> "RC" == Ricardo Carvalho <[EMAIL PROTECTED]> writes:
RC> I've also tried to do it using different contexts, but it still
RC> doesn't work. I've done like this:
RC> [default]
RC> exten => secretary_extension,1,Dial(SIP/secretary_extension)
RC> exten => boss_extension,1,Dial(SIP/secretary_ext
I tried to look at the code in Trixbox but when the option 'confirm' is
selected in the follow me properties screen, no code is generated and the
call goes dead. Is there a trick to get the code generated?
_
From: Philippe Lindheimer [mailto:[EMAIL PROTECTED]
Sent: Saturday, March 17
Hello,
On most SIP phones a conference call is done on the phone and is limited to 3
participants. Polycom phones has a configuration option to use a conference
server instead of the internal conferencing feature. I guess I need some
conference server; any experience with such a server which ca
Hi everybody,
I have a problem which I cannot eliminate on my own. Has anybody any idea
for the following:
I am using the asterisk-version from Debian-Testing (1.2.13) with the
latest chan_capi (also tried an older version).
When using the Capi-Channel, everything works fine except from the sou
Rizwan Hisham wrote:
Im using asterisk1.4.0 . declaring type=peer solves the problem. but if
anybody knows why its not working for type=friend, plz share.
Please try 1.4.1, this should be fixed.
--
Tomislav Parcina
[EMAIL PROTECTED]
___
--Bandwidth
17 mar 2007 kl. 12.21 skrev Rizwan Hisham:
Hi,
I have declared my sip users call-limit=2 and type=friend. When any
user recieves a waiting call while already in a conversation, the
peer call counter is set to 2.The problem is that, the counter is
not reset to zero after hangup and becoz o
may be this will help :)
http://www.celliax.org
Good luck
___
--Bandwidth and Colocation provided by Easynews.com --
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Michiel ten Hagen wrote:
I also see the need for an asterisk only wiki.
That is the reason I have started one at http://www.asterisk-wiki.org/
the information from voip-info.org is being added now as a starting
point.
Ofcourse the success of this wiki will depends on the information
posted.
Is there any technical difference between a T1 cable and a cat5e patch
cable as far as using them with Digium T1/E1 cards? Can PRI circuits
terminating at a smart jack connect successfully to Digium cards using
straight through CAT5e cables? If so, are they using all of the pins in
the cable?
Than
You can use a patch cable, yes. The T1 will look to use pins 1,2,4
and 5 while Ethernet will typically use 1,2,3 and 6 provided you're
not using POE or something simliar that requires additional pins.
On 3/18/07, Jeronimo Romero <[EMAIL PROTECTED]> wrote:
Is there any technical difference betwee
A common Cat5 straight through cable will work fine.
T1s use 1 and 2 (rx tip and ring) and 4 and 5 (tx tip and ring) for signals.
A T1 loopback plug would be wired 1 to 4 and 2 to 5.
They come in handy for testing T1 cards or for providing a hard loop
for the telco.
Tom
At 05:42 PM 3/18/200
When phone A registers, I want phone B to ring, when picked up, it should
call phone A and connect the phones.
Translated: When GF in Mexico powers up laptop where soft iax-phone
registers automatically, I want to talk to her asap :-)
How to?
Leif
___
I assume that I would need to cross these pins over if I were going from
t1 card to t1 card. Is this correct?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tom
Sent: Sunday, March 18, 2007 7:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussi
Yes. At that point, you're looking for a T1 "cross-over".
The pinout is as follows:
1
4
RX/Ring/- <-->TX/Ring/-
2
5
RX/Tip/+ <-->TX/Tip/+
4
1
TX/Ring/- <-->RX/Ring/-
5
2
TX/Tip/+ <-->RX/Tip/+
3
3
Shield/Ret
So a regular cross over cable wouldn't work?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of BJ Weschke
Sent: Sunday, March 18, 2007 10:35 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] T1 cable for Digium T1/E
Leif Neland wrote:
> When phone A registers, I want phone B to ring, when picked up, it should
> call phone A and connect the phones.
>
> Translated: When GF in Mexico powers up laptop where soft iax-phone
> registers automatically, I want to talk to her asap :-)
>
> How to?
I don't really kn
It would be cool if you could add some kind of login script capability to nodes
in sip.conf and iax.conf.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Philipp Kempgen
Sent: Sunday, March 18, 2007 11:16 PM
To: Asterisk Users Mailing List - Non-Commerci
Correct. Because Ethernet cross-over cables are crossing over 1,2,3
and 6; no 1,2,4 and 5.
On 3/18/07, Jeronimo Romero <[EMAIL PROTECTED]> wrote:
So a regular cross over cable wouldn't work?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of BJ Weschke
Se
Hi Artur,
Just follow the information Moises had recommended you and for sure this will
work. The default configurations that was exampled in the document is just
fined and suited with Nortel Meridian. Just be sure that your Nortel MFC Card
is installed and working in good condition with up to
Hi Giorgio,
I guess it will be more benefitial to all old version users to read some
information regarding new version like * 1.4. In this way, they will be
encourage, or probably have an idea whether to upgrade or not based on all the
concerns that was posted. I for one still using 1.2.13 but
Learn to search - this question has been asked and answered at least 100
times
There are also all sorts of cable references on the Internet
Can you spell "google" ?
Jeronimo Romero wrote:
So a regular cross over cable wouldn't work?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:
Jeronimo Romero wrote:
> It would be cool if you could add some kind of login script capability to
> nodes in sip.conf and iax.conf.
>
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Philipp Kempgen
> Sent: Sunday, March 18, 2007 11:16 PM
> To: Aste
Just noticed that no matter what the line condition is, zttool always
reports "OK", so it's pretty useless. (In contrast, I'd get "Red alert" if I
unplug the line connecting to an X100P.)
I'm using zaptel 1.2.15 on Linux 2.6.15-28 (also tested on 2.6.10).
Yuan Liu
___
The revision on this card was new and my version of zaptel (1.2.11) did
not support my pri signaling for that card. I upgraded zaptel through
svn and all was good.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir
Cohen
Sent: Wednesday, February 28,
I'm not sure what problem you are having, it should be fine on a proper freepbx
install (and as best I know trixbox installs it properly although I don't use
trixbox so can't say first hand). It should work fine, I know plenty of people
who use it - I use it regularly. Feel free to try and catch
jeronimo
there is no difference...
Jeronimo Romero wrote:
Is there any technical difference between a T1 cable and a cat5e patch
cable as far as using them with Digium T1/E1 cards? Can PRI circuits
terminating at a smart jack connect successfully to Digium cards using
straight through CAT5e
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