Rich Adamson wrote:
Looks like a couple of problems here. I don't believe the Cisco phone
handles md5, so remove that line.
As I told before, tried 3 different approaches:
1) password; md5;
2) password, no md5;
3) no password, no md5.
Only the third one worked. Trying to give SOME security, I
C F wrote:
how are you telling the cisco what the password is? TFTP?
TFTP (SIPmacaddress.cnf)
you will not see anything on * CLI unelss you do sip debug
And after sip debug I saw (among other lines):
[...]
Retransmitting #5 (NAT):
SIP/2.0 407 Proxy Authentication Required
[...]
SIP/2.0 401
Tom wrote:
What times are others seeing for the load when you reboot a phone?
About the same here, but I don't care as I never reboot my phone (about
once every month or two).
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Vicky Shrestha wrote:
I have tried a lot of things to make broadvoice work with asterisk , but I
failed each time.
I had some problems here, mainly because I was trying to use g729 and
broadvoice will only accept g711. Other than that, configuration itself
took about 10~15 minutes with some
Sys Admin wrote:
couldnt agree with u more !!
And, please, add another one to the list: PLEASE TRIM THE ^*[EMAIL PROTECTED]
MESSAGE. TIA.
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Brian Dingman wrote:
The FWD - Vonage interconnect has been down for some time now. Vonage
claimed there was a secuity issue and pulled the plug. No word when/if
it will ever be working again.
So I'm guessing that FWD - Packet8 falls into the same problem? Not
working here for a couple of
I found a thread [1] last month about the poor/crappy g729 quality on
Sipura units. Anyone noticed an improvement or the quality is still poor?
If the Sipura firmware/g729 offers no quality yet, who else is offering
a dual channel g729 ATA? I heard about Uniden, but I have no reports
about
Chris Lee wrote:
Has anyone else upgraded to 7.4 and found that the date time no
longer appears on the phone?
This problem was pointed at the SIPPhoneReleaseNotes7_4.pdf file.
What I noticed is that when the phone lost the internet connection the
date/time will no longer be present on the phone.
William Suffill wrote:
According to the small print in the bottom graphic:
http://www.sipura.com/products/spa2100.htm
The SPA 2100 would give u 2 ports + 2 RJ45 as well as 2 G729
When I was placing an online order, I found this:
support for two concurrent calls using the G.729 codec (in a
Brian West wrote:
exten = _8001133[12345789]XX.,1,Dial(SIP/france-gateway,60,tr)
or
exten = _8001133[1-57-9]XX.,1,Dial(SIP/france-gateway,60,tr)
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Dave Green wrote:
Following a top posted thread is a pain.
not trimming the useless part of a reply is another pain...
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I'm trying to find some live examples on how to use the h, H and g
parameters on the dial command
(http://www.voip-info.org/wiki-Asterisk+cmd+dial)
Any ideas? I was testing with the code below but after pressing *
nothing happens (only after a long pause the goodye file was played)
[testset]
Tzafrir Cohen wrote:
BTW: did I mention that we have binary packages for standard Debian
Sarge kernels in our apt source?
zaptel is the only package that never worked for me from apt-get. I need
to download, compile and install the kernel (specially because the
original debian install is pre
Matthew Boehm wrote:
[...] In the meantime, get a Sipura 2100, supports 2 729 calls and
has both WAN/LAN ports.
I was told that the Uniden DTA200 also supports 2 g729 calls. I'm buying
one to test. Street price around US$ 90.
Another one with dual g729 channels is MTA V102. Street price US$ 100.
Roger Schreiter wrote:
But when dialing a number, I get:
Feb 2 09:44:45 NOTICE[20380]: chan_sip.c:7296 handle_request: Unable to
create/find channel
After I installed my Digium g729 license, I'm trying to place a call
from my Cisco 7960 and I'm receiving the same error:
Feb 19 09:47:06
Olaf Klein wrote:
Why not just kill yourself, fucking wannabe spammer? DIE DIE DIE
This is *REALLY* offtopic, but Isamar is the founder of Brazilian
AntiSPAM - http://antispam.org.br/ and later http://spambr.org/
Does it matter here? I don't think so, but calling he (or even me) a
spammer is
Max wrote:
Pessoal estou querendo montar um servidor SIP para fazer testes [...]
wrong list. For Portuguese mailing list please subscribe to
http://groups.yahoo.com/group/asteriskbr/
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Matthew Boehm wrote:
Is there a way for asterisk to notify you of this? Send an email? Send a
page? Call you?
Nagios (I believe now is called NetSaint) can do this and much more.
But you must have the power to configure it... after that, Nagios can
send you an email, a pager, even call you and
Paul A Brown wrote:
Anyone had a Cisco 7970 working with Asterisk?
As 7970 uses SCCP, you can do it with asterisk. I did it with 7960.
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I'm trying to place a call from my Cisco 7960 and I'm receiving this error:
Mar 1 06:19:44 NOTICE[3060]: chan_sip.c:7399 handle_request: Unable to
create/find channel
Mar 1 06:19:58 NOTICE[3060]: chan_sip.c:7399 handle_request: Unable to
create/find channel
I can't place calls, but I can
Guy Decarpentrie wrote:
Try to configure your Cisco type=friend in your sip.conf
It is already type=friend
[1234]
type=friend
username=1234
auth=md5
secret=supersecret
deny=0.0.0.0/0.0.0.0
permit=my_ip/255.255.255.255
canreinvite=no
reinvite=no
host=dynamic
dtmfmode=rfc2833
qualify=1800
Christian faucher wrote:
I read that, using a modem,I can use a standard phone line, and
convert that as input for Asterisk PBX, right?
Not that simple, not every modem, but yes.
Also, where can I get VOIP phones?
eBay
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After fighting with a Unable to create/find channel [1] [2], I gave up
on my previous installation and rebuild my asterisk from CVS-Head. I
guess the Debian package available today is broken somewhere (after a
previous broken release made with an old libpri package), but now I'm
having another
My Cisco 7960 is working well with * using SCCP, but I want to change it
to SIP.
Can anyone here help me on how/where I can buy a SIP image? I contacted a
few Cisco partners in the US and some replied will not sell 1 copy/can't
handle a small contract and others ignored me.
Thanks, Hermann
On Wed, 25 Feb 2004, Ernest W. Lessenger wrote:
With recent CVS builds I've been able to specify 7000 and 7001-7200 as the
call parking lot. I haven't tried any other numbers.
The parking lot is assigned by the user or by the system?
I found that my * is assigning 'lot' 701 for my parked calls
I'm running * with a 7960/Skinny. I'm seeing several pages with
SIPDefault.cnf config file, but as I'm not running SIP for this phone
(yet), it is useless now.. Is there any Skinny/SCCP Default.cnf also?
Actually, I'm trying to enable the extra 7960 features, like directory,
services etc...
On Thu, 26 Feb 2004, Jeremy Jones wrote:
I _think_ my problem has to do with the Dial Plan settings on the SIP
configuration page. Anyone familiar with these things? By default, the
dial plan setting reads: 1xx|x.T.
This is my dialplan for Packet8 / 8x8:
exten =
Anyone here with experience on the Cisco ATA 188 and *?
Is it as good as ATA 186?
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I can't start *. I'm receiving the following error:
[chan_zap.so] = (Zapata Telephony w/PRI)
== Parsing '/etc/asterisk/zapata.conf': Found
Feb 29 19:01:46 WARNING[1024]: chan_zap.c:673 zt_open: Unable to specify channel 1: No
such device or address
Feb 29 19:01:46 ERROR[1024]: chan_zap.c:5324
On Thu, 5 Feb 2004, Tim Sailer wrote:
Does anyone have the zaptel modules built for Debian 2.4.24 kernel?
Someone here is running * on debian?
I tried to follow every howto page I found, but all ended with the same
problem:
Mar 3 02:21:51 WARNING[16384]: chan_zap.c:673 zt_open: Unable to
On Thu, 5 Feb 2004, Tim Sailer wrote:
Does anyone have the zaptel modules built for Debian 2.4.24 kernel?
After trying and trying to compile and make Asterisk run on a Debian
box, I gave up and picked another HD with RH 9 on it. No headaches. Only 1
build was necessary to build and run *.
I
I'm trying to deploy an IVR with 12 analog lines (using VoiceTronix
OpenSwitch 12 - http://www.voicetronix.com/hda.htm) and 5 SIP phones to
handle these calls - if IVR prompts weren't enough to help the customers.
I'm still researching, nothing decided so far.
Which should be the 'best' hardware
My ZAP calls are being dropped after 7 seconds. The only info I can find
is:
Mar 12 14:03:08 WARNING[98311]: Maximum retries exceeded on call
[EMAIL PROTECTED] for seqno 102 (Response)
Mar 12 14:03:11 WARNING[98311]: Maximum retries exceeded on call
[EMAIL PROTECTED] for seqno 102 (Request)
I
On Mon, 15 Mar 2004, Joshua McAdam wrote:
So far I have managed to upgrade the firmware, but I am not sure what the
cfg.txt should contain as I have tried a few I found from searches of the
list and also on the wiki.
I found this:
On Sat, 13 Mar 2004, Brian Buhrow wrote:
address which contains letters -- do I pretend I'm dialing a name and use
the numbers associated with the letters of the MAC address? When I try to
do this, it doesn't reset, and tells me my numbers are invalid.
Any suggestions on how to restore
On Mon, 15 Mar 2004, stan wrote:
Is anyone using a 3com 3CNJPSE to power a 7960G?
I have a couple of 7960Gs and 3CNJPSEs but no combination appears to
work. Both phones work fine with a cisco power cube. I get a 47.6V
reading across pins 5 and 8 on the patch cable coming from the 3CNJPSE.
I
On Mon, 15 Mar 2004, Matthew Marlowe wrote:
I can confirm 1.0.4.53 is bad as well. :) 1.0.4.50 has been working
fine for me.
I received the 1.0.4.54 firmware. So far, so good. No new problems.
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On Mon, 15 Mar 2004, stan wrote:
Is anyone using a 3com 3CNJPSE to power a 7960G?
Forgot to mention that I also have a a 7960G and I tried to use a Compaq
PoE (http://www.compaq.ca/english/business/mobile/wireless/poe.asp) and a
3com Network Jack NJ100
On Mon, 15 Mar 2004, Matthew Marlowe wrote:
Out of everyone using the 7960 currently, what would you say is the best
firmware to use w/ asterisk?
I'm using SIP 6.2.00.
What's the most compatible / stable?
Cisco is not like Grandstream. Grandstream released an image that didn't
work
On Sat, 5 Jun 2004, Tony Hoyle wrote:
Also, what is the code of the $8 support option and who sells it (it
seems cisco don't sell direct to end users)? The cheapest I've seen is
$100 and if it's that kind of price I'll just see how far I can get with
the default firmware.
Search the list.
On Tue, 8 Jun 2004, Chris wrote:
I'm trying to build an IVRs. anyone here can
spare a sample extensions.conf? or maybe
a link.
http://www.voip-info.org/wiki-Asterisk+tips+ivr+menu
http://www.voip-info.org/tiki-index.php?page=Asterisk+config+extensions.conf
On Tue, 8 Jun 2004, Chris wrote:
I'm trying to build an IVRs. anyone here can
spare a sample extensions.conf? or maybe
a link.
I found the example I think is one of the best to learn about IVR:
http://www.voip-info.org/wiki-Asterisk+Telemarketer+Torture
On Sat, 12 Jun 2004, Jacob Hunter wrote:
I have a list of all my local prefixes(free) on my POTS. Is there a
way to integrate that so * decides
if it is going to use iax or POTS? There is about 60 prefixes..
1831-XXX
If I understood what you are asking, just do this:
; your first prefix
On Mon, 14 Jun 2004, Shoval Tomer wrote:
Can I use a Wildcard X100P to connect an outgoing line jack (on the
Panasonic) to Asterisk, so I can route calls from the PBX to Asterisk,
and calls from Asterisk to the PBX?
I have an * under a Panasonic KX-TD816, as an extension for Panasonic,
On Wed, 16 Jun 2004, Nicholas Bachmann wrote:
You might try reading http://www.caliburn.nl/topposting.html -- it
explains why people don't like top posting.
Or read this quote:
A: Because we read from top to bottom, left to right.
Q: Why should i start my reply below the quoted text?
- --
On Tue, 15 Jun 2004, Lars Boegild Thomsen wrote:
Since only one of the asterisk servers are on a known IP, the two
systems on dymanic IP registers at the one in Europe.
Just one question: is there any reason not to use a dyndns name for these
two dynamic boxes? I believe they are PPPoE xDSL, so
On Thu, 17 Jun 2004, [ISO-8859-15] Robin Calmegård Siurua wrote:
I can't compile Asterisk on a Debian machine.
What is wrong? :/
debian... :-(
I was only able to compile asterisk when I gave up on doing it by myself
and decided to use the debian package (.deb).
Do it via apt-get. Remember to
I followed the instructions at http://www.opencall.org/instructions.html
and
http://lists.digium.com/pipermail/asterisk-users/2003-October/025094.html
I was able to compile spandsp (./configure ; make ; make install),
manually patched asterisk apps/Makefile (/usr/src/asterisk/apps), as the
On Tue, 22 Jun 2004, Dean Collins wrote:
Hi Brian, I have been using a X100P to Packet8 ATA connection for about
3 months, it works fine apart from needing the occasional reset.
Working well here also, 2 Packet8 ATAs and no reset necessary so far.
___
On Sun, 27 Jun 2004, Edwig Knol wrote:
Only No content
http://web.voiphk.net/ appears to be back up now.
The site is not Mozilla-friendly. If you are using a Mozilla-based
browser, point it to http://web.voiphk.net/ instead of
http://www.voiphk.net/
On Mon, 2004-06-28 at 02:02, Samantha (Femtech) wrote:
Is there a cron that I con do to replace this, as the fx0 card doesnt
hang up properly
I had the same problem here, and fixed within zapata.conf by adding
these lines:
busydetect=1
busycount=5
Try reading this also:
On Thu, 1 Jul 2004, chouck wrote:
First being, how do I setup asterisk to point to another asterisk server
and make all the lines which should be PSTN or POTS go directly to
another existing asterisk server by using accounts?
You are looking for IAX - Inter Asterisk eXchange.
Also, Is there
On Fri, 9 Jul 2004, Greg Boehnlein wrote:
[...] who is responsible for the Debian packages?
I believe the responsible is Mark Purcell = msp at debian dot org
I sent an email last week and received no reply so far...
asterisk*CLI show version
Asterisk 0.7.2 built by msp at dell dot purcell dot
On Mon, 12 Jul 2004, AsteriskList wrote:
what does the command NOTRANSFER in IAX.CONF?
where do i find asterisk´s commands?
usually do a google search...
About notransfer:
http://www.mail-archive.com/[EMAIL PROTECTED]/msg42262.html
That prevents IAX from transferring call to remote Asterisk,
How I can retrieve the original ${EXTEN} value when falling into the
exten = i,1,whatever context??
I'm trying to implement this extension rule:
exten = i,1,NoOp(${EXTEN})
exten = i,2,Wait(1)
exten = i,3,Playback(vm-extension)
exten = i,4,SayDigits(${EXTEN})
exten = i,5,Playback(is-invalid)
On Wed, 14 Jul 2004, xfastjackx wrote:
I will receive my CISCO 7960G tomorrow. [...] so could please someone
send me the SIP-firmware?
Search the list:
http://www.google.com/search?hl=enlr=ie=UTF-8q=site%3Alists.digium.com+cisco+firmware+sip
--
Lista asterisk em portugues:
Steve Kann wrote:
[...] I've gotten 270 already: [...]
I've got only 1. But... what is the main issue now? Is this topic just
another (endless) troll or someone is trying to get some config help for *?
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Damon Estep wrote:
[...] Contains a link you need for firmware.
Correct URL is http://www.voip-info.org/wiki-Asterisk+phone+cisco+ATA18x
URL:http://www.voip-info.org/wiki-Asterisk+phone+cisco+ATA18x
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nkb wrote:
I was wondering if I could use IXAy to forward my call via the internet
to my destination, something of similar function to SIPURA 3000?
The IAXy is similar to the Sipura 1000 or 2000, or the Cisco ATA 18x...
You can use it to connect to a VoIP server with the IAX2 protocol
(instead
nkb wrote:
So, do I still need to have an Asterisk server connected to my IAXy even
after I've made provision for it?
You can only connect IAXy to an asterisk server. Yours or from a VoIP
provider.
Like, can I just carry this IAXy
around(after provision) and just plug into any broadband
Joseph wrote:
After somebody records a message asterisk notifies me and encloses the
WAV file. Though I'm not sure if this is a WAV format. I can not play
it.
How to play received message?
Did you try to use Windows Merdia Player?
In other hand, if you are receiving a .GSM file, you can use the
I noticed that I'm no longer able to receive calls from PSTN to my
SipGate DID number.
I changed the sip.conf and extension.conf as per
http://www.voip-info.org/tiki-index.php?page=Sipgate but the problem
remains...
However, I can receive calls from another sipgate user. The problem is
only
Federico Gonzalez wrote:
I have an Asterisk with one local Cisco ATA and one remote Cisco ATA
connected to the Asterisk, the remore connection is a satellite link
with an 900ms delay.
This is the same delay I have here. Never less than 900, sometimes over
1500 ms.
Check
Michael Graves wrote:
[...] Although there have
been a few (very few) times when I've notcied a brief pause after
dialing and found that it had in fact dialed out on the last possible
option.
[...]
The problem of your approach is that if you are out of credit with the
first provider, your call
Eric Wieling aka ManxPower wrote:
What company are you using for your service?
Intelsat. But I'm not using it point-to-point as I'm not the primary
contractor of this channel - I'm buying internet access.
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Is there any winblows softphone available offering g729 *and* IAX?
I couldn't find any http://www.voip-info.org/wiki-VOIP+Phones
The best choice should be dIAX, but it is only GSM.
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On Thu, 25 Mar 2004, Anton Tinchev wrote:
Some troubles with dtmf sending.
I tested here (I'm preparing a report to send to support at diamondcard
dot us) and I found that they only support dtmfmode=info. Before I was
using dtmfmode=rfc2833. Using a Cisco 7960G phone. I don't know if this
On Mon, 29 Mar 2004, Iain Stevenson wrote:
- cheap option - Grandstream Budgetone - works well but current firmware is
buggy
Which one? I'm running one the latest image available at
http://www.grandstream.com/BETATEST/ (b14p4.54.zip) and my * and my GS are
working OK.
The 4.53 was buggy, but I
On Fri, 2 Apr 2004, Joshua Colp wrote:
http://join.nufone.net/ has some information, but it's under development.
You can signup through there though.
After filling all the information, I got an error message:
Something is seriously wrong
Column 'emailAddress' cannot be null
Can someone
I found some old messages regarding a possible pkt8 DTA bypass. Anyone
is using Packet8 with Asterisk?
==
http://www.dslreports.com/forum/remark,7292324~root=voip~mode=flat
Got Softphone Working with Packet8
Friendly name: {Anything you'd like.}
SIP domain: packet8.net
SIP proxy:
I'm trying to add 2 FXO cards but when the second is added, asterisk stop
to respond (zap channel):
app_dial.c:545 dial_exec: Unable to create channel of type 'Zap'
modprobe zaptel and modprobe wcfxo didn't return any error.
ztcfg -vv is reporting only 1 card:
Zaptel Configuration
On Fri, 9 Apr 2004, Stephen Karrington wrote:
I can't read German. Can you outline the cost for me? Thanks.
http://www02.sipgate.de/user/tarife.php
Tarife NATIONAL
Deutschland* - E 1,79Ct/Min (US$ 0.021637)
Cellular: E 22,90Ct/Min (US$ 0.276815)
They have a plan which includes 1000
On Mon, 12 Apr 2004, Mark Phillips wrote:
I tried,
exten = s,1,Zapateller(answer|nocallerid)
exten = s,2,Privacymanager
exten = s,3,Dial(a bunch of SIP extensions)
But then every call was answered regardless of CID and the tones were heard.
I tried the sample I found at:
On Wed, 14 Apr 2004, Markus Mayer wrote:
The 1st box shows the original callerID, on the 2nd box callerID shows
the callerID of the 1st box. Apparently the 1st Asterisk box replaces
the original callerID with its own.
Try this one:
On Mon, 19 Apr 2004, Michael Welter wrote:
Every half hour I get -- MARK -- in the syslog. Is this normal behavior?
This has nothing to be with asterisk, but with your linux installation.
Yes, it is a normal behavior and it is harmless... It is just a half hour
stamp to your syslog...
On Wed, 21 Apr 2004, Altus Snyman wrote:
Do you have a copy for me,the page seems to be closed and it redirect me
to http://swpat.ffii.org/ and I cant read that
http://www.linphone.org/linphone.php
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On Thu, 22 Apr 2004, Paul Tyreman wrote:
I am guessing the phone that I get won't come with that as it was used
with the cisco call manager software in the past. Can I still use this
phone with Asterisk, or have I waited my money ?
Every Cisco software embedded with their hardware is valid
On Fri, 23 Apr 2004, Johnson-Perkins, Robert wrote:
I have just got 3 Cisco 7960 phones which I would like to connect to
Asterisk...
However they seem to have v3 SCCP firmware.
The same question, posted a few hours before:
http://lists.digium.com/pipermail/asterisk-users/2004-April/044025.html
On Fri, 23 Apr 2004, Paul Tyreman wrote:
What website do I have to go to in order to buy a SIP image update ?
When I bought mine, I did a Google search on their part number:
SW-SM-UL-7960 (Cisco SIP license for 7960 IP Phone)
Also, read this message:
On Fri, 23 Apr 2004, Paul Tyreman wrote:
So are you telling me that to be legal, I need to pay $105, but could
get away with $8 ?
*IF* your phone qualifies for service contract (which is US$ 8), yes.
You still will have an illegal copy, and you can also be charged later for
all the software you
On Fri, 23 Apr 2004, Mark Olliver wrote:
I seam to have a problem working out how to get my X100P to answer after
1 ring. Currently it is working fine and connects to the switchboard
menu correctly but just does it after 4 rings, which I would prefer if
we could reduce.
Try this:
On Fri, 23 Apr 2004, Paul Tyreman wrote:
I have three questions to ask about this:
1) How do I know if my phone qualifies for a service contrct ?
When you (try to) buy your service contract, you will need to give the
model and serial number of the item you are trying to include into your
On Fri, 23 Apr 2004, Rubens Zupelli Filho wrote:
I'm using a Debian kernel 2.4.22 with all (second the archives/docs)
pre-reqs packages.
Quick and dirty: instead of trying to compile/build, just do this:
apt-get -t sarge install asterisk/unstable
Remember to have the appropriate entry for
Sorry to ask this here but I believe that it is the best place to receive
a feedback...
I would like to know if anyone is using SNOM 100 / SNOM 200 phones with *,
and the overall impression about these phones...
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On Mon, 10 May 2004, Administrator wrote:
i have found a webmin module on the astersik ftp server!
It is broken. Forget it.
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On Fri, 30 Apr 2004, Mark Elkins wrote:
Looking at pbx.c - I'm not sure if I should change the end time (ie
midnight) to either 23:59 -or- 00:00.
it is 23:59
23:59 will work - but what happens to calls then between 23:59 and
midnight?
23:59'59 is still 23:59 mainly because you are not
Is it possible to strip some numbers from the *end* of a number?
I know that ${EXTEN:1} will remove 1 position from the beggining... but
how to remove N numbers from the end?
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On Thu, 27 May 2004, Fabio Donaggio wrote:
[EMAIL PROTECTED] src]# cvs login
-bash: cvs: command not found
Anyone can help me??
Yes. First of all, you need to install CVS.
http://www.fluidthoughts.com/howto/cvs/install/
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On Thu, 27 May 2004, Chris Sullivan wrote:
Your rates page still has no useful information. Can you guys at least
put some rates up somewhere?
They have their rates available at
http://www.diamondcard.us/exec/voip-rep-acc-type?secRel=/secondary/corporatepriRel=/templatessecId=corporate
It is
On Thu, 27 May 2004, Tony Mountifield wrote:
No more SIP, No more IAX. It was a damn good IAX client... too bad its crap
now.
Are you sure?
http://www.virbiage.com/firefly/download/ still says the following:
[...]
I just download the latest version (1.7 Build 3532) and they are no
On Thu, 27 May 2004, Harry Flink wrote:
www.cvshome.org is home for CVS but the site is currently down.
Is down due to security issues:
. TA04-147A - CVS Heap Overflow Vulnerability (US-CERT)
http://www.us-cert.gov/cas/techalerts/TA04-147A.html
. Advisory 07/2004 - CVS remote vulnerability
On Sat, 29 May 2004, Bartek Kania wrote:
I am having problems with the PlayTones application and VoIP softphones.
I have the following in my extensions.conf:
exten = 123,1,Answer
exten = 123,2,PlayTones(Busy)
exten = 123,3,Hangup
But when I connect with gnophone(IAX) or
On Sun, 30 May 2004, Isamar Maia wrote:
From where can I download the portuguese sounds?
For those who speak Portuguese, please join a list I just created to
discuss the Portuguese prompts. I believe there are enough people here to
start this task.
Group home page:
On Sat, 6 Nov 2004, Damon Estep wrote:
Alternatively, if you think it can be done by someone with the right
skills let me know where the best forum to post a bounty would be.
http://www.voip-info.org/tiki-index.php?page=Asterisk+bounty
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Arun Kumar wrote:
I've upgraded my server to asterisk-1.2.22 from 1.2.10 after that my
DeadAGI scripts are not working properly. Like after hangup I used to do
some more work now its not working.
Try, at your own risk, this:
Jeng Yu wrote:
I would like to hear if anyone out there in Asteriskland has used the
Dock-N-Talk (DNT) box to connect cell phones to Asterisk box.
The only problem I noticed is that after a random amount of time the box
will lost contact/synch with the cell phone. I'm using DockNTalk for
about
Julian Lyndon-Smith wrote:
however, I get no errors, but still get the default Allison sounds
for the digits. Anyone got any clues on what I'm doing wrong ?
1) Create a directory named your_country_iso_code (AR|MX|ES|ETC) [1]
under the main sounds directory (/var/lib/asterisk/sounds/ ???);
Crazy Boy wrote:
If IPhone is released in India, Can you tell me any Apple authorized
showroom in Hyderabad (Andhrapradesh, India)?
Oh gosh... another troll... Google IS your friend:
http://www.google.com/search?q=apple+iphone
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--Bandwidth and
Matt Brown wrote:
Does anyone have any experience with a GSM card, preferably Quad Span
(4 GSM modules or higher) for use in the UK. I have seen the
Junghanns* version but I am not keen on the limitation of having to
use a BriStuffed version of Asterisk.
I'm buying this one to test:
Michelle Dupuis wrote:
We're looking at a large wifi phone deployment, and we're looking for
wifi phones that:
1. Are SIP compliant (Asterisk friendly)
2. Provision capable (ideally TFTP of own MAC address)
3. Industrial quality (no cheap plastic stuff).
4. Well documented (and none of
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