I'm connecting a T1 PCI card to a Nortel Option 61 switch T1 card. My
Sangoma A102D shipped with 2 T1 cables - which I assume are straight
through. Do I need to make crossover cables for this scenario?
As people answered here you need a crossed cable; Note that T1/E1 cables are
different
Hello,
A few days ago I've posted two questions about Polycom phones: How to access
corporate phone directory from the phone and how to use a conference server
with it. After I got zero responses I tried openning a support call in
Polycom's site. Here are the replies I got from them:
-
On Fri, Oct 26, 2007 at 04:52:07PM -0800, Mojo with Horan Company, LLC wrote:
I don't have T1 but it seems that the first time I run ztcfg (or in
fact, the zaptel startup script runs it for me) it fails.
What distribution is it?
RHEL4 / CentOS4 has an early udev version that seems to react
Dear all
I have small lan and i have configure hardphone with my
asterisk with one E1 PSTN line now i have configue to use canreinvite=yes in
sip.conf
If i user conreinvite=no then my RTP goes throgh asterisk means asterisk come
in media path
and if i user
In what amount of time does 100 Trying message have to be
sent to asterisk? I see asterisk retransmitting the INVITE
message multiple times before receiving the 100 Trying message.
The INVITEs are retransmitted based on a timer T1, which starts at a default
of 500 ms and then exponentially
On Sat, 27 Oct 2007, satish patel wrote:
My all phone in LAN not behind the NAT so guessest me what option would
be best for my setup
That depends on what you think the best option for your setup is. :)
No, seriously. On the one hand, in a LAN environment, it's probably
easier for
Hi,
Yes, i have used it for T.38 faxing.
Thanks,
Vivek
On 10/26/07, Nasir Iqbal [EMAIL PROTECTED] wrote:
Hi,
Have you tried Callweaver http://www.callweaver.org
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Well, unfortunately for you, that is the exact opposite of
the philosophy of the Asterisk codebase. Every attempt is
made to genericize the channel driver interface so that you
do not need to know the details of the underlying driver.
Where we have failed to do so in the past, we are
http://www.faqs.org/rfcs/rfc3398.html
The conversion is lossy. More than 1 SIP cause code is
mapped to a Q.931 cause code (in Asterisk at least). See
hangup_sip2cause() in chan_sip.c
True. The conversion is lossy in that respect and most of the times
semantically incorrect simply because
Hi friends,
We have are getting SIP/2.0 407 Proxy Authentication Required on Invite pakcet
once Nokia E65 trying to dial number. But it can recive well from other caller.
We tried to disable secrete and it worked fine. But we have lot of users and
disabling secrete is risky.
Interesting thing
Ok..so how would the CALLED and CALLERID ID be presented to Asterisk when
using PRI signaling.
Mike
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Lyle Giese
Sent: Friday, October 26, 2007 5:54 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
john beaman wrote:
For pinout info, check out: http://www.asteriskdocs.org/cables/
John Beaman
Telecom Specialist II
Voice Telecommunications Services Department.
Good Samaritan National Campus
605-362-3331
[EMAIL PROTECTED] 10/26/2007 4:01:29 PM
Michelle Dupuis wrote:
Sir,
I have configured chanspy and extenspy to listen call on any extension but in
both case i am unable to hear voice only silence is there.
Rajeev.
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asterisk-users mailing list
Michelle Dupuis wrote:
Ok..so how would the CALLED and CALLERID ID be presented to Asterisk
when using PRI signaling.
When you are using a PRI, you'll see something like:
Accepting call from '248xxx' to '734xxx' on channel 0/1, span 1
So, for the inbound, you'd have an entry that
The same as any other zap channel does. That is part of the magic of
the zaptel drivers.
Lyle
Michelle Dupuis wrote:
Ok..so how would the CALLED and CALLERID ID be presented to Asterisk
when using PRI signaling.
Mike
CCMANAGER 0.5 released!!
NOTE:
this is a previous alpha release, maybe there is some customization to
do on the settings files,
i can't write a clear and complete howto at the moment
I don't have released upgrades in the last months but the project is still alive
i'm too busy at the moment, i'm
Ok - that's great. I see how the destination number will match to the exten
value, but how do I access the from number '248xxx'?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Doug Lytle
Sent: Saturday, October 27, 2007 9:44 AM
To: Asterisk
I've got a few Cisco 7921 wifi phones to use with an Asterisk pilot. For
the purpose of the pilot (i.e. low investment) I want to configure the
phones from the keypad.
Each phone shows settings locked! whenever I try to edit the network
profiles. I can't seem to unlock them! Hopefully there
Michelle Dupuis wrote:
Ok - that's great. I see how the destination number will match to the exten
value, but how do I access the from number '248xxx'?
exten = s,1,GotoIf($[${CALLERID(number)} = 248xxx ]?2:3)
Doug
--
Ben Franklin quote:
Those who would give up Essential
I've got a few Cisco 7921 wifi phones to use with an Asterisk pilot. For
the purpose of the pilot (i.e. low investment) I want to configure the
phones from the keypad.
Each phone shows settings locked! whenever I try to edit the network
profiles. I can't seem to unlock them! Hopefully
Alex Balashov wrote:
On Sat, 27 Oct 2007, satish patel wrote:
My all phone in LAN not behind the NAT so guessest me what option would
be best for my setup
That depends on what you think the best option for your setup is. :)
No, seriously. On the one hand, in a LAN
On Saturday 27 October 2007 08:14:05 Torbjörn Abrahamsson wrote:
Well, unfortunately for you, that is the exact opposite of
the philosophy of the Asterisk codebase. Every attempt is
made to genericize the channel driver interface so that you
do not need to know the details of the
I *STRONGLY* recommend that you do NOT use realtime extensions. If you
want a dynamic dialplan, the correct way to do it is to segregate your
logic
and your data (via something like func_odbc), not to stick all of your
logic
into a database.
--
Tilghman
I'm confused by your statement
Pat Phelan posted this on Facebook - thought the SIP functionality would
interest some people here as well.
Jajah inks large Japanese dealJajah inks large Japanese dealJajah inks large
Japanese deal (Pat Phelan http://www.facebook.com/profile.php?id=754567533 )
Michelle Dupuis wrote:
Ok..so how would the CALLED and CALLERID ID be presented to Asterisk when
using PRI signaling.
The CALLING and CALLED numbers are sent automatically during the call setup.
CALLING NAME is usually sent right after the call setup happens.
On Saturday 27 October 2007 11:07:32 JR Richardson wrote:
I *STRONGLY* recommend that you do NOT use realtime extensions. If you
want a dynamic dialplan, the correct way to do it is to segregate your
logic and your data (via something like func_odbc), not to stick all of
your logic into a
Hello,
I want to use a 4 port ISDN card from EICON/DIALOGIC in our asterisk server.
The system is a Ubuntu 7.10 with kernel 2.6.23.1. The compilation of the
kernel finishes without any problems. I have downloaded and installed
the deb-source package that EICON/DIALOGIC offers. Th installation
Tilghman Lesher wrote:
I *STRONGLY* recommend that you do NOT use realtime extensions. If you
want a dynamic dialplan, the correct way to do it is to segregate your logic
and your data (via something like func_odbc), not to stick all of your logic
into a database.
Should this be taken
Russell Bryant wrote:
Alejandro Cabrera Obed wrote:
Dear all, I have Asterisk 1.4.13 and I need to use encryption among
Asterisk and my SIP users, and with the RTP data interchanged among
users. I prefer the use of ZRTP/SRTP because we use Twinkle and
X-Lite/Zfone as our voip clients and they
Or just generate config files (or parts of config files) from a
database dynamically.
On Sat, 27 Oct 2007, Brian Capouch wrote:
Tilghman Lesher wrote:
I *STRONGLY* recommend that you do NOT use realtime extensions. If you
want a dynamic dialplan, the correct way to do it is to segregate
The only place where it is reasonable to customize is in the
specification of the channel in the configuration file.
That is where
you would customize, for example, whether DTMF is inband,
SIP INFO, or
RFC 2833, as well as what codecs will be negotiated for that
particular
On Sat, 2007-10-27 at 19:49 +0200, Stefan Guenther wrote:
Hello,
I want to use a 4 port ISDN card from EICON/DIALOGIC in our asterisk server.
The system is a Ubuntu 7.10 with kernel 2.6.23.1. The compilation of the
kernel finishes without any problems. I have downloaded and installed
the
On Saturday 27 October 2007 13:10:37 Brian Capouch wrote:
Tilghman Lesher wrote:
I *STRONGLY* recommend that you do NOT use realtime extensions. If you
want a dynamic dialplan, the correct way to do it is to segregate your
logic and your data (via something like func_odbc), not to stick
DL == Doug Lytle [EMAIL PROTECTED] writes:
DL Michelle Dupuis wrote:
Ok - that's great. I see how the destination number will match to
the exten value, but how do I access the from number '248xxx'?
DL exten = s,1,GotoIf($[${CALLERID(number)} = 248xxx ]?2:3)
That works, of course,
That did the trick! It appears that all of the config is retrieved from a
.cnf.xml file, so there wasn't much more I could do at the phone level other
than set the networking parameters.
MD
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Yehavi
I want to use a Fritz AVM ISDN card to create a switch which is
connected to 4 analogue extensions.
I believe I need a 4 port FXS module for that, are there any cheap but
reliable options out there?
Are there some guides that go through the whole process?
/voipfc
I have a customer who wants the Polycoms to display the CallerID name of
the person they called on the phone they are calling from.
The receiving phone gets CID just fine, but the calling phone doesn't
display a name. For instance, if you dialed extension 3000, the Polycom
Displays 3000(3000)
I have added directory creation support from CSV as well as a bug fix.
V0.0.3 is available http://www.wintrisk.com/ppt.html
Yours,
Michael Munger, dCAP
404-438-2128
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
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On Sat, 27 Oct 2007, Stefan Guenther wrote:
Hello,
I want to use a 4 port ISDN card from EICON/DIALOGIC in our asterisk server.
The system is a Ubuntu 7.10 with kernel 2.6.23.1. The compilation of the
kernel finishes without any problems. I have downloaded and installed
the deb-source package
On Sat, 27 Oct 2007, Patrick wrote:
On Sat, 2007-10-27 at 19:49 +0200, Stefan Guenther wrote:
Hello,
I want to use a 4 port ISDN card from EICON/DIALOGIC in our asterisk server.
The system is a Ubuntu 7.10 with kernel 2.6.23.1. The compilation of the
kernel finishes without any problems. I
I am trying to get distinctive ringing going again with these phones,
depending on the outside line the call comes in on.
I had a working 1.0.x Asterisk setup using:
SetVar(ALERT_INFO=http://127.0.0.1/Bellcore-dr2)
Which used the short quick rings.
In Asterisk 1.4, I have tried several things,
Michael Munger wrote:
I have a customer who wants the Polycoms to display the CallerID name
of the person they called on the phone they are calling from.
You'll want to see this bug:
http://bugs.digium.com/view.php?id=8824
Doug
--
Ben Franklin quote:
Those who would give up
Hello,
I am experiencing difficulty registering my Snom 320 phone with Asterisk
1.4.13, and have been receiving the same transport error messages on the
phone as described in this forum post:
http://forums.digium.com/viewtopic.php?p=40554highlight=sid=b6d7fd216103dcdafb0b995aff03f07f
Are there
I've been looking around for an example of a method of reading back a caller
ID value, but I haven't found anything that doesn't use Festival. I'd rather
not resort to the Mr. Roboto voice if I can avoid it.
Playback of the numbers one at a time is perfectly fine, so I'd like to use
the default
On Saturday 27 October 2007 11:19:05 pm arkda wrote:
I've been looking around for an example of a method of reading back a
caller ID value, but I haven't found anything that doesn't use Festival.
I'd rather not resort to the Mr. Roboto voice if I can avoid it.
Playback of the numbers one at a
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