On Sun, 7 Dec 2003, David Coulson wrote:
NOTICE[114696]: File chan_iax2.c, Line 4581 (socket_read): Rejected
connect attempt from 216.234.116.189, requested/capability 0x4/0x4
incompatible with our capability 0xff03.
incompatible codecs, check your allow lines in iax.conf, nufone does GSM
Hi Wasim,
This is Zebble from IRC.
Please sign me up!
Wade Weppler
WW Works Inc.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Sunday, December 07, 2003 2:32 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users]
My apologies to the list for this getting here! I'm off to RTFM on how
to use bleedin' e-mail!!!
-wade
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wade J.
Weppler
Sent: Monday, December 08, 2003 2:23 AM
To: [EMAIL PROTECTED]
Subject: RE:
Hi,
- Original Message -
From: Andrew Thompson [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, December 08, 2003 7:06 AM
Subject: Re: [Asterisk-Users] DIAX 0.9.6 now available- some fixes included
- Original Message -
From: Dan [EMAIL PROTECTED]
To: Asterisk Users
Hi,
There is any other user of DIAX with this problem?
Thanks,
Dan
___
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The original mail with the Makefile and my comments has been held up waiting
for mdoeration, it's probably a little to big. Here is just the makefile
without all my previous ranting.
From the normal zaptel source, do a normal build, backup tones.h and the
makefile, do a make clean, moves tones.h
On Sun, Dec 07, 2003 at 10:57:33PM -0600, Tilghman Lesher wrote:
Prefix is an older application which was more useful prior to being
able to manipulate variables (the days of BYEXTENSION instead of
${EXTEN}). Instead, do:
Dial(SIP/[EMAIL PROTECTED])
I have a smilar problem : I have a
Hi Tony
The configuration looks fine to me. Did you check the log of your tftp
server? Do the phone config files get loaded correctly? Do check also
the Settings/Status/Status Messages of your phone for any errors.
Sascha
---
Sascha Knific
proxy1_address: `129.82.44.223
it that ' really there? that could be it
Thanks,
Will
- Original Message -
From: Sascha Knific [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, December 08, 2003 6:07 AM
Subject: AW: [Asterisk-Users] Phone Unprovisioned Message in IP 7940 ?
Hello tony,
Did you pass the tftp server options in Dhcp server?
I know this might sound silly but maybe you forgot that :)
Monday, December 8, 2003, 2:21:09 AM, you wrote:
tb Hello all,
tb I am newbie to Telephony world (IP and PSTN). Please excuse
tb me if you find my questions very dumb.
Hi,
Project: Call Center Features
Project TurnOver Time: Thursday 11 Dec 2003.
Amount: $250 (negotiable)
Scenerio:
US T1 Asterisk ServerIAX2, GSM/iLBC/G.729 Asterisk Server --- Agents:Agents areusing X-Lite/GS SIP phones.
Features:
1.SIPScan: Like ZapScan, Admin/s want to Zap the client call
Hi,
Is there IAX client in Applet JAVA which can be
embeded in a web page ?
Best regards
Rattana
- Original Message -
From: Dan [EMAIL PROTECTED]
To: Asterisk Users [EMAIL PROTECTED]
Sent: Monday, December 08, 2003 3:49 AM
Subject: [Asterisk-Users] DIAX to DIAX call and disconnecting after 50-60
sec.
Hi,
There is any other user of DIAX with this problem?
Thanks,
Dan
Yes, my
On 08/12/03 13:29, Rattana BIV wrote:
Is there IAX client in Applet JAVA which can be embeded in a web page ?
Nope. But I'm working on a Java IAX2 library that would let you easily
build one. It'll be a little while yet, though. :)
Regards,
Alastair
Hi,
- Original Message -
From: Andrew Thompson [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, December 08, 2003 3:30 PM
Subject: Re: [Asterisk-Users] DIAX to DIAX call and disconnecting after
50-60 sec.
- Original Message -
From: Dan [EMAIL PROTECTED]
To: Asterisk
Hi,
- Original Message -
From: Rattana BIV
To: [EMAIL PROTECTED]
Sent: Monday, December 08, 2003 3:29 PM
Subject: [Asterisk-Users] IAX clients
Hi,
Is there IAX client in Applet JAVA which can be embeded in a web page ?
I hope that version 0.9.7 of DIAX will contain an ActiveX
Exciting, ground floor opportunity to play a
significant role in the development of a new company offering PBX/VoIP solutions
and services. With Asterisk as the core PBX software, our products and services
help our customers realize the advantages of fully converged voice and data.
RedFone
[EMAIL PROTECTED] wrote:
I have a smilar problem : I have a default context for an interface,
where I'd like to prefix all incoming calls DID numbers (basically,
the telco sends the last 4 digits dialed, I want to fully qualify my
E164 number before doing extensions processing).
I don't
On Sun, Dec 07, 2003 at 10:56:38AM -0500, Troy Settle wrote:
-Original Message-
From: Walker Haddock
Sent: Thursday, December 04, 2003 7:54 PM
To: [EMAIL PROTECTED]
We have an installation with 9 inbound voice channels (one is
the fax) and 768K data. It is a Hybrid
On Mon, Dec 08, 2003 at 08:58:07AM -0600, Barton Hodges wrote:
The lines in a context get reordered. If you want to force the order
of those lines, put the exten lines in separate contexts and include
them... something like this:
[some-context]
include = prefix
include =
Dan,
is DIAX opensource?
If so where can I find the sources?
Thanks.
Hector.
- Original Message -
From: Dan [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Lunes, 08 de Diciembre de 2003 11:16 a.m.
Subject: Re: [Asterisk-Users] IAX clients
Hi,
- Original Message -
From: Rattana
Guys,
You can do the same thing w/ the builtin application LookupCIDName.
That's exactly what it was designed for. You just store the information
in database family 'cidname' and use it the same way. Search the
archive or google for examples.
jl
Dan wrote:
Hi,
- Original Message -
It is currently free, but unfortunately I don't believe it is open source.
--
Jonathan Moore
Director of Technology
Winfield Public Schools
Office 620.221.5100
Fax 620.221.0508
Quoting Hcqm [EMAIL PROTECTED]:
Dan,
is DIAX opensource?
If so where can I find the sources?
Thanks.
Hector.
[EMAIL PROTECTED] wrote:
On Mon, Dec 08, 2003 at 08:58:07AM -0600, Barton Hodges wrote:
The lines in a context get reordered. If you want to force the
order
of those lines, put the exten lines in separate contexts and
include them... something like this:
[some-context]
include = prefix
At 11:04 AM -0800 12/5/03, Bob Knight wrote:
Greg Boehnlein wrote:
On Thu, 4 Dec 2003, Bob Knight wrote:
Steve Dolloff wrote:
I would be seriously wary of putting a DS3's worth of voice traffic on a
TNT. I don't believe they are rated to handle that much voice. The
APX1000 would be a much better
On Mon, 2003-12-08 at 09:15, Nicolas Bougues wrote:
On Mon, Dec 08, 2003 at 08:58:07AM -0600, Barton Hodges wrote:
The lines in a context get reordered. If you want to force the order
of those lines, put the exten lines in separate contexts and include
them... something like this:
On Mon, 8 Dec 2003, John Todd wrote:
- a high-density T1 termination system that can handle 8 T1's in a
very small amount of rackspace. DS3 de-muxing onboard would be
optimal, since anyone with 8 T1's is probably getting a DS3 delivery
method, and removing the M13 mux from the rack
Hi,
- Original Message -
From: Hcqm [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, December 08, 2003 4:37 PM
Subject: Re: [Asterisk-Users] IAX clients
Dan,
is DIAX opensource?
If so where can I find the sources?
No, DIAX is not open-source, but it is and will be available as
On Mon, 8 Dec 2003, Jason A. Pattie wrote:
Brian West wrote:
| Also i'm an SBC victim also.. I feel sorry for you SBC is evil.
What about McLeod USA?
They aren't necessarily evil, just incompetent.
--
Dave Weis I believe there are more instances of the abridgment
[EMAIL
I constantly get the following error messages in
/var/log/asterisk/messages:
Dec 8 10:52:57 WARNING[1009521664]: File chan_iax.c, Line 3324
(iax_ack_registry): Received unsolicited registry ack from '192.168.0.1'
Dec 8 10:52:57 WARNING[1009521664]: File chan_iax.c, Line 4181
(socket_read):
I am have trouble getting caller id to work here is my current mgcp.conf am
I missing something?
=
Mgcp.conf
[general]
port=2427
[Egraph-1]
;dlink 104s-1
host = 12.151.207.2
context = local
line = aaln/1
callerid = jim's office 1 321
On Mon, Dec 08, 2003 at 11:05:17AM -0600, Steve Dolloff wrote:
Local server:
register = [EMAIL PROTECTED]
;
[voip2p]
type=peer
host=dynamic
port=4569
trunk=no
qualify=yes
context=IAX
Remote server:
register = [EMAIL PROTECTED]
;
[voip1p]
type=peer
host=dynamic
port=4569
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[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
--- Andrew Thompson [EMAIL PROTECTED] wrote:
DIAX 0.9.6 gave me a little trouble tonight on WinME. (My wife's .laptop, I
don't use ME, honest!) I'll have to fiddle with it some more to get a list
of exactly what's up, but this is what I remember:
It started up in what I think was German. I went
On Mon, Dec 08, 2003 at 10:27:15AM -0600, Dave Weis wrote:
What about McLeod USA?
They aren't necessarily evil, just incompetent.
Any sufficiently advanced incompetence is indistinguishable
from malice
-- Jamie Reid
--
/~\ The ASCII Ribbon Campaign
\ /No HTML/RTF in
Hi,
- Original Message -
From: Kevin Bockman [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, December 08, 2003 7:39 PM
Subject: Re: [Asterisk-Users] DIAX 0.9.6 now available- some fixes included
--- Andrew Thompson [EMAIL PROTECTED] wrote:
DIAX 0.9.6 gave me a little trouble
Hi Sean,
I spoke to Atool about his project and assured him AGI on asterisk was the
way to go. I quoted $1000 for the programming and he may be interested,
again, thanks for the lead.
I'm in Philly but I'm only 2-3 hrs away from the DC market place. If you
would be so kind to pass my
John Todd wrote:
At 11:04 AM -0800 12/5/03, Bob Knight wrote:
Greg Boehnlein wrote:
On Thu, 4 Dec 2003, Bob Knight wrote:
Steve Dolloff wrote:
I would be seriously wary of putting a DS3's worth of voice
traffic on a
TNT. I don't believe they are rated to handle that much voice. The
APX1000
Hi TC,
I followed your instructions. Today I managed to lock up one of our
Asterisk boxes with an incoming SIP call.
It locks up right after seeing I this line:
Dec 8 13:48:34 WARNING[4101]: File chan_sip.c, Line 456 (retrans_pkt):
Maximum retries exceeded on call [EMAIL PROTECTED] for seqno
Hello,
I've continued to have lockups since last week and this is really beginning
to drive me crazy. Could this be a problem with specific hardware or a
specific setting in the configuration files that causes these lockups for
some and not others? If everyone that is experiencing Asterisk
Hi,
I have chan_sip.c version 1.259 do I still need the patch.
I can now get calls from sipphone.com but they drop after 5 seconds.
Regards
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Leif Madsen
Sent: 01 December 2003 18:39
To: [EMAIL
I've continued to have lockups since last week and this is really beginning
to drive me crazy. Could this be a problem with specific hardware or a
specific setting in the configuration files that causes these lockups for
some and not others? If everyone that is experiencing Asterisk lockups
On Mon, Dec 08, 2003 at 07:46:50PM -, David J Carter wrote:
Hi,
I have chan_sip.c version 1.259 do I still need the patch.
yes.
--
/~\ The ASCII Ribbon Campaign
\ /No HTML/RTF in email
X No Word docs in email
/ \ Respect for open standards
Hi all!
I updated my snom200 to 2.02t and now MOH from *
don´t works anymore... only the MOH from snom server and if i clear the MOH
server field in the phone i have noMOH at all..( with the transfer button,
moh playsusing a extension).
Someone with that problem?
I downgrade to 2.01s but
- Original Message -
From: Ariel Batista [EMAIL PROTECTED]
To: Asterisk User List [EMAIL PROTECTED]
Sent: Friday, December 05, 2003 3:14 PM
Subject: [Asterisk-Users] Help with setup IpDialog Sip Phones.
I just got 2 IpDialog phones for use with my Asterisk system. I have been
able
Hi Mark,
Here are some details:
1. Asterisk is still locked, if you need anything let me know please.
2. Asterisk box locked up when processing 1 incoming SIP call (there we no
other calls at the time).
3. Seems to lock up because it receives no ACK to a STATUS 200 OK message
during call
At 12:23 PM 12/8/2003, listas iPfone
[EMAIL PROTECTED] wrote:
I updated
my snom200 to 2.02t and now MOH from * don´t works anymore... only the
MOH from snom server and if i clear the MOH server field in the phone i
have no MOH at all..( with the transfer button, moh plays using a
extension).
- Original Message -
From: Carl Youngblood [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, December 05, 2003 10:19 AM
Subject: Re: [Asterisk-Users] Internet-to-phone gateway?
I'm looking for a solution that allows me to queue up many outbound
voice messages that will be sent to
Hi
all,Has anyone have an idea why, if you capture the files on a Asterisk
network (ex with Ethereal) you always see the communication between the two sip
phones( hard or soft) passing through the asterisk server (on UDP
layer)
Isn't SIP
a protocol that (after that it has established the
SIP control messages goes always through the server
(port 5060) , only RTP media streams is p2p .
you can see RTP passing not p2p but by * server if:
* the phone doesn't supports reinvites
or
* set in sip.conf canreinvite=no in the user definition
Matteo.
Il lun, 2003-12-08 alle 22:17, Wim
Wim Venneman wrote:
Has anyone have an idea why, if you capture the files on a Asterisk
network (ex with Ethereal) you always see the communication between the
two sip phones( hard or soft) passing through the asterisk server (on
UDP layer)
Yes.
Isn't SIP a protocol that (after that it has
You're right for pure SIP configurations , but Asterisk acts here as media gateway
and treats all of the media comm. e.g. to eventually communicate with other types
(like PSTN, H323, AIX, ... the voicemail app.)
Michael Devenijn
IT Manager
DKMA
Schaarbeeklei 636
B-1800 Vilvoorde
Tel.: +32 2
Brancaleoni Matteo wrote:
SIP control messages goes always through the server
(port 5060) , only RTP media streams is p2p .
you can see RTP passing not p2p but by * server if:
* the phone doesn't supports reinvites
or
* set in sip.conf canreinvite=no in the user definition
or if the both ends
Greetings,
I have been experimenting with Asterisk for a few weeks and finally decided to
take the plunge and purchase a few DIDs for inbound calling. Our attempts at
IAX/IAX2 connectivity with VoicePulse have been less than successful. We get
Registration Refused errors from Asterisk whenever we
AGI's are resulting in unusual behaviors. Can someone please tell me
if this is my inappropriate use of AGI's, inappropriate use of
Time::HiRes, or a bug with *:
I call this script twice:
#!/usr/bin/perl
use Time::HiRes qw( gettimeofday );
($seconds, $microseconds) = gettimeofday;
$hirestime
On Mon, Dec 08, 2003 at 10:20:43PM +0100, Brancaleoni Matteo wrote:
SIP control messages goes always through the server
(port 5060) , only RTP media streams is p2p .
you can see RTP passing not p2p but by * server if:
* the phone doesn't supports reinvites
or
* set in sip.conf
[default]
exten = _1NXXNXX,1,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN}
change that to
exten = _1NXXNXX,1,Dial,user:[EMAIL PROTECTED]/[EMAIL PROTECTED]
Regards,
Andrew
___
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[EMAIL PROTECTED]
I am have trouble getting caller id to work here is my current mgcp.conf am
I missing something?
=
Mgcp.conf
[general]
port=2427
[Egraph-1]
;dlink 104s-1
host = 12.151.207.2
context = local
line = aaln/1
callerid = jim's office 1 321
Dear all,
I'd like to know if there is a way for multiple asterisk servers to
share a common SIP and/or IAX registry.
The setup I imagine would be something like :
- several asterisk servers called sip1.isp.com, sip2.isp.com, ...
- a DNS alias sip.isp.com pointing to all the addresses (thus
Brancaleoni Matteo wrote:
SIP control messages goes always through the server
(port 5060) , only RTP media streams is p2p .
you can see RTP passing not p2p but by * server if:
* the phone doesn't supports reinvites
or
* set in sip.conf canreinvite=no in the user definition
or if
On Mon, 2003-12-08 at 15:31, John Todd wrote:
AGI's are resulting in unusual behaviors. Can someone please tell me
if this is my inappropriate use of AGI's, inappropriate use of
Time::HiRes, or a bug with *:
I call this script twice:
#!/usr/bin/perl
use Time::HiRes qw( gettimeofday );
At 01:32 PM 12/8/2003, you wrote:
Greetings,
I have been experimenting with Asterisk for a few weeks and finally
decided to
take the plunge and purchase a few DIDs for inbound calling. Our attempts at
IAX/IAX2 connectivity with VoicePulse have been less than successful. We get
Registration
Hi all,
I have asterisk installed with chan_capi, an isdn fritz card
and a snom 200 phone connected to the asterisk box. The asterisk version is
CVS-11/26/03.
When someone calls me I see his msn number (caller id) on the display of my
snom phone.
But, when he hides his phone number and he calls
On Mon, 8 Dec 2003, John Todd wrote:
AGI's are resulting in unusual behaviors. Can someone please tell me
if this is my inappropriate use of AGI's, inappropriate use of
Time::HiRes, or a bug with *:
I'd say inappropriate use on Time::HiRes. Microseconds increment from 0
up to 999,999
Hello list!
I only can make successful calls if I disable gsm with disallow=gsm. As soon as I allow gsm the following appears at the console. There are much much more Lines with
File dsp.c, Line 1198 but I cut them for a better survey :
- Log Start -
Asterisk Ready.
Another way to handle this
Local server:
register = [EMAIL PROTECTED]
;
[voip2p]
type=peer
host=dynamic
port=4569
trunk=no
qualify=yes
context=IAX
Remote server:
register = [EMAIL PROTECTED]
;
[voip1p]
type=peer
host=dynamic
port=4569
trunk=no
Title: IAXTEL down?
Can not make any calls.
# telnet 69.73.19.178 5036
Trying 69.73.19.178...
telnet: connect to address 69.73.19.178: Connection refused
DEBUG[1133718080]: File chan_sip.c, Line 3407 (build_route): build_route: Contact hop: sip:[EMAIL PROTECTED]:5060
-- Executing
Title: IAXTEL down?
telneting won't work - telnet uses TCP, IAX is
UDP.
- Original Message -
From:
Alexander Romanov
To: [EMAIL PROTECTED]
Sent: Tuesday, December 09, 2003 10:12
AM
Subject: [Asterisk-Users] IAXTEL
down?
Can not make any calls
.
#
On Mon, 2003-12-08 at 16:28, James Golovich wrote:
On Mon, 8 Dec 2003, John Todd wrote:
AGI's are resulting in unusual behaviors. Can someone please tell me
if this is my inappropriate use of AGI's, inappropriate use of
Time::HiRes, or a bug with *:
I'd say inappropriate use on
IAXTEL down?sorry about the HTML, not sure how that turned on
- Original Message -
From: Adam Hart
To: [EMAIL PROTECTED]
Sent: Tuesday, December 09, 2003 10:17 AM
Subject: Re: [Asterisk-Users] IAXTEL down?
telneting won't work - telnet uses TCP, IAX is UDP.
- Original Message -
What about Asterisk log?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adam Hart
Sent: Tuesday, 9 December 2003 10:29 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] IAXTEL down?
IAXTEL down?sorry about the HTML, not sure how that turned on
Should be up now. Found/hopefully fixed a really essoterric iax2
deadlock.
Mark
___
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James Golovich [EMAIL PROTECTED] wrote:
Change your sprintf to:
$hirestime = sprintf(%d%06d, $seconds, $microseconds);
This will make it so that microseconds will always be 6 characters long
or change it to something like:
$hirestime = sprintf(%d.%d, $seconds, $microseconds);
So there will
I want to change the word 'asterisk' into another description. Is this
possible?
show application SetCallerID, SetCIDName and SetCIDNum -- that should help.
Regards,
Andrew
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Thanks Mark
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark Spencer
Sent: Tuesday, 9 December 2003 11:10 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] IAXTEL down?
Should be up now. Found/hopefully fixed a really essoterric iax2
deadlock.
Andrew Kohlsmith wrote:
[default]
exten = _1NXXNXX,1,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN}
change that to
exten = _1NXXNXX,1,Dial,user:[EMAIL PROTECTED]/[EMAIL PROTECTED]
Doing it that way will cause your username and secret to be spread all
over many log files and even in your
Can not make any calls.
# telnet 69.73.19.178 5036
Trying 69.73.19.178...
telnet: connect to address 69.73.19.178: Connection refused
As mentioned yesterday on this list, iax v1 (which is udp port 5036) has
been discontinued. iax v2 (udp port 4569) replaced it, and is now
functional via
Did you make sure to put the in- in front of your register =
in-login:[EMAIL PROTECTED]
Ernest W. Lessenger wrote:
At 01:32 PM 12/8/2003, you wrote:
Greetings,
I have been experimenting with Asterisk for a few weeks and finally
decided to
take the plunge and purchase a few DIDs for inbound
On Mon, 8 Dec 2003, John Todd wrote:
The PM3 LIVES ON DUDE! :) I'm all about Livingson, and have refused
to put the Asscend stuff in my data center. Seriously, Jake over at
portmasters.com is doing some good stuff with the PM3. Now that
we've got control of ComOS, it is just a matter of
Thanks alot. I found the problem, with SIP firmware image was 2.* large size
SIPDefault.cnf leads to Buffer overflow, I kept firmware_version field and deleted
rest. That helped in upgrading the firmware to higher version.
Thanks
Tony
On Mon, 08 Dec 2003 Sascha Knific wrote :
Hi Tony
The
I've attempted to get zaptel to compile with no success. zaptel.c won't
compile because of many references to devfs, which from my understanding
was stipped down to nothing in 2.6 in favor of udev. Following is a
snipit of some of the devfs errors:
[EMAIL PROTECTED]:~/zaptel# make -C
UPDATE:
We were able to consistently reproduce this problem using a Grandstream
phone with buggy firmware.
Mark Spencer logged into our Asterisk and identified the issue. He said it
was a typo in an ast_mutex_lock. After fixing it, the problem seems to have
been solved. We have now repeated
This a re-submittal hoping for some
input:
We are using an MGCP configuration. There seems
to be some incompatibilities between our Netergy T2 VOIP chip and Asterisk.
This is how our client gateway Vendor sees it:
1. The first call is initiated. (CRCX) The interesting
thing
TeleSIP wrote:
And by the way, do not use firmware 1.0.4.18 on GS phones. It contains a
nasty SIP Port bug.
Where do you guys come up with this Grandstream firmware that isn't on
their website?
Are their back channels that the rest of us are kept out of? This is
the umpteenth reference to
Anyone has an good/bad experience setting up Asterisk to work with them?
Or it's incompatible?
___
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We are a service provider and are constantly being handed new firmware by
Grandstream to test on our network. Every time we ask when it will be
offically on their web site they answer soon. But a week or two later
there is a new version that needs to be tested. (and so far all versions
have
sniff sniff.. sounds like an ad to me. maybe not. If its sip it should
work.
bkw
On Tue, 9 Dec 2003, Alexander Romanov wrote:
Anyone has an good/bad experience setting up Asterisk to work with them?
Or it's incompatible?
___
Asterisk-Users
im using 1.0.4.24 and i havent seen any problems yet!
http://www.grandstream.com/TEMP/FIRMWARE/
Miguel
On Tue, 2003-12-09 at 04:11, TeleSIP wrote:
We are a service provider and are constantly being handed new firmware by
Grandstream to test on our network. Every time we ask when it will be
- Original Message -
From: Dan [EMAIL PROTECTED]
To: Asterisk Users [EMAIL PROTECTED]
Sent: Friday, December 05, 2003 4:59 AM
Subject: [Asterisk-Users] DIAX 0.9.6 now available- some fixes included
Hi all,
A new version (0.9.6) of DIAX is available for download at:
Hello
I am getting ready to install chan_h323. Just updated my * with the latest
code from CVS (12/08/03). I was reading the Readme file and confused.
Quoted from the README
NOTICE: Whatever you do, DO NOT USE distrubution specific installs
of Open H.323 and PWLib. In fact you should check to
If you have problems use the latest ones. Otherwise use whats listed
because thats what JJ has tested and is known to work.
bkw
On Mon, 8 Dec 2003, SW wrote:
Hello
I am getting ready to install chan_h323. Just updated my * with the latest
code from CVS (12/08/03). I was reading the Readme
Hi,
Our firm has developed two applications that I
thought might be of interest to members of this list
as both run over Asterisk:
The first is a calling card application that covers
needs in that area: scratch number generation, call
termination via least-cost route (i.e. multiple
termination
Is it possible to log the CallerID of
an inbound call including the time to a log / text file? Also the
same for outbound? ie., dialed number and time?
Thanks.
Regards,
Steven Thomas
Title: Message
This is all
handled by CDR - it can be stored into a database or a text
file.
Its part of
the Asterisk base system.
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steven
ThomasSent: Tuesday, 9 December 2003 4:27 PMTo:
On Tue, 9 Dec 2003, Steven Thomas wrote:
Is it possible to log the CallerID of an inbound call including the time
to a log / text file? Also the same for outbound? ie., dialed number and
time?
/var/log/asterisk/cdr-csv/Master.csv
use SetCallerID in the dialplan if you want to play with
Are they apps that load into asterisk? or agi scripts?
bkw
On Tue, 9 Dec 2003, [iso-8859-1] Kita B. Ndara wrote:
Hi,
Our firm has developed two applications that I
thought might be of interest to members of this list
as both run over Asterisk:
The first is a calling card application that
In response to the postings by Andrew Kohlsmith and Ernest W. Lessenger:
Andrew,
I modified the exten line in extensions.conf as you suggested. Unfortunately,
It still does not work...
Ernest,
I spent approx. 4 hours reading list archives (and anything else Google served up) on
how to
Is there a company website? or just a free yahoo email address?
- Original Message -
From: Kita B. Ndara [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, December 09, 2003 4:01 PM
Subject: [Asterisk-Users] (no subject)
Hi,
Our firm has developed two applications that I
Hi Andrew,
- Original Message -
From: Andrew Thompson [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, December 09, 2003 7:02 AM
Subject: Re: [Asterisk-Users] DIAX 0.9.6 now available- some fixes included
- Original Message -
From: Dan [EMAIL PROTECTED]
To: Asterisk
Well if it links to asterisk and or used any of its code as a base it
can't be sold without a comercial lic. for asterisk. Thats my
understanding of the GPL. If its sold then all the source has to go along
with it right?
bkw
On Tue, 9 Dec 2003, Adam Hart wrote:
Is there a company website? or
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