Yes, it is possible. I'm using PtP and TE mode at home with chan_misdn.
Hans
Ralf Mueller schrieb:
Hello,
can someone on the list confirm, that it is possible to connect a FritzCard to an
Anlagenschluss, when I use the mISDN driver?
I have read a number of posting and articles, that this is
HiWhen i ran the below command on vicidial dialer:[EMAIL PROTECTED] ~]# tethereal -i eth0 -a duration:300 -w sample.capCapturing on eth0320167147496 packets droppedon net i found: When i ran Acterna PVA-1000 on
sample.cap it showed Max Jitter about 20 % and packet loss and echo as major cause of
hello
all
Isit possible to
send special informations to a phone after it registered?
i want to send some
config infos to the phone after it registered to the *.
Is that possible?
And if yes how?
regards
rene
___
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2006/4/23, Armin Schindler [EMAIL PROTECTED]:
On Sun, 23 Apr 2006, Olivier Krief wrote: 2006/4/21, Armin Schindler [EMAIL PROTECTED]:But if you want to forward a call (which was already accepted by Asterisk)
to another CAPI application, it is not possible. (Well, Eicon has a special driver
Olivier Krief wrote:
When writing receiving a fax over CAPI, do you mean receiving a fax
over CAPI with Asterisk and processing it with spandsp ?
No, with a full Eicon Diva 4BRI card, it does hardware faxing. Instead
of using rxfax (which uses spandsp), you'd use capicommand(receivefax)
Hi!
I am using asterisk with two tdm400p cards.
Sometimes (one call out of ten), when a call comes in and is taken,
there is some terrible noise for a short time in the line (for about a
second).
Both partys can hear the noise. And sometimes the call has to be hung
up, because the noise doesn't
On Mon, 24 Apr 2006, Olivier Krief wrote:
2006/4/23, Armin Schindler [EMAIL PROTECTED]:
On Sun, 23 Apr 2006, Olivier Krief wrote:
2006/4/21, Armin Schindler [EMAIL PROTECTED]:
But if you want to forward a call (which was already accepted by
Asterisk)
to another CAPI
Hi,
I have read many postings but still can not understand - is it
possible the X100P to detect a polarity reverse, when the call is answered and
when it ends?
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On Thursday 20 April 2006 19:22, Kevin P. Fleming wrote:
A better solution is to set the PRI hangup cause before dropping the
incoming call; if you set the hangup cause to 'number not assigned'
then your telco's switch will play its normal intercept message to
the caller.
Thank you! This
hi all,
is there a kind of application can let asterisk call out
fellows, and invite them to come to join the meetme.
these fellows do not need to call in asterisk , just wait for a call.
3x
Hi Armin!
Armin Schindler wrote:
On Fri, 21 Apr 2006, Klaus Darilion wrote:
Hi!
I've forgotten to ask an important question:
Does Diva Server V-4BRI with Asterisk support BRI P2P and P2MP mode?
Yes, and each port can be configured separately.
I guess also NT/TE can be configured for each
You can just call them from your dialplan and make them join a meetme room.If by application you mean a frontend, you can use web meetme (juste search for web meetme) to invite new participant from a web browser.
b.en.qOn 4/24/06, welemon lee [EMAIL PROTECTED] wrote:
hi all, is there a kind of
On Mon, 24 Apr 2006, Klaus Darilion wrote:
Hi Armin!
Armin Schindler wrote:
On Fri, 21 Apr 2006, Klaus Darilion wrote:
Hi!
I've forgotten to ask an important question:
Does Diva Server V-4BRI with Asterisk support BRI P2P and P2MP mode?
Yes, and each port can be
Welcome to the Asterisk users community!
Asterisk is the leading Open Source Telephony platform,
with support both for classical telephony and IP telephony.
Asterisk.org is a fast moving project. New code is added every
day.
Our community is also growing
In article [EMAIL PROTECTED],
welemon lee [EMAIL PROTECTED] wrote:
hi all,
is there a kind of application can let asterisk call out
fellows, and invite them to come to join the meetme.
these fellows do not need to call in asterisk , just wait for a call.
You could try adapting
1. u need to schedule the call -- u can do it with something like this:
http://www.voip-info.org/wiki/view/Asterisk+tips+wake-up
2. just call all the participants. check out the GOTO or G in Dial()
application. it will send the called peer to an extension u want. u just
need to make them
After my ongoing experience of being a customer of Currys (part of
Dixons Group PLC), I can only suggest that NOONE SANE should ever
purchase anything from any part of Dixons Group PLC, including Currys,
Dixons PCWorld MasterCare.
They are a total waste of time/money. Liars, Corporate liars,
Hi all,
Where can we find a roadmap of asterisk 1.4 release ?
Harry
--- Olle E Johansson [EMAIL PROTECTED] a écrit :
Welcome to the Asterisk users community!
Asterisk is the leading Open Source Telephony
platform,
with support both for classical
Hi list!
I'm trying to setup a new * server with a TE210P in a Dell 2850 box. I'm
using zaptel version 1.2.5, the linux flavour I'm using is CentOS 4.
In a previous thread I read about the results I should expect from
zttest. On my home box (using the crappy Asus A7V600) I got really bad
Hello,
What's wrong ?
make install
.
options torisa base=0xd
alias char-major-196 torisa
alias wcfxs wctdm
alias wct2xxp wct4xxp
if [ -d /etc/modutils ]; then \
/sbin/update-modules ; \
fi
depmod: *** Unresolved symbols in
/lib/modules/2.4.27-2-386/misc/wctdm24xxp.o
depmod: ***
I think it is correct. Isn't that why they call it a Smart Jack? I've
only ever seen a regular cat5 cable used from the Smart Jack to the
device (router/PBX/CSU/DSU/whatever).
I believe the point of the smart jack is, amongst other things, to allow
for the use of readily available cables.
I
I've just completed downloading and installing [EMAIL PROTECTED] on my already
running
server, and came to view the config areas, however I already have Apache
installed, and operating on the server, so viewing port 80 takes me straight
to the relevant apache served page. Is there a way to
Hi Friends,I want to implement VOIP PBX service in my office. I have 10 computers and a server. All computers are Pentium IV processors with 512 MB RAM. All employee computers have Windows 2000 Professional OS and Server computer Windows 2000 Professional and Fedora Core 5 Linux OS. I have a VOIP
Are there some instructions how to solve problems that produce some typical
error messages in asterisk? For example, if I don't use iax, dundi or mysql
logging, every time I start asterisk I'll get several error messages. How What
can I do to disable loading those files?
Here re some error
Hi,
I am currently running several * boxes on 1.0.9 with HFC chipset ISDN
modems using i4l's hisax driver and chan_modem.
Will I be able to use my existing chan_modem setup with 1.2.4 or 1.2.7
or will I need to change it to use bristuff or chan_capi?
I want to do the upgrade with as little
Yes its possible, just create different contexts for each organisation.
Bails
Michiel van Baak wrote:
On 13:40, Thu 20 Apr 06, Douglas Garstang wrote:
Does AMP also let you split up each charity so that each only has access to
manage their own content? That seems to me to be a pretty big
Hi,
I wish to use the manager API to make an outbound call to a sip
url,subsequently play a prompt and hangup.Any hints on how to acheive
this/feasability will be much appreciated.
Regards,
Ajit
___
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Marnus van Niekerk wrote:
Hi,
I am currently running several * boxes on 1.0.9 with HFC chipset ISDN
modems using i4l's hisax driver and chan_modem.
Will I be able to use my existing chan_modem setup with 1.2.4 or 1.2.7
or will I need to change it to use bristuff or chan_capi?
I want to do
I just started another html process running as user asterisk and group
asterisk and changed the listen directive to another port. I am not
running asterisk at home, but freepbx which is very similar, so I am
told.
on Monday 04/24/2006 James Nunnerley([EMAIL PROTECTED]) wrote
I've just
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
Please has anyone on this list had experience with getting Quintum
equipment to talk to Asterisk? Specifically a D3000 in my case.
It is refusing to register and I'm out of ideas.
Any help appreciated.
Neil
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--- [EMAIL PROTECTED] a écrit :
Hello,
What's wrong ?
make install
.
options torisa base=0xd
alias char-major-196 torisa
alias wcfxs wctdm
alias wct2xxp wct4xxp
if [ -d /etc/modutils ]; then \
/sbin/update-modules ; \
fi
depmod: *** Unresolved symbols in
--- [EMAIL PROTECTED] a écrit :
Hello,
I read
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+SIPredirect
I wish to configure asterisk as a redirect server.
I have badly understood this command .
ASTERISK
|
sip agents nated
--- [EMAIL PROTECTED] a écrit :
Hello to all,
Is there an how-to for asterisk and setting up a
t38
fax gateway (SIP) ?
I look at http://bugs.digium.com/view.php?id=5090 to
patch asterisk chan_sip.c file.
What are the next steps to get a t38 fax gateway
with
asterisk ?
Regards
Hi all
I noticed that most caller are quite confused by the standard voicemail
announcement text. Especialy as the number read is the 'internal' number.
Callers often hang up because they think having called the wrong number when
they hear the announcement.
Is there a way (like in many other
Tomislav Parčina wrote:
Are there some instructions how to solve problems that produce some typical
error messages in asterisk? For example, if I don't use iax, dundi or mysql
logging, every time I start asterisk I'll get several error messages. How What
can I do to disable loading those
Rich Adamson wrote:
snip
For short runs, the use of cat5 vs proper T1 cables isn't likely to
have any impact unless there is a fair amount of induction from
electrical noise, etc. That can take the form of florescent fixtures,
transformers, older CRT monitors, etc, etc.
On longer runs, the
Hi Folks,using this:exten = x,1,Playback(audio,noanswer)exten = x,2,Answerexten = x,3,BackGround(out)exten = x,103,HangupI'm not billed and remain connected, but the file 'audio' is not played...well I do not ear it.
But after, it pass correctly to answer and I can
Leo Ann Boon wrote:
Rich Adamson wrote:
snip
For short runs, the use of cat5 vs proper T1 cables isn't likely to
have any impact unless there is a fair amount of induction from
electrical noise, etc. That can take the form of florescent fixtures,
transformers, older CRT monitors, etc, etc.
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
Hi,
We got following surplus for sale:
TE210P $700
TE410P $1100
TE411P $1950
Bundle (All 3 cards) please make an offer :)
Cards not used except for development testing.
___
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Asterisk-Users
I've noticed that when app_queue.so is reloaded(or just a reload command is
used) that all queue members that were paused are automatically unpaused. Is
there a workaround for this? (Note, I use statically defined callback agents).
--johann
___
I have configured telasip DID with following entried in
sip_custom.conf [telasip] username= (fake) type=peer
secret=x quality=yes nat=yes insecure=very fromuser=
host=gw4.telasip.com #disallow=all #allow=ulaw #allow=alaw
fromdomain=gw4.telasip.com context=from-telasip and Register
Thank you.
It SEEMS to be working fine now as-is with the cranked-up registration time.
When the time comes to tinker with it in the future - I will probably try
working with groups again, or even work something out with astdb. (and,
most likely, end up breaking something that seems to work
Yes, this is possible, but a management nightmare.
-Original Message-
From: bails [mailto:[EMAIL PROTECTED]
Sent: Monday, April 24, 2006 5:16 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Announcement System for a Charity
Yes its
sorry for my english, I did not explain myself correctly. I mean I
downloaded the file Today, never meant to say that the file was
uploaded Today. I know the file is recent enough because i looked
for a change in mfcr2.c source that I know was put there recently.
Regards
On 4/22/06, Anton Krall
2006/4/24, Armin Schindler [EMAIL PROTECTED]:
When using a card with onboard DSPs (or even the software fax of AVM Fritzbinary-only driver) you can do faxing with the CAPI interface. That meansyou don't get the audio data stream, you get the fax-data instead which can
be save in a file.In that
My telco used cat5 as well for the demarc to CPE. It's also with noting that
many channel banks, such as my Atlas, and zapata.conf itself also have
parameters to allow you to tune the gains to compensate for cable signal
loss. I've never had to touch them, and my CPE is about 300 feet from the
PRI
On Mon, Apr 24, 2006 at 04:46:29PM +0530, Ajit spake thusly:
Hi,
I wish to use the manager API to make an outbound call to a sip
url,subsequently play a prompt and hangup.Any hints on how to acheive
this/feasability will be much appreciated.
I'm no expert, but it looks simple enough to me -
Benoit Panizzon wrote:
Hi all
I noticed that most caller are quite confused by the standard voicemail
announcement text. Especialy as the number read is the 'internal' number.
Callers often hang up because they think having called the wrong number when
they hear the announcement.
Is there
Yes it is possible - check out the Asterisk manual or nice book from
O'Reilly - Asterisk PBX (The Furute of telephony)
Marcel
Crazy Boy wrote:
Hi Friends,
I want to implement VOIP PBX service in my office. I have 10 computers
and a server. All computers are Pentium IV processors with 512 MB
For hardware check out this page:
http://www.digium.com/en/products/hardware/
Marcel
Crazy Boy wrote:
Hi Friends,
I want to implement VOIP PBX service in my office. I have 10 computers
and a server. All computers are Pentium IV processors with 512 MB RAM.
All employee computers have
when did you trash it
since they changed to freepbx they have added a new permissions based login and they have also split out users and devices.
Very nice for setting up a company with different divisions.
You can give the support extensions to the support manager to deal with.It is well worth
Hi all,
I am running 1.2.7.1 asterisk on FC3. Every thing works except dtmf
detection on my Zap lines. I am using a TE411P with isdn NI2. Thnx.
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To UNSUBSCRIBE or
Eric ManxPower Wieling wrote:
Davi-Ann wrote:
When I set asterisk to to sequence the lines as Ground Start the
system is not starting. It is giving the following error Invalid
Argument 22
Do you have any ideas about this.
Any help or assistance appreciated.
I don't think Digium's
On 4/24/06, Johann [EMAIL PROTECTED] wrote:
I've noticed that when app_queue.so is reloaded(or just a reload command is
used) that all queue members that were paused are automatically unpaused. Is
there a workaround for this? (Note, I use statically defined callback
agents).
That sounds
Alexander Lopez wrote:
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.
Good thing he doesn't work for a
RTFM
On 4/24/06, Benoit Panizzon [EMAIL PROTECTED] wrote:
Hi all
I noticed that most caller are quite confused by the standard voicemail
announcement text. Especialy as the number read is the 'internal' number.
Callers often hang up because they think having called the wrong number when
On Mon, 24 Apr 2006, Rich Adamson wrote:
Alexander Lopez wrote:
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
Hello everybody,
does anybody use P100P FXO card on POTS lines in Slovakia, Bohemia
(Czech rep.), Poland, Hungary...?
I need to know if those cards work especially in Slovakia or if you can
reccomend FXO cards for Slovak POTS lines.
Thanks,
Marcel
___
You can't use round robin DNS. Round robin DNS will cause every SIP packet to
potentially go through a different static path, which will break things.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Sent: Saturday, April 22, 2006 5:27 AM
To: 'Asterisk Users
Hi,
I have an URA that
says to my customers: "Dial 1 for support, dial 2 for sending a
fax".
This URA starts with
g729, but when the call is transferred to the RxFax, it should be converted into
g711, for the fax to work.
Is there a way to
solve this???
Thank
you!Dov
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,
Hi,
Well, I have a big problem with Asterisk, my problem is that when I'm in a
conversation, using zap channels, in a moment the line has a interferce
that produce a sound in the conversation, this sound is a electratical
sound I think, I was reading about that and I found that the utility
Using an SMP kernel will fix the interrupt sharing, you could also disable
hyperthreading and set runlevel 3. FWIW I almost exclusively use Poweredge
850 for my * servers with a third party sata raid controller if raid is
required. Never had any problems.
Craig
- Original Message -
I'd like to increase the speed of the Asterisk sound files. Miss Alison talks a
bit slow.
I can use sox to increase the speed, but then the pitch changes and she starts
to sound like a chipmunk. Any audio experts out there know how I can increase
the speed a little bit, and change the pitch
Essentially true, but the impedance of a T1 cable is different from Cat5
cables, which is one of the primary factors in limiting distance. Has
nothing to do with the twists.
Shielded vs non-shielded has to do with the environment, and how much
electrical noise there is near the T1 cable.
How about using LVS?
http://www.ultramonkey.org/3/topologies/lb-overview.html
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Douglas Garstang
Sent: lunes, 24 de abril de 2006 17:12
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
Hi,
I dont want to have in the SIP HEADER the CALLERID(name) (the Display Name)
for the initial INVITE to an SIP proxy.
If I use SET(CALLERID(name)=) the display-name is asterisk.
I want to have the SIP HEADER like this: FROM:
sip:CALLERID(number)@domain.tld
thanks
best regards
Thomas
Thanks for the hints and tips.
While you are familiar with the 2850, I am using the PERC raid controller
but guess this shouldn't make any real difference.
I used the middle PCI slot for the TE210P, do you use any particular slot.
I will disable HyperThreading and the box was already running
Well, for a start, there's a single director, which means a single point of
failure. Really, I wonder why they even bother.
-Original Message-
From: Sergio García Murillo [mailto:[EMAIL PROTECTED]
Sent: Monday, April 24, 2006 9:44 AM
To: Asterisk Users Mailing List - Non-Commercial
Has anyone had any luck getting listed in directory
assistance when your number is ported from
For example, I have an asterisk box for a client, that is
also shared with another client in the same building. The CLEC provided PRI and
numbers (including the ported #s from Verizon) as the
On Monday 24 April 2006 11:42, Ken Godee wrote:
cat5 is NOT T1 cable and if any telco/vendor tried
to install it in my location I'd have them pull it and
put in the proper cabling.
T1 cable is generally Cat3 is it not? That's certainly how the old T1s loops
were run between the CO and the
On Mon, 24 Apr 2006, Douglas Garstang wrote:
You can't use round robin DNS. Round robin DNS will cause every SIP packet to
potentially go through a different static path, which will break things.
Huh? Has this happened to you in practice?
--
Aaron Daniel
Computer Systems Technician
Sam
Had a Pent 4 server running fc3 crash (kernel panic) and am rebuilding
from scratch. I installed FreePBX (CentOs) from scratch and asterisk was
running, but had not yet been configured. It too crashed with a kernel
panic. Ran memtest for 24 hours; no errors or issues uncovered.
I then noticed
[EMAIL PROTECTED] wrote:
Where can we find a roadmap of asterisk 1.4 release ?
Harry... please use proper mailing list etiquette when posting to these
lists. It is very tiresome to see you quote an entire long message,
without changing the subject, and insert a one-line unrelated comment at
the
Rich Adamson wrote:
Oh, and if shielded T1 cable is used, the shield at each end of the
cable must be grounded. (Let's see how many can figure out how to do
that via an rj45 plug. ;)
You use shielded plugs and jacks, of course :-) That is why the
TE405P/TE410P have shielded jacks (as of about
Andrew Kohlsmith wrote:
Insulation (especially such thin insulation) does not prevent crosstalk.
Distance, shielding and tighter twists do.
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
Well, are you running an SMP box, or is it just hyperthreaded?
I know there are issues with running an SMP kernel on a machine that's
only HT.
On Mon, 24 Apr 2006, Rich Adamson wrote:
Had a Pent 4 server running fc3 crash (kernel panic) and am rebuilding from
scratch. I installed FreePBX
Thomas Winter wrote:
Hi,
I dont want to have in the SIP HEADER the CALLERID(name) (the Display Name)
for the initial INVITE to an SIP proxy.
If I use SET(CALLERID(name)=) the display-name is asterisk.
Just a guess, try:
SET(CALLERID(name)= )
Doug
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: Aaron Daniel [mailto:[EMAIL PROTECTED]
Sent: Monday, April 24, 2006 10:06 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Connecting to a cluster of SIP servers
On Mon, 24 Apr 2006, Douglas Garstang wrote:
On Monday 24 April 2006 11:12, Douglas Garstang wrote:
You can't use round robin DNS. Round robin DNS will cause every SIP packet
to potentially go through a different static path, which will break things.
Um... The media gateways do not do a DNS lookup for every packet they send
out... At
Andrew Kohlsmith wrote:
On Monday 24 April 2006 11:42, Ken Godee wrote:
cat5 is NOT T1 cable and if any telco/vendor tried
to install it in my location I'd have them pull it and
put in the proper cabling.
T1 cable is generally Cat3 is it not? That's certainly how the old T1s loops
were run
On Monday 24 April 2006 12:13, Kevin P. Fleming wrote:
Minor point: isn't it safer to only ground the shield on one end?
Yes, you *never* shield both ends. That can cause ground loops and add to the
long list of what the..? head-scratching problems that telephony has.
As to WHICH end to
On Mon, Apr 24, 2006 at 09:41:40AM -0600, Douglas Garstang spake thusly:
I'd like to increase the speed of the Asterisk sound files. Miss Alison talks
a bit slow.
I can use sox to increase the speed, but then the pitch changes and
she starts to sound like a chipmunk. Any audio experts out
-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
Hi All:
I used FreePBX to configure Asterisk, and tables are create in MySQL by
using FreePBX install script.
I created two x-lite softphone accounts by using FreePBX, they are
stored in table sip as friend.
I followed wiki doc to edit the extconfig.conf file.
I can not get those two softphone
Hey, all. I'm trying to set my CID such that, internally, I see a
four-digit extension (which is also handy when checking VM), but
externally, I see the full 10-digit number. So I plugged these lines into
my extensions.conf:
exten = _XXX,1,GotoIf($[ ${CALLERIDNUM} != 1625]?4:2)
exten =
Ok,
Im not a developper but what do you think of both a
wish list .
Harry
To answer your question: there is no roadmap for
1.4. We just began the
'scheduled release' cycle with this release, and we
are still trying to
feel our way into the process and learn how much
work we can
Huh? Has this happened to you in practice?
It sure has. Polycom phone queries DNS for domain.com and gets round robin IP
of 192.168.10.1. It sends a REGISTER request to that IP. Asterisk at
192.168.10.1 sends back a 407 Proxy Auth required. The polycom phone then
queries DNS again and gets
Kevin P. Fleming wrote:
Andrew Kohlsmith wrote:
Insulation (especially such thin insulation) does not prevent crosstalk.
Distance, shielding and tighter twists do.
Ever looked at the underground cable in the street outside your
building? If it's more than 20 years old, it's
On Monday 24 April 2006 12:39, Alexander Lopez wrote:
And if you don't believe the 'high-end' part brush up against a 66-block
while your well grounded, you will be singing in the high-end!!! At
that voltage I think the differential created by the twists would cancel
anything including small
On Monday 24 April 2006 12:42, Rich Adamson wrote:
The T1/E1 interface spec's are typically 75 ohm balanced (BNC, E1), 100
ohm balanced, etc.
Ahh yes, this is true. Is that a typical spec for even POTS lines?
I've never bothered to check to see if cat5 cables use the appropriate
mating
Rich Adamson wrote:
I've never bothered to check to see if cat5 cables use the appropriate
mating twisted pairs or not. Since the pinouts are different for cat5 vs
T1 cables, I'd have to guess a single strand is used from two different
twisted pair groups. That wouldn't be cool, but in short
I've never bothered to check to see if cat5 cables use the appropriate
mating twisted pairs or not. Since the pinouts are different for cat5
vs
T1 cables, I'd have to guess a single strand is used from two
different
twisted pair groups. That wouldn't be cool, but in short runs it
probably
Alexander Lopez wrote:
6-8 spans? That's the number that I have been trying to get, and why the
limit. Is it X-talk?
I think so. I've had clients before who had to have spans brought in via
different routes even though the pairs in the underground cable were in
otherwise acceptable condition.
Hi,
I've got this PXA270 board set-up with Linphone 1.2.0 and am trying to get
linphonec to work with Asterisk.
I have the echo test working, but when I dial in to this, to voicemail or
anything else using Playback() to play a sample, I hear nothing for ages (10-15
secs) and then
When someone calls into our asterisk server over a PSTN line, dials an
extension and then hangs up, the SIP phone related to the given
extension will ring about 4 or 5 times before asterisk shows that the
channel has been hung up in the console. This isn't such a big deal
on its own, but what's
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