Joe,
This is highly implementation specific. Perhaps I can give you some
pointers to help you out. BTW, if you just happen to be in Texas, I can
provide you with a list.
Regular 911 calls are answered by a PSAP. Voip calls also goto a PSAP,
but are handled differently. In fact, in most regions
Hi All
I have a strange problem using IAX2. When placing a call to my IAX clients
(firefly) via the Asterisk dialplan all works great. However trying to
initiate a call via the manager interface to the IAX client using the
following command results in an error:
Action: Originate
Channel:
check under your network settings that you have all the codecs selected
and obviously type IAX
Jason Penton wrote:
Hi All
I have a strange problem using IAX2. When placing a call to my IAX clients
(firefly) via the Asterisk dialplan all works great. However trying to
initiate a call via the
hi,
Am Do, den 17.06.2004 schrieb oi geli um 1:13:
I am trying to install the Modified Prepaid App. I
have installed PostgeSQL, created the tables, etc.
Make Install runs ok. The when I try to launch
asterisk (asterisk -vgc), it fails to run. I get
the following errors,
1st error:
I can make call in
to the asterisk server listen to voice mail, and do the echo test. When make a
call I get no audio inbound or outbound. When making incoming call I can leave a
valid voice message, but when then extentions pick up again no audio inbound or
outbound.I am using Xten liteand
Hi Adam
Done all that but still the same problem.
Do you have any other ideas?
Cheers
Jason
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adam Hart
Sent: 17 June 2004 08:29 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] IAX2 no
iax2 debug is your friend, looks at the capibilities asterisk is sending
in it's NEW message
Jason Penton wrote:
Hi Adam
Done all that but still the same problem.
Do you have any other ideas?
Cheers
Jason
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
hi all,
i am trying to get my TE410P (see previous posts) working in Austria
(telekom Austria - i am still waiting for an answer for my questions).
my /etc/zaptel.conf looks like
span=1,1,0,ccs,hdb3,crc4,yellow
span=2,2,0,ccs,hdb3,crc4,yellow
bchan=1-15,17-31
John Bittner wrote:
Hi,
I have it working great. I have debian running on it with music on
hold disabled. I setup 10 cisco 7960 phones and tested the 4801 with
calls on all 10 phones at the same time through voicepulse with no
issues. I ran top with all the phones running and I was only up
Hi,
Is there a way to accept SIP calls from
unregistered gateways?
autocreatpeer=yes seems to disable checking
credentials but the originating gateway is still required to register itself
with a username and password (which can be anything since it won't check
it).
I like to be able to
On Thu, 17 Jun 2004, Wolfgang Pichler wrote:
... on the card i can see the two leds pulsing red (i think thats the
yellow alaram - or i am wrong) ?
Are you sure it is not a red alarm? That would indicate a loss of link.
I think you can check with the command zttool.
Are you sure the cables
I've got Zaphfc working running Asterisk v. 0.7.2
Then I have tried with Asterisk V. 1.0 and the latest from CVS - with
no succes. Has anybody got zaphfc working with newer version than 0.7.2
zaphfc is in bri-stuff from www.junghanns.net --- or in a patched version
at
Is there a way to register with or call the firefly network from an Asterisk
server. It would be pretty cool if you could gateway calls onto it.
Have a nice day,
--
Martijn van Oosterhout
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Hi Adam
Thanks - Here are the two attempts:
This is the first one where * dials firefly via the dialplan (which works
fine):
Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW
Timestamp: 1ms SCall: 4 DCall: 0 [146.231.125.65:4569]
VERSION : 2
Am Do, den 17.06.2004 schrieb Peter Svensson um 9:43:
On Thu, 17 Jun 2004, Wolfgang Pichler wrote:
... on the card i can see the two leds pulsing red (i think thats the
yellow alaram - or i am wrong) ?
Are you sure it is not a red alarm? That would indicate a loss of link.
I think you
Hi,
Actually, you the Geode CPU mentioned below is a 5x86 (486 platform) at
233 MHz. If you take Pebble (http://www.nycwireless.net/pebble/), which
is a downstripped Debian ( 64 MB) on a readonly ext2 filesystem, you
should be grand. Installing asterisk + some extra stuff will probably
What is in your config file ?
zaptel.conf ?
also, check the crc4 settings
and
maybe the wire you are using is wrong since some equippment needs
crossed wires, some needs straight wires. Crossed would be 1-4 2-5
cheers
Michael
On Thu, 2004-06-17 at 10:28, Wolfgang Pichler wrote:
Am Do, den
On 16/06/2004 at 22:53 Jay Milk wrote:
You're correct -- I believe I pointed out in my original post that there
is a $200+ difference between a cordless Cisco with/without software.
And that's plain ridiculous. Plus, the phone alone isn't worth $500 in
hardware -- so we're obviously dealing
On Thu, 17 Jun 2004, Wolfgang Pichler wrote:
Are you sure the cables are correct?
Have you set the jumpers on the card to E1 and not left them on T1?
The jumpers are on E1 - the cables should be ok (they are working with
other hardware) - and the card is directly connected to a simens
Hello,
I understand there's no possibility to have asterisk configuration
(sipusers, extensions, voicemail) in LDAP right now. I'm thinking
about put the (sipusers, extensions, voicemail) info in LDAP and then run
some synchronization script on the asterisk server which will build up
appropriate
Am Do, den 17.06.2004 schrieb Michael Bielicki um 10:32:
What is in your config file ?
i've already posted my config in my first post - but here is my
/etc/zaptel.conf
span=1,1,0,ccs,hdb3,crc4,yellow
span=2,2,0,ccs,hdb3,crc4,yellow
bchan=1-15,17-31
dchan=16
Martijn van Oosterhout wrote:
Is there a way to register with or call the firefly network from an Asterisk
server. It would be pretty cool if you could gateway calls onto it.
Have a nice day,
You can register and dial out with * like on other IAX services. You
can verify it by changing the
I think there is a odbc driver for ldap. at least I remember that I saw
one a while ago. You could combine that with ast_data and off you fly
Just my 2 cents (EUR)
Michael
On Thu, 2004-06-17 at 10:41, David Hajek wrote:
Hello,
I understand there's no possibility to have asterisk
Throw out the yellow.
Check if for sure the other side is using crc4.
On Thu, 2004-06-17 at 10:45, Wolfgang Pichler wrote:
Am Do, den 17.06.2004 schrieb Michael Bielicki um 10:32:
What is in your config file ?
i've already posted my config in my first post - but here is my
/etc/zaptel.conf
Am Do, den 17.06.2004 schrieb Peter Svensson um 10:38:
On Thu, 17 Jun 2004, Wolfgang Pichler wrote:
Are you sure the cables are correct?
Have you set the jumpers on the card to E1 and not left them on T1?
The jumpers are on E1 - the cables should be ok (they are working with
other
HI ALL;
Is asterisk voicemail service can be run under H323
or it just run under SIP.
mohammad
Can I just pay you to fix it for me.
I cant see anywhere where I use the debug
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael
Manousos
Sent: Wednesday, 16 June 2004 11:55 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] oh323
Have you
On Thu, 17 Jun 2004, Wolfgang Pichler wrote:
but the same cable works great with an other hardware (a Parlay i60)
You could try a loopback plug to make sure your TE410P does not have a
damaged receiver.
Peter
--
Peter Svensson ! Pgp key available by finger, fingerprint:
[EMAIL PROTECTED]
At 16:49 16/06/2004 -0400, Eric wrote:
I upgraded my two asterisk boxes today to the latest cvs (up from 5/3/04).
These two boxes talk to eachother via sip, not iax. Since the upgrade, I
get the error Failed to authenticate on INVITE trying to make calls to/from
either box. Removing the secret
Hi all,
I have a box running asterisk with T410 connected to a Nortel DMS 100 switch
and another box running SER with grandstream phones on it
So if there is a call from the pstn it goes from the Nortel to the asterisk
and then to the SER box and finally to the phones.if the phone is busy or
the
Did you compile the channel driver with the sources of the running
Asterisk? This is happening because of a mismatch between the include
Asterisk files used to compile asterisk-oh323 and the running Asterisk.
Make sure that you have removed any previous version of Asterisk
(including header files
On Tue, 2004-06-15 at 17:44, Mark Elkins wrote:
On Tue, 2004-06-15 at 11:43, Shaun Ewing wrote:
This is an issues with DTMF clamping, you need to use
chan_capi to get DTMF
working correctly.
That's the last thing I wanted to hear :-(
The jist of this is that i4l does not allow
The disconnect issue also still exists (for me) with 4.55 firmware. I
can use the uniden to call another local sip phone (with
canreinvite=no), and leave both phones off the hook for as long as I
like. However, if I use the Uniden to call an PSTN number (via tdm400p
fxo), then the uniden will
What is the easiest way to guarantee everything is gone
rm -f /usr/lib/asterisk
is there anything else
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael
Manousos
Sent: Thursday, 17 June 2004 7:32 PM
To: [EMAIL PROTECTED]
Subject: Re:
Hi!
I will use it as simple ivr ...get the call from fxo gateway port ..give
some options and rings the recepcionist phone.
I have a x100p here and the thin client have a pci slot...maybe i can use
it...maybe...not...i will test
The main reason is to free a p4 2.0 ..that is runing * now... i
David Hajek
Sent: Thursday, June 17, 2004 2:41 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] LDAP synchronization script
Hello,
I understand there's no possibility to have asterisk configuration
(sipusers, extensions, voicemail) in LDAP right now. I'm thinking
about put the
And rm -rf /usr/include/asterisk
Michael M. Saunders wrote:
What is the easiest way to guarantee everything is gone
rm -f /usr/lib/asterisk
is there anything else
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael
Manousos
Sent: Thursday, 17 June 2004
Hi,
has anyone succesfully installed such scenario ?
I'm having problem with Award bios mb pc's... it do works with others,
what's your idea ?
Tnx !
--
Best regards,
Alessio mailto:[EMAIL PROTECTED]
___
Asterisk-Users
This is a problem I pointed out to Digium a while back, but I am not sure Markster
understood the issue and I didn't really have the time to follow it up. It does need
fixing though, as it is a major drawback in the current architecture.
Rgds
Tim
-Original Message-
From: [EMAIL
Please can you explain in more details as to what your problem is? I have 2 cards
working in one PC, but have had problems with Dell motherboards.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alessio Focardi
Sent: 17 June 2004 11:41
To: [EMAIL
google for it :)
http://lists.digium.com/pipermail/asterisk-dev/2003-November/002299.html
On Jun 17, 2004, at 1:05 AM, listas iPfone wrote:
Hi All,
I have a thin cliente here that i want to run asterisk:
- National Semicondudor Geode GX1 266MHz Geode 266MHz single chip
- NS Cx5530a
On Thu, 2004-06-17 at 04:40, Mark Elkins wrote:
On Tue, 2004-06-15 at 17:44, Mark Elkins wrote:
On Tue, 2004-06-15 at 11:43, Shaun Ewing wrote:
This is an issues with DTMF clamping, you need to use
chan_capi to get DTMF
working correctly.
That's the last thing I wanted to
I'm planning a system, just need to know if it works with Telstra's
network.
Cheers,
Clint
Sydney, Australia
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options
Send traces.
- Original Message -
From: Aimable [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, June 17, 2004 6:28 AM
Subject: [Asterisk-Users] Problems with PRI with T410 messages
Hi all,
I have a box running asterisk with T410 connected to a Nortel DMS 100
switch
and
These are the three cheap SIP phones that I've used.
Grandstream BT10x $65/street
Number only LCD
Zultys ZIP 2 $100/retail
No LCD
Uniden UIP 200 $120/retail
PoE, built-in switch
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
In a related story, the IRS has recently
Hello Robinson,
Thursday, June 17, 2004, 12:42:21 PM, you wrote:
RTW Please can you explain in more details as to what your
RTW problem is? I have 2 cards working in one PC, but have had
RTW problems with Dell motherboards.
voice is out of sync, it syncs for some second if I run something over
I'm planning to incorporate this (native and dynamic) LDAP for my own
system on short term. Do you have any LDAP design in mind?
Stefan
On Thu, 17 Jun 2004, Jeremy Jones wrote:
David Hajek
Sent: Thursday, June 17, 2004 2:41 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] LDAP
This is a problem I pointed out to Digium a while back, but I am not sure
Markster understood the issue and I didn't really have the time to follow it
up. It does need fixing though, as it is a major drawback in the current
architecture.
Rgds
Tim
Hi all,
I have a box running asterisk with
Hi Alessio
Yes, the problems you report do seem similar to the issues I had. I found on the
Dells that the audio prompts were very choppy and played slower than normal.
Occasionally there would be 'bursts' oav a second or so of 'good' audio.
I also suspected IRQ issues but the Dell Mobos had
I think I'll use something from this article -
http://www.marko.net/asterisk/archives/0205/0006.html
-David
-Original Message-
From: Stefan de Konink [mailto:[EMAIL PROTECTED]
Sent: Thursday, June 17, 2004 1:12 PM
To: David Hajek
Cc: [EMAIL PROTECTED]
Subject: RE:
Hi,
How to configure our ZAPATA.CONF in case that the PSTN in Bulgaria
is based on Siemens equipment?
Now my configuration is:
[channels]
language=en
busydetect=no
when is yes I have problems with answering of FXO when FXS line is
open
callprogress=no
when is yes
Hi
That rescue disk sugestion seems to be very good...
Let´s see if i undestood:
1. burn the rescue iso
1. copy the rescue disk to a hard drive
2. compile asterisk
3. copy all to the flash disk
It is that simple?
Miklos
- Original Message -
From: Klaus-Peter Junghanns [EMAIL
Hello Robinson,
Thursday, June 17, 2004, 1:19:12 PM, you wrote:
RTW Hi Alessio
RTW Yes, the problems you report do seem similar to the issues
RTW I had. I found on the Dells that the audio prompts were very
RTW choppy and played slower than normal. Occasionally there would
RTW be 'bursts' oav
Im having problems with a new install of Asterisk (I
had to reinstall because hard drive failed). Ive used debian net install
this time and for some reason WS FTP will not connect using SFTP (it keeps
coming back with username and password fail) but when I use Putty to connect
with the
On Thu, 17 Jun 2004, listas iPfone wrote:
1. burn the rescue iso
mount -o loop -t iso9660 /file /mnt/loop
1. copy the rescue disk to a hard drive
cp -dpR /mnt/loop/* /new/location
2. compile asterisk
make PREFIX=/new/location install (check if asterisk don't copy all
development non-sence)
Yeah that 'old' message discribes VERY MUCH what I'm doing at the moment.
Though there should be an 'application' part and an universal 'user' part.
For example the meetme is application specific, should be in the Asterisk
tree. But the extentions should basically be templates part of the
Now what is the normal behavior and how can I set it so that * behaves
normally?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
Sent: Thursday, June 17, 2004 2:06 PM
To: [EMAIL PROTECTED]
Subject: Asterisk-Users digest, Vol 1 #4186 - 11 msgs
Send Asterisk-Users
First of all, this is all too vague information you guys are providing
here.
When you state problems like this, you should be more specific.
A) What card are you using (there are lots of HFC-S cards out there).
B) What distribution, asterisk-version (stable, HEAD, what date if HEAD)
are you
Your
WSFTP program may only have SSH1 but your Debian server may only have
SSH2.
Look
in /etc/ssh/sshd_config
Make sure you have
'Protocol 1'
I do not recommend this setting as it is not
secure. I use F-Secure SSH Client w/Debian and like
it.
TL
P.S.
Please take this question to a
has anyone succesfully installed such scenario ?
Yes, see just my e-mail from today. It's in the mailing list archive, see
http://lists.digium.com/pipermail/asterisk-users
BTW: it's always good to check mailing list archives :-)
I'm having problem with Award bios mb pc's... it do works with
Dean,
This really has nothing to do with Asterisk. I suspect
you'll get better response by posting to a Linux oriented list. Check your
distribution vendor's website, as I'm sure they will have
links.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dean
I do not believe you are correct. We see CALL PROCEEDING in both directions as part of
the normal ISDN call setup process. See trace below.
Asterisk sends 'CALL PROCEEDING' followed immediately by 'ALERTING'. CALL PROCEEDING
is normally an acknowledgement to a SETUP. See Q931 below:
3.1.2
You could use winscp3 (it comes from the putty family). It has support
for scp and sftp.
On Thu, 2004-06-17 at 08:11, Todd Lieberman wrote:
Your WSFTP program may only have SSH1 but your Debian server may only
have SSH2.
Look in /etc/ssh/sshd_config
Make sure you have
'Protocol
HS a) an IRQ problems, see cat /proc/interrups
HS b) a mainboard problem (because usually you've to change the mainboard to
HS change the BIOS)
HS In case of a), try disabling built-in peripherals of the board, e.g. the
HS second serial port, usb host etc. That should make IRQs free. You can
HS
Filezilla
SFTP,
FTP, SSL-FTP
Works
on every linux distro I've tried as well as cygwin and several other encrypted
file transfer servers both Win32 and Unix-based
http://filezilla.sf.net/
MATT---
-Original Message-From: Todd Lieberman
[mailto:[EMAIL PROTECTED]Sent:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris Lee
Sent: Tuesday, June 15, 2004 6:34 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Asterisk-Users List Etiquette
Kevin Walsh wrote:
Steven Critchfield [EMAIL PROTECTED] wrote:
A: Because we read the question in the previous message.
Q: Why should I post my reply above the quoted text?
--
Troy Settle
Pulaski Networks
http://www.psknet.com
866.477.5638
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Hermann
Matt, thanks for your suggestion another
kind soul just suggested it about 10 minutes ago and it is already working like
a charm, as for those that dont think this is an asterisk problem phooey
;)
Night all.
Dean
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
On Thu, 2004-06-17 at 09:21 -0400, Troy Settle wrote:
..snip..
However, my preference is for top posting. The reason, is that in order to
read my message here, you had to scroll through ~70 lines of previous
discussion. Stuff that you've /already/ read since you've been following
this
Hi Jason,
Thanks for your reply. I didn't really want to use the insecure option,
that defeats the purpose of using a password :)
I was, however, able to specify user= in my sip.conf entity and that
solved the problem I was having.
Thanks again.
- Eric
On Thu, 17 Jun 2004 10:17:54 +0100
On Thursday 17 June 2004 09:21, Troy Settle wrote:
However, my preference is for top posting. The reason, is that in order to
read my message here, you had to scroll through ~70 lines of previous
discussion. Stuff that you've /already/ read since you've been following
this thread.
That's
Hello, ereyone!
I have just installed Asterix on my FreeBSD (-current) box
I'm planning to use it as H323 PBX for softphones
Currently I'm stuck in transfering a call to another machine
running H323 client
When I define forwarding address as H323/ip$192.168.1.77|20|r
Asterisk will crash
On Thu, Jun 17, 2004 at 05:28:00PM +0400, AK wrote:
Hello, ereyone!
I have just installed Asterix on my FreeBSD (-current) box
I'm planning to use it as H323 PBX for softphones
Currently I'm stuck in transfering a call to another machine
running H323 client
When I define forwarding
-Original Message-
From: Gonzalo Servat
Sent: Thursday, June 17, 2004 9:34 AM
Sorry to butt into this thread, but I think this is where you went
wrong. There was absolutely no need to quote 70+ lines of text to say
what you had to say. You're supposed to quote the relevant bits
Folks
I am getting the following error as of today after updating both
asterisk and asterisk-addons. These are both under /usr/src.
Any ideas?
dora-debian:/usr/local/src/asterisk-addons# make
./mkdep -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql `ls *.c`
cc -fPIC -I../asterisk
What is this? Day Three?
What is the standing record on this list for flame wars?
You guys need to do a sanity check. These posts are nothing more than
SPAM and Ive just added to it.
I feel so dirty now.
[EMAIL PROTECTED] 6/17/2004 9:08:09 AM
-Original Message-
From: Gonzalo
On 17 June 2004 Eric Wieling wrote:
These are the three cheap SIP phones that I've used.
Grandstream BT10x $65/street
Number only LCD
Zultys ZIP 2 $100/retail
No LCD
Uniden UIP 200 $120/retail
PoE, built-in switch
Are there any online retailers that carry the Uniden UIP
Hi,
I posted the same problem yesterday/day b4?
Add CFLAGS+=-I../asterisk/include to the top of the Makefile
--
[ Nick Luckcuck | [EMAIL PROTECTED] ]
[ Junior Software Developer | Motorola ]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Santiago
Ahh, of course :-) A little fiddling around with expect and I can reboot it from a
webpage now :-)
Thanks.
Best Regards
Michael
On Wed, 16 Jun 2004 14:32:15 -0500
Roger [EMAIL PROTECTED] wrote:
Michael Løjtnant wrote:
Hi,
I have seen a couple of scripts that should be able to
Hi People,
I know that this is a Digium forum, and actually i will buy cards now from Digium too.
But a have just a question.
For test purposes and of course save some money a buy from Ebay a " Mercury M/N: AMI-IA92 card."
With this card Asterisk work well - my linux appear like "Tiger Jet card".
I see in the archives a brief thread between Barton and w
last November 2003 about streaming to the Internet. Id like to use an Asterisk to
mediate multiple VOIP calls originated from the Internet to the studio to be
mixed then passed out to an encoding PC thence back to Internet
Being new to this list i must tread carefully but
Who cares where the answers are so long as they are helpful and to the
point.
If i ask a question it's just nice to get a good clear and concise answer.
Makes no odds to me where the answer is in the reply.
Simon
-Original Message-
On Thu, 2004-06-17 at 09:23, Troy Settle wrote:
A: Because we read the question in the previous message.
Q: Why should I post my reply above the quoted text?
You are assuming that everyone subscribed to the list is reading you
particular thread. If they're not, but are mostly just skimming
Hello All,
I have downloaded and installed spandsp and downloaded rxfax, etc and
rebuilt asterisk with app_rxfax. I have added the following to my
extensions.conf:
[macro-faxreceive]
; ${ARG1} - sendto e-mail
exten = s,1,Wait(2)
exten = s,2,Answer
exten =
We're having a strange problem when an external call is transferred to
an external line. Once the transfer happens, the other line gets opened
but as far as we can tell the number never gets dialed. The person being
transferred gets an extremely loud squealing noise and usually
disconnects. Both
On Jun 17, 2004, at 4:48 AM, Stefan de Konink wrote:
It is that simple?
Probably you want something that actually boots the system too. I don't
know if the ISOLINUX pakage supports a LILO kind of thing, but I guess
it
does. That should be in the MBR of your flash disk and you could
probably
boot
I am having a problem with SJphone registration, having read the list
and wathced it for a while for similar problems. I just can't seem to
figure out the problem.
I tryed to follow a tutorial from
http://www.voip-info.org/tiki-index.php?page=Asterisk+phone+sjphone,
but in SJphone (SIP tab), I
On Thursday 17 June 2004 10:21, Simon wrote:
If i ask a question it's just nice to get a good clear and concise answer.
Makes no odds to me where the answer is in the reply.
Precisely -- this is what this mini flame thread is all about.
Many of us believe that top posting, not trimming, etc.
I do not believe you are correct. We see CALL PROCEEDING in both
directions as part of the normal ISDN call setup process. See trace
below.
Asterisk sends 'CALL PROCEEDING' followed immediately by 'ALERTING'. CALL
PROCEEDING is normally an acknowledgement to a SETUP. See Q931 below:
Luckcuck Nick-LCKN001 [EMAIL PROTECTED] wrote:
I posted the same problem yesterday/day b4?
Add CFLAGS+=-I../asterisk/include to the top of the Makefile
Alternatively (and IMHO, better), make sure you do make install in
asterisk BEFORE trying to do make in asterisk-addons.
Cheers
Tony
--
Hi Jorge!Our application rom version is 4fxosip.102boot version is boot.104I think we need to upgrade the app rom to version 103.I get intowelltech ftp server and founda file called 4fxosipN2004_05_17.BIN. Do you know ifthat isthe last version for the 3804?I solvedsome of theproblems I had.1. I
Joe Baptista wrote:
I understand that most VoIP providers allow for 911 calling but that 911
service is not the same as that available to PSTN.
From what I understand a 911 Call Will Go To A General Access Line at the
Public Safety Answering Point (PSAP). This is different from the 911
Emergency
Of course, big brother wants his say in the matter.
http://www.wired.com/news/politics/0,1283,63884,00.html?tw=wn_2polihead
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Rui,
Create a profile (I've used both Simple SIP and calls through SIP),
then click on it. This should enable the Initialize button.
It opens a window with the fields:
Proxy Domain
Account
Password
CallerID
I am using SJPHone on windows however. Look around in your Linux SJphone
for the
Notice there need not be ANY telco POTS lines.
I wonder if there is a group discussion of this type of functionality.
Would the LINE OUT/IN from Asterisk to analog MIXER console be PC
Sound cards or something more discrete like a form of telco line
cards?
We do not need the
On Thu, 2004-06-17 at 08:20 -0700, Deepak Malhotra wrote:
Hello
Any idea or code on How to allow users to change their voice mail
password over the Phone.
The only way io know is to change in voicemail.conf file and restart
asterisk.
Try dialing your voicemail extension, enter your
On Thursday 17 June 2004 11:38, John Fraizer wrote:
If you have PRI service into your * server, it is possible - though not
always easy - to set the ALI database information specific for each ANI
(DID number) that you use. I do this with our PRI's. Depending on
which number we present to the
To use Asterisk as platform for such a system you probably want to have a
Alsa enabled card which supports routing of multiple channels in and out.
So Asterisk is like the intermediate 'engine' that routes the signal. (Or
sort some soft-mixer). A user is then placed in a Meetme room and the
hold
I see in the archives a brief thread between Barton and w last November
2003 about streaming to the Internet. I'd like to use an Asterisk to
mediate multiple VOIP calls originated from the Internet to the studio
to be mixed then passed out to an encoding PC thence back to Internet
Hi,
I'm a newbee to all this asterisk stuff, and after
reading a fair amount of docs at the voip-info wiki, I
was wondering if it will be possible to work out
something as the following scheme:
A B
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