Ron Wellsted wrote:
What route is left for guy with a few Cisco phones in Europe?
Piracy?
I looked around for nearly a year for a contract after a kind soul got
me the images (the closest I got was a site in the US who were prepared
to sell me the CON-SNT-CP7960 for £8 ... with £150
Bob Goddard wrote:
It doesn't arrive. It's all done instantly via email.
There's a whole package apparently (hence the £150 postage I was quoted,
although I suspect they just weren't interested in selling).
Even the entry on voip-info.org says it takes two weeks... Once you buy
it the request
Henry Devito wrote:
If you call Cisco contract support. 1-800-447-9347 and give them the
serial number used when you purchased the smartnet they will give you
the contract number over the phone. If the contract was sold properly
No serial number was asked for.. I just explained that I just
Hi,
I've been trying to setup SMS on asterisk - would be useful to have for
things like server outages, email from important customers, etc.
I can send SMS with no issues, although I have to send it over the Zap
line.. none of the VOIP providers will route the call. It arrives on my
mobile
Wilson Pickett wrote:
You are sending the extra digit to say which mailbox the message is
for, right? In this country, if you do not send that digit, it will
try to vocalize the message during the calls.
I'm dialling 17094009, as instructed in the BT documentation. Where
does the extra digit go?
Wilson Pickett wrote:
Quoting the wiki at http://www.voip-info.org/wiki-Asterisk+cmd+Sms
appended to the end. The telco can define a default sub address (9 in
the UK) which is used when the extra digit is not appended to the end.
It says there's a default anyway. Note smsq doesn't send one (I
Tony Hoyle wrote:
response). Has anyone got this working in the UK? Do I have to set a
country specific setting?
OK I got it working... there's a timeout in app_sms.c that just isn't
long enough for the BT implementation - the app gives up long before the
message centre has had time to respond
Eric Rees wrote:
MemTotal: 2074808 kB
MemFree:417420 kB
Buffers: 39396 kB
Cached:1547124 kB
SwapCached: 0 kB
Active: 471180 kB
That's a total memory usage for the entire OS of only 107MB:
(Total-Free)-Cached.
Tony
Paul wrote:
I recently hooked up my sipura IP phone and set it up as an SIP device to
POTS line. After approximately 1 minute, the quality turns horrible and the
person can no longer hear me, but I can faintly here them. There is a lot of
static on the line, it almost sounds like an electronic
Eric Wieling wrote:
Do you have a SIPura?
Not any more.. don't like them. I had two duffs in a row and haven't
taken the risk since.
They didn't used to come with documentation - an A5 sheet on how to
enter the sip gateway, that was it. The rest was down to guesswork,
which is probably how
used has not required 3a-3d. It
looks like a real hack to do so.
It anyone working on implementing this?
Tony
--
All your code belongs to Santa
Tony Hoyle [EMAIL PROTECTED] Key ID: 104D/4F4B6917 2003-09-13
Fingerprint: 063C AFB4 3026 F724 0AA2 02B8 E547 470E 4F4B 6917
Phone(FWD): (0845 004 5566
genuinely surprised anyone can survive without this feature.
Tony
--
All your code belongs to Santa
Tony Hoyle [EMAIL PROTECTED] Key ID: 104D/4F4B6917 2003-09-13
Fingerprint: 063C AFB4 3026 F724 0AA2 02B8 E547 470E 4F4B 6917
Phone(FWD): (0845 004 5566) 413300
in aure.
Tony Hoyle [EMAIL PROTECTED] Key ID: 104D/4F4B6917 2003-09-13
Fingerprint: 063C AFB4 3026 F724 0AA2 02B8 E547 470E 4F4B 6917
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE
sapientum fixa est in aure.
Tony Hoyle [EMAIL PROTECTED] Key ID: 104D/4F4B6917 2003-09-13
Fingerprint: 063C AFB4 3026 F724 0AA2 02B8 E547 470E 4F4B 6917
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk
Adam Goryachev wrote:
Plus consultative transfer calls
Well yes... pbx's job though (usually, although apparently not always...).
Plus speaker phone
No allowed to use them as they disturb people working.
Plus conferencing
We have a conferencing phone which is a huge triangular thing with lots
of
Nik Martin wrote:
You really should be using analog phones with asterisk. You'll be a
hero to your boss, because the phones wont cost a pile. All you need to
Since we already have the phones, they won't cost anything :)
add is a channel bank for the analog phones, and asterisk. There is a
Eric Wieling wrote:
Why are you even looking at VoIP? Analog ports and phones are pretty
cheap. They are not pretty, but they are cheap and all the smarts are
in the PBX.
Free calls to the US, basically, since the leased line is dirt cheap to
run. ie. the purpose of the exercise is to save
.
I'd rather use X-Lite than MSN Messenger...
The advantage that MSN Messenger has is that it's installed on every
machine (whether you like it or not!) so it's a zero effort (and zero
cost) fallback.
Tony
--
All your code belongs to Santa
Tony Hoyle [EMAIL PROTECTED] Key ID: 104D/4F4B6917 2003-09
Philipp von Klitzing wrote:
So also everyone has a headset plus a soundcard, I assume? ;-
Well everyone has a soundcard... There's a load of headsets hanging
around the office that get used at various times.
Tony
--
All your code belongs to Santa
Tony Hoyle [EMAIL PROTECTED] Key ID: 104D
at it).
All companies seem to suffer more or less from bad management - there's
not much that those of us on the coalface can do about it.
Tony
--
All your code belongs to Santa
Tony Hoyle [EMAIL PROTECTED] Key ID: 104D/4F4B6917 2003-09-13
Fingerprint: 063C AFB4 3026 F724 0AA2 02B8 E547 470E 4F4B 6917
[EMAIL PROTECTED] wrote:
I hear the exact same noise on 2 units I purchased a few months ago.
I've been in contact with sipura support and they are willing to try
RMA'ing one of my units.
As soon as I can get to the site with the sipura, I'll be sending it in.
I'll post my results to the list.
Kevin Walsh wrote:
Try playing with the FXS Port Impedance in the Regional settings.
Mine is set to 600. I imagine that a mismatch here could cause that
sort of thing.
I've just been through them all... I can change its properties (some
have a lot of hum, some have a lot of hiss).
The true
I've just managed to get hold of a cisco 7940, which looks nice but I'm
unable to make it actually do anthing...!
All the online manuals say things like see your network administrator
which isn't a whole lot of use.
First thing I think I need to do is work out how to set the TFTP server
IP as it's
Storer, Darren wrote:
http://tinyurl.com/37fe4
Unfortunately those instructions don't seem to relate to
my phone (eg. there's no option 6 on the 'Settings' menu).
I've found some other documents which seem to help but am unable to
change any of the settings even in the unlocked state - it all
Hermann Wecke wrote:
Search the list. Look for sip 7960 firmware.
http://google.com/
sip 7960 firmware site:lists.digium.com
That's no help.. read all of them. The best I can find out is the $8
price on the wiki is bogus and should be removed as it's misleading.
The cheapest smartnet is
but never thought they'd do it with a phone... luckily I've found there
are other ways to get hold of the firmware.
Tony
--
All your code belongs to Santa
Tony Hoyle [EMAIL PROTECTED] Key ID: 104D/4F4B6917 2003-09-13
Fingerprint: 063C AFB4 3026 F724 0AA2 02B8 E547 470E 4F4B 6917
Phone(FWD): (0845
.
Tony
--
All your code belongs to Santa
Tony Hoyle [EMAIL PROTECTED] Key ID: 104D/4F4B6917 2003-09-13
Fingerprint: 063C AFB4 3026 F724 0AA2 02B8 E547 470E 4F4B 6917
Phone(FWD): (0845 004 5566) 413300
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http
shoulders (for now).
Tony
--
All your code belongs to Santa
Tony Hoyle [EMAIL PROTECTED] Key ID: 104D/4F4B6917 2003-09-13
Fingerprint: 063C AFB4 3026 F724 0AA2 02B8 E547 470E 4F4B 6917
FWD: 413300
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http
none of my messages are arriving on the list... just testing.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
. The second one
needs an extra field for the authuser - when I try to dial out I just get:
May 17 01:03:45 NOTICE[1110916016]: chan_sip.c:5059 handle_response:
Failed to
authenticate on INVITE to 'Tony Hoyle
sip:[EMAIL PROTECTED];tag=as4afae981'
I think this means it's using the wrong username
. The second one
needs an extra field for the authuser - when I try to dial out I just get:
May 17 01:03:45 NOTICE[1110916016]: chan_sip.c:5059 handle_response:
Failed to
authenticate on INVITE to 'Tony Hoyle
sip:[EMAIL PROTECTED];tag=as4afae981'
I think this means it's using the wrong username somewhere
in case the debian
package was
broken.
I still get the error:
May 17 23:20:27 NOTICE[1110916016]: chan_sip.c:5059 handle_response:
Failed to
authenticate on INVITE to 'Tony Hoyle
sip:[EMAIL PROTECTED];tag=as5c348356'
Relevant chunks here of data are:
[pipecall]
type=peer
secret=
username
tmpm wrote:
Youre making it now..
Sorry... I actually didn't expect it to work.
My first resend from yesterday came through (twice) but my second
one doesn't (including the output from sip debug) - it seems the list
quietly drops long messages (14K in this case). I put it on
Manuel Wenger wrote:
Hi Tony,
Try adding fromuser=x, maybe username= isn't enough... Just a guess, it
already solved a few problems for me.
I've tried fromuser=, username= and some fromdomain= combinations -
unfortunately I'm not 100% sure what they change, and the error message
stays
I'm finding I can't run two festival commands in the same connection. Given
the following:
exten = 555,1,Answer
exten = 555,2,Wait(1)
exten = 555,3,Festival(mary had a little lamb)
exten = 555,4,Wait(1)
exten = 555,5,Festival(she also had a duck)
exten = 555,6,Hangup
Calling 555 gets the first
if the extensions had a
netmask/allowable IP setting like the iax.conf file uses, but there isn't one
documented...
Tony
--
Te audire no possum. Musa sapientum fixa est in aure.
Tony Hoyle [EMAIL PROTECTED] Key ID: 104D/4F4B6917 2003-09-13
Fingerprint: 063C AFB4 3026 F724 0AA2 02B8 E547 470E 4F4B
on to state all 3 of those manufactures no longer support it.
I wonder if the low cost geographic VOIP numbers support it?
Tony
--
Te audire no possum. Musa sapientum fixa est in aure.
Tony Hoyle [EMAIL PROTECTED] Key ID: 104D/4F4B6917 2003-09-13
Fingerprint: 063C AFB4 3026 F724 0AA2 02B8 E547 470E 4F4B
passwords... the recommendation always
seems to be make username==extension number, though.
Tony
--
Te audire no possum. Musa sapientum fixa est in aure.
Tony Hoyle [EMAIL PROTECTED] Key ID: 104D/4F4B6917 2003-09-13
Fingerprint: 063C AFB4 3026 F724 0AA2 02B8 E547 470E 4F4B 6917
is in their database, it calls
the VOIP number directly, otherwise it calls the POTS number
It's an interesting idea. Of course having a huge database of
names/addresses/phone numbers can be quite lucrative too.
Tony
--
Te audire no possum. Musa sapientum fixa est in aure.
Tony Hoyle [EMAIL
it myself (it means
trusting an external database to produce a least cost route.. I'm just not
that trusting).
Tony
--
Te audire no possum. Musa sapientum fixa est in aure.
Tony Hoyle [EMAIL PROTECTED] Key ID: 104D/4F4B6917 2003-09-13
Fingerprint: 063C AFB4 3026 F724 0AA2 02B8 E547 470E 4F4B 6917
the 'local' target this afternoon so I have everything copied/pasted).
Tony
--
Te audire no possum. Musa sapientum fixa est in aure.
Tony Hoyle [EMAIL PROTECTED] Key ID: 104D/4F4B6917 2003-09-13
Fingerprint: 063C AFB4 3026 F724 0AA2 02B8 E547 470E 4F4B 6917
Billy Huddleston wrote:
'local' target? What's that?
http://www.voip-info.org/wiki-Asterisk+local+channels
It's like a subroutine, so you can use it to call bits of the dial plan
that get repeated a lot, like dialing FWD after first setting the caller ID.
(AFAIK anyway... not tried to get them
Dean Collins wrote:
Hi Tony, it is the same duane - lol you are hardly allowing it to
perform least cost routing, it just does one check for ip to ip call
then drops back to whatever you have written on your asterisk.
So eg. if I've registered 3 different sip providers and an IAX provider,
plus a
Duane wrote:
If there is an IAX2 or SIP or H323 NAPTR record in DNS, this increments
the dial plan by +1, if it's a TEL (i.e. the talking clock in china) it
increments by +51, and increments by +101 if it fails, so unless you
tell it to dial the number at +51 it won't use any TEL fields from
Karl Dyson wrote:
Well, I have a USR015630B, which, according to the FAQ supports (UK)
CLI. It supports the at #cli command, but no matter what I try, it will
not pick up the caller id. Lucky I already had it and didn't buy it
soley for this purpose! My caller display unit (unfortunately a CD60 --
(to see if you somehow see the pulse that comes just before
the ring - I always wondered why phones tended to blip a second before
starting to ring these days.. now I know).
Tony
--
Te audire no possum. Musa sapientum fixa est in aure.
Tony Hoyle [EMAIL PROTECTED] Key ID: 104D/4F4B6917 2003-09-13
no possum. Musa sapientum fixa est in aure.
Tony Hoyle [EMAIL PROTECTED] Key ID: 104D/4F4B6917 2003-09-13
Fingerprint: 063C AFB4 3026 F724 0AA2 02B8 E547 470E 4F4B 6917
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman
Tony Hoyle wrote:
The CID specs look really simple... I'll definately have a go at
implementing something like it. Just need to find a spec sheet for the
Intel chipset in the FX100/FX101 (to see if you somehow see the pulse
that comes just before the ring - I always wondered why phones tended
,
probe: wcfxs_init_one,
--
All your code belongs to Santa
Tony Hoyle [EMAIL PROTECTED] Key ID: 104D/4F4B6917 2003-09-13
Fingerprint: 063C AFB4 3026 F724 0AA2 02B8 E547 470E 4F4B 6917
Phone(FWD): (0845 004 5566) 413300
___
Asterisk-Users mailing
Amaury Jacquot wrote:
Tony Hoyle wrote:
Guess what... BT actually charge a monthly rental fee (£1.50 or ~$2.40
per month) for providing this information. Gotta love state
monopolies hmm? Hope they don't take too long provisioning it...
heh, they'll probably mess up some other setting
the wire data format - you might be able to grab the data
but then not be able to make any sense of it..
Tony
--
All your code belongs to Santa
Tony Hoyle [EMAIL PROTECTED] Key ID: 104D/4F4B6917 2003-09-13
Fingerprint: 063C AFB4 3026 F724 0AA2 02B8 E547 470E 4F4B 6917
Phone(FWD): (0845 004 5566
David J Carter wrote:
Tony,
Lost some of the mails on this topic somewhere.
Does this need the BT50 mod or will the X100p now output the Caller ID?
It's to allow the X100P to output the caller ID.
My soldering skills just weren't up to the BT50 mod :)
Tony
--
All your code belongs to Santa
Tony
DTMF though).
Tony
--
All your code belongs to Santa
Tony Hoyle [EMAIL PROTECTED] Key ID: 104D/4F4B6917 2003-09-13
Fingerprint: 063C AFB4 3026 F724 0AA2 02B8 E547 470E 4F4B 6917
Phone(FWD): (0845 004 5566) 413300
___
Asterisk-Users mailing list
[EMAIL
Chris Stenton wrote:
Tony,
Are you going to submit the patches to the cvs head?
http://bugs.digium.com/bug_view_page.php?bug_id=0001719
Tony
--
All your code belongs to Santa
Tony Hoyle [EMAIL PROTECTED] Key ID: 104D/4F4B6917 2003-09-13
Fingerprint: 063C AFB4 3026 F724 0AA2 02B8 E547 470E 4F4B
Scott Brooks wrote:
Has anyone ported the ztdummy module to 2.6? I don't really want to dive into
it that far if someone already has.
http://www.nodomain.org/asterisk/ztdummy.diff
:)
Tony
--
Te audire no possum. Musa sapientum fixa est in aure.
Tony Hoyle [EMAIL PROTECTED] Key ID: 104D/4F4B6917
would work on 2.4
though (maybe worth a try if someone's got some spare time...)
If you run zttest with it loaded you get about 99.98% accuracy.
Tony
--
Te audire no possum. Musa sapientum fixa est in aure.
Tony Hoyle [EMAIL PROTECTED] Key ID: 104D/4F4B6917 2003-09-13
Fingerprint: 063C AFB4 3026 F724
if asterisk is getting the incoming CID. It
could just be the phone not displaying it.
Presumably you have signed up with BT to have caller ID sent on your line?
Tony
--
Te audire no possum. Musa sapientum fixa est in aure.
Tony Hoyle [EMAIL PROTECTED] Key ID: 104D/4F4B6917 2003-09-13
Fingerprint
the same that goes
into the cdr-csv file.
Tony
--
Te audire no possum. Musa sapientum fixa est in aure.
Tony Hoyle [EMAIL PROTECTED] Key ID: 104D/4F4B6917 2003-09-13
Fingerprint: 063C AFB4 3026 F724 0AA2 02B8 E547 470E 4F4B 6917
___
Asterisk-Users mailing
at 4642 (offset -148 lines).
Hunk #2 FAILED at 4681.
1 out of 2 hunks FAILED -- saving rejects to file
channels/chan_zap.c.rej
bash #
The patch is against the HEAD branch not the stable one.
Tony
--
All your code belongs to Santa
Tony Hoyle [EMAIL PROTECTED] Key ID: 104D/4F4B6917 2003-09-13
prefer my phone to
start ringing immediately).
Try this patch. It also enables distinctive ring detection even if
usecallerid=no. It's not well tested yet (well, at all actually since I
don't have access to distinctive ring...)
Tony
--
All your code belongs to Santa
Tony Hoyle [EMAIL PROTECTED
(or may not)
work better.
Tony
--
All your code belongs to Santa
Tony Hoyle [EMAIL PROTECTED] Key ID: 104D/4F4B6917 2003-09-13
Fingerprint: 063C AFB4 3026 F724 0AA2 02B8 E547 470E 4F4B 6917
Phone(FWD): (0845 004 5566) 413300
___
Asterisk-Users mailing list
used that broke the date handling I
see... I'd have thought they would have come up with a better one by
now - many sites will be unable to apply it because it renders many
clients incompatible.
Anyway, this is OT for this list :)
Tony
--
All your code belongs to Santa
Tony Hoyle [EMAIL
it around slightly as (I think) the last one will result in
the phone getting the ringtone noise when they pick up.. it was
bypassing the actual reading of the data, whereas it's better to read
and ignore it. I put that one on the web page.
Tony
--
All your code belongs to Santa
Tony Hoyle [EMAIL
run asterisk with in verbose mode does it log anything useful?
Tony
--
Te audire no possum. Musa sapientum fixa est in aure.
Tony Hoyle [EMAIL PROTECTED] Key ID: 104D/4F4B6917 2003-09-13
Fingerprint: 063C AFB4 3026 F724 0AA2 02B8 E547 470E 4F4B 6917
Tony Hoyle wrote:
That's the way the code is written (which is why you have 3 values in
the dring entry I think). There's an assumption in the code that it has
3 rings to compare with.
Actually it's worse than that... there is actually no terminator in the code.
It's looking for a distinct
Tony Hoyle wrote:
I've changed the patch to fix the buffer overrun, plus a hack to only
look for the dring values you specify, thus:
btw. It would be great if someone in the US can test these changes to make
sure I haven't broken the CID/DR on their side by doing this.
Tony
--
Te audire
as it's
neater).. the zaptel side though is stable.
Tony
--
Te audire no possum. Musa sapientum fixa est in aure.
Tony Hoyle [EMAIL PROTECTED] Key ID: 104D/4F4B6917 2003-09-13
Fingerprint: 063C AFB4 3026 F724 0AA2 02B8 E547 470E 4F4B 6917
___
Asterisk
Tony Hoyle wrote:
I'll probably do some tidying up (change ukcallerid to callerid=uk as
it's neater).. the zaptel side though is stable.
OK... now uses usecallerid=uk (or usecallerid=us for symmetry).
Tony
--
Te audire no possum. Musa sapientum fixa est in aure.
Tony Hoyle [EMAIL PROTECTED] Key
the dtmf decoder (ast_dsp_new() and friends).
It's harder than doing it for the UK but not so much harder that someone
couldn't knock it together in a day or two...
Tony
--
Te audire no possum. Musa sapientum fixa est in aure.
Tony Hoyle [EMAIL PROTECTED] Key ID: 104D/4F4B6917 2003-09-13
Fingerprint
.
Tony Hoyle [EMAIL PROTECTED] Key ID: 104D/4F4B6917 2003-09-13
Fingerprint: 063C AFB4 3026 F724 0AA2 02B8 E547 470E 4F4B 6917
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update
Philip Edelbrock wrote:
18 17.161118 Grandstr_05:a9:bf - BroadcastARP Who has 206.228.191.144?
Gratuitous ARP
19 17.609869 3com_96:2f:eb - Grandstr_05:a9:bf ARP 206.228.191.144 is at
00:10:4b:96:2f:eb
20 20.155260 206.228.191.144 - 206.228.191.7 DHCP DHCP Decline -
Transaction ID
Carlos Chavez wrote:
I have been using Sixtel from the beginning of the year and service was
getting worse and worse. Yesterday I tried to access the website to get the
CDR and I got an error saying that the domain no longer exists. I checked the
whois and it says that the domain is on
Rich Adamson wrote:
Just a data point... tftp works just fine in RHv9 and FC3 with remote
7960's. Images, config files, etc, get transferred correctly every time,
and the 7960's are between elcheapo firewall boxes.
If you really want to restrict who can access the tftp server, run one
of the
Balaji NJL wrote:
-- Call accepted by 65.39.205.121 (format ulaw)
-- Format for call is ulaw
It's working fine..
== No one is available to answer at this time
-- Executing Goto(SIP/200-d345,
s-NOANSWER|1)
in new stack
..but you're not answering the phone, or it's offline.
Try
Jason Walker wrote:
the SEPDefault.cnf binary file that is refered to on voip-info under the
Cisco 12sp+/30VIP page. I am open to anything needed to get these
phones working as the company i work for is non-profit and dont have
much of a tech budget and I can get them for around $30 each.
[EMAIL PROTECTED] wrote:
I get the same on all cid-unavailable or private calls.
Seems that asterisk is not picking up the cid, so it is just saying
asterisk, probably somewhere in the cid coding.
It would be nice to have it show as unavailable though.
In the [default] section of sip.conf
Bruce Leetch wrote:
I've purchased a TDM11B card and have installed it in the box. Windows
sees it as a PCI Simple Communications board. A Linux lspci doesn't
show anything even vaguely resembling this card. This troubles me.
VMWare is a virtual machine and has nothing to do with the
Michiel van Baak wrote:
I have put this in my dhcpd.conf to make sure my cisco
phones connect to my TFTP server:
server-name 192.168.2.1;
I'd be surprised if that worked... the server name is for.. um.. the
name of the server :)
Try:
option tftp-boot-server code 150 = ip-address;
option
Pavel Jezek wrote:
Hi,
asterisk will negotiate codecs for both parties independently (use sip
show peer peer and look for codec order entry), so, if you have
prefered codec g729 for your sip phone/peer, asterisk will use them
(regardles of codec setting for other party - if codecs does not
Erick Weber V. wrote:
For me to
Works for me...
Tony
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
Don Fanning wrote:
What settings are people using? I've seen the ones from dslreports but
I'm in that lucky group of people that paid the 1 euro just to have it
no longer work. Even after I setup a additional account over the
weekend it still doesn't work.
[voipbuster]
Don Fanning wrote:
That's what I have as well... What codec are you running with
connections to it?
My full settings (there's no g729 codec btw.. so that doesn't really count).
[general]
delayreject=yes
disallow=all
allow=g729
allow=gsm
allow=ulaw
allow=alaw
jitterbuffer=yes
Don Fanning wrote:
CALLED NUMBER : 1516308
Is that a valid number? AFAIK all voipbuster numbers have to start with
0 as there's no local dialing.
Tony
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
Too many people with misconfigured autoresponders...
Latest is Make Zuzlak, who has announced he'll be annoying everyone
until August 22nd.
If people are going on holiday please do one of 3 things:
1. Don't use an autoresponder
or 2. Use one that isn't broken.. ie. knows what the Precedence:
canuck15 wrote:
To start ztmonitor in quantitative mode you do the following.
Change line 261
fprintf(stderr, Usage: ztmonitor channel num [-v] [-f FILE]\n);
to
fprintf(stderr, Usage: ztmonitor channel num [-v | -f FILE]\n);
Err.. changing that does absolutely nothing to the active part
Sherwood McGowan wrote:
since delete is a reserved word, what do you name a column in your
voicemail options table to allow setting of the delete option for
realtime voicemail? Anyone?
[delete] should work, or on some databases 'delete'.
Tony
___
Scott Henderson wrote:
You can't use the same extension on multiple line buttons but you can
use different extensions on different line buttons.
Actually you can, and the 7960 does the 'right thing'.. surprised me too.
Tony
___
Asterisk-Users mailing
[EMAIL PROTECTED] wrote:
Maybe not the place for this but thought I'd post the info for others.
I purchased a cisco 7960 off ebay and needed to convert to SIP for *.
I know * supports SCCP but I wont go into that here. I'd read on
voip-info.org that a contract could be purchased for approx
trixter http://www.0xdecafbad.com wrote:
does anyone have knopsterisk for download, I assume that because its GPL
the creator of that iso cant restrict spreading it. A friend wanted it
to play on a box and the only thing I can find with google is the
knopsterisk.com site which wants $10 to get a
Ronald Wiplinger wrote:
A PHP program could find the remote address and the remote proxy
address. I guess a java program could find out what IP addresses the
user has and what neighbor addresses the users has (arp) and the java
program could make a DHCP request. If the user just submit his data
Isamar Maia wrote:
Good programmer is who makes the things working well *as planned* in the
time-limit planned beforehand, having good results for the *business* in
the end-of-the-day.
The rest doesn't matter.
Actually no. That confuses short term business objectives with quality
of programming.
Ronald Wiplinger wrote:
Hmm, I have never heard about that, but as far as I expierenced with
DHCP you would be assigned the same IP address.
Many providers force an IP change on each query (and an sometimes even
more often) to differentiate between the dynamic 'home' accounts and the
static
Hi,
I've been looking at the problem of the default caller ID. When a call
comes in with no CID or witheld it's always set to 'asterisk' which is
what the phone displays. I've been looking for an option to change that.
The only place I can find is DEFAULT_CALLERID in chan_sip.c. This is
Waldo Rubinstein wrote:
I'm planning on setting up some remote agents and before doing so, I
did some simple PING tests to measure latency. The average latency I
got was 250ms. Does anyone have experience in terms of quality of calls
when there is such high latency? Can anyone comment?
[EMAIL PROTECTED] wrote:
This comes in with a price tag of £56 ( $100 ). It has been 6 weeks since
I purchased the contract. Still no news! and no access!
The UK vendors don't seem to like dealing with these contracts.
I waited two months. It took legal threats before the vendor (Lanway)
Juan Pablo Abuyeres wrote:
lspci -v says:
02:08.0 Communication controller: Unknown device d161:0410 (rev 02)
Flags: bus master, medium devsel, latency 32, IRQ 52
Memory at dd20 (32-bit, non-prefetchable) [size=128]
There's nothing in the PCI device database
Juan Pablo Abuyeres wrote:
and there's nothing for vendor 79de at pcidatabase.com, and the kernel
module loads well. Maybe there's more than one identifier for T410P ?
There's nothing for 79de at pci-sig either
(http://www.pcisig.com/membership/vid_search/search_form/process) and
every PCI
Jon Creasey wrote:
Tony,
I'm havin a similar issue i'm in the UK using x100p with the patch for
CID and get the following. Any ideas
Did you change line as mentioned earlier? Without that patch incoming
SMS won't work at all.
You might need to use an even higher pause... might be worth
Terry H. Gilsenan wrote:
And as for cell phones being cheap, you have a receiver pays setup! How good
is that, then you have so many competing Telcos that sometimes you just
I believe that's unique to the US, the idea of paying for actually
receiving calls... don't know why they stand for
Neil Bullock wrote:
Hi all,
Am looking for everyones advise/recommendations.
I have am setting up a network of both office and home based workers.
The office workers will be on the same network as the Asterisk box so no
NAT hassles there. However, the home workers are on their own DSL
1 - 100 of 126 matches
Mail list logo
