ignorpat is your friend
On 9/30/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Sun, Sep 30, 2007 at 02:34:01AM -0700, bilal ghayyad wrote:
Dear List;
How can I place a call via Zap/g1 (group) but need to
determine the line (FXO port)
that will go via it?
Simply don't use groups. Use
Hi
I'm looking for a iax client that will run on my htc tytn (windows ce) ..
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Hi everyone,
I'm having an odd problem with one way RTP on SIP to SIP calls.
I have two SIP servers, one is an Asterisk and the remote SIP server
is a Nortel SIP server.
When a call comes to the Nortel server through the PSTN and is routed
to the Asterisk, audio is fine. Two way RTP and no
Chiming in here, I had to return a Polycom to VOIPSupply and the
turnaround time was
basically immediate with no questions asked. They've always done us
right here. OTH, I did have a bad glitch with ATAComm and it took a
while for them to resolve the issue.
That's horrible. I don't buy too many
What version of ODBC does asterisk 1.4 need?
--
-
Chris Stinson
Network Operations Center
ISDN-Net, Inc.
615-221-4200 x103
[EMAIL PROTECTED]
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Is this a SIP connection or a SIP-T one? Not sure (don't have access to
my previous life docs :-), but this seems to be a Session Server Trunks
doing SIP-T, not sure is the configuration you want...Have you tried to
contact their support ?
PS: this c:
Hi
I've configured my asterisk and voicemail all works fine but I want to
restrict call time to voicemail that is when user calls voicemail he
can use voicemail system only for a max of 5 min that is after five
minutes asterisk should disconnect the call.
thanks
Arun
Hi all,
I'm having a problem with latest version of Asterisk.
When I put someone on hold or if I dial an extension with music on hold the
call hangs up after a few seconds when MUOH has changed file to play. Any
thoughts?
Many thanks,
Christian
___
You are right, the remote server is a SIP-T.
I haven't had any problems connecting it to regular SIP servers
thusfar though. Also like I mentioned, I don't have this one-way RTP
problem with an earlier version of Asterisk.
Thanks for your reply,
Örn
On 10/1/07, Julio Arruda [EMAIL PROTECTED]
I do not believe that the web page referenced below states that you need a
license to use Cisco phones with any pbx other than Call Manager. It only
states that you are required to have a license regardless of the protocol
used and their documentation is specifically aimed at Call Manager
Julio,
It seems you had something going there; I disallowed ISUP messages on
the SIP-T server and now I have two way audio.
Thanks a lot for your help!
Best regards,
Örn
On 10/1/07, Örn Arnarson [EMAIL PROTECTED] wrote:
You are right, the remote server is a SIP-T.
I haven't had any problems
Hi,
Many thanks all for the useful tips - I've gone with a (simple!) mySQL table
with a flag in it, indicating the day/night mode, adding the following into
the dialplan:
[external]
; other stuff in here, excluded for clarity
; Include the SJS phone line controls
include = sjs_ctrl
[sjs_ctrl]
In trying to verify licensing requirements I called Tech-Data and spoke to
the Cisco licensing reps there (my company is set up as a reseller through
Tech-Data) and was informed by them that a license for Cisco VoIP phones is
only required if connecting it to a Call Manager or any other Cisco
Andrew Joakimsen wrote:
That's horrible. I don't buy too many IP phones these days, but can
anyone suggest a place better than the scumbags at VoIP supply?
http://www.pcp.ch/ or http://www.digitec.ch/
/Per Jessen, Zürich
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Hi,
I'm trying to register SIP phone with an asterisk serve, failing miserably.
The server is sending 401 Unauthorized responses to the registration
attempts, but every time the phone is re-REGISTERing without authorization.
I'd think this was a problem with the IP phone, except... the very
Just a guess in fact..but..
I'm sure others would love to know how is the NGSS (SST now ?) config
for this purpose, as well as your sip.conf and etc (one note, you are
running SN09 or ISN09 ?
Not sure, but this also would help others out there.. :-)
Örn Arnarson wrote:
Julio,
It seems
Hi Arun -
I've configured my asterisk and voicemail all works fine but I want to
restrict call time to voicemail that is when user calls voicemail he can
use voicemail system only for a max of 5 min that is after five minutes
asterisk should disconnect the call.
Do you mean that you want the
Vahan Yerkanian wrote:
Andrew Kohlsmith wrote:
On Saturday 29 September 2007 18:43:59 Andrew Joakimsen wrote:
That's horrible. I don't buy too many IP phones these days, but can
anyone suggest a place better than the scumbags at VoIP supply?
I don't know about you, but I've had nothing but
On 9/30/07, Andrew Kohlsmith [EMAIL PROTECTED] wrote:
I don't know about you, but I've had nothing but very good results with
VOIPSupply. I didnt do huge business with them, but I have purchased new and
refurb polycoms from them without so much as an ounce of pain.
Ditto - I've never had a
Hi all, I have 2 asterisk box connected with IAX trunk.
One box have connected a SIP phone and the second have a TDM card with
one analog phone.
When from SIP phone I try to park the call from analog phone with #700
the call is correctly parked but in the second asterisk I see this log:
--
I am having problems with SIP Registration. There has been an article
about the issue
(http://www.asteriskguru.com/archives/asterisk-users-sip-registration-problem-w-sbc-vt96867.html
) but I am not able to apply the patch.
I am using AsteriskNow beta6.
The message I am having is:
[Oct 1
Good point. Here goes.
I am running ISN09 (recently upgraded). Actually the upgrade caused a
lot of problems and now the CS2K has to be datafilled so that the
Asterisk trunks are Q764 and not Q767, lest the calls fail.
Additionally the NGSS/SST had to be patched up to date to fix another
issue.
Sorry for the spam, but there was a typo. I was running ISN09, but the
upgrade was to ISN09u, which I am currently running. That was the
upgrade that caused the interoperability problem with Asterisk that I
mentioned.
On 10/1/07, Örn Arnarson [EMAIL PROTECTED] wrote:
Good point. Here goes.
I
ignorepat continues dialtone after a leading digit has been dialed on
FXS ports. How does ignorepat help this guy?
Al lists wrote:
ignorpat is your friend
On 9/30/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Sun, Sep 30, 2007 at 02:34:01AM -0700, bilal ghayyad wrote:
Dear List;
How can
Hi, also I have called Cisco suport to ask how to use SIP protocol on Cisco
7941G (and my Astersik), the their answer is the following:
..SIP Firmware for the 7941G phone only works with Call Manager 5.x. You
must have CCM 5.x to use this firmware, is needeful to buy a CCM
license for use
On Sun, 2007-09-30 at 10:49 -0400, Eric B. wrote:
Thanks for the advice everyone. Will continue reading TFOT and get started!
For what it's worth, the second edition of Asterisk: The Future of
Telephony is now available as a free PDF from
http://openbooks.oreilly.com/. (It's obviously
I just got SIP firmware images from Cisco for installation on 7970G.
Cisco requires you buy a SmartNet account (about $15, no other
dependencies apply) that entitles you to download a SIP firmware image
file from their protected support website. The 7970G now needs a
different image than
Matthew Rubenstein wrote:
I just got SIP firmware images from Cisco for installation on 7970G.
Cisco requires you buy a SmartNet account (about $15, no other
dependencies apply) that entitles you to download a SIP firmware image
file from their protected support website. The 7970G now
I was told 7941G were sold with SIP firmware factory installed.
Does anyone know this to be true or not ?
Regards
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Matthew,
Did you keep any hardcopy of licensing terms (when downloading SIP firmware)
?
This way we might double check if CCM license is mandatory to connect a
Cisco SIP phone to an Asterisk server.
Beside that, Cisco SIP phones require menu localization files to come from
CCM.
Did you run into
http://bugs.digium.com/view.php?id=10535
It is quite serious, costing us money and ill will
from our customers.
Yes, we are still running 1.2.
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On Mon, 2007-10-01 at 11:44 -0500, Jason Parker wrote:
Matthew Rubenstein wrote:
I just got SIP firmware images from Cisco for installation on
7970G.
The way I understand it, that $15 doesn't actually even give you the
right to
use the SIP firmware. It only gives you the right to
Eric,
It's a huge learning curve, but you'll soon see light at ahead even
before you know a lot. Get the book and start playing. You won't be
sorry!
On 9/30/07, Eric B. [EMAIL PROTECTED] wrote:
Jim Canfield [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
Eric B. wrote:
site
I'm having an error when I try to ./configure asterisk using
--with-odbc=/usr/lib. Below is the version of each application and the
./configure below that. Any help would be appreciated.
unixODBC-2.2.11-7.1
unixODBC-devel-2.2.11-7.1
mysql-connector-odbc-3.51.12-2.2
mysql-5.0.22-2.1
Contents of
On Mon, 2007-10-01 at 19:02 +0200, Olivier wrote:
Matthew,
Did you keep any hardcopy of licensing terms (when downloading SIP
firmware) ?
This way we might double check if CCM license is mandatory to connect
a Cisco SIP phone to an Asterisk server.
I haven't seen any such mandate,
You should probably post that question on the Asterisk business forum.
--
Eric Chamberlain, CISSP
Chief Technical Officer
Voxilla - http://voxilla.com/
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Andrew Joakimsen
Sent: Saturday,
I am having some hardware problems with the Linux machine where I have
Asterisk installed and want to use a different machine.
Assuming I install Asterisk on machine number 2, is it possible to just move
files over from the old machine to the new machine and the new machine will
behave like the
On Mon, 2007-10-01 at 12:52 -0500, Chris Stinson wrote:
I'm having an error when I try to ./configure asterisk using
--with-odbc=/usr/lib. Below is the version of each application and the
./configure below that. Any help would be appreciated.
The autoconf magic in Asterisk looks for a shared
The libtool-ltdl package is installed.
On 10/1/07, Jared Smith [EMAIL PROTECTED] wrote:
On Mon, 2007-10-01 at 12:52 -0500, Chris Stinson wrote:
I'm having an error when I try to ./configure asterisk using
--with-odbc=/usr/lib. Below is the version of each application and the
./configure
On 10/1/07, Robert DeVries [EMAIL PROTECTED] wrote:
Anyone have a list of the files that would need to be moved? (Obviously the
*.conf files in the Asterisk directory, I can think of some others, but if
someone ever did a list that would be a great help.)
You'll probably want to move the
I believe libtool-ltdl-devel is what you need.
On Mon, 2007-10-01 at 13:22 -0500, Chris Stinson wrote:
The libtool-ltdl package is installed.
On 10/1/07, Jared Smith [EMAIL PROTECTED] wrote:
On Mon, 2007-10-01 at 12:52 -0500, Chris Stinson wrote:
I'm having an error when I try to
I'm not sure I did it right, but I always just moved the following:
/etc/asterisk/*.conf
/var/spool/asterisk
/var/lib/asterisk
/usr/lib/asterisk (may be unnecessary; only for non-typical modules; see
below)
And I haven't had any problems, assuming all required modules are in the
new
If this is on a RedHat-type system (EL, Fedora, but also CentOS), make
sure you have a symlink in place for libltdl.so. Even though I also
had the libtool-ltdl package installed, it only provided libs and
links for /usr/lib/libltdl.so..3.1.4 and libltdl.so.3. It did not
create a symlink to a
Razza wrote:
On 27/09/2007, Eric B. [EMAIL PROTECTED] wrote:
For starters, what is the difference btwn the 1.2 and 1.4 branches of
Asterisk? I can't seem to find a document that describes the changes.
Anyone?
Not much/Lots
Depends what you're looking for. Important considerations for us in
I didn't have libtool-ltdl-devel. Once I install the devel package, it
finished the configuration. Thanks James, Jared and Kai-Uwe for the
responses.
On 10/1/07, Kai-Uwe Jensen [EMAIL PROTECTED] wrote:
If this is on a RedHat-type system (EL, Fedora, but also CentOS), make
sure you have a
Unfortunately 1.2 is no longer getting bug fixes (except for security
fixes). You will have to manually apply the patch for 1.2.
Yes the 1.2 maint policy sucks for many people, including me.
Doug wrote:
http://bugs.digium.com/view.php?id=10535
It is quite serious, costing us money and ill
Do you mean, when people call VoiceMailMain to _check_ their messages
they need to be cut off after five minutes? For this, I'd put an
absolute timeout before the call to VoiceMailMain.
I'm using asterisk 1.4, and the following syntax works for me:
; Set absolute timeout to five minutes (300
In my experience, and theoretically by design, it doesn't matter what
codec you are using when you call a meetme conference.
Moj
Mark Quitoriano wrote:
Hi,
is there a way to use g729 in meetme?
Thanks!
At 14:14 10/1/2007, Eric \ManxPower\ Wieling wrote:
Unfortunately 1.2 is no longer getting bug fixes (except for security
fixes). You will have to manually apply the patch for 1.2.
Yes the 1.2 maint policy sucks for many people, including me.
Hmmm. Many people believe that 1.4 is still
Jason Kincaid [EMAIL PROTECTED] writes:
Hi,
I'm trying to register SIP phone with an asterisk serve, failing miserably.
The server is sending 401 Unauthorized
responses to the registration attempts, but every time the phone is
re-REGISTERing without authorization. I'd think this
was a
In article [EMAIL PROTECTED],
Mojo with Horan Company, LLC [EMAIL PROTECTED] wrote:
Do you mean, when people call VoiceMailMain to _check_ their messages
they need to be cut off after five minutes? For this, I'd put an
absolute timeout before the call to VoiceMailMain.
I'm using
I have both units on my desk here, the server is on the local 224 subnet and
the phone is on 220 subnet (IP 192.168.220.31).
My PC is on the same jack as the phone, sharing a hub, so I can sniff packets
with ethereal. My PC can see the 401 unauthorized packets so therefore the
phone can
Hi list,
Have somebody tried a tor3e board on a intel x86_64 ?
I have installed one but I have no audio on it, but, installing on a x86 32
bits server it works fine.
I'm using asterisk-1.4.11 and zaptel-tor3-1.4.5.1.tar.gz.
Ard
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Hello Gurus
I've installed my Asterisk server for testing on the company I work the
setup or the approach let's call it is:
1 Asterisk Server fully configured and with some SIP extensions setup on two
cities A and B.
2. One local PSTN line connected thru a x01p card to call local phone
numbers
Doug wrote:
At 14:14 10/1/2007, Eric \ManxPower\ Wieling wrote:
Unfortunately 1.2 is no longer getting bug fixes (except for security
fixes). You will have to manually apply the patch for 1.2.
Yes the 1.2 maint policy sucks for many people, including me.
Hmmm. Many people believe
Eric ManxPower Wieling wrote:
Doug wrote:
At 14:14 10/1/2007, Eric \ManxPower\ Wieling wrote:
Unfortunately 1.2 is no longer getting bug fixes (except for security
fixes). You will have to manually apply the patch for 1.2.
Yes the 1.2 maint policy sucks for many people, including
My understanding is:
Smartnet: service contract basically allows you to download the
newest sw release.
Besides that you can buy phones without a license. Presumably as
spares But you must buy a SIP license to technically be allowed to
use that software that can be obtained from Smartnet.
I
Just buy the Linksys SPA962's they work better than the cisco phones
in a NAT env.
/b
On Oct 1, 2007, at 6:13 PM, Andrew Joakimsen wrote:
My understanding is:
Smartnet: service contract basically allows you to download the
newest sw release.
Besides that you can buy phones without a
but is there a way to use g729 codec in meetme?
On 10/2/07, Mojo with Horan Company, LLC [EMAIL PROTECTED] wrote:
In my experience, and theoretically by design, it doesn't matter what
codec you are using when you call a meetme conference.
Moj
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
For the archives, the quick fix was to change:
p[2] = 0x80 + (type4) + plan;
to:
p[2] = 0x80;
Problem now resolved and system working well.
- --
Kind Regards,
Matt Riddell
Director
___
Hi to everyone!
I have succerfully instaled my new Asterisk 1.4 on my debian etch.
I have my users in sip.conf like this:
[200]
type=peer
host=dynamic
context=home
secret=200
callerid= 200
dtmfmode=rfc2833
nat=yes
[EMAIL PROTECTED]
disallow=all
allow=ulaw
I can make calls in my LAN but i can´t
Ok Let me chime in on this one.
If you can use ulaw/alaw because you'll end up with tandem encoding
which will make the conference sound worse to some people.
All audio coming in will get transcoded to signed linear and pushed
down into zaptel then back up and out to the conference
As long as you have some g729 codecs installed, Asterisk will do this
fine.
PaulH
On Tue, 2007-10-02 at 07:37 +0800, Mark Quitoriano wrote:
but is there a way to use g729 codec in meetme?
On 10/2/07, Mojo with Horan Company, LLC [EMAIL PROTECTED]
wrote:
In my experience, and
Hi everyone,
I'm trying to get a Sangoma A101D-X card talking to a Sasktel PRI
(Megalink) circuit and having some trouble getting it to handshake.
Thanks for any help or suggestions to diagnose this problem.
The problem is that Asterisk has trouble bringing up the link. I see
the following
The only time I have had this problem is when there was a version
mismatch between Zaptel and Asterisk. Once I resolved that issue
(latest asterisk + latest zaptel + reasonably recent wanpipe) everything
worked for me.
Alvin Austin wrote:
Hi everyone,
I'm trying to get a Sangoma A101D-X
Hi
suffered that issue since I started that´s the course oif all of us
newbies, noone is willing to help/and even answer, I don't even know if my
messages are being read on this list cause not evena google for it i've
received.
I'm now acroos the rive with that problem you're being the victim of
As soon as I saw channel '24 as D-channel' my guess is that the
card/config is set up as T1, when you need E1.
PaulH
On Mon, 2007-10-01 at 18:23 -0600, Alvin Austin wrote:
Hi everyone,
I'm trying to get a Sangoma A101D-X card talking to a Sasktel PRI
(Megalink) circuit and having some
Hello Mark,
On 10/2/07, Mark Quitoriano [EMAIL PROTECTED] wrote:
but is there a way to use g729 codec in meetme?
You have to buy a G.729 license for each channel which I believe is at USD
10.00 if I'm not mistaken. Then, make sure that your machine is fast enough
for transcoding. But the
Since the channels have to be mixed together by Asterisk, passthrough
can't be supported in this case.
In other circumstances, passthru works fine.
PaulH
On Tue, 2007-10-02 at 10:02 +0800, GNUbie wrote:
Hello Mark,
On 10/2/07, Mark Quitoriano [EMAIL PROTECTED] wrote:
but is there
I've recompiled with the latest svn sources for zaptel, libpri, and
Asterisk. Wanpipe is 3.3.0.p4.
Switched the T1 cable. Same result.
(It's a Sasktel Megalink T1/PRI circuit)
CLI shows:
~~
== Primary D-Channel on span 1 up
== Primary D-Channel on span 1 up
Alvin Austin wrote:
I've recompiled with the latest svn sources for zaptel, libpri, and
Asterisk. Wanpipe is 3.3.0.p4.
Switched the T1 cable. Same result.
Hmn -- when you recompiled, did you
1. clean out all the source directories?
2. remove the binaries?
3. recompile in the right order?
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Hash: SHA1
What we do is the following:
Our CPE (Customer premises equipment) registers via IAX with all of our
servers at the same time (with qualify turned on for the links).
All of the servers first try to reach numbers via local IAX links.
If this fails
Correction, on FXO port not FXS,
second, read his email first:
Also, how it will be possible to assign an dedicated
line (connected to FXO) to an
button on the Polycom IP Phone or Broadtel IP Phone,
so if user select that button
then he will be sure that his outside call will be via
that specific
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Terry Giufre-Sweetser wrote:
Dear List,
Has anyone found or written a status panel application, windows or
linux, that uses SIP notifies and subscriptions, to gather the status of
SIP extensions from Asterisk?
And displsy nicely on a GUI?
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Olivier wrote:
Strange !
We successfully used SuperMicro boards without any IRQ problems.
What is SuperMicro's reply, concerning this IRQ problems ?
They sure have interest to solve this or at least explain why it can't be
done.
It's not the
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Santiago Aguiar wrote:
I'm using the Originate Action on the Asterisk Manager to place calls
between two extensions in async mode.
Is there any way to cancel the Originate Action before I get the
OriginateResponse action? I'm unable to perform a
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