I had some weird flaky-ness after upgrading to the latest. Did a format
file system and let it reload from scratch. Works like a charm.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Scott Higginbotham
Sent: Monday, October 02, 2006
CALLERID(number) is invalid use CALLERID(num)
[Description]
Gets or sets Caller*ID data on the
channel. The allowable datatypes
are all, name,
num, ANI, DNID, RDNIS.
SNIP
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You can parse the Variable BEFORE sending to the conf.
Ie:
Exten = _8700X,1,Set(${DB(conf${EXTEN}/lastin)=${CHANNEL})
It will always be the last one in.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Time Bandit
Sent: Sunday,
Try This
exten = 5481,1,NoOp(${TIMESTAMP} paging Group 1 Office Page)
exten = 5481,2,DIAL(IAX2/5480/w1||)
SNIP
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You can use:
Exten = _X.,1,Goto(inbound,s,1)
snip
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Qualify does what the name implies qualifies the connection' It pools
every 60s but it calculates he time it took for the packet to reach the
end device. If the endpoint has a latentcy than the qualify parameter,
* considers the endpoint unreachable. This does not however address the
point you
This may not solve your problem but try
adding
canreinvite=no
to your sip.conf definitions.
Snip
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I'll feed the Troll!!!
Mark deserves this, he has given us, all of us, a way to make and/or
save money. Kudos to him and the staff at Digium. I, for one feel Mark
owes me nothing but I still feel like I owe him and the project much of
my uncompleted work.
Way to GO!! Mark. May the extra cash
I would use a cron job to set the number
in the AstDB such as
Asterisk rx database add
weeklynumber 301212 every Sunday PM, Monday AM or whatever works
for you
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dan Brummer
Sent: Monday, August 07, 2006
]
[mailto:[EMAIL PROTECTED] On Behalf Of Alexander Lopez
Sent: Monday, August 07, 2006
11:15 AM
To: Asterisk
Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] By
week extension dialing
I would use a cron job to set the number
in the AstDB such as
Asterisk rx database add
Try switching the wait and answer:
Wait 3
Answer
What kind of interface is this?
Zaptel, SIP, IAX, FXO, PRI??
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of shawn bright
Sent: Monday, August 07, 2006 9:49
PM
To: asterisk mailing list
Subject:
Prime
Murder
Suspect?
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of C F
Sent: Monday, August 07, 2006 11:27 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Hotels...
Interesting you
This might be what you're seeking;
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+ChanIsAvail
If the phone rings, then the channel IS available. The solution is to
disable call waiting on the SIP device.
The s option needs to be used:
s - Consider the channel unavailable if
Make sure you can access the file on your FPT server. Also make sure
that you did not fry the Ethernet port(s) on the phone.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Stas Khromoy
Sent: Wednesday, August 02, 2006 9:50 AM
To:
There currently exist no such option. But
you are free to try to add it.
SNIP
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Use a variable that is set when the call
comes in such as:
Exten = s,n,Set(OUTSIDECALL=1)
Then in your dial macro test for variable existence
and change ring via alert info or other distinctive ring methods. It is unfortunate
that it is heavily dependant on technology of the channel
Make sure the binary you downloaded MATCHES your machine.
snip
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SNIP
type=friend
echo=no
Any suggestions ?
One final note, 'echo' is not a valid option.
Thanks
Joshua Colp
Digium
ECHO=no Now that's a cool option
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If what you are asking for is a conference, you can use MeetMe and
transfer the participants to that MeetMe extension.
I you want it to be triggered by say the * sign then look at the
featuremap in features.conf. Using an AGI and redirect can do this for
you. Use the wiki @ www.voip-info.org for
If by ringing duration you mean how long a device will ring, then look
at options to Dial
If you mean how long the ring sounds to the callee look at
indications.conf
Alex
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Zenone
Sent: Wednesday, July 26,
Make sure you have enough CPU bandwidth on both sides IPsec has to
encrypt every little packet.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Stephen
Bosch
Sent: Tuesday, July 25, 2006 11:25 AM
To: Asterisk Users Mailing List - Non-Commercial
Cheaper depends on the age of the switch
at the CO you are connecting to. In some parts of Tenn.. I cannot get a PRI but a T1 will be
just fine, alas NO CID, or enhanced features but a T1 nonetheless.
I my old CO PRIs had to be FXed (Brought
in from another CO) My CO was a 1A and but
Make sure your mail system is working.
Try mail [EMAIL PROTECTED]
From the os command prompt
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Dean @ INKnBITs
Sent: Monday, July 24, 2006 8:02 AM
To: Asterisk Users Mailing List - Non-Commercial
Take it back to the old-skool!
Use biff..or a newer version ebiff,
Yamb, etc. etc.
Snip
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-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Douglas
Garstang
Sent: Thursday, July 20, 2006 12:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Redundant Ethernet
We're using OSPF...
Is That?
Oh
Shit
I call and immediately identify this as a test call.
I state the following. My Nane, and the fact that I am the PBX tech,
(engineer confuses them). I ask them to confirm my address and call back
number I provide to them. If all is OK I thank them and hang up. I do
not think it is a false call if
I have had mixed results with Modems the pass through Asterisk. I can
recommend a solution that will always work however. We purchased an
Atlas 550 from Adtran, It 'splits' our PRIs into T1, PRI, BRI, and or
POTS. It is NOT a trivial purchase but it is a great product. We also
use it to provide
After Following this post, I have come to realize... That you may NO
LONGER NEED KY!!!.
Please take this off the list, the bandwidth consumed by this getting
ridiculous. Kevin Fleming (Digium) has already asked for this to be
taken off line. Please respect the wishes of those that fund the list.
Lists.digium.com
Whatever deal Digium may have struck with Easynews is not the issue
here.
Snip.
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easynews?
Snip.
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Define invite.
Yelling across the office saying, Yo,
Dude! Dial ${CON}, works as an invite for me!.
If what you want is an automated invite
look at callout files and using creative dialplan options to Meetme App.
Snip
Hi,
recently im working on using meetme application in
me to do. tell me if there is any help on the
internet about this.
On 7/6/06, Alexander Lopez [EMAIL PROTECTED] wrote:
Define invite.
Yelling across the office saying, Yo, Dude! Dial
${CON}, works as an invite for me!.
If what you want is an automated invite look at callout
Yep, forgot bout that.
Or you could use
web-meetme, it has this feature.
On 7/6/06, Alexander Lopez [EMAIL PROTECTED] wrote:
Snip, snip.
Chop Chop.
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asterisk-users
Your problem is intermittent. It is probably Network related as if you
reboot that problems may or may not comeback.
In addition to the lspci stuff requested. Have you checked your
fiberlink. Is it possible that something or someone is saturating the
link with Virus/Spy/PtP Ware???
SIP doesn't
Isnt DOCSIS a network layer 1 or 2?
I
TCP/ip would run on top of a DOCSIS network
SIP on top of TCP/ip
DOCSIS specifies downstream traffic
transfer rates between 27 and 36 Mbps over a radio frequency (RF) path in the
50 MHz to 750+ MHz range, and upstream traffic tranfer rates
otherwise store the information that is needed for the phone?!
CS
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Alexander Lopez
Sent: Thursday, June 29, 2006 4:01 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial
]
[mailto:[EMAIL PROTECTED] On Behalf Of
Alexander Lopez
Sent: Thursday, June 29, 2006 4:01 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: RE: [Asterisk-Users] SNOM Softphone on windows 2000
W2K had problems with Security (Surprising huh
Snip
On Fri, 2006-06-30 at 10:44 +0100, Steve Langstaff wrote:
trixter aka Bret McDanel wrote:
Lastly, and probably the least effective, is you can watch channel
usage
and when someone exceeds 5 run over to their desk and smack them
with
a
rotten fish.
I have used these in the past, with only one issue. The T1 line was at
the end of its tolerances as far as length from the repeater. The surge
suppressor ntroduced enough resistance to make the T1 bounce, like
Tigger.
Having the Telco put in a repeater closer to our facility made the
problem go
Ah, the problem is that you are connecting FXO to FXO. The IAXy provides
dialtone and o does your Intercom system. You can try to use an FXO to
FXS converter or simply replace it with an FXO adapter.
I would also check the documentation on your intercom device. There may
be a way to switch the
W2K had problems with Security (Surprising huh?) You may need to grant
write access for the user to the Folder where SNOM is installed. I don't
think SNOM is writing to the registry if so you will need to open
permissions up on those keys in the hive.
If they are on the same network you can do the following:
arp -a | grep $IPADDRESS |awk '{print $4}'
you may need to adjust awk(ed) position due to you distro.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Casey Boone
Sent:
What is you AGI written in??
-Original Message-
snip
Doug.
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Use __ALERT_INFO and IAX...
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Douglas Garstang
Sent: Monday, June 05, 2006 6:43 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] IAX Passing
You can look at the change logs
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Erick Perez
Sent: Thursday, May 25, 2006 1:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk codec
-Original Message-n to a non-US dummy the following phrases I have
What is US48?
I assume by US48 they mean RJ48 which is a 8 conductor modular jack
with
signal from the phone company on 12 and signal to the phone company
on
45.
Don Pobanz
You are kidding right???
I would write an AGI to place the needed files in the Voicemail
directory, and conversely remove them.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Matt
Sent: Wednesday, May 24, 2006 4:12 PM
To: Asterisk Users Mailing List -
the
ACD light on their phone.
On 5/24/06, Alexander Lopez [EMAIL PROTECTED] wrote:
I would write an AGI to place the needed files in the Voicemail
directory, and conversely remove them.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:asterisk-users-
[EMAIL PROTECTED
Are the 800 numbers you have new (post-outage) of existing
(pre-outage)??
SNIP
It is working, I have a 800-number with them.
jens
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We suffered no outbound downtime that was a 'show-stopper' proof
positive that JJ was and always will be there, however the TF REsporgs
are still killing me. You are working your butt off and I appreciate it,
good luck and hope to hear ringing in my ears soon. :-)
Alex (OpSys)
-Original
I could not find anything so I whipped up something myself using an
M-Audio FireWare Box, It gives my 4 distinct analog audio channels as
well as two inputs. It also has SPDIF (optical and Coax). I then paired
it with the SNOM softphone. I am also able to record to an external DAT
or MD, or feed
I have the same problem, Switched to
HylaFax and IAXModem and had MUCH better luck. MUCH better being defined as not
a single usable fax to only missing about 1%. Not bad.
Spandsp or rather RXFax works on a few
machines I hae quite well but on others it does not work at all, I have had
] On Behalf Of Armin Schindler
Sent: Friday, May 19, 2006 12:54 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] FAX over PRI
On Fri, 19 May 2006, Alexander Lopez wrote:
I have the same problem, Switched to HylaFax and IAXModem and had
MUCH
better luck
Is it Phone - ShoreTel - Asterisk
- PSTN ???
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anthony Cennami
Sent: Friday, May 19, 2006 1:47 PM
To: Doug
Cc: Asterisk
Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] PRI
dialing IVR
over PRI
If this works reliably
while rxfax+spandsp does not, wouldn't this point the blame at rxfax as opposed
to spandsp?
IAXmodem uses spandsp the same way rxfax does right?
On 5/19/06, Alexander
Lopez [EMAIL PROTECTED]
wrote:
I have the same problem, Switched to HylaFax
You are correct, That is why the PAGE()
Application was made. It creates a MeetMe room, calls the Technologies on the
list and transfers the calls to the temp MeetMe.
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Lacy Moore - Aspendora
Sent: Friday, May
DSL works by using the frequencies above 4k that were unused in POTS loops of
yesterday. Load Coils, Bridge Taps, and DC taps are all devices added to lines
to increase their reach and stability, unfortunately, they are DEADLY for DSL.
Other problems can effect DSL service, and cause it to be
BellSouth will provide Static IPs for home users, staring with the
Extreme product. (3MBit) and up. No extra charge for this, included in
the package.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Andres
Sent: Tuesday, May 09,
Look at the arguments for MeetMe,
MeetMe does not usually require a PIN
unless you use the P option.
You could do this for example.
[outside]
exten = 8600,1,Meetme(1234|P|321)
[inside]
exten = 8600,1,Meetme(1234)
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
The line build out value is a power level that is set
based on the distance from the Device to the T-1 service provider's gateway. If
the device is close by, the gateway requires less power and the line build out
value is lower; if the device is far away, the gateway requires more power
See:
http://www.sineapps.com/news.php?rssid=1130
snip...
I have gotten intercom working on my office phones (Linksys SPA-942s),
but I have noticed that if someone is in a call, it places the call on
hold and sends the intercom audio to the person holding the phone that
is being paged. I'd
] On Behalf Of Eric ManxPower Wieling
Sent: Saturday, May 06, 2006 2:23 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Info
Alexander Lopez wrote:
The line build out value is a power level that is set based on the
distance from the Device to the T
Title: Messaggio
Under Advanced make sure this is set:
Call join on Xfer (2 calls): to off
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tommaso Calosi
Sent: Thursday, May 04, 2006 4:02
AM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users]
I have a client that 'NEEDS' (his words not mine) to make sure that all
faxes, emails, calls, and mail are archived. Phone and email are simple,
Mail depends upon the integrity of the mail room, Faxes however can be
sent from anyone. They would like this as they recently had an issue
with a fax
The problem with the Dell's is their incompatibility with the TigerJet
Chipset, I have had problems with the SC 4X0 line of machines, they are
known to have issues.
Alex
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Dovid Bender
This is a very KGB / NSA / InterPOL / CIA type question, but
if I have a recorded file (G.711, no compression) can I feed it into standard
in of an application and have it recreate the fax that was send?
I dont know enough about the Fax handshaking to
understand this.
Use the Local channel and add the agents using that IE:
Member Local/[EMAIL PROTECTED]
Snip
Hello,
I'm experimenting with Asterisk for possible use in a call center.
I'm trying to figure out how to run applications when an agent answers
a call in the queue. I see that the queue itself
Acually, I have no time on my hands, but this was the
thought while in the shower this AM. Thought was the following. I needed to have
one fax sent to me and a customer at the same time. I know that I can recieve
and resend to both but I want to be able to 'snoop'.
From: [EMAIL
You da' Man!!!
I'll try this.
In spandsp there is a program in the tests directory called
fax_decode.
It isn't very sophisticated, as it is intended for my test
work, rather than general decoding. It is able to decode some
FAX audio from a wave file, though.
There are some expensive
-Commercial Discussion
Subject: Re: [Asterisk-Users] Running applications when a queued call
isanswered
Alexander Lopez [EMAIL PROTECTED] writes:
Use the Local channel and add the agents using that IE:
Member Local/[EMAIL PROTECTED]
Thanks Alexander,
That works, but it's backwards
Yes. I'd like to do something like:
Ringing()
SendURL(http://example.com/${EXTEN}.html)
SayDigits(${EXTEN})
Wait(5)
That's close to what you suggest, but Asterisk on its own announces
first then sends the URL with no wait, so the agent is left scrambling
to see who
I was about to post a bug, It hasn't worked for me since CVS 11/01/05!!!
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of William Piper
Sent: Tuesday, May 02, 2006 11:17 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Are you seriously trying to run 4 cards in one system? The odds of
getting
that working are about the odds of Angelina Jolie showing up on my
doorstep
ready to whisk me off tobut I digress...you will have serious
interrupt
issues trying to get 4 cardss working in one system. I am
But code quickly, as the quality produced is inversly related to the
amount of ${Insert_Your_Fav_Booz_bottle_brand_here} in your system.
Grab your fav. bottle of
${Insert_Your_Fav_Booz_bottle_brand_here} and get
working on it.
--- TV JOE [EMAIL PROTECTED] wrote:
How are the calls coming into the PBX. PRI? If so add a Wait(1) before
your try ringing the SIP channel.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Hall, Eric M.
Sent: Monday, May 01, 2006 12:37 PM
To: Asterisk Users Mailing
Looks like your D-channel is down.
Ztcfg reports all is ok, b/c as far as iut
is concerned, it is talking to your card just fine. LibPri handles the PRI
implemetaton.
Since you are able to see the pri commands
from the CLI, Isdn supprt is linked into your asterisk core.
Call your
His PRI span is showing down, If you
forget to add the ${EXTEN} as you said it would show as connecting and he _should_ get an intercept from the telco.
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Colin Anderson
Sent: Monday, May 01, 2006 2:53 PM
To:
. If
my telco says everything is ok, what should I look at next?
AFAIK this PRI was in working condition
before I moved it to the asterisk test machine.
Thanks!
-Dan
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alexander Lopez
Sent: Monday, May 01, 2006 11:53
AM
M.Sent: Monday, May 01, 2006 4:34 PMTo: Asterisk
Users Mailing List - Non-Commercial DiscussionSubject: RE:
[Asterisk-Users] CallerID Name problem
Do you wait before or after the answer? Do you even need the
answer?-Original Message-From:
Alexander Lopez [mailto:[EMAIL
01, 2006 6:29 PM
To: Asterisk
Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users]
CallerID Name problem
That worked GREAT
Thank you so so MUCH for your help!!
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alexander Lopez
Sent: Monday, May
You explained this very well thank you!!, We discussed (Astricon 2005 Anaheim)
having LibPri either wait 1 second before passing the call on to asterisk, or
waiting until CNAME was received, both ideas were not good as it will introduce
delays for all instead of just those that needed it.
For
It's a little crude but you can
1: Use VLAN(ing) on the Cisco Switch to segment the traffic on an
addition 'LAN'.
2: Low Budget, Add a NIC on a separate network with the NAS.
3: Give me a bit, It'll come to me! :-)
SNIP!!
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You can use the __Variables They are passed along the IAX2 channel
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Steve Totaro
Sent: Friday, April 28, 2006 9:24 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
At $10.00US per concurrent channel, it is better to buy, than to
complain. Do you complain i someone gives you a new car but you have to
pay for the gas?? (Bad example with Oil prices going high, but you get
the point)
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
Look at the joinempty option in queues.conf
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kevin Savoy
Sent: Wednesday, April 26, 2006
5:01 PM
To: 'Asterisk
Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Status
of Queue
Is
there
Add an Answer() as your first step in your dialplan and see if that
help.
snip
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Same results here with my PPC-6700 nice phone no processing power, I
found that my EVDO card on Laptop works great with SIP softphones.
Unfortunately, I have to agree. I was very pumped about being able to
use VoIP over WiFi on the PPC-6700 (which has a 416 MHz cpu), but the
phone's
Use the local channel to call the agent first, and if there is no answer, log
them out.
From: [EMAIL PROTECTED] on behalf of Kerry Garrison
Sent: Tue 4/25/2006 2:38 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users]
I was once told by a lineman that the cables they use didn't have that
many twists in them because it wasn't needed, and that the extra twists
would effectively use more cable and thus cost and weigh more than
triple what they do now. He told me that with the number of twists in
the Cat 5 cable it
Good thing he doesn't work for a cable manufacturer as that's
a total crock of crap that even an inexperienced person
should be able to detect. (You can't twist two wires to make
them weight three times as much, or cost three times as much.)
He may have started out as an underground lineman,
Unless you're going for some kind of distance record, standard Cat5
will
work
without any issue on any modern installation. As I said, I'm pretty
sure
(not 100%, but close) that the T1 specification is only Cat3, since
it's
standard BellCore wire and they don't run your T1 loops (which
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Kevin P. Fleming
Sent: Monday, April 24, 2006 12:14 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: Pinouts for T1/E1 crossover cable WAS RE:
Ever looked at the underground cable in the street outside your
building? If it's more than 20 years old, it's probably
paper-insulated
gel-filled cable, with an _extremely_ thin amount of insulation
between
the conductors and _zero_ insulation between the pairs. T1s seem to
work
just fine
You will not be able to determine what number was DIALED unless you have
DID service from you phone company. CF's suggestion is your best bet,
unless you move over to DID service.
Alex
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of C
Please:
1 Follow the suggestions that are sent out on EVERY Digest and
edit your subject line.
2 Trim your posts to only include the relevant information, the
list is quite large and brevity is a plus as smaller message distribute
faster than larger ones.
3 Make sure your
Are using g.729, by any chance?
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Chris Mason (Lists)
Sent: Sunday, April 23, 2006 8:10 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Zap -
Can't anyone stop self-promotion and tell the poor guy what he needs.
A T1/E1 X-over cable using an RJ-45 (8-cond.) is pinned out as follows:
1 - 4
2 - 5
3 - NU
4 - 1
5 - 2
6 - NU
7 - NU
8 - NU
NU = Not Used
I have not in my experience seen any problems with using a Good Quality
Cat5 vs. Cat 3
/E1 crossover cable WAS RE: [Asterisk-Users]
whatcable to connect a legacy PBX to a TE410P ?
Alexander Lopez wrote:
Can't anyone stop self-promotion and tell the poor guy what he needs.
Seems to me that SOME self promotion belongs on the biz list, and for
those considered in the inner circle
Http://bugs.digium.com
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Marco
MoutaSent: Wednesday, April 19, 2006 10:38 AMTo:
Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re:
[Asterisk-Users] Music on Hold bug? User disconnect Sip user
I think he wanted now instead of not. Changes the whole meaning of the
question!!!
snip
I not want to add a playback of a file (Please waite
while you are
being transfered) before transfering the call to the cell phone.
Snip, snip
I think he wanted to say:
I NOW want to add a
snip
Can Meetme be made to work with G.729? (I gather not)
IIRC, MeetMe does it 'mixing' using SLIN (Signed Linear, * should
transcode to/from g.729 to SLIN.
If a call comes in (internally or externally), the call comes in as a
G.729 call, which then re-negotiates to a G.711u call when if
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