Re: [Asterisk-Users] Re: Red Alarm TE110P
You're lucky you didn't let the smoke out of your card - some HDSL units in the USA have some serious voltage/current on the pair that goes into the telco side to power the unit. Glad to hear it turned out okay. Tim - Original Message - From: Remco Barende [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, May 24, 2005 6:33 PM Subject: RE: [Asterisk-Users] Re: Red Alarm TE110P Hi all! *Very* happy to report that it is working now :) Indeed for a Dutch (KPN) PRI it seems that you must always connect the TE110P to the HDSL desktop unit, not directly to the telco line! For the connection HDSL - TE110P a standard ethernet patch cable did the trick. Thanks for all that replied, especially Ron Arts!! Remco On Tue, 24 May 2005, Peter Svensson wrote: On Tue, 24 May 2005, Remco Barende wrote: On Tue, 24 May 2005, Huddleston, Robert wrote: OK, but being from Europe I haven't got a clue what an American SmartJack is for :) Would that mean that I would have to hook up the TE110P to the HDSL device? If so, what sort of cable would be needed for that? HDSL is not the same as a E1. Sometimes E1:s are tunneled over HDSL to extend the range without the need for midspan repeaters. Does the HDSL device have an E1 port? If so, connecting to it using a standard ethernet cable should work. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Vonage goes to .ca
Hopefully they won't mandate a specific amount of Canadian content per call. ;) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of TC Sent: Tuesday, April 13, 2004 7:40 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Vonage goes to .ca FYI http://www.itbusiness.ca/index.asp?theaction=61sid=55298 did not like this by line in the story the CRTC has said it will likely regulate voice over IP the same as other phone services. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ATA 186 Registration!!!!
In your sip.conf file, where you're describing your ata-186, make sure that your username is the same as the [description] above the phone. Mine starts out like [cisco] type=friend username=cisco secret=* nat=yes host=dynamic My UID0 on the ata-186 is 'cisco' and the password matches the secret in the sip.conf file. I couldn't get it to register either until the bit in brackets matched the username, then it worked fine. I have both channels of my ata 186 registering as different extensions. Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Erick Weber V. Sent: Monday, February 23, 2004 8:24 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] ATA 186 Registration I'm tring to register my ATA to * and I getting the following message: Feb 23 18:13:04 NOTICE[1125329600]: chan_sip.c:5405 handle_request: Registration from 'sip:[EMAIL PROTECTED] user=phone' failed for 'xxx.xxx.xxx.xxx' I don't know what's wrong an why it register as user=phone??? Coul some one help me Thanks Erick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] agi scripting in perl - dealiing withunexpected disconnects gracefully / spurious DTMF
Thanks - that gave me the basis for a couple of google searches.
Near the top of the script I put in
$SIG{HUP} = \exitGracefully;
and I added a subroutine that looks like this:
sub exitGracefully {
exit(0);
}
It now kills itself off without needing to be killed and take * with it.
THANKS!!!
-Tim
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steven
Critchfield
Sent: Wednesday, February 18, 2004 8:00 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] agi scripting in perl - dealiing
withunexpected disconnects gracefully / spurious DTMF
On Wed, 2004-02-18 at 11:11, Tim Petlock wrote:
If everything is entered correctly and perfectly the script works and
the call goes through. However, I've found that if I enter only a
partial calling card number and then hang up the script will
continue
to run with the perl process that it lives inside of taking up lots of
processor time and for each failed call you get one of those
disconnected processor-hungry processes.
When a user disconnects and asterisk hangs up the line, it closes it's
side of the pipes it uses to talk to AGI. If you don't handle a null
read from the pipe as a hangup, then it writes commands to asterisk and
then gets a null read back and loops hard and fast.
Maybe if I get some more time later I'll read the rest of that lengthy
post and see if there are more answers for you.
--
Steven Critchfield [EMAIL PROTECTED]
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[Asterisk-Users] agi scripting in perl - dealiing with unexpected disconnects gracefully / spurious DTMF
$try 3) {
$pin = $AGI-get_data(plscrdnum, 1, 8);
# debug --
$AGI-exec('SayDigits',$try);
# debug --
$units = check_pin($pin);
if($units eq undef) {
$try++;
# we could do an invalid pin warning here... :P
} else {
# reset try because we want to make sure we try to get a
valid
# X digit number from the user.. its set at 7 now.. but
you can
# change that to say 11.
#
$try = 0;
$timeout = $units * 60;
# use the pin as the account code so we can track this
$AGI-exec('SetAccount',$pin);
$AGI-stream_file('balance');
$AGI-exec('SayDigits',$units);
$AGI-stream_file('minutes');
$AGI-exec('Wait','1');
while(length($target) != 11) {
if($try = 3) {
$AGI-stream_file('vm-goodbye');
$AGI-hangup();
exit(0);
}
$target = $AGI-get_data(plsdial1, 1,
12);
$try++;
}
$AGI-exec('AbsoluteTimeout',$timeout);
# maybe a prompt saying connecing your call here.
# this could be a config option or even dynamic
depending on info associated
# with the pin. Many Many options here.
$provider = 'IAX2/[EMAIL PROTECTED]/';
$dialstring = $provider . $target;
$AGI-exec('Dial',$dialstring);
$AGI-hangup();
exit(0);
}
$try++;
}
# user screwed up so lets just say goodbye and let them try again later.
$AGI-stream_file('vm-goodbye');
$AGI-hangup();
exit(0);
sub check_pin($pin) {
my $query = SELECT units FROM pins WHERE pin='$pin' LIMIT
1;
my $sth = $dbh-prepare($query);
$sth-execute
|| die(Couldn't exec sth2!);
# need to be graceful here also
my $units = $sth-fetchrow_hashref;
#
# Now we subtract usage from this pin.
# Lets ditch all calls under 6 seconds.. you can change this.
# Also when you clean the database you need to be sure that you
remove the pin from
# pins database. Check TODO list above.
#
my $query = SELECT SUM(CEILING(billsec/60)) AS used FROM
cdr WHERE accountcode='$pin' and billsec 6;;
my $sth = $dbh-prepare($query);
$sth-execute
|| die(Couldn't exec sth2!);
# need to be graceful here also
my $used = $sth-fetchrow_hashref;
$units-{units} = $units-{units} - $used-{used};
$sth-finish;
if($units-{units} 0) {
return $units-{units};
} else {
return undef;
}
}
Thanks!
Tim Petlock
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RE: [Asterisk-Users] Re: Need to interface to BRIs
What is the advantage of having zaptel timing? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Klaus-Peter Junghanns Sent: Monday, February 16, 2004 11:34 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Re: Need to interface to BRIs The FritzCard has CAPI drivers and does NOT provide zaptel timing. The quadBRI PCI has zaptel drivers and does provide zaptel timing. Am Mo, 2004-02-16 um 14.41 schrieb Master Abi: Does the Fritz!Card PCI and Quad BRI also provide timing like the Digium Zaptel cards? Matteo Brancaleoni wrote: Il lun, 2004-02-16 alle 12:49, Cees de Groot ha scritto: Klaus-Peter Junghanns [EMAIL PROTECTED] said: we have a 4 BRI solution for Asterisk, the quadBRI PCI ISDN. One thing I'd like to know about this card: Echo Cancellation? I've replaced by Fritz!Card PCI by a Diva Server 2M, and the difference is remarkable... since is zaptel based, it shares same zaptel routines for EC, as far as I know. Matteo. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dialing 800 numbers with VOIP
Hm. After seeing all the people who say it works, I thought - maybe I forgot to dial 9 in front of the number and that's why the call failed. So I looked up the Wells Fargo toll free number again and tried it. Failed. SIT tones and We're sorry, your call did not go through. Will you please try your call again later? The recording has nothing at the end that might give some clue who was generating it either. Okay, I thought, maybe it's a regional number and that's why. Nope, for further testing I looked up toll free numbers that for sure would be nationally dialable. They fail exactly the same way. Perhaps it's something in my Nufone account setup? I don't know. The SIP and IAX debug messages on my console don't look appreciably different from those where calls complete so I know the call is getting there. It's impossible for any recording to be generated on my * box because my sound driver has yet to work, it's not coming from there. I'm certainly not out to bash Nufone - I live in Uruguay and have a 64k internet connection. Vonage barely works because the codec they've chosen and the speed I've bought don't match up. (I imagine that it would work better if I had a 128k connection - but 64k is USD$42 per month and the providers prices increase exactly proportional to the speed you want.) There was about a 1.5 second delay in the other party hearing what I was saying and only one person could talk at a time. Using a different Cisco ATA to connect to * and then GSM over IAX to Nufone sounds night and day different. No delay, minimal distortion, two-way conversation - this is good stuff. -Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Greg Boehnlein Sent: Tuesday, February 10, 2004 12:50 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Dialing 800 numbers with VOIP On Mon, 9 Feb 2004, Tim Petlock wrote: [ Long Explanation Deleted ] Nufone and Voicepulse would have to maintain some number of trunks with an ILEC or CLEC to complete toll-free calls. I dial 800 numbers all the time from my Nufone account without problem. Hell, my DID through Nufone -IS- an 800 number! -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Spurious DTMF tones heard by the person being called
I have one issue - spurious inband DTMF detection results in the caller hearing the odd burst of DTMF at random. My calls are going from a Cisco ATA 186 running Version: v2.16.1 ata18x (Build 030709a) using the g.711 codec with the AudioMode parameter set to 0x11241124, to a default-installed Asterisk box (running a recent build of Asterisk - I did the CVS checkout 2/6 or 2/7) whose only configuration changes were to add the extension for my ATA and to add the IAX configuration stuff for Nufone. Is that enough information for anyone to hazard a guess where the spurious DTMF is coming from? My understanding is that 0x11241124 sets up both channels of the ATA to use out-of-band DTMF and that should rule the ATA out as the cause. Calling a phone number and then making DTMF entries works the way this is configured and I don't really want to break that either. Does anyone have any idea where the issue is coming from? Tim ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Termination - Cuba
There was an article in El Pais' (Uruguay) Sunday paper two weeks ago about internet access in Cuba. Tourists can buy cards (priced in US dollars) that allow dialup access. According to the article, the call to the dialup number can only be placed from phone accounts that are billed and paid in US dollars. If your phone is billed and paid in Cuban pesos - you can't even complete the call to the modem bank. The article said that internet access is very sought after but hard to come by. The article said that computers, fax machines and photocopiers are not legal to buy or sell without a government license. I googled ETECSA (the name of the phone company in Cuba) and came up with this link to their rate sheet: http://www.infocom.etecsa.cu/nuevas_tarifas_u.htm Given that from the rate sheet I can see that the COUNTRY has embraced the concept of having an intranet, I think you're going to be hard-pressed to get what you're looking for. (Interestingly, while visiting Cuba three years ago I needed to call the USA - I went to the local ETECSA office and paid dollars in advance for the number of minutes I wanted to talk - and nearly fell over when they handed me a 3watt Motorola bag phone [AMPS cellular] across the counter to use to place the call.) Good luck in your search! Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Daniel Bichara Sent: Tuesday, February 10, 2004 5:15 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Termination - Cuba Hi, I am looking for a VoIP (SIP or Asterisk) termination at Cuba. High traffic. Thanks in advance, Daniel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dialing 800 numbers with VOIP
The business reason is that Voicepulse and Nufone probably have a combination of other VOIP business relationships and direct T1's to long distance resellers and those are used to complete calls. I have never experienced a case where a direct T1 to a long distance carrier could complete toll free calls. Not to say it doesn't exist, but I haven't seen it. Reason? When you dial a toll-free number there's an immediate database lookup prior to the call even starting to ring - to determine where (which carrier) to route that call. Local exchange carriers - whether ILEC or CLEC - are positioned and more importantly motivated to do so. They want all calls to complete from their dialtone. They want and are equipped to stay on top of and book the settlement income from all the other carriers they do business with. A long distance carrier provides you with a T1 channel directly into one of their switches. They have no incentive to maintain the realtime links to a database to facilitate calls that have to be handed off through some possibly non-existent interconnect to another carrier so that other carrier can bill their customer while that call free-loads on a channel in and out of their switching facilities. I bumped into the toll-free limitation with Nufone today when I was trying to call a bank. I don't live in the USA so I can't just walk to a payphone either - I had to locate a regular phone number of a bank branch and 'splain to them why I wanted to be transferred. They did so and it wasn't that painful. It didn't occur to me to add FWD to my config to complete such calls - but I will when I can figure out how to do it securely without giving anyone who dials my FWD number access across my Asterisk box to outbound calls through Nufone. Nufone and Voicepulse would have to maintain some number of trunks with an ILEC or CLEC to complete toll-free calls. -Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Isamar Maia Sent: Tuesday, February 10, 2004 4:45 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Dialing 800 numbers with VOIP Use fwd.pulver.com On Mon, 9 Feb 2004, Matt Lawson wrote: Hmm. Both Voicepulse and Nufone don't seem to be able to dial out 800 numbers. Are 800 numbers treated differently somehow? Or is there a business reason for disallowing them? It makes the ringing sound but never connects. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] can't compile asterisk
My compile of asterisk is bombing out in /usr/src/asterisk/res . I'm running debian with kernel 2.4.21 (custom). Zaptel and libpri appear to make okay. Any suggestions? When I reboot I don't see any indication that the zaptel modules are loading. I don't know if the final bits of a successful asterisk make do anything with that so I'm not sure if I have deeper problems than that. If you're using debian, which packages of readline and openssl have to be installed? Tim Petlock ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] re: unresolved symbols in /lib/modules/2.4.18/misc/zaptel.o
THANKS! I was missing PPP support in my kernel, thinking that I didn't need it with Ethernet present. It compiled and is running. now I just ;) have to configure it. Tim Petlock [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
