that is SWEET!!!
- Original Message -
From: Philipp von Klitzing [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, June 08, 2004 9:02 AM
Subject: Re: [Asterisk-Users] Re: Grandstream 1.0.5.0 Firmware: SIP Register
option gone
Hi!
Did you try out the new ring tones? One of
I used something like this...
LOAD DATA INFILE '/tmp/cdr.csv' INTO TABLE cdr FIELDS TERMINATED BY ','
ENCLOSED BY '' LINES TERMINATED BY '\n' (accountcode, src, dst, dcontext,
clid, channel, dstchannel, lastapp, lastdata, calldate, pickup, hangup,
duration, billsec, disposition, amaflags);
I had
That's good to know. However, if I call into my broadvoice number and
pick up the call, I can send and receive DTMF audibly. Shouldn't that
fall under inband signaling and should be taken care of by the
appropriate dtmfmode setting?
-Original Message-
From: [EMAIL PROTECTED]
Hello!
I have download the latest CVS and recompiled zaptel, libpri and asterisk.
I have included these two new contexts into my extensions.conf.
Have I pasted them in the right place?
Do I need to include them into my [local] conext?
[smsdial] ; create and send a text message, expects
Hi,
- Original Message -
From: Christopher Wall [EMAIL PROTECTED]
I have been lookin for someplace to lean how to dump all of the call
transactions into a sql database. Can anyone provide me any assistance?
If you want to import Master.csv, take a look at my mail posted on 2-Jul on
Hi people.
I have 2 cisco 7940, the first is behind a NAT and the second is behind
2 NATs (The one for the LAN and one for Wireless).
The two phones worked perfectly until Friday. For some reason the
second one stoped registering (chan_sip) and the today none would work.
Any ideas on what
I changed my php to
fputs($socket, "Channel: Zap/g1/17094009\r\n");
and now I get different output hooray!
== Manager 'cron' logged on from 10.0.0.3 -- Executing SMS("Zap/1-1", "0151707||07950160728|LoveU") in new stack -- Executing SMS("Zap/1-1", "0151707") in new stack == Manager
Ive got similar probs Mark, and no one either here (unless I havent got
thru the pile yet) or on the IRC channel last nite answered. Ive simply got
no response when I try to use Iaxtel to call anywhere. My distant end is
experienceing the exact same thing. I also tried FWD to Iaxtel, and it
Hi
Is it possible to turn down a 2 users conference into a normal call ?
I don't think that the redirect manager api command can do it.
The new Meetme CLI commands detailed in the Wiki don't seem of
any help.
I'm willing to make it a normal call because I need an access
to the transfer
I worked out I was doing something wronggg!
i changed tthis in the PHP
fputs($socket, "Channel: Zap/g1/17094009\r\n");
and now I get ...
== Manager 'cron' logged on from 10.0.0.3 -- Executing SMS("Zap/1-1", "0151707||07779510643|LoveU") in new stack -- Executing SMS("Zap/1-1",
This problem with Grandstream was fixed long ago by Mark.
You have to change your sip entry from friend to peer and enable option
insecure=very to make early dialing working.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Stephen R. Besch
Sent:
Thanks for verifying that...thats what I thought...took two days to verify
it...
At 13:21 6/8/2004, you wrote:
same with their 700 network
w
- Original Message -
From: Mark Musone [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, June 08, 2004 11:24 AM
Subject: [Asterisk-Users]
Can we see your extensions.conf
- Original Message -
From: John Campbell [EMAIL PROTECTED]
To: asterisk-users [EMAIL PROTECTED]
Sent: Tuesday, June 08, 2004 3:55 PM
Subject: Re: [Asterisk-Users] Don't want a ring before voice menu
I should have been clearer in my description of the
Heh..yea, I made sure I did a search through the archives before posting
it :) (not that I'm complaining)
The weird thing though is that I _am_ able to call digium's iaxtel
number..
-Mark
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of tmpm
Sent:
I have had similar troubles and doing a modprobe -r zaptel then
re-loading the zaptel modules seems to cure it. ( if you unload the
wcfxo ztdummy wct1xxp etc it leaves the zaptel module loaded.)
On Tue, 2004-06-08 at 07:25, Rich Adamson wrote:
I've been playing with two pieces of hardware:
John,
If your FXS lines are ringing, it's because your dialplan spells out
that the line is to ring. If you don't want it to ring, don't Dial that
Zap line.
Otherwise If you have an analog phone attached to the same line as
the FXO card, you will always hear some amount of ring. Since there
I just uploaded a beta CallerID program.
It talks through the Asterisk Manager .
Pretty self expanatory for setup and configure.
Please Let me know what you think.
http://www.easyhomenetworks.com/AstRec/
If you feel our programs are useful please make a donation. We dont plan
on SELLING these
On Tue, 8 Jun 2004, Jay Milk wrote:
That's good to know. However, if I call into my broadvoice number and
pick up the call, I can send and receive DTMF audibly. Shouldn't that
fall under inband signaling and should be taken care of by the
appropriate dtmfmode setting?
I feel better knowing
Hi people,
There is any way to control silence detection in Zaptel ??
I have a x101p card, and sometimes the sounds dont come.
I notice that this is a Silence Detection in the card how can i avoid this ?
Another things, g.729a has silence detection ? Or Asterisk do this ???
Using a sip i can
Most of what I have found on the net, and the documentation with the card
says you need a v before the number to make it a voice call.
I tried it without the v, and the calls still aren't going through, but now,
I get:-
-- Executing Ringing(SIP/PHONE2-b960, ) in new stack
-- Executing
Darren Edmundson wrote:
I apologise if I misunderstood your aims, however surely it is infinitely
better to try and get asterisk standards compliant by default (preferably
before the 1.0/1.1 release) rather than resort to promoting DNS hacks to
get around what should be an easy enough bug to fix.
Jan Janak wrote:
In your own sandbox, feel free to do whatever you want. If the companies you
In my own sandbox I'm promoting enum, I can't dial SRV records from my
SIP hard phone, I don't want to have a laptop to make calls, I want
something small, and can dial a number easily without
Semi, offtopic question:
What do you communicate with your as5300 for and how do you do it? We
also use an as5300 for data dialin. If I could get more use out of it
that would be great
Jeremy Jones wrote:
I can't speak for general cases, but I know when I've tried to set
videosupport=yes, my
An interesting article for those needing ammunition to sell Asterisk within
their organisation or to others:
Is open source IP telephony ready for prime time? Yes
by Zenas Hutcheson, St. Paul Venture Capital
Network World, 06/07/04
http://www.nwfusion.com/columnists/2004/0607faceoffyes.html
On
Achilles Bochoris wrote:
Hello,
I understand these licensing issues very well. I don't reside in the
US, so I assume that there is no problem, especially for
testing/development, and not commercial use.
What I was asking however, is whether there is an alternative G.723.1
library which
Hi,
hskim wrote:
I heard that asterisk support r2 signaling.
I'm try to test r2 using e100p.
How should I configure zaptel.conf, zapata.conf?
And if I want to modify source for customization, where should I start?
Thanks.
Asterisk does not support R2 right now. You will find some R2 code
Kevin P. Fleming wrote:
Steve Underwood wrote:
spandsp doesn't try to reimplement all of HylaFAX. It reimplements
only one piece - the T.4/T.30 code. I have a half implemented
spandsp as class 1 fax modem which I put aside. People are using
spandsp happily for things like fax to e-mail.
Scott Nelson wrote:
My office is investigating using an Asterisk PBX and also going to a VOIP
provider for our main phone connections, but one of the tricky things is that
we need to have outbound and inbound modem calls (fax too).
I see a lot of talk about faxes but no mention of modems on
Having spent the better part of an hour searching the archives and voip-info
I hesitantly ask what appears to be an obvious question but one I cannot
find an answer for.
Using Grandstream phones it seems that the only way to support Call Parking
is to enable # transfers (i.e. use T in the dial
I mean CPU loading. HylaFAX only does 1D coding (unless that changed
very recently) and the ECM is brand new. The features you list may be a
lot less well tested than you think. :-) Also, only a tiny fraction of
FAX machines can even support ECM.
Steve,
HylaFAX supports 1D MH, 2D MR, and 2D
Darren Nickerson wrote:
Steve,
HylaFAX supports 1D MH, 2D MR, and 2D MMR.
The last time I looked (a few months ago) it supported those file
formats, but only supported 1D transfers on the wire.
ECM is new in HylaFAX, but already seems more robust than the implementation
one finds in most
The last time I checked on a big FAX server, only a few percent of the
calls used anything but basic 9600bps non-ECM operation. When I look in
the shops, hardly any of the FAX machines - other than the low selling
high end laser models - support anything fancy. If you are dealing
purely with
Steve Underwood wrote:
If you want to FAX over IP you need to be *very* careful if you want it
to be reliable. You cannot use anything other than A-law or u-law as the
codec. However, even using those, any data slips will kill the FAX
operation. If the two boxes are on the same LAN it tends to
Darren Nickerson wrote:
What can I say? That's not our reality. Every little HP OfficeJet el-cheapo
multifunction inkjet device does ECM, MMR and V.34 (up to 33.6) speed faxing
these days. That's reflected in a large number of greater than 14,400 speed
connections (typically 28.8). They're not
Olle E. Johansson wrote:
Enum doesn't replace SRV records at all.
I'm not suggesting it does, however sending mail to a host with only an
A record will still be delivered if the host is configured to accept
mail for the domain just like sip servers will accept the call if it's
configured to accept
Has anyone implemented HOBIC SMDR output from *?
Can someone point me to the Bell HOBIC specification?
Thanks,
--
Michael Welter
Introspect Telephony Corp.
Denver, Colorado
+1 303 674 2575
[EMAIL PROTECTED]
www.introspect.com
___
Asterisk-Users mailing
Hey all,
I have an as5300 I use for dial in customers, we have 4 PRIs on it.
We have a few free channels on it. I'm wondering if I setup SIP on the
as5300 I can have asterisk use the free channels for dial out.
I'd still have to use my TDM04B for incoming calls, but at least I can
expand my
At 7:06 PM -0700 on 6/8/04, George Pajari wrote:
An interesting article for those needing ammunition to sell Asterisk within
their organisation or to others:
Is open source IP telephony ready for prime time? Yes
by Zenas Hutcheson, St. Paul Venture Capital
Network World, 06/07/04
John Todd wrote:
I am not saying that this is good or bad, actually. It's neutral. The
purpose of Open Source is not to defeat commercial implementations of
the same features, but to provide a better solution for some people
who want to get in there and make things work exactly they way they
At 9:59 PM -0600 on 6/8/04, Michael Welter wrote:
Has anyone implemented HOBIC SMDR output from *?
Can someone point me to the Bell HOBIC specification?
Thanks,
--
Michael Welter
Introspect Telephony Corp.
Denver, Colorado
+1 303 674 2575
[EMAIL PROTECTED]
www.introspect.com
Sounds like you're
search for app_valetparking and hope its still out there somewhere :)
bkw
- Original Message -
From: George Pajari [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, June 08, 2004 8:21 PM
Subject: [Asterisk-Users] Sending # and Asterisk Transfer Conflict
Having spent the better
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
Dear All.
I'm very new to CT, but attracted by
asterisk.
I plan to start learn to build IVR, based on
asterisk.
Do I need FXO/FXS card to start building simple IVR
box ?
Can I use OEM X100P - FXO PCI Card ( http://www.digitnetworks.com/store/product_info.php?products_id=28
) to start build
Steve,
I'm going to use e100p for an ivr system.
Currently local telco only supports r2 for E1.
If you release the code, it will be very helpful for me.
Best regards,
Hong
- Original Message -
From: Steve Underwood [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, June 09, 2004
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