as an FXS, or FXO ?
If
so, can you provide an example.
It
might be the easiest way to test Asterisk's collaboration with the PSTN (or
analog phones) and provid
(B
(B
(BI've been using asterisk-0.5.0.
(B
(BI've been reading about the externip param (it
(Blooks like it is only available in the lastest releases).
(B
(BCould someone tell me the version number (or tag)
(Bto check out of CVS so I can get this functionality?
(B
(B(And, if its no
Steven Critchfield wrote:
- I need to be able to generate large amounts of audio in realtime,
conference people together but then only play an audio file to one
person within the conference, etc. I don't think AGI is flexible
enough to do this.
Meetme is doing the conferencing re
Hi,
Does anybody know how to properly execute a transfer (without using the
|Tt option) from a GS100? Scenario:
1. I call from X-PRO (ext 1100) to Grandstream (1101).
2. Grandstream answers. Call is established.
3. Press [TRANSFER] on the Grandstream. X-PRO caller is put on hold.
Grandstrea
Olle, www.voip-info.org still resolve to 192.168.168.3 from here, and many
other places (like our branch office, my home dial-up account, my parents
dial-up account)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Olle E.
Johansson
Sent: Monday, November
I just pulled down the newest CVS and recompiled.
FWD (free world dialup) works now from *, and I AM behind a NAT. I've nearly
given up on the xten lite, iaxcomm sounds better. I'll be trying the other win
app thats up-and-coming on the list later.
It seems to have broken iptel, but that's no
On Mon, 3 Nov 2003 17:15:21 -0600
Don Pobanz <[EMAIL PROTECTED]> wrote:
> You also need to verify that you are using loop timing and not
> internal timing. (Your telco will provide timing)
>
> in zaptel.conf you should have something like
> span=1,1,0,ccs,hdb3,crc4
> where the second 1 says to use
On Sun, 2 Nov 2003 16:21:48 +0300, Shoval Tom wrote
> Either it's not working, or I don't know what I'm doing.
> It's giving me the error "sox: effect '.gsm' is no known!
>
> Let's say I need to convert file 1.wav to 1.gsm.
> How do I apply this command to it?
>
[snip]
> #!/bin/sh
> for i in *
> There is someone on this list who has the specification for the IPW driver?
> I want to provide full support for it in DIAX Phone, but Actiontec does not
> answer to my mails.
>
> Dan
Dan, you might want to also look into the
Internet Power Phone 2000 - Standard Telephone Adapter
http://www.e
Hi guys,
This is a two part question about the Voicemail application:
Firstly, is there a way built in to the current app which would allow me
to have Asterisk call a phone number everytime a certain mailbox
receives a new voicemail? I know about the email and pager
notifications that are alr
> > Sometimes I receive a Red Alarm in my E1 trunk (E&M immediate
> > start
> > signaling), and just few seconds after this, all alarms are
cleared.
> >
> > This problem ocurrs many times/day, and if are calls in progress,
> > these calls just hang-up.
> > Could it be an asterisk b
It doesn't care about the phones. If you phones are behind nat use nat=yes
for each defined account.
Martin
On Tue, 4 Nov 2003, Shoval Tom wrote:
> Will extern IP work if I had multiple phones connected behind NAT?
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECT
Paul Crick wrote:
It's more of an ADSI/Voicemail problem than phone specific I think? Or is it
only affecting the 480s? I know I had a problem a while back with having the
phone lock up and keypad become unresponsive, but with a newer version of
Asterisk the problem went away.
Since I'm only testi
Will extern IP work if I had multiple phones connected behind NAT?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Martin Pycko
Sent: Monday, November 03, 2003 8:35 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Asterisk behind LinkSys NAT Routing
Dan, any chance getting a look at the code?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Peer Oliver
schmidt
Sent: Monday, November 03, 2003 8:20 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] New IAX software phone (for WIndows platform)
Jerem
Hi Dan,
Nice goingsome testing feedback. Testing your new client with
voicemail - after entering password (4 digits), 1 for new messages, then no
further digits can be sent. So far everything else OK.
Regards,
Paul
- Original Message -
From: "Dan" <[EMAIL PROTECTED]>
To: <[EMAIL PROT
For some of these noise problems, it is good to do a jitter analysis. I
found it to be the cause of the problems I was having with my Cisco phone
and a X-Lite client. Also, I found that the X-Pro client is better at voice
delivery than the X-Lite client.
Anyway, if you have tcpdump available on
I'm replying to my own post because it is not clear.
1) STUN can find all the firewalls between Aterisk
and whatever else. It can also find what your public
IP address it.
2) THere is an easy to use opensource STUN library
available but I'm not yet sure how acivly developed it is
currently. It
I'd suggest your telco doing loopup and checking the circuit.
regards
Martin
On Mon, 3 Nov 2003, Eduardo Goncalves wrote:
> Hi list,
>
> Sometimes I receive a Red Alarm in my E1 trunk (E&M immediate start
> signaling), and just few seconds after this, all alarms are cleared.
> This p
On 03/11/03 20:03, Steven Critchfield wrote:
Sounds like you really need a C programmer and get into the guts
of asterisk. Can't get more flexible than having the source code
yourself to do anything you want. You could add your DSP routines into
the dsp.c file and call them when needed. You can al
--- Robert L Mathews <[EMAIL PROTECTED]> wrote:
> At 11/3/03 10:00 AM, Martin Pycko <[EMAIL PROTECTED]> wrote:
>
> >> Is "externip" and new parameter??
> >
> >It's new. It prevents asterisk from putting the private IP in the
> messages
> >that asterisk sends with SIP.
>
> Does it take an IP addr
Hi,
There is someone on this list who has the specification for the IPW driver?
I want to provide full support for it in DIAX Phone, but Actiontec does not
answer to my mails.
There is any windows driver available for the Digium's S100U USB interface?
Anyone knows if it can provide full functiona
It's not for phones, it's for asterisk behind a NAT.
Martin
On Mon, 3 Nov 2003, Robert L Mathews wrote:
> At 11/3/03 10:00 AM, Martin Pycko <[EMAIL PROTECTED]> wrote:
>
> >> Is "externip" and new parameter??
> >
> >It's new. It prevents asterisk from putting the private IP in the messages
> >tha
I have used G723.1 (although unlicensed) with Asterisk. The info is even
in the Makefile, just drop in a few files in your source directoy,
uncomment something in the Makefile and instant G723.1 support...
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROT
Hello all,
I am thinking of giving my employees incentives - I want them to speak
with my clients more (on the phone) - and in order to do so I need to
measure it! I have had a look through the master csv file which asterisk
produces - but I am not sure how to interpret it.
I am looking for
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
WipeOut wrote:
| Has the bad quality started just recently? Has it ever worked nicely for
| you??
It depends. :)
| If either of these is "yes"..
|
| What has changed in your setup? Have you recently upgraded to a newer
| CVS??
I have a fairly recent
Hi list,
Sometimes I receive a Red Alarm in my E1 trunk (E&M immediate start
signaling), and just few seconds after this, all alarms are cleared.
This problem ocurrs many times/day, and if are calls in progress,
these calls just hang-up.
Could it be an asterisk bug? Or may
It's more of an ADSI/Voicemail problem than phone specific I think? Or is it
only affecting the 480s? I know I had a problem a while back with having the
phone lock up and keypad become unresponsive, but with a newer version of
Asterisk the problem went away.
__
> I don't plan on using it. I will use mine, which is created in wxWindows
> and C++ and will run on Winsucks, UN*X and Mac.
>
> Yes, someday it will get released, maybe even the code if people are nice.
I think it's just a race to see who makes the first GPL'd or LGPL'd IAX
client for Windows (or
Hi,
As Freelance programmer/consultant I'm looking for project/task of IVR/CTI/CRM/IP-based, my skils are as following,
1 Dialogic-based CTI/IVR software programming 2 Intervoice IVR development 3 Siebel CRM integration and development 4 IBM DirectTalk and WebShpere Voice(Vo
At 11/3/03 10:00 AM, Martin Pycko <[EMAIL PROTECTED]> wrote:
>> Is "externip" and new parameter??
>
>It's new. It prevents asterisk from putting the private IP in the messages
>that asterisk sends with SIP.
Does it take an IP address, like "externip=1.2.3.4"? And does it then
force the SIP messa
Hi Dave,
- Original Message -
From: "David J Carter" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, November 03, 2003 10:28 PM
Subject: RE: [Asterisk-Users] DIAX Soft phone v0.9.1 is available for
downlaod...
> Hi Dan,
>
> Just downloaded 0.9.1. Works fine on test set up inte
Hi,
- Original Message -
From: "Florian Overkamp" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, November 03, 2003 10:18 PM
Subject: Re: [Asterisk-Users] DIAX Soft phone v0.9.1 is available for
downlaod...
> ...
> Cool, that would pretty much do it. Now, in earlier mails and
Title: Call waiting on X100P
I have Asterisk setup in a SOHO environment.
I have 2 X100P cards at Zap/1 and Zap/2.
I have 1 TDM400P card with Zap/3 - Zap/5.
I have subscribed to callwaiting, callerid and calleridcallwaiting from Qwest on the 2 PSTN lines - Zap/1 and Zap/2.
My problem
> - Should I implement IAX or IAX2? What's the main difference, other
> than IAX2 supporting trunking (which according to the docs needs a
> Zaptel timing source).
IAX2 without any question. You will not be required to run trunk mode in
your case, especially if you're just doing it loca
I don't want to rain on Dan's parade, but I'd like to call everyone's attention
to an existing project.
Steve Kann has developed a crossplatform IAX/IAX2 library, and there are a few
clients available for it.
I have written iaxComm using the wx toolkit. It compiles and runs under Windows
XP using
Are you guys using voicemail2 ?
Martin
On Mon, 3 Nov 2003, Philipp von Klitzing wrote:
> Hi!
>
> > The voicemails "servermail" and "fromstring" variables should change
> > default
> > values when email voicemail notification gets received by user.
> >
> > I change it, but received mail still sho
It's for setting asterisk box with SIP support behind a NAT.
You need to do port redirection of eg. 5060 and then setup
externip=ip_of_your_nat_gateway
Martin
On Mon, 3 Nov 2003, Andrew Thompson wrote:
> According to the source, it goes in the general section of sip.conf:
>
> } else if (!strcase
Download the new code and see in asterisk/configs/sip.conf.sample
It can't be easier than that.
Martin
On Mon, 3 Nov 2003, listas iPfone wrote:
> Hi!
>
> How to use that externip new parameter?
>
> Where in sip.conf and what is the format?
>
> thanks
>
>
> - Original Message -
> From: "
On Mon, 2003-11-03 at 13:49, Alastair Maw wrote:
> On 03/11/03 18:02, Alastair Maw wrote:
> >>> I'm implementing a Java-based IVR server (and yes, I know Asterisk does
> >>> IVR, and no, it's not flexible enough to do what I want and no, it
> >>> doesn't integrate well with the Java systems we have
Martin Pycko wrote:
It's new. It prevents asterisk from putting the private IP in the messages
that asterisk sends with SIP.
Hmmm. According to the sip.conf example:
[general]
externip = 200.201.202.203 :Address that we're going to put in SIP messages if we're
behind a NAT
Does this apply on
If you're using the TDM400P card, VMWI is currently broken...
> -Original Message-
> From: PBX [mailto:[EMAIL PROTECTED]
> Sent: Monday, November 03, 2003 3:17 PM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] MWI - I know this has been discussed in
depth
> already
>
> Let post this
Hi Dan,
Just downloaded 0.9.1. Works fine on test set up internally.
I get my WAN IP dynamically and have used DynDNS.org for updating a URL for
the home network. Could the registration look for this rather than a fixed
IP address?
Regards, and keep up the good work for us non techies to use.
D
Hi,
At 21:17 3-11-2003 +0200, you wrote:
- solved some bugs reported by the users;
- no more crashes when exit from the system tray;
- no need to restart application after a registration change;
- single form for the registration data. You can choose not to register in
the registration window (abl
Brian Schrock wrote:
Everyone,
I think I have a way of doing it
exten => _911,1,ChanIsAvail(Zap/1)
exten => _911,2,Dial,Zap/1/911
exten => _911,3,Hangup()
exten => _911,103,SoftHangup(Zap/1-1)
exten => _911,104,Wait(1)
exten => _911,105,Goto(1)
The only thing I would add is a variable to set
According to the source, it goes in the general section of sip.conf:
} else if (!strcasecmp(v->name, "externip")) {
if (!(hp = gethostbyname(v->value))) {
ast_log(LOG_WARNING, "Invalid address for externip keyword: %s\n", v->value);
} else {
memcpy(&__ourip, hp->h_addr, sizeof(__ourip));
use_exter
Don't get me wrong, I think its great too that he has spent the time on it,
and has something working.
I am just trying to be helpful sharing my experience with the platform. We
had what we thought was something working too, but when we started testing
on all the actual supported windows o/s va
Let post this question.. Because I must be real slow... The following is
my config on this...
group=1
context=default
signalling=fxs_ks
channel => 1
context=local
signalling=fxo_ks
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
echocancel=yes
echocancelwhenbridged=yes
r
On Mon, 2003-11-03 at 12:02, Alastair Maw wrote:
> On 03/11/03 16:35, Jeremy McNamara wrote:
>
> >> I'm implementing a Java-based IVR server (and yes, I know Asterisk does
> >> IVR, and no, it's not flexible enough to do what I want and no, it
> >> doesn't integrate well with the Java systems we h
Let post this question.. Because I must be real slow... The following is
my config on this...
group=1
context=default
signalling=fxs_ks
channel => 1
context=local
signalling=fxo_ks
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
echocancel=yes
echocancelwhenbridged=yes
r
> exten => _911,1,ChanIsAvail(Zap/1)
> exten => _911,2,Dial,Zap/1/911
> exten => _911,3,Hangup()
> exten => _911,103,SoftHangup(Zap/1-1)
> exten => _911,104,Wait(1)
> exten => _911,105,Goto(1)
That is pretty much exactly it, except I don't use _, and I play a file and
sleep(5) before what you're
On 03/11/03 18:02, Alastair Maw wrote:
I'm implementing a Java-based IVR server (and yes, I know Asterisk does
IVR, and no, it's not flexible enough to do what I want and no, it
doesn't integrate well with the Java systems we have, etc. hence my
doing this).
Are you mad? What is not flexable eno
Update...
Thank you for everyone that replied to this. I received a new 350 today
and plugged it in... Works great. Problem appeared to be the other
phone.
Geoff
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul Crick
Posted At: Thursday, October 30
i've tried to set dial = 0/1500 in the indications.conf but still
getting a dial tone.
is this a bug with *? or did i do something wrong.
thank you,
patrick
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/aste
the above-message keep popping up every second during a conversation
between a
zap(fxs) channel and sip channel.
* eventually hung after a long while
we can talk to each other and we can ring one another without any problem.
i've had x-lite and x-pro register with * without this problem.
furth
On Monday 03 November 2003 12:13, Steven Sokol wrote:
> Hi. I am looking for the proper syntax for the Cut application. I
> am working on a "Feature Code" extension that drops a caller
> directly into a voicemail box. Here is what I have:
>
> exten => _55.,1,Answer()
> exten => _55.,2,Cut(VMEXT=
as promise, at:
http://www.laser.com/dante
or
http://www.geocities.com/tdanro
To be short...
Version 0.9.1 is available.
What's new:
- solved some bugs reported by the users;
- no more crashes when exit from the system tray;
- no need to restart application after a registration change;
- s
> I was willing to give vb a chance at one time, but won't touch it any
> more.
I am not defending VB. I won't touch it either (I use XWT for all my
cross-platform user interface type stuff, any web monkey can be proficient
in it in a very short time) -- I am defending Dan's work in actually
p
Rich and I have updated the Wiki page "Asterisk rollout tips" with advice on how to
plan
and implement your Asterisk rollout. This page is based on many discussions on the
mailing list, so don't be surprised if your comment or thought is included in the
text. Thank you for your input!
http://www.v
How much arrogance...
I'm very sad that they are such people on this otherwise great distribution
list.
Maybe you are a big *nix expert , but this does not give you the right to
talk like that with the other ones.
I have never seen Mark or any other really big *nix specialist to talk like
that with
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