Honestly I can't see all these NAT woes people speak of... I have * on a
public ip .. sip.conf entries with nat=yes load em up.. and they work. So
I have yet to see why everyone has SO MANY problems.
bkw
On Tue, 28 Oct 2003, Christopher Stephens wrote:
> Hello everyone and welcome to my first p
Crank down your reg int on the 7960 to like 120
telnet into the phone and issue:
reg 1 1
reg 1 2
reg 1 3
reg 1 4
reg 1 5
reg 1 6
Its just confused.. if you wait long enough it will correct itself. This
is why you crank the reg int. down so it fixes itself faster. :P
bkw
On Mon, 27 Oct 2003, Ph
It works in /usr/local/bin/ now also.
On Mon, 27 Oct 2003, CW_ASN - Gus wrote:
> MPG123 is not included in Asterisk...
> Download the package:
>
> http://www.mpg123.de/cgi-bin/sitexplorer.cgi?/mpg123/precompiled/
>
> Install using:
>
> rpm -ivh mpg123-0.59q-1.i386.rpm
>
> Copy the file mpg123 fro
> 2) Transcoding: To be avoided at all times
>
> Transcoding is the conversion of a voice stream with one codec to a voice
> stream with another codec (e.g. G.729 to G.7.23). Transcoding
> dramatically degrades the voice quality. It has to be avoided at all
> times.
I really dont know what they ha
If I understand correctly the Sipura people are the same guys that made
the Cisco ATA (Komodo phone) or what ever. I'm going to get one of the
Sipura SPA-2000 to use and abuse with * I have seen the web
interface.. John over at Chagres was nice enough to let me login to one
and look around a
I just did some testing[sip] with the guys at stealthtele.com with * and
everything went great... thinking setting up an account with them sometime
soon... He said they were working on IAX but not sure how far out that
would be Has anyone else checked them out?
bkw
dtmfmode=inband is evil anyway. :P
On Sun, 26 Oct 2003, CW_ASN wrote:
> I had similar problems, and were related to dtmfmode=inband in sip.conf
>
>
> - Original Message -
> From: "duncan" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Sunday, October 26, 2003 9:15 PM
> Subject: Re:
Yes its called CN. www.bkw.org/~brian/cisco/ata.html
check audiomode and connectmode
RFC 3389 - Real-time Transport Protocol (RTP) Payload for Comfort Noise
(CN)
bkw
On Sun, 26 Oct 2003, Phillip Jackson, Director of IT wrote:
> Hello all,
>
> Things are going well. I've even unlocked that ex
Yep.. does the same with my credit card machine.
On Sun, 26 Oct 2003, Andrew Kohlsmith wrote:
> > Has anyone had any luck getting a ReplayTV DVR box to connect
> > through an Asterisk box? Mine seems to dial just fine, but can't
> > negotiate a connection. I am using:
>
> I would suggest NOT us
Did you try fxsls or gs?
On Sun, 26 Oct 2003, Jean-Philippe Lord wrote:
> Hi All...
>
> I'm currently trying to have an extension of my Lucent Partner phone
> system connected to Asterisk using an X100P. The issue I'm having is
> that the Lucent Partner analog port connection have different ring
Is it an error or is it a WARNING?
On Fri, 24 Oct 2003 [EMAIL PROTECTED] wrote:
> when I dial out from my Cisco phone I get this error
>
>
> File channel.c, Line 2258 (ast_channel_bridge): Didn't get a frame from
> channel: SIP/210.9.49.216-c26e
>
>
> Regards Mick
>
>
> My 'sip.conf' file reads:
> [general]
> port=5060
> bindaddr=0.0.0.0
> context=default
>
> [sjphone]
> username=name
> secret=password
> host=dynamic
> defaultip=192.168.1.120
username= can go... the part in [] is the username or is on
> How many hobbyists/hackers/etc. would have any NEED to have T-1 hardware
> in their house?
Its not that they need it.. its that they want it.
bkw
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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 0
> 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.
>
> Reg
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
> >program
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
> > put
> 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 t
> 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 someti
John,
I want the tftp configs done like cfgMACADDRESS.txt or compile
them into a binary form like the ATA's use. And stop trying to rip us for
the GAPS system. WHAT A RIP. It makes cisco so worth the extra cash!
Config refresh similar to the ATA.. refresh config every x seconds.
bkw
O
Yes i'm using one of the workarounds.. but you can't do a native transfer
to the parking extension. # transfer yes.. but that is NOT a native sip
transfer.
On Mon, 20 Oct 2003, Juan J. Sierralta P. wrote:
> On Mon, 2003-10-20 at 16:44, Brian West wrote:
> > > - I coul
> - I couldn't get Asterisk call-parking to work with this phone, transferring
> to extension 700 doesn't work(and it works fine with my SNOM200) maybe just
> a config change on my end, but I couldn't figure it out
cant do native sip transfers to parking.
__
Good job.. now that the cat is out of the bag i'm sure you will get alot
of requests or ideas and maybe code!
bkw
On Mon, 20 Oct 2003, Steve Underwood wrote:
> Hi all,
>
> I would like to announce the availability of an initial test version of
> a totally software FAX facility, suitable for use
You mean 5.3. I'm currently running 5.3 on my 7960.
bkw
On Sun, 19 Oct 2003, Tomica Crnek wrote:
> I am using 5.03 image on 7940 and 7960 and it is ok
>
> - Original Message -
> From: "Andy Powell" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Sunday, October 19, 2003 1:03 PM
>
You must use GSM with iaxtel and Voicepulse for now... I talked to the guy
from voicepulse and they said ilbc might be turned sometime in the future.
But not sure.
bkw
On Tue, 14 Oct 2003, Stig Hess wrote:
> I'm having trouble configuring these services the way I want. Basically I
> prefer using
register => abatista:[EMAIL PROTECTED]/114 doesn't work in iax.conf
also you are sending the full 917009965342
you should only send ${EXTEN:1} strip that 9 off.
bkw
On Mon, 13 Oct 2003, Ariel Batista wrote:
> Hello I am still having problems with IAXTELL and FWD configuration. I get the
> fo
Are you using the recommended pwlib and openh323 tarballs?
bkw
On Mon, 13 Oct 2003, CW_ASN wrote:
> Hi all:
>
> I've got some core dumps when I use chan_h323. I dial an extension using
> h323, routed thru an E100P (like a H323-ISDN_PRI gateway). Sometimes *
> hangs, sometimes not. The client use
Record(/tmp/testing:gsm)
Thats what I use.. and it works.
bkw
On Thu, 9 Oct 2003, Lists wrote:
> If I do something like
>
> exten => 1,1,Record(/someplace/somefile|gsm)
>
> It does not record I end up getting
> -- Executing Record("SIP/mlh-04d0", "|gsm") in new stack
>
> exten => 1,1,Record(
show channels?
On Thu, 9 Oct 2003, duncan wrote:
> So whats the best way to find the maximum number of concurrent calls in
> this setup:
>
> IAX2 Trunk using GSM over a 512k internet line.
>
> thanks
>
>
> duncan
>
> ___
> Asterisk-Users mailing list
>
> I've implemented a bit of a workaround.
>
> I've setup the dial plan 2 in my system as the call park prefix.
>
> When you want to park a call, you blind transfer to 2 where is your
> extension (eg: 27011). The call is parked, and you will immediately receive
> a call announcing the p
> If I dial # while in a call nothing happens. I was transfering using
> the 7960 transfer function which gives me a dial tone and then I dial
> 700 which gives me a busy tone I also tried to dial #700 but as soon as
> you push # on a 7960 it dials since # its used to signal the end of the
>
Nope mine sounds fine.
On Wed, 8 Oct 2003, Adam Rothschild wrote:
> Hello,
>
> Anyone else having problems with the Cisco 7940/7960 (5.3 firmware)
> and the latest CVS build, placing conference calls from the phone?
> I've noticed the party on the Cisco phone's side will sound very
> garbled, and
Yes but you can't do native sip tranfers to parking. Thats what I want.
And thats what I was talking about. You can't say use a Cisco 7960 and
hit transfer then dial 700 then transfer. WONT WORK.
bkw
On Tue, 7 Oct 2003, Andrew Joakimsen wrote:
> You need to enable transfer:
>
> Dial
> Dialing
Not yet.. but I sure wish we could... :)
On Tue, 7 Oct 2003, Juan J. Sierralta P. wrote:
> Hi,
>
> It is posible to put a call in the parking lot with a SIP phone as a
> Cisco 7960 ?
> Anyway, how can I put a call park on a FXS line ? Is there any magic
> digits ?
>
> --
> Juanjo sin
I dont see it as a bug.. I see why it don't work.. and why people think it
should. Enable # transfers.. and setup call parking to get around this.
Also if you conf very much look at app_meetme.
bkw
On Tue, 7 Oct 2003, Juan J. Sierralta P. wrote:
> On Tue, 2003-10-07 at 12:09, Brian We
exten => h,1, will not work if you park a call then pick it back up. You
are flipping the call direction from what Mark told me. Whats wrong with
CDR data? is that not good enough to tell call lenght?
bkw
On Tue, 7 Oct 2003, mattf wrote:
> The way I worked around this is to log the "uniqueid"
> I´m having a similar problem with my 7960 when I receive two incoming
> calls I cannot join them.
ya you can't join them. That sucks.. but you can park one call, go back
to call number 1. Press conf. Dial the parking orbit.. then press join!
bkw
___
> use
> mailbox=500
>
> instead of [EMAIL PROTECTED]
[EMAIL PROTECTED]
since he doesn't have his stuff in the default context
bkw
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Works fine on my 7960 with 5.3 firmware.
bkw
On Mon, 6 Oct 2003, Babak Pasdar wrote:
>
>
> Hello,
>
> I am trying to conference two or more calls on a Cisco 7940 phone. When I have one
> inbound call and one outbound (I initiate the second call by pressing conference) I
> get the join button
Why do stuff the hard way?
; used to record prompts
exten => 205,1,Wait(2)
exten => 205,2,Record(/tmp/asterisk-recording:gsm)
exten => 205,3,Wait(2)
exten => 205,4,Playback(/tmp/asterisk-recording)
exten => 205,5,Wait(2)
exten => 205,6,Hangup
On Mon, 6 Oct 2003, Stuart Mackintosh wrote:
> I ha
Or you can use safe_asterisk to start * then asterisk -r to connect
bkw
On Thu, 2 Oct 2003, PJ Welsh wrote:
> on Thu, Oct 02, 2003 at 02:53:00PM -0500, Andy Hester wrote:
> > This probably has an easy solution, but I found it yet. How can I get out
> > of a remote console after using ssh to get
Strange.. I still can't get iaxtel to work unless the [iaxtel] entry is at
the bottom of the file.
bkw
On Thu, 2 Oct 2003, Mark Spencer wrote:
> The location of the "guest" / "iaxtel" section having to be at the end is,
> as it turns out, a configuration error on iaxtel. I hope to have it
> str
Hrm blew away my src tree checked it out again.. and I don't see this
problem anymore.
bkw
On Wed, 1 Oct 2003, Brian West wrote:
> Its not fixed. I still have the issues with fwd using the latest cvs
>
> On Wed, 1 Oct 2003, John Todd wrote:
>
> > >Can anyone confirm t
Its not fixed. I still have the issues with fwd using the latest cvs
On Wed, 1 Oct 2003, John Todd wrote:
> >Can anyone confirm that the SIP updates in CVS have fixed the channel
> >leakage and the codec negotiation problem that was happening a few days
> >ago?
> >
> >Thanks
> >dave
> >
> >
> >-
Any nat involved? and what codec's are you trying?
On Tue, 30 Sep 2003, Kevin wrote:
> When I dial with my Grandstream 101 telephone to another sip phone or
> Zap FXS, the call rings, but no audio is passed. Eventually the call
> gets disconnected. The same thing happens if I dial the Grandstre
Have a way to specify it in the src? I would like to try the 8k between a
few servers and see how it sounds.
bkw
On Tue, 30 Sep 2003, James Golovich wrote:
>
>
> On Tue, 30 Sep 2003, WipeOut wrote:
>
> > Whats the default SPEEX bitrate set to in Asterisk?
> >
>
> The default bitrate for speex (
Sep 2003, WipeOut wrote:
> Brian West wrote:
>
> >You can't anymore MySQL was ripped from Asterisk because the client libs
> >are GPL.
> >
> >bkw
> >
> >On Tue, 30 Sep 2003, Manoj K Gupta wrote:
> >
> >
> >
> I still don't get
You can't anymore MySQL was ripped from Asterisk because the client libs
are GPL.
bkw
On Tue, 30 Sep 2003, Manoj K Gupta wrote:
> Hi list,
>
> I am trying a scenerio where the * will take the email and mailbox number from the
> Mysql database for sendming mail to a voicemail user. I have seen v
http://www.loligo.com/asterisk/current/
I'm sure he has a few in his sip.conf examples
On Mon, 29 Sep 2003, Leif Madsen wrote:
> Hi all,
>
> I would like people to email me at 'leif at hacklocalhost dot com' some
> example configuration files for VoIP providers which * can register
> with. I am
+0100
> WipeOut <[EMAIL PROTECTED]> wrote:
> >*This message was transferred with a trial version of
> >CommuniGate(tm) Pro*
> >Brian West wrote:
> >
> >>>I have found the CDR in general to be a problem, We use a
> >>>system that
> >>&
> I have found the CDR in general to be a problem, We use a system that
> allows a user to simply click a phone number in a web page and then PHP
> drops a call file into the /outgoing directory.. These calls are not
> logged at all.. not in the text file or the MySQL..
Can I make calls thru your
ie a perl script. :)
On Mon, 29 Sep 2003, Mark Evans wrote:
> >> As far as I am aware the CDR logging can only log
> >> to MySQL or CSV text file..
>
> It doesn't matter, we could write an automatic import routine which can
> import that CVS into anything :)
>
> Regards
>
> Mark
>
>
> ___
> Personally, I *love* MySQL, and I'm a bit surprised by their sudden change
> from public domain (and maybe LGPL) to GPL for their client libraries...
Who can we bug at mysql to see if we can get that changed?
bkw
___
Asterisk-Users mailing list
[EMAIL
> Then you haven't been around Open Source software long, have you.
> There are many "wars" on going some going on many years now.
Yes but I have never paid much attention to all this drama.
bkw
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http://l
I ment BSD lic. mybad. :P
On Mon, 29 Sep 2003, Brian West wrote:
> I recall someone saying that hdparm is embeded in the codec and
> Registration binaries.. and that is a violation of the GPL. But thats
> voiceage's doing. Anyone care to shed some light on this?
>
> bk
I recall someone saying that hdparm is embeded in the codec and
Registration binaries.. and that is a violation of the GPL. But thats
voiceage's doing. Anyone care to shed some light on this?
bkw
On Mon, 29 Sep 2003, Eric Wieling wrote:
> On Mon, 2003-09-29 at 09:40, Mark Spencer wrote:
>
> >
I would also like some more info on this whole mysql being taken out of
the core asterisk install. I understand its because of the dual lic. that
digium has.. gpl and comercial... why can't mysql be non-existant in the
comercial version. Then mysql would be compatible with asterisk?!? Or am
I wr
w that 210 lives on BOXB and to route the call over the IAX
> trunk to BOXB
>
> Lee
> ----- Original Message -
> From: "Brian West" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Friday, September 26, 2003 10:56 PM
> Subject: Re: [Asterisk-U
What must be done for everyone software to be able to play on the same
playground without the parents watching?
bkw
On Fri, 26 Sep 2003, Mark Spencer wrote:
> Just FYI, MySQL stuff has been pulled from Asterisk since apparently now
> the client libraries are under GPL and not LGPL (and thus are
Just a heads up.. you can't loop switch statements
ie
BOX A switch => BOX B
BOX B switch => BOX A
show dialplan will show the switch but not the dialplan of the remote
switch.
bkw
On Fri, 26 Sep 2003, Lee Goodman wrote:
> Hello
>
> I want to setup a IAX trunk between 2 asterisk servers. I als
Some ISO firmwares have bugs.. we ran into this and had to downgrade to
get it to work correctly.
bkw
On Fri, 26 Sep 2003, Bartosz Jozwiak wrote:
> I have fixed i already.
> And still it does not want to work :(
>
>
> - Original Message -
> From: "Sean Figgins" <[EMAIL PROTECTED]>
> To:
I used it with my 7960 also. all you do is set the Preferred Codec to
g729 and put the below in your [general] section.. and its hould just
work.
disallow=all
#allow=g723.1
allow=g729
allow=ilbc
allow=gsm
allow=ulaw
On Fri, 26 Sep 2003 [EMAIL PROTECTED] wrote:
>
> >
> > I've used it with the C
Dream on!... as of now they dont.
On Thu, 25 Sep 2003, Gary wrote:
>
> I would probably be interested except when will their products actually
> support GSM codecs ??
>
> On Wed, 24 Sep 2003 21:34:36 -0500 (CDT), Dave Weis wrote:
>
> >
> >On Thu, 25 Sep 2003, Aaron Martin wrote:
> >> Does anyone
http://www.chagres.net/products/voip/phones.html
bkw
On Thu, 25 Sep 2003, Aaron Martin wrote:
> Does anyone know of any reliable supplier for Grandstream phones?
>
> I tried dealing with David Li from Grandstream, but after emailing him an order in
> August, and asking how he wanted payment, I
http://voipstore.atacomm.com/shops/Browse.aspx/27934028032-27934130944.htm
bkw
On Wed, 24 Sep 2003, Mike Hjorleifsson wrote:
> Does anyone sell a preinstalled asterisk server ?
>
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.di
has a T1 adapter on a high-density high-density
> voice port adapter?
>
> BTW... Because I am lazy, what does plar do?
>
> -Sean
>
> On Wed, 24 Sep 2003, Brian West wrote:
>
> > This is simple to do..
> >
> > voice-port 1/0/0
> > connection plar
t; > Reply-To: [EMAIL PROTECTED]
> >
> >
> > That is about what I have been seing for help. Has anyone any clue what
> > to di with a 2600 that has a T1 adapter on a high-density high-density
> > voice port adapter?
> >
> > BTW... Because I am lazy, what
y
> voice port adapter?
>
> BTW... Because I am lazy, what does plar do?
>
> -Sean
>
> On Wed, 24 Sep 2003, Brian West wrote:
>
> > This is simple to do..
> >
> > voice-port 1/0/0
> > connection plar
> > !
> > voice-port 1/0/1
> >
This is simple to do..
voice-port 1/0/0
connection plar
!
voice-port 1/0/1
connection plar
!
dial-peer voice 1000 voip
max-conn 4
destination-pattern
req-qos guaranteed-delay
codec g711ulaw
ip precedence 5
no vad
session target ipv4:x.x.x.x
!
in h323.conf set the context=b
their really isn't much fixed between 4.4 and the 5.x stuff but at the
time thats all I had. So I put that on the phone. So far everything
works like a champ. Not one problem.
4.4
http://www.cisco.com/en/US/products/sw/voicesw/ps2156/prod_release_note09186a008016096f.html#63943
Resolved Caveat
I have tried for over a month off and on to get iaxtel for inbound to
work... and tonight after alot of troubleshooting we noticed this:
iaxtel inbound will use the last entry in your iax.conf to auth against.
So if [iaxtel] is at the top and say [voicepulse] at the bottom. An
inbound call will t
Here's one thats way out in left field... don't use call pickup! :P
Problem solved sorta!
bkw
On Mon, 22 Sep 2003, Jared Smith wrote:
> On Mon, 2003-09-22 at 15:42, Manuel Marín García wrote:
> > Please help! When I try to place a call pickup from a cisco phone 7960
> > using *8 the call is
[EMAIL PROTECTED] needs to fix their spam filter. Please stop using it or
learn to configure it.
+ 1 Sep 22 AntiSpam UOL (6828) RE:Re: [Asterisk-Users] MS Outlook
Thanks,
Brian
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://li
I second that... I have received a load of virii from people on this
list..
Received: from torch.junct.com (sootbox.junct.com [65.168.64.10])
by www.bkw.org (8.11.6/8.11.6) with ESMTP id h8MIcEJ06998
for <[EMAIL PROTECTED]>; Mon, 22 Sep 2003 13:38:14 -0500
Received: from wdxmvur (u
Its fairly simple.. meetme isn't that big you can find where the hooks are
its commented in the code.
bkw
On Mon, 22 Sep 2003, Chee Foong wrote:
> Hello,
>
> Is there a asterisk developer guide/source code doc or something like that?
>
> I want to see if I can implement the admin menu function f
Just on a side note can you please put a realname in your name field on
your email client. Everytime I see "Asterisk PBX" I think gee more
voicemail.
bwk
On Sun, 21 Sep 2003, Asterisk PBX wrote:
> My partner found it!!
>
> Problem solved...
>
> The error was a syntax error in the zapata.conf
>
I bet your jack is wired backwards.. :) Try checking that out.
bkw
On Sun, 21 Sep 2003, Asterisk PBX wrote:
> The echo canceller algorithms aren't doing anything. We get extreme
> echo during the conversation, it appears even before the call connects,
> the echo is there...
>
> This only happe
--dport 5060 -j
> ACCEPT
> -A INPUT -s x.x.x.x -p udp -m udp --dport 1:2
> -j ACCEPT
> -A INPUT -s x.x.x.x -p tcp -m tcp --dport 22 --syn -j
> ACCEPT
>
> ...
>
> Sunny
>
> --- Brian West <[EMAIL PROTECTED]> wrote:
> > I'm trying to get some iptables
I'm trying to get some iptables rules that work with asterisk but for some
reason I keep blocking everything and or locking myself out of the box..
mybad does anyone have any configs they would like to share that allow
asterisk and ssh from x ip?
TIA
bkw
__
Try gdb asterisk /etc/asterisk/core.2035
bkw
On Sat, 20 Sep 2003, Dan Fernandez wrote:
> Last week I did a CVS update and since then I haven´t been able to run asterisk
> normally. The strange thing is that I have even go back to previous versions (0.5.0)
> and I am seening the same problems.
RFC 3389 - Real-time Transport Protocol (RTP) Payload for Comfort Noise
(CN)
Turn off CN and you will be fine.
bkw
On Sat, 20 Sep 2003, Don LeBlanc wrote:
> Hello All,
> We are running Asterisk on a linux server as a SIP proxy with Cisco ATA 186's at the
> subscriber end. For long distance we
Doesn't matter it should still work. Here is a hint.. dont use
passwords/secrets it will then work!
bkw
On Fri, 19 Sep 2003, Xisco wrote:
> That's true if always there to connect two asterisk servers, but I'm doing
> some proves in order to connect one asterisk server with another SIP server.
>
Mark,
I added one to the bug report. Hope that helps.
bkw
On Fri, 19 Sep 2003, Mark Spencer wrote:
> I"ll need a backtrace.
>
> Mark
>
> On Fri, 19 Sep 2003, Dave Cotton wrote:
>
> > Using CVS update from 11:00 CET today * crashes at this point.
> >
> > == Parsing
> > '/var/spool/aster
Ya I just got my phone and upgraded it to 5.3 without a problem. It works
perfect with *
bkw
On Wed, 17 Sep 2003, Travis Johnson wrote:
> Yes. 30 phones in production environment. No problems so far. :)
>
> Travis
>
>
> At 08:21 PM 9/17/2003 -0500, you wrote:
> >Anyone running the 5.x firmware
Nope works fine here...
NUFONE ROCKS!
bkw
On Wed, 17 Sep 2003, Peter Pauly wrote:
> Is anyone else having trouble dialing 800 numbers
> through Nufone? I'm getting the SIT tones no matter
> what number I dial. Normal long distance works fine.
> I don't think it's my dial plan (it was working pr
Anyone running the 5.x firmware on their 7960's with asterisk?
bkw
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Ya fire your network admin :P Firewalls shouldn't be blocking cvs and
such if they do then your admin is way too anal.
bkw
On Wed, 17 Sep 2003, Dan Austin wrote:
> So I've been trying to pay attention, but I hadn't seen any updates on
> SourceForge.
>
> I inferred from the thread I could get a
Thats all going to depend on the speed of your DSL...
bkw
On Wed, 17 Sep 2003, Senad Jordanovic wrote:
> Hi,
>
> What are real life bandwith stats for * supported codecs?
> Is it true one can run 6-32 conversations over DSL, as stated in this list?
>
>
> Senad
>
>
> _
Just as simple to call your telco and have those turned off then its not
an issue ever!
bkw
On Wed, 17 Sep 2003, Ariel Batista wrote:
> I would like to prevent * from dialing 900 and 976 numbers. I setup the following
> settings in extensions.conf. But this does not seem to work! I don't know
6KHz != 6kbps
bkw
On Tue, 16 Sep 2003, Alex Zarubin wrote:
> Is there a way to play adpcm, 6KHz in asterisk? If yes, where can we get
> this codec?
>
> Thank you.
>
> Alex Zarubin
>
>
>
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Don't forget nufone can do outbound for 2.9 cents a min also.
bkw
On Mon, 15 Sep 2003, Steve Haehnichen wrote:
> -=> On Tue, 16 Sep 2003 00:07:39 +0200, Michael Koehler <[EMAIL PROTECTED]> said:
>
> > You get a Budgetone for free at Nikotel if you charge your account
> > there with 100 bucks. Th
blem with wcfxs and wcfxo drivers was resolved by loading
> in a specific order, which i find strange...
>
>
>
> On Sun, Sep 14, 2003 at 11:46:41AM -0500, Brian West wrote:
> > I never has problems with CVS I think the issues were with zaptel and
> > the order in which
I never has problems with CVS I think the issues were with zaptel and
the order in which the wcfxo and wcfxs were loaded. Not totally sure.
bkw
On Sun, 14 Sep 2003, Timothy Soos wrote:
> Hello All,
>
> There have been some reports of person(s) checking out the latest version of
> Asterisk u
Have you tried:
exten => _9,1,Dial(OH323/${EXTEN:[EMAIL PROTECTED])
bkw
On Sat, 13 Sep 2003, Michael Manousos wrote:
> Cerrajetto wrote:
> > Hello:
> >
> > I am testing Asterisk with oh323.
> >
> > My question is: can Asterisk route some calls thru a second h323 gateway (a
> > h323 <->
Accually you can MAKE * stand in the call path.
bkw
On Thu, 11 Sep 2003, Timothy Soos wrote:
> On Thursday 11 September 2003 02:26 am, John Todd wrote:
> > >Hello All,
> > >
> > >Is it possible to monitor and record a SIP to SIP call? If so, how?
> > >
> > >I gathered from some previous posts t
> issue. If they are using Asterisk is it not possible to record calls
> automatically. I have not reviews the CALEA requirements, must access be
Yes it is very possible to record calls with *. I record all in and
outbound calls.
bkw
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${CHANNEL} doesn't work because it contains their uniqueid on the end such
as SIP/111-asdf
bkw
On Thu, 11 Sep 2003, Adam Goryachev wrote:
> I use a single queue for all incoming calls, and different people login at
> different times to handle the calls, however, quite often, people forget to
> l
ory, clearly shows they made
> one and tested. Second its trivial to make one, if you see what is wrong
> in the code.
>
> Original advisory should have been posted here at the date of release,
> or announced by someone, but it wasn't... I guess some people are too
> busy, can
Also it wasn't a proven exploit. They said it "could allow an attacker to
obtain remote and unauthenticated access". And if pigs "could" fly I
would be a rich man!
bkw
>
> Read the security vulnerability. It referenced CVS as of a certain
> date. If you aren't keeping up with CVS changes, wh
Dude,
NuFone so totally ROCKS... I have yet to have any issues. The
last issue I had wasn't even related to NuFone.. but this stupid Nachi
worm nailing our routers and causing packets to be dropped. Other than
that the call quality is excellent. Customers can't tell the diffrence.
bkw
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