I want to create a CGI that will allow me to make a call when a user
clicks on a URL in a webpage. I believe I need to create a file in
/var/spool/asterisk/outgoing that defines the number I want to call and
the phone I want to connect it to but I see no way to define the phone
number I want to
Hi List,
I had a wonderful meeting with GS's President last week
and he is very interested in feedback on what top features,
functions, bugs the community would like to see in upcoming
firmware.
Please keep in mind that adding new features take time
to develop, test and such.
So please rate your
I want to create a CGI that will allow me to make a call when a user
clicks on a URL in a webpage. I believe I need to create a file in
/var/spool/asterisk/outgoing that defines the number I want to call and
the phone I want to connect it to but I see no way to define the phone
number I want to
Hello all,
I'm planning to setup a linux box + asterisk as a h.323 conference
bridge.
My goal is to allow incoming h.323 calls with a voice menu that says
welcome, and enter 1 thru 4 for public rooms or enter 5 then code for
private rooms, or such.
How much CPU power will Asterisk require to
John Brown (CV) wrote:
Things like ring tones and fixing call waiting are already
on the list. :)
Lets also keep the replys away from gripes and complaints
and more towards constructive comments.
1. More volume out of the speakerphone, and better range of the headset
volume. I guess it would
John Todd wrote:
I want to create a CGI that will allow me to make a call when a user
clicks on a URL in a webpage. I believe I need to create a file in
/var/spool/asterisk/outgoing that defines the number I want to call and
the phone I want to connect it to but I see no way to define the phone
Hello
Is it possible, (without hacking the source), to change the code for call pickup
because my SIP gateways uses * key as End-Of-Dial.
If I have to hack the source can somebody tell me where to look?
Mvh
Mickey Binder
Comflex A/S
Tlf.: 43997102
John Brown (CV) wrote:
Hi List,
I had a wonderful meeting with GS's President last week
and he is very interested in feedback on what top features,
functions, bugs the community would like to see in upcoming
firmware.
Please keep in mind that adding new features take time
to develop, test and
Hi,
Does anyone have any instructions or pointers to get VM2 to work with MySQL?
I can't seem to find any docs or how-to's on this..
Later..
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
On Mon, 20 Oct 2003, John Brown (CV) wrote:
Hi List,
I had a wonderful meeting with GS's President last week
and he is very interested in feedback on what top features,
functions, bugs the community would like to see in upcoming
firmware.
It goes without saying that consultative transfer
On Mon, 20 Oct 2003, John Todd wrote:
9 - Buttons. The 102 model I have absolutely SUCKS as far as the
buttons go. I have to pretty much press them like manual typewriter
keys to get them to work. Any lateral force causes them to bind up.
10 - button response. Even when I _do_ manage
On Tue, 21 Oct 2003, rnc Info Lists wrote:
9 - ability to switch back and forth between speakerphone and handset
The Grandstream seems to have a strange method of working when it comes to
speakerphone. I would expect the speakerphone button to just switch on
and off the speaker, however it
On Mon, 20 Oct 2003, John Todd wrote:
9 - Buttons. The 102 model I have absolutely SUCKS as far as the
buttons go. I have to pretty much press them like manual typewriter
keys to get them to work. Any lateral force causes them to bind up.
10 - button response. Even when I _do_ manage to
I don't have a single client that runs 10Mbps ethernet in their offices anymore and
to
tell them that the phone will downgrade their network speed to 10Mbps
puts them off the phone straight away..
Hey WipeOut,
Maybe I am missing something here but why would it downgrade their network
Hi all,
I would like to forward an incoming call to my mobile with the incoming callerID after
ten seconds, I have tried with this:
exten = 910,1,Dial,SIP/[EMAIL PROTECTED]|10
exten = 910,2,Dial,CAPI/526$(CALLERIDNUM):bmymobilenr|20
exten = 910,3,Voicemail,u226
exten = 910,102,Voicemail,b226
On Tue, 21 Oct 2003, Low, Adam wrote:
Maybe I am missing something here but why would it downgrade their
network speed to 10mbps, its very rare to find a 100bT switches these
days that don't also support 10bT. In a switched ethernet network there
would be no performance loss for the other
JanM wrote:
Hi all,
I would like to forward an incoming call to my mobile with the incoming callerID after ten seconds, I have tried with this:
exten = 910,1,Dial,SIP/[EMAIL PROTECTED]|10
exten = 910,2,Dial,CAPI/526$(CALLERIDNUM):bmymobilenr|20
exten = 910,3,Voicemail,u226
exten =
Bah, I replied directly instead of to the list. :-(
1 = Nice to have some day
10 = Got to have it right now
10 - Fix SIP disconnection problem
9 - Ringtones (downloadable?)
8 - ILBC
8 - MUCH MORE professional looking case (this includes dropping the four
red LEDs beneath the white plastic
how do I prevent people from calling as soon as I restart the * server ? cos' this
will result (I assume)
in pri channels getting blocked. Because of those few calls that's taken during
restart results in
those few pri channels not to get properly restarted. I need something like 1~2
minutes
On Tue, 21 Oct 2003, Low, Adam wrote:
Maybe I am missing something here but why would it downgrade their
network speed to 10mbps, its very rare to find a 100bT switches these
days that don't also support 10bT. In a switched ethernet network there
would be no performance loss for the other
John Brown (CV) [EMAIL PROTECTED] said:
Please keep in mind that adding new features take time
to develop, test and such.
1. (8) Higher speakerphone volume - the current volume is inadequate;
2. (8) Lower DTMF volume - I usually use the volume at its highest
setting (see 1.), the DTMF tones
On Tue, 21 Oct 2003, rnc Info Lists wrote:
Michael,
How would you be able to connect all phones in a room to one socket? The
Ethernet specificiation has a limit to the number of hubs/switches that
can be inline. (or at least it used to). The only way I can see to
connect all phones to one
6 - 2.5mm headset jack
6.5 - when a headset is connected the ringer should NOT come through the
headset... damn that is annoying on softphones...
Regards,
Andrew
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
On 21/10/2003 11:14, Andrew Kohlsmith wrote:
[...]
6 - POE (12V-48V input range)
[...]
5 - integrated 100mbit switch ***capable of sustaining 100mbit***
[...]
+1 on both of these points. The power brick is cheap and nasty. POE would be
a huge plus. A 100mb bridge would make the phone a lot
+1 on both of these points. The power brick is cheap and nasty. POE would
be a huge plus. A 100mb bridge would make the phone a lot more attractive
in an office full of cables.
I specifically stated a wide POE range because let's face it, with the power
requirements that phone has, a
I love to have on my GS, GSM codec, scale = 10
- Original Message -
From: rnc Info Lists [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, October 21, 2003 1:48 AM
Subject: Re: [Asterisk-Users] Survey: Grandstream improvements.
7 - Ringer volume control
4 - plug in
Yes, just press pound twice.
Mark
On Mon, 20 Oct 2003, Louis-David Mitterrand wrote:
Hi,
This morning I found myself stumped when a remote interactive system
asked me to enter some identification followed by the # key, and my
local Asterisk interrupted with Transfer?.
Is there a way to
Andrew Kohlsmith wrote:
Why not specify a TFTP server/config filename via DHCP? It's already
standard and would work very well.
This would need to be optional, what if a phone was deployed remotely
where you have no control over the DHCP.. then you would need to specify
the config file
Hi!
I defer to your knowledge in this area, but I would be interested to know
what the limit is in terms of the number of devices that can be put
inline.
Correct me if I am wrong:
5 switches on 10 Mbit/s
2 switches on 100 Mbit/s (for the same segment)
Note: Switches slow down your
I know I am posting to myself all the time here but as i didnt find any
info on this when I was looking it may help others..
I have just been playing with the retrieve_sip_from_mysql.pl..
Some notes..
You must create an entry with the keyword account and the value will
be what you want between
Philipp von Klitzing wrote:
Hi!
Correct me if I am wrong:
5 switches on 10 Mbit/s
2 switches on 100 Mbit/s (for the same segment)
Note: Switches slow down your network... cable length matters as well, of
course.
Philipp
IIRC it was 5 HUB's on 10Mbps and 2 HUB's on 100Mbps, I seem to
You'll want to #include it. This leaves the burden of the [general] and
any static configs on sip.conf but allows the script to blindly write out
from the database to sip_additional.conf
in sip.conf:
#include sip_additional.conf
Steve
On Tue, 21 Oct 2003, WipeOut wrote:
I know I am posting
Hi,
Some calls I make trough my PSTN asterisk gateway just hangup
after some minutes. Even if I'm using sip or iax. I have callprogress=no
busydetect=no in my zapata.conf.
Anyone help? Or tell me what to look at /var/log/asterisk/debug. I
didn't find anything wrong.
Are you manually updating the mySQL tables or do you have a web app. to do
that?
Robert
Steve Creel wrote:
You'll want to #include it. This leaves the burden of the [general] and
any static configs on sip.conf but allows the script to blindly write out
from the database to sip_additional.conf
It's documented somewhere for extensions.conf, and I was delighted to see
that it is a function of the config parser, so yes - it's available in the
other files.
On Tue, 21 Oct 2003, WipeOut wrote:
Steve Creel wrote:
You'll want to #include it. This leaves the burden of the [general] and
any
rnc Info Lists wrote:
Are you manually updating the mySQL tables or do you have a web app. to do
that?
Robert
I am manually updating the DB as I have just started playing with the
files today.. but using phpmyadmin its really easy to do..
Later..
It goes without saying that consultative transfer has to be a 10 and I am
sure I am not alone in saying so. Other things are niceties, but when
selling to business this is an expected basic minimum.
I fully agree with that. on my list, 'supervised transfer'
is the more (software) feature
John,
I second Brian's comments. After setting up 20 GS phones using their
somewhat odd web interface, I would really appreciate a more rational
provisioning system for small to medium installations. I would add the
following:
cfgEveryone.txt:Generic setup for all phones.
Hello everyone:
I have a few questions about IAX. As I understand IAX is used for Inter-*
communications.
1. How do the asterisk boxes communicate? Is it over IP or some other mechanism
(T1 trunk?)
2. If I have a scenario where I'd like to use * as a PBX in a Small to medium
enterprise and at the
Hi
Thank you everyone for your help.
I got Asterisk to stop crashing by installing a 2.4.22 kernel.
Best regards,
Ívar Ragnarsson
-Original Message-
From: Michael T Farnworth [mailto:[EMAIL PROTECTED]
Sent: 17. október 2003 22:21
To: '[EMAIL PROTECTED]'
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
How can I check if (i.e.) my provider is requesting me to hide the callerid?
I.e.
(Telco)E1/PRI---Zap(Asterisk1)IAX---IAX(Asterisk2)SIP---EP
Now, if a call comes from the Telco with CLI screening prohobited to
Asterisk1, where the call is
Channel: Zap/g2/14109850123
MaxRetries: 2
RetryTime: 60
WaitTime: 30
Context: extensions
Extension: 84
Priority: 1
JT
From your example you would dial the outbound line and once connected
dial the internal extension.. does this not mean that the external
person would hear the phone
I have a Nortel phone on my desk right now. IF the handset is picked up
and you press the speaker button, it does not hang up but switches back
to the handset instead.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Robert Hajime
quote who=Michael T Farnworth
On Tue, 21 Oct 2003, rnc Info Lists wrote:
Michael,
How would you be able to connect all phones in a room to one socket?
The
Ethernet specificiation has a limit to the number of hubs/switches that
can be inline. (or at least it used to). The only way I can
Since quite a few people in the Grandstream improvements. thread have
requested support for other low bandwidth codecs. I thought I would post this link.
http://www.globalipsound.com/newsroom/releases.php?newsID=46
___
Asterisk-Users mailing
Hi,
I did say this was a first test release :-) I can't be held responsible
for libtiff being empty on your machine, but they other issues are my
fault. I have put a new tarball up, which should need nothing more than
libtiff for the library to work. All the other libraries you had issues
I never had this problem. As all the PBX phones (currently NorTel
Meridian) that I have used work that way. (Speaker button turns on the
speaker, use hook button to switch back to handset.)
Agreed. One thing that would be nice though is to emulate the meridian's
use of the handsfree button
I have a Nortel phone on my desk right now. IF the handset is picked up
and you press the speaker button, it does not hang up but switches back
to the handset instead.
Not with my Meridian system. Just tested to verify:
handset onhook + handsfree/mute pressed: handsfree (goes off-hook)
and the bits 1,2,4 For the queue skillmask just keep multing the number by 2 1 = sales 2 = tech level 1 4 = tech level 2 8 = tech level 3 16 = advanced problems 32 = coperate to allow a queue member to be allowed to take the call just add up all the numbers that go with his skills and set that as
Can anybody give me some directions how i could compile Asterisk for SuSE
8.2 of 9.0. I have a lot of problems compiling.
I have installed the readline-dev ncurses package etc.
Or maybe somebody has a rpm for SuSE available?
With kind regards,
Martin Temmink.
Ashley Jones wrote:
Channel: Zap/g2/14109850123
MaxRetries: 2
RetryTime: 60
WaitTime: 30
Context: extensions
Extension: 84
Priority: 1
JT
From your example you would dial the outbound line and once
connected dial the internal extension.. does this not mean that the
external person
On Tuesday 21 October 2003 01:07, John Todd wrote:
Hi List,
I had a wonderful meeting with GS's President last week
and he is very interested in feedback on what top features,
functions, bugs the community would like to see in upcoming
firmware.
5 - Weight. Phone should weigh more. I'm
Personally, I just wire every jack the same way back to the patch panels,
4pr cat5 or better, terminated in an rj45. Back at the panel wire the blue
pair to your analog telephony stuff, and the org/grn to your networking.
Then if you plug in an rj11 you get a phone line, if you plug in a
On Tue, 2003-10-21 at 01:30, Herc wrote:
Quoting Steven Critchfield [EMAIL PROTECTED]:
Your problem basically comes from the fact that in the analog world, a
busy signal is a audio only signaling. Asterisk could do this detection
if it was directly touching the phones, or if it was
- Original Message -
From: WipeOut [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, October 21, 2003 3:32 AM
Subject: Re: [Asterisk-Users] Auto-dial from webpage
John Todd wrote:
I want to create a CGI that will allow me to make a call when a user
clicks on a URL in a
Does anyone have a quick and dirty script for defragmenting mailboxes?
i.e.:
-rwx--1 root root80553 Oct 20 16:27 msg.gsm
-rw-r--r--1 root root 218 Oct 20 16:27 msg.txt
-rwx--1 root root 781164 Oct 20 16:27 msg.wav
-rwx--
Is there a trick to getting into a zhone z-plex 10 through the serial
interface? I tried using a couple of terminal programs the other day and
didn't get a login.
I have a port that quit working and it doesn't appear to be wiring related.
I had another case like this the other
We have a 10 and we need it yesterday (as well as many other people who don't
even know it). We have a Bug report at GS. The problem is with STUN and
changing IP Addresses. It happens like this:
1. Phone does a STUN query and registers fine.
2. If the public IP Address changes sometime
Hi, thanks for you reply. I'll send you till the end of the week more
info on how to download and use it.
Best regards,
Paulo Mannheimer
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Lal, Deepak
(Contractor)
Sent: sexta-feira, 17 de outubro de 2003
I did say this was a first test release :-) I can't be held responsible
for libtiff being empty on your machine, but they other issues are my
fault. I have put a new tarball up, which should need nothing more than
libtiff for the library to work. All the other libraries you had issues
I'm trying to come up with good jitterbuffer related settings for my
Asterisk boxes.
I ran 4 pings for about 2 days from my main Asterisk server to remote
Asterisk servers. During that time there were some large file uploads
which caused the max rtt to be quite large.
Here are the results:
Hi, thanks for you reply. I'll send you till the end of the week more
info on how to download and use it.
Best regards,
Paulo Mannheimer
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Thomas
Wienecke
Sent: sexta-feira, 17 de outubro de 2003 17:43
To:
Hi, thanks for you reply. I'll send you till the end of the week more
info on how to download and use it.
Best regards,
Paulo Mannheimer
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of James
Coberly
Sent: sábado, 18 de outubro de 2003 14:49
To: [EMAIL
Hi, thanks for you reply. I'll send you till the end of the week more
info on how to download and use it.
Best regards,
Paulo Mannheimer
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Josh
Roberson
Sent: sábado, 18 de outubro de 2003 01:21
To: [EMAIL
My issue is not the encoding of the digits into the data stream, but
the ability of the device to recognize the keystrokes. I use INFO,
as well, after the usual failed experiments with inband and RFC2833
encoding. It just seems like there is some hardware issue that is
not fast enough to
I alwasy laff at those DISCLAIMERS on email... funny they are at the
bottom.
bkw
On Tue, 21 Oct 2003, Low, Adam wrote:
I don't have a single client that runs 10Mbps ethernet in their offices anymore
and to
tell them that the phone will downgrade their network speed to 10Mbps
puts them
On Tue, 2003-10-21 at 09:44, Ryan Tucker wrote:
Does anyone have a quick and dirty script for defragmenting mailboxes?
[snip]
Note the gap between 0003 and 0009. This is caused by a somewhat common
situation, and it tends to bite us somewhat often. :-)
If not, if I get a chance, I'll
[extensions.conf]
exten = 123456,1,SetVar,SIP_CODEC=ulaw
exten = 123456,2,Dial(${TRUNK}/${EXTEN})
The problem is with the SetVar function, the debug shows that the
function is executed, but after that, * sends the media capability to
the phone with g729 as preferred codec.
SIP_CODEC
On Tuesday 21 October 2003 12:01 pm, Maik Schmitt wrote:
We tried to use it witch our AVM Fritz!Card with chan_capi but
asterisk always crashes after our fax-machines shows the ID of the
soft-fax (12345678). Here's a backtrace:
#0 0x417546a4 in TIFFWriteBufferSetup () from
On Tue, 2003-10-21 at 16:22, Steve Underwood wrote:
I did say this was a first test release :-) I can't be held responsible
for libtiff being empty on your machine, but they other issues are my
fault. I have put a new tarball up, which should need nothing more than
libtiff for the library
I know this is going to sound like a strange question, but here goes:
Does anyone know of a SIP softphone that has either a button or a
programmable soft-key to send the current call to VoiceMail?
I'd appreciate any ideas you might have.
Jared Smith
--- Jared Smith [EMAIL PROTECTED] wrote:
On Tue, 2003-10-21 at 09:44, Ryan Tucker wrote:
Does anyone have a quick and dirty script for defragmenting
mailboxes?
[snip]
Note the gap between 0003 and 0009. This is caused by a somewhat
common
situation, and it tends to bite us
I'm using the SNOM 200 latest SIP beta (so that I can have the GSM codec,
etc). Everything seems to be working fine, but the music on hold doesn't
play when I use the HOLD button on the snom. Any suggestions?
Thanks,
--Ernest
___
Asterisk-Users
At 09:53 AM 10/21/2003, you wrote:
I know this is going to sound like a strange question, but here goes:
Does anyone know of a SIP softphone that has either a button or a
programmable soft-key to send the current call to VoiceMail?
Here is what I use with a SNOM 200...
exten =
No I think he means on the phone.. like a softkey to do it.
On Tue, 21 Oct 2003, Ernest W. Lessenger wrote:
At 09:53 AM 10/21/2003, you wrote:
I know this is going to sound like a strange question, but here goes:
Does anyone know of a SIP softphone that has either a button or a
programmable
There is a C Library function that will return a unique
file name. (see man mkstemp)
That's the best way to go. It is generally a
bad design to encode any information in a file name. Better to
simply use the file's date/time stamp to order the messages.
I was speaking with tclark on IRC
As this is a separate project, shouldn't it have it's own mailing list and web
site. ie. sourceforge.
It's OK to announce it, but if everyone added posts about other software that
turned into support and maintenance of said software, then this list is going
to become unusable.
Title: RE: [Asterisk-Users] A software FAX modem
The same problem with tif_dir.h is on RH9, make fails because of that.
Alex Zarubin
-Original Message-
From: Dave Cotton [mailto:[EMAIL PROTECTED]]
Sent: Tuesday, October 21, 2003 11:45 AM
To: Asterisk List
Subject: Re:
Jared Smith wrote:
I think the best long-term solution is to [ask|beg|pay|coerce|convince]
someone to fix the way voicemail messages are numbered to avoid race
conditions.
Here's a thought: don't use the filenames to determine the order the
messages were left, use ctime or mtime on one of the
10 Fix call waiting tone.
9Fix the tftp configs so that I can host my own provisioning server.
Or make a command prompt based tool kit, so that I can use
Gaps with out writing a http screen scraper.
4 Having the Conference button do something would be cool.
John Brown (CV) wrote:
Its still broken... hrm
#0 0x420743da in _int_realloc () from /lib/i686/libc.so.6
#1 0x42073416 in realloc () from /lib/i686/libc.so.6
#2 0x477ea074 in _TIFFrealloc (p=0x477b7d98, s=20) at tif_unix.c:189
#3 0x477b82a7 in t4_rx_start_page () from
/usr/lib/asterisk/modules/app_rxfax.so
#4
Interesting... I see that you are not just reordering the lines but
putting the information in different places. Can you give an example of
what the Context would actually look like for this? I tried this with
Context: extensions and that did not work.
elrod
WipeOut wrote:
John Todd wrote:
That's very close to my suggestion. It is scalable but
only to a point. As soon as you are so big as to require
multiple Asterisks servers you will have the same problem
as the guys who run large e-mail servers. Te first step
would be to NFS mount the mail dir from an NFS server
running some
The current pthreads based locks don't work across mutiple
servers so something needs to be done once you move out of
the small office environment. The maildir design would work
for up to a few thousand users
I like DBMS based designs as they make web based interfaces
easy to implement and would
On Tue, 2003-10-21 at 13:38, Adam Williams wrote:
The current pthreads based locks don't work across mutiple
servers so something needs to be done once you move out of
the small office environment. The maildir design would work
for up to a few thousand users
I like DBMS based designs as they
On Tuesday 21 October 2003 10:52, Brian West wrote:
We have a 10 and we need it yesterday (as well as many other people who
don't even know it). We have a Bug report at GS. The problem is with
STUN and changing IP Addresses. It happens like this:
1. Phone does a STUN query and
Shaun
Thanks
I finally got it working
You where correct. Removing the callerid from the iax.conf file allowed my
DID callerid to show up on the destination phones. I have no idea why the
callerid field in the iax.conf would overwrite the inbound callerid (I
thought it was for outbound
That's very close to my suggestion. It is scalable but
only to a point. As soon as you are so big as to require
multiple Asterisks servers you will have the same problem
as the guys who run large e-mail servers. Te first step
would be to NFS mount the mail dir from an NFS server
running
Title: RE: [Asterisk-Users] A software FAX modem
Found tif_dir.h, make and install look OK. Now it's a coredump:
#0 0x4b4ba652 in _TIFFFax3fillruns () from /usr/lib/libtiff.so.3
(gdb) bt
#0 0x4b4ba652 in _TIFFFax3fillruns () from /usr/lib/libtiff.so.3
#1 0x4b4074d4 in t4_rx_end_page () from
10. Auto answer option on 2nd line appearance. To support paging over the
phones.
Lee
- Original Message -
From: John Brown (CV) [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, October 20, 2003 10:38 PM
Subject: [Asterisk-Users] Survey: Grandstream improvements.
Hi List,
10. Auto answer option on 2nd line appearance. To support paging over the
phones.
That would be very cool. Voice Call I think it's called on the Meridian
system.
DND would be nice too (just return busy)
Regards,
Andrew
___
Asterisk-Users mailing
Martin,
Thank you for replaying. That's exactly what I am trying to do, but the
call never gets answered because is dropped before that due codec
incompatibility.
Please see what the debug shows with my comments in line.
Regards,
Luis
On Tue, 2003-10-21 at 11:36, James Sizemore wrote:
9Fix the tftp configs so that I can host my own provisioning server.
Or make a command prompt based tool kit, so that I can use
Gaps with out writing a http screen scraper.
So I'm not the only one who wrote an http screen
On Mon, 2003-10-20 at 20:38, John Brown (CV) wrote:
So please rate your ideas on a scale of 1-10
10 - Fix the TCP/IP stack. The phones don't work with certain switches
(i.e. the one at my house), and occasionally do other weird things
(although they fixed the MAC address takeover bug,
1st. - I'm from Poland, we don't have (yet, and hopefully forever) software
patents.
Is there any free g.729.1 implementation for asterisk? I want to use it for my
private use (dialing into inet-PSTN gateway), and I don't want (now) to buy
codec, as I don't know if I will be using this service
On Tue, Oct 21, 2003 at 09:32:44AM +1000, Paul Liew wrote:
Sorry, to repost - but I left a /* comment - here it is again
Paul
--- chan_sip.c.save 2003-10-20 21:51:52.0 +1000
+++ chan_sip.c 2003-10-21 09:26:41.0 +1000
@@ -959,7 +959,9 @@
return 0;
Witold Krecicki wrote:
1st. - I'm from Poland, we don't have (yet, and hopefully forever) software
patents.
Is there any free g.729.1 implementation for asterisk? I want to use it for my
private use (dialing into inet-PSTN gateway), and I don't want (now) to buy
codec, as I don't know if I
Agreed, don't drive up my shipping cost. light is good.
Tilghman Lesher wrote:
I'd have to respectfully disagree. If this is really a problem I'd
suggest taking advantage of the mounting bracket on the bottom
and either attach the phone to the desk or attach a sheet of lead.
-Tilghman
Steve Creel wrote:
You'll want to #include it. This leaves the burden of the [general] and
any static configs on sip.conf but allows the script to blindly write out
from the database to sip_additional.conf
in sip.conf:
#include sip_additional.conf
Eureka! ...is this #include construct a general
go to the Vovida site http://www.vovida.org/ and checkout
the Open G.729(A) Initiative
There is an open source g.729 there
--- Witold Krecicki [EMAIL PROTECTED] wrote:
1st. - I'm from Poland, we don't have (yet, and hopefully forever)
software
patents.
Is there any free g.729.1
1 - 100 of 144 matches
Mail list logo
