Is called name delivery supported by Asterisk and SIP?
On various PBX's, if you dial an extension (or a phone number stored in an
internal database), the caller's phone will display the called party's name
on the caller's phone.
This is really handy when you're dialing extensions you don't
I ran into the same problem until I found the answer:
http://lists.digium.com/pipermail/asterisk-users/2004-March/040488.html
Either you have 'signaling forward unconditional' inside voice service
voip or in a dial-peer.
IPTel SEMS, Asterisk and many other SIP Implementations (including IP
Hi Joe,
The Polycom IP phones support this, however currently there is no
support for it in *.
I don't think the SIP RFC requires support for this.
- Brent
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Joe Greco
Sent: Wednesday,
--On Wednesday, October 13, 2004 16:04 +1000 James Bean [EMAIL PROTECTED]
wrote:
a) Ensure you actually have the callerid service provided to your line,
this is usually an extra charge from telstra (AFAIK)
Yep my analog handset on the line (not through asterisk) displays the
callerid of the
James - I have the same problem, and tried a some of the same ideas. No
result.
But at least we both know that a few people in Australia are using Asterisk!
Later,
PaulH
-Original Message-
From: James Bean [mailto:[EMAIL PROTECTED]
Sent: Wednesday, 13 October 2004 4:05 PM
To:
Hi All.
How to receive multiple pages with rxfax ?
Here is what I have:
exten = 10,1,Setvar(FAXFILE=/var/spool/asterisk/fax/${UNIQUEID}.tif)
exten = 10,2,Setvar([EMAIL PROTECTED])
exten = 10,3,rxfax(${FAXFILE})
exten = 10,4,system(/usr/local/sbin/mailfax ${FAXFILE} ${EMAILADDR}
${CALLERIDNUM}
I may have missed something here but couldn't you just do this with a
bit of bash/perl/etc using 'externnotify=' option in voicemail.conf file?
I do this to set MWI via OAI (CTI) on a NEC switch without having to
'integrate' heavily. If you just need those bits you could probably just
On Wed, 2004-10-13 at 17:00, Vladyslav wrote:
Hi All.
How to receive multiple pages with rxfax ?
Here is what I have:
exten = 10,1,Setvar(FAXFILE=/var/spool/asterisk/fax/${UNIQUEID}.tif)
exten = 10,2,Setvar([EMAIL PROTECTED])
exten = 10,3,rxfax(${FAXFILE})
exten =
At 02:10 13.10.2004, you wrote:
On 13-Oct-2004, Dee Lowndes wrote:
If you compiled 0.9.1 on the same system make sure you remove all old
source dir's, /var/lib/asterisk and that X is installed. I did this and
it all installed perfectly well on my slack 10 system.
I also had this same problem
Vlad,
That's because jpeg does not support multiple pages. Use pdf instead:
#!/bin/sh
FAXFILE=$1
RECIPIENT=$2
FAXSENDER=$3
/usr/local/bin/tiff2pdf $FAXFILE | mime-construct --to $RECIPIENT
--subject Fax from $FAXSENDER --attachment fax.pdf --type
application/pdf --header From: [EMAIL PROTECTED]
I'm in the process of setting up the X100P card and I am getting
continuous tone on the X100P but only if plugged into the POTS
line. Here is what I have so far:
# lsmod
Module Size Used by
wcfxs 26912 0
zaptel223460 1 wcfxs
crc_ccitt
You want to use the wcfxo module with the X100P. wcfxs is for the
TDM400P card.
---
Ilia Mirkin
[EMAIL PROTECTED]
On Wed, 2004-10-13 at 03:43, Neil Cherry wrote:
I'm in the process of setting up the X100P card and I am getting
continuous tone on the X100P but only if plugged into the POTS
Not bad at all... I've been trying since September 28th.
I really hate companies with an attitude like that and I'm not even an
end-user, how are you supposed to promote the products of such a
producer to your customers..
Did you send your e-mail to [EMAIL PROTECTED] or another address?
On
On Tuesday 12 October 2004 22:58, Rich Adamson wrote:
Adding resistance to one side of the line only begs for problems
as it creates a tip-ring imbalance that will cause echo, etc,
when other imperfections exist.
If that approach works at all for anyone, its addressing a symptom
and not the
At the time of writing there is no GSM codec in the 190 !
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: 13 October 2004 02:55
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] SNOM 200
Good day all
We have a voicetronix openline4card in a new system
On our old system we had a zaptel card and if a user want to pickup a
remote call he just go *8
How do I do this with a voictronix card?
Please Help
___
Asterisk-Users mailing list
[EMAIL
Hello,
I'm having the following problem in my asterisk config.
I have a little voice menu, with two options,
The welcome message looks like that,
1- press 1, to dial an extension
2- press 2, to speak with an operator.
If I press 1, I get the following message
Dial the extensión number
Quoting Brian Roy [EMAIL PROTECTED]:
Knowing that we are decided to make the move to VoIP, can somebody
tells me
the feasibility of deploying such a solution in an environment that
has the
following technical requirements:
- 250 Users for the Headquarter (100 Mb LAN)
- Around 50
Yeah I have callerid=asreceived in my zapata.conf still nothing
unfortunately.
James
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael
Loftis
Sent: Wednesday, 13 October 2004 4:31 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Its getting pretty well spread here with several ISP's/Telco's offering
IAX connectivity for cheap calls.
It's growing, I hope we can just sort out the callerid thing :-).
Although I could name the line it comes in on so it doesn't just say
asterisk when the call comes in.
James
Where is the cheapest place to buy grandstream phones ?.
And the other day I posted questions about security fir SIP, is the only
solution a vpn ?.
Isn't there SSL integrated in SIP ?.
/Hitete
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
This may be a nastey way of doing it, I'm fairly new to all this * stuff. But crazy
hacks are my chosen style of coding :-P
This MAY work better for you, but this is how I would do it:
Remove include = default
Replace
exten = s,2,Wait,Ttr,200
With
exten =
Hello,
We have a Asterisk CVS-HEAD-08/13/14-12:00:00-BRI-stuffed-0.1.0-RC4a and we
have problem with fax.
zapata.conf:
group = 1
signalling = bri_net
channel = 1,2
channel = 4-5
group = 2
signalling = bri_cpe
channel = 7-8
channel = 10-11
Before install asterisk we have a Panasonic PBX
Hi
I'd like to share a weird and awkward experience...
I have an Asterisk server which connects to various other Asterisk
servers using IAX2 peering through IPsec tunnels.
Recently this server has started to show some weird behaviour. For
example, you would be able to dial out and the console
Hmm,
Didn't think about unloading the driver, sounds like a plan.
I will give it a go when I get home.
Thanks
Ian
From: [EMAIL PROTECTED]Sent: Wed 13/10/2004 02:17To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] TDM01B Goes missing after reboot
On
I don't try the perl script. here is what I expect
from asterisk and sql database for example.
one asterisk pbx per office, several offices,one sql
server.I want to admin all sip conf offices from sql
server
I create one sip table per office on my database
server.
each pbx office get his sip
Ok in order to add a conf file in sip.conf we need to
load app_realtime
harry
--- Brian Wilkins [EMAIL PROTECTED] a écrit :
I believe retrieving in real-time is being worked on
and should be done soon.
Developers are almost finished working on RealTime.
include = sip_additional.conf in
I'm sure you've considered it, but having distributed asterisk services
dependent upon one instance of SQL Server at remote location always being
available seems a weak point in the design. If the SQL Server node is not
available, all asterisk users will be affected.
Have you considered using
On Wed, 13 Oct 2004 10:48:39 +0200, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Where is the cheapest place to buy grandstream phones ?
I have heard that SIPphones.com are about to sell them for $49 or $59
a piece but that may be just a rumour or it may be an offer limited to
those over the age
I agree you a database server must be available for
any applications.
but pbx office get conf from database with perl script
so pbx keep sip config not like ser getting conf from
sql server
harry
--- Bill Seddon [EMAIL PROTECTED] a écrit :
I'm sure you've considered it, but having
Hello all,
I am tring to change the default language in Asterisk, exactly for the
Voicemail messages.
I trying with the option Language=fr in the voicemail.conf global
section, without success.
I trying with the Setlanguage(fr) in the extensions.conf global section,
but without success too.
Hi list!
I set up a dual server config as outlined in the excellent howto at
voip-info.org by JR.
When I try to call an extension on the other server however the call is
not getting through. This is what appears on the asterisk server that
should forward the call to the remote server:
Oct 13
I am trying to setup a SNOM 190 with my asterisk box but having a few
problems
When a call comes in it connects and rings and I can talk no problems...
If I try to call out with the phone I get...
NOTICE[-165364816]: chan_sip.c:7561 handle_request: Unknown SIP command
'PUBLISH' from
El 13/10/2004, a las 12:48, ismaelg escribió:
How could I change the default Languaje for Voicemail?
I have got a /var/lib/asterisk/sounds/fr/ with all the sounds, i have
a letter and diggits directory too.
Any clue will be appreciated.
Mine is running fine, try it.
exten =
On Wed, 13 Oct 2004, Remco Barende wrote:
Hi list!
I set up a dual server config as outlined in the excellent howto at
voip-info.org by JR.
When I try to call an extension on the other server however the call is
not getting through. This is what appears on the asterisk server that
Hi,
is there any more info about securing IAX calls or better said remote iax
extensions ? I feel much more comfortable using IAX.
Regards,
Robert.
- Original Message -
From: Benjamin on Asterisk Mailing Lists [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial
On Wed, 13 Oct 2004 13:07:59 +0200 (CEST), Remco Barende
[EMAIL PROTECTED] wrote:
I omitted the port numbers from the howto because they seem to apply to
IAX1 only which has been obsoleted.
Did you check if chan_iax.so loaded when Asterisk starts?
Did you verify the hosts can see each other
dont think you understood the posters question.. he was asking if *
could be run over a ssh tunnel. not running admin commands via ssh
cli.
On Tue, 12 Oct 2004 23:05:42 -0400, Andrew Thompson
[EMAIL PROTECTED] wrote:
Christopher Jacob wrote:
Anyone ever set up Asterisk to use SSH
spkao wrote:
Wonder if anyone has experienced this. I setup the SIP on * and I found that
it will accept all calls does not matter if the username or secret matches
any
client definition in sip.conf or not.
I thought that was fixed months ago.. You are either running an older
Asterisk or you
On Wed, 13 Oct 2004 13:21:32 +0200, Robert Rozman [EMAIL PROTECTED] wrote:
is there any more info about securing IAX calls or better said remote iax
extensions ? I feel much more comfortable using IAX.
I presume you mean to say you want to encrypt the calls so they cannot
be eavesdropped on
On Wed, 13 Oct 2004 07:33:48 -0400, Jason Price
[EMAIL PROTECTED] wrote:
dont think you understood the posters question.. he was asking if *
could be run over a ssh tunnel.
Did you understand the question, then?
What does it mean to run * over a tunnel?
The OP might have meant to ask if IAX
It works!!! Thanks Steve and Benjamin for the suggestions. I'll try and
see how the WIKI thing works and put a comment.
On Wed, 13 Oct 2004 [EMAIL PROTECTED] wrote:
On Wed, 13 Oct 2004, Remco Barende wrote:
Hi list!
I set up a dual server config as outlined in the excellent howto at
Pedro
You probably need to disable echo cancel when bridged. Can't recall the exact
zapata.conf line. I had problems faxing through Asterisk until I disabled echo
cancelling on bridged Zaptel calls.
Rgds
Tim
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On
On Wed, 13 Oct 2004, Jason Price wrote:
dont think you understood the posters question.. he was asking if *
could be run over a ssh tunnel. not running admin commands via ssh
cli.
Which I have done, and it does work, more or less.
However - tunelling UDP over SSH which uses TCP is not a
Thanks for the response... Of course you can SSH in to a machine and run the
Asterisk CL. That is not what I am asking about. Specifically I am asking
about tunneling. (ie establish an SSH session between my machine and the
server, initiating a tunnel on the SIP/IAX ports, and connecting a client
On Tue, 12 Oct 2004, Geoff Nordli wrote:
Is this where we get to vote for our favorite router software? I choose
Bering-uClibc
(http://leaf.sourceforge.net/mod.php?mod=userpagemenu=910page_id=36). It
comes with a ton of packages, and you can easily configure it to boot from
HDD, or Compact
I have two * servers that are connected with IAX2. Each server has one
pots line attached to it, the users connected to it can dial out by
dialing a 9 and then the telephone number. So far nothing spectacular.
Is it possible however to use the remote POTS line if the local POTS line
is in use?
Christopher Jacob wrote:
Thanks for the response... Of course you can SSH in to a machine and run the
Asterisk CL. That is not what I am asking about. Specifically I am asking
about tunneling. (ie establish an SSH session between my machine and the
server, initiating a tunnel on the SIP/IAX ports,
Just my 2p.
But might it not be a better idea to push for proper secure SIP support.
However this requires a number of steps in the * dev:
- TCP Support for SIP
- TLS Support for SIP
- SIPS Support
- Secure codec support via * (SRTP - http://www.voip-info.org/wiki-SRTP)
tho transcoding is
Jon Radon wrote:
Thanks for bringing this up again Jay.. I wonder how the people
working on the code are doing.. if they've had the time.
The Update:
At the moment we have testapplication connectivity with the Nokia 6310i
and the Jabra headset. With the side note that this connectivity for the
Vladyslav wrote:
Hi All.
How to receive multiple pages with rxfax ?
Here is what I have:
exten = 10,1,Setvar(FAXFILE=/var/spool/asterisk/fax/${UNIQUEID}.tif)
exten = 10,2,Setvar([EMAIL PROTECTED])
exten = 10,3,rxfax(${FAXFILE})
exten = 10,4,system(/usr/local/sbin/mailfax ${FAXFILE} ${EMAILADDR}
Alex Barnes wrote:
Else would VPN's with IPSec or whatever incur less overhead
IPSec VPNs use UDP and IPSec protocols (you can just think of both as
udp) rather than TCP for transport so I would think they have less
overhead as you call it. TCP based tunnels (like SSH tunnels) do all
Hi.
Thank you all for your replies.
Now I do converting into pdf file and it's ok with multiple pages.
tiff2ps -a ${FAXFILE} | ps2pdf - ${FAXFILE//tiff/pdf}
On Wed, 2004-10-13 at 15:39, Steve Underwood wrote:
Vladyslav wrote:
Hi All.
How to receive multiple pages with rxfax ?
Here is
Pavel Jezek wrote:
my favorite alternative to cisco 7912G/7940G is Intracom's Netphone
http://www.intracom.com/en/products/terminal_equip/netphone.htm
Mine is the Polycom Soundpoint IP 500.
begin:vcard
fn:Eric Wileing
n:Wileing;Eric
email;internet:[EMAIL PROTECTED]
tel;work:504-899-1387 x2120
Hi, I have installed Asterisk and it seemed to go well except that i can
not dial out nor in.
This scenario should be plain and simple, but there has to be a small
detail i am missing.
I am trying to call with softphones via Asterisk. Softphone and Asterisk
are behind same firewall. Where SIP/RTP
Is it possible however to use the remote POTS line if the local POTS line
is in use? (sort of fail-over?).
http://www.voip-info.org/wiki-Asterisk+cmd+ChanIsAvail
... JG
--
Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net
We call it the 'one bite at the apple' rule.
Ilia Mirkin wrote:
You want to use the wcfxo module with the X100P. wcfxs is for the
TDM400P card.
On Wed, 2004-10-13 at 03:43, Neil Cherry wrote:
I'm in the process of setting up the X100P card and I am getting
continuous tone on the X100P but only if plugged into the POTS
line. Here is what I
On Oct 13, 2004, at 8:07 AM, Vladyslav wrote:
Hi.
Thank you all for your replies.
Now I do converting into pdf file and it's ok with multiple pages.
tiff2ps -a ${FAXFILE} | ps2pdf - ${FAXFILE//tiff/pdf}
On Wed, 2004-10-13 at 15:39, Steve Underwood wrote:
Vladyslav wrote:
You can also cut to the
When i run asterisk with a H323 communication :
0:40.519 H225 Answer:9734528 H323CreateLogicalChannel -
forward channel
0:40.520 H225 Answer:9734528 H323Found capability:
G.711-ALaw-64k{hw} 1
0:40.521 H225 Answer:9734528 RTP Found existing session 1
On Wed, 13 Oct 2004 13:39:38 +0100, Alex Barnes
[EMAIL PROTECTED] wrote:
But might it not be a better idea to push for proper secure SIP support.
*proper* *secure SIP*
That will win you the gold medal for the double oxymoron of the year :-)
rgds
benjk
--
Sunrise Telephone Systems, 9F
First Thanks to brian for work on valetpark it seems to work really well
I was working on some apps using ValetParking and having good success but
was wondering when you think valetparking will make it into the
CVS/releases? So, I can build around it with a little more confidence.
Thanks
On Wed, 13 Oct 2004, Joe Greco wrote:
Is it possible however to use the remote POTS line if the local POTS line
is in use? (sort of fail-over?).
http://www.voip-info.org/wiki-Asterisk+cmd+ChanIsAvail
Thanks! I had not found this link but this is only for local Zap
interfaces I guess? Can I
No you don't.
You had it right in that last email. 1 db server, multiple * boxes. Make 1
sip table on the db server for each location. Then on each seperate * box,
run the perl script to generate a new sip for that * box. Pretty simple.
Matthew
- Original Message -
From: harry gaillac
got a favorite alternative for Cisco 7940G or 7960G?
Thanks,
Matthew
- Original Message -
From: Eric Wieling [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Wednesday, October 13, 2004 8:12 AM
Subject: Re: [Asterisk-Users] Re: cisco
On Wed, 13 Oct 2004, Joe Greco wrote:
Is it possible however to use the remote POTS line if the local POTS line
is in use? (sort of fail-over?).
http://www.voip-info.org/wiki-Asterisk+cmd+ChanIsAvail
Thanks! I had not found this link but this is only for local Zap
interfaces I
Apparently you did not read my entire message.
I specifically stated it would be non-optimal and a bare-bones solution.
While 600 ohms may be the characteristic impedance of the wire run, a
mismatch at either end will change the impedance of the entire path to
some value that is the related to
On Wed, 2004-10-13 at 16:00, Matthew Boehm wrote:
got a favorite alternative for Cisco 7940G or 7960G?
Have a look at the Polycom IP500 or IP600.
I'm not affiliated.
Regards,
Patrick
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Matthew Boehm wrote:
got a favorite alternative for Cisco 7940G or 7960G?
Thanks,
Matthew
- Original Message -
From: Eric Wieling [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Wednesday, October 13, 2004 8:12 AM
Subject: Re:
I think the Polycom phones do it via a lookup in the directory stored
in the phone. At least, that's how I read it from the Admin Guide.
On Oct 12, 2004, at 11:19 PM, Brent Franks wrote:
Hi Joe,
The Polycom IP phones support this, however currently there is no
support for it in *.
I don't think
On Wed, 2004-10-13 at 03:37, ismaelg wrote:
Hello,
I'm having the following problem in my asterisk config.
But if I wait a moment after this message I get this message again
1- press 1, to dial an extension
2- press 2, to speak with an operator.
Asterisk repeat the
Patrick wrote:
On Wed, 2004-10-13 at 16:00, Matthew Boehm wrote:
got a favorite alternative for Cisco 7940G or 7960G?
Have a look at the Polycom IP500 or IP600.
I'm not affiliated.
The IP500 is a very good alternative to the 7940G:
- one more line appearance
- easier access to DND feature
- can
I have posted to the list about a week ago asking if anyone had got asterisk
successfully connected to verso's class 5 call manager, I received no replys
and I have found nothing. Not surprised, their SIP support is brand new, and
they have been releasing sofware updates almost every other day. To
Great, this is getting me excited!
-Original Message-
From: Stefan de Konink [mailto:[EMAIL PROTECTED]
Sent: Wednesday, October 13, 2004 7:41 AM
To: Jon Radon; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Bluetooth Bounty
Jon Radon
Hi all,
You can download a prerelease of DIAX 0.9.9a from the following location:
http://www.geocities.com/tdanro/diax/diax099a.zip
What's new in this version:
- midi file as ringin signal (polyphonic)
- configurable audio latency
- configurable keyboard support (USB phone keyboard) - by config
I have the same problem. I've got 35 IP500s. I get the echo but can't trace
the blame to one thing.
It only does it part of the time, and it can be on a SIP-SIP call,
IAX-SIP call, or PR-SIP call.
Jody N. Rudolph
Heartland Communications Internet Services, Inc
[EMAIL PROTECTED]
-Original
ok i agree you but what's app_realtime how does it
work?
harry
--- Matthew Boehm [EMAIL PROTECTED] a écrit :
No you don't.
You had it right in that last email. 1 db server,
multiple * boxes. Make 1
sip table on the db server for each location. Then
on each seperate * box,
run the perl
On Wed, Oct 13, 2004 at 07:12:12AM -0700, Chad Scott wrote:
If you wanted to fix an impedance mismatch the right way, you'd use a
matching network. In it's simplest form, you could use a transformer
to convert the 150 ohms impedance at the jack to 600 ohms for the
equipment. You could
You should either not convert (IMHO, this
is not the best solution, as it is difficult to get a decent TIFF
viewer) or convert to another format which does support multiple pages
in a single file (think pdf).
http://www.hylafax.org/links.html#viewers
A tiff viewer is like standard on any
Hi,
I can call iax extension from local iax extension by number or by name.
But from outside (iaxphone) I cannot call something like this
[EMAIL PROTECTED] or better [EMAIL PROTECTED] ?
Is this possible to have and possibly also for iax extensions ?
What should I do to get this working ?
include = sip_additional.conf in [general]
Ok Just for the sake of some poor soul 4 months from now that keeps trying
to do an include and reads this info thinking its correct.
The proper way is:
#include somefile.conf
Or
#include /full/path/to/a/file.conf
No = and you MUST have a # in
Maybe I'm just doing this wrong. Is the FXO card (X100P) used to
connect to the telco pots line?
--
Linux Home Automation Neil Cherry [EMAIL PROTECTED]
http://home.comcast.net/~ncherry/ (Text only)
http://hcs.sourceforge.net/ (HCS II)
Hi Dan,
Did you release the source for DIAX? I'm trying to build a drop-on
component for MS .NET (2005) and I've been looking for a good starting
place. I spent some time with IAXClient and a few other from wiki, but most
are Linux specific..then there's X10, but it's commercial.
Jon Bebeau
Yes,
- Original Message -
From: Neil Cherry [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Wednesday, October 13, 2004 10:54 AM
Subject: [Asterisk-Users] Telco POTS - FXO ?
Maybe I'm just doing this wrong. Is the FXO card (X100P)
NO it won't go in CVS. We have a few options ... 1. Try to work in most (if
not all) the features into the internal parking. 2. Keep it up to date with
latest cvs which is what we do now. www.asterlink.com/svp
bkw
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
Correct. Line as in Wall Jack not as in Phone. You have to connect
your FXO card with a RJ11 cable between your telephone wall socket and
the RJ11 Port in the FXO card.
(You will connect a Analog Phone if you have a FXS card. If you
connect between wall jack and fxs card. You can potentially
On Wed, 13 Oct 2004, Joe Greco wrote:
On Wed, 13 Oct 2004, Joe Greco wrote:
Is it possible however to use the remote POTS line if the local POTS line
is in use? (sort of fail-over?).
http://www.voip-info.org/wiki-Asterisk+cmd+ChanIsAvail
Thanks! I had not found this link but this is only for local
Hi Jon,
- Original Message -
From: Jon Bebeau [EMAIL PROTECTED]
Sent: Wednesday, October 13, 2004 5:52 PM
Hi Dan,
Did you release the source for DIAX? I'm trying to build a drop-on
component for MS .NET (2005) and I've been looking for a good starting
place. I spent some time with
Using 10/12/04 cvs of asterisk and spandsp.0.0.2pre4
After changing line 86 in app_rxfax for new callerid info i got a clean
compile.
Using tiff-v3.5.7 straight from the tiff site and compiled manually no
packages.
I am getting half pages. The first half of page 1 will be fine then it
goes blank.
Anyway we could talk you into releasing the source? I would love to see
wider codec support. And the ability to launch the URL sent with the IAX
call.
bkw
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Dan
Sent: Wednesday, October
OK I had an odd inquiry during hte setup of a PRIDoes asterisk
need/support L1 Flags? I can't even seem to figure out what that means
I thought that ISDN required exchanging capabilities and that's the
nearest I can come to what they mean by L1 Flags. The switch is a DMS-100
on
If I could ask a question about a unique asterisk implementation.
If I wanted to take a BOXA (master) and connect 10 or20 other boxes (slaves)
all running asterisk with the connection being wireless IP using serial
modems. Is this likely to work?
I am not trying to have MANY conversations going.
See http://www.opencall.org/faq/x26.html
Rodger Lewis wrote:
Using 10/12/04 cvs of asterisk and spandsp.0.0.2pre4
After changing line 86 in app_rxfax for new callerid info i got a clean
compile.
Using tiff-v3.5.7 straight from the tiff site and compiled manually no
packages.
I am getting half
ok but if i add or remove variables from database.
Does the perl script overwrite the conf file ?
for example i remove a phone so i run the perl script
on pbx in order to update config file.Is it a problem
for real time calls ?
harry
--- Brian West [EMAIL PROTECTED] a écrit :
include =
The bounty is bogus, the offerors are not serious, and they should take
it off the wiki.
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Glad to hear you guys are making progress. :) I also have a t68i and
M3000 headset, so if you need any help testing just ask.
On Wed, 13 Oct 2004 09:26:02 -0500, Jay Milk [EMAIL PROTECTED] wrote:
Great, this is getting me excited!
-Original Message-
From: Stefan de Konink
The perl script will overwrite the existing conf file. I've had bad
experiences with constant reloading. Maybe you want to schedule your updates
through a crontab.
On Wednesday 13 October 2004 03:28 pm, harry gaillac wrote:
ok but if i add or remove variables from database.
Does the perl
I don't understand your targeted market. Is your software available for
people who have their own asterisk servers and if so why a limit on the # of
usable ports?
Gary
Our already made solutuons are designed for just such scenarios.
Have a look at
- Original Message -
From: Glenn Dalgliesh [EMAIL PROTECTED]
Subject: [Asterisk-Users] ValetParking
To: [EMAIL PROTECTED]
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=iso-8859-1
First Thanks to brian for work on valetpark it seems to work really well
I was working on
There is a correct way of doing this within SIP but I don't know what it is.
I do know that you can fudge it like this
Including fred in the default config
[fred]
exten = fred,1,Macro(stdexten,,SIP/)
Given that SIP/ exists in sip.conf of course !
d
-Original Message-
We have the SMDI interface running to a DMS 10. If anyone is interested let
us know. The code would need a little clean up to get released.
Kent
On 10/13/04 10:29 AM, Michael Welter [EMAIL PROTECTED] wrote:
The bounty is bogus, the offerors are not serious, and they should take
it off the
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