currently thats not possible unless you speciffy the async flag, in
that case Event: OriginateSuccess or Event: OriginateFailed event
will be launched with the uniqueid
Regards
On 12/4/06, Rodrigo Gonzalez [EMAIL PROTECTED] wrote:
My code is using phpagi-asmanagerbut what is sent is...
I dont think the registration will be the problem, but the media
communication, for that you could use an Application Layer Gateway
(ALG), you can check netfilter.org for more information.
Regards
On 12/1/06, Jason Kim [EMAIL PROTECTED] wrote:
Hi,
I installed asterisk with oh323.
My
I think you can, using the manager Originate action and the Bridge()
application on the dial plan. The bridge application is still not in
trunk ( AFAIK ), or you can try to test the framework were working on
to have complete control over the asterisk channels using a PHP
routing daemon. The page
You may want to give a try to this document I wrote about mfcr2 with Asterisk.
http://moy.ivsol.net/unicall/mfcr2-asterisk-unicall-0.2-english.pdf
Regards
On 11/24/06, Alyed Tzompa [EMAIL PROTECTED] wrote:
Hello!
I'm tryuing to bring up an R2 connection but eventhough I've followed the
If the call is generated, the problem has nothing to do with the PHP
file. By the way, instead of looking into GET and POST, you may want
to use REQUEST global variable.
Regards
On 11/19/06, zero massive [EMAIL PROTECTED] wrote:
Hi all!
Can anyone shed some light on a problem with a php to
in google
about Makefiles
4. What actually the asterisk do with patch channels_makefile.patch?
Apply the patch to the Makefile code
Desperately need help.
Everyone does.
Best Regards and good look
Moises Silva
--
Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org
compatible with Asterisk 1.4?
I don't know how much the channel API changed from 1.2 to 1.4.
The patch for the Makefile I guess I could fix it by hand
BarZ
Moises Silva wrote:
libunicall, spandsp, libmfcr2 are independent from Asterisk version,
the only thing you need to adapt for each Asterisk
On 11/4/06, Stephen Bosch [EMAIL PROTECTED] wrote:
Stephen Bosch wrote:
Moises Silva wrote:
try enabling DTMF debugging on logger.conf for the console, and tell
us here waht do you see
This is what comes out on the console, with IP addresses removed:
-- Call accepted by xx.xx.xx.xx (format
should be natively included into Asterisk, as at least
half of the world uses E1...
BarZ
Moises Silva wrote:
Yes, you are right, as I said, you need to adapt chan_unicall.c to the
1.4 * version, it should not be hard, let me know if you have problems
and may be i will be able to help you.
Kind
libunicall, spandsp, libmfcr2 are independent from Asterisk version,
the only thing you need to adapt for each Asterisk version is
chan_unicall.c
Best Regards
On 11/3/06, Barzilai Spinak [EMAIL PROTECTED] wrote:
Is there any way to compile Unicall's libraries (mfcr2, spandsp,
chan_unicall,
try enabling DTMF debugging on logger.conf for the console, and tell
us here waht do you see
On 11/3/06, Stephen Bosch [EMAIL PROTECTED] wrote:
Hi, everybody:
As part of a paging macro I'm using SendDTMF to send digits to the
called party.
The section looks like this:
exten = s,1,Wait(0.5)
You can make RTP pass through Asterisk, or not. Look in voip-info.org
about Native Bridge and sip.conf canreinvite option. And may be
this page will be usefull too:
http://www.voip-info.org/wiki/index.php?page=Asterisk+SIP+not-proxy
Regards
On 10/31/06, Mike Williams [EMAIL PROTECTED] wrote:
of course you can always use http://cacti.net/download_cacti.php
On 10/31/06, Andrew Joakimsen [EMAIL PROTECTED] wrote:
Check out voip-info.org, there are quite a few GUIS some even generate nice
graphs!
On 10/31/06, omar parihuana [EMAIL PROTECTED] wrote:
Hi Folks,
I would like to
Learn how a patchfile/Makefile works, and fix the patch. Actually the
Makefile patch never has applied cleanly in my experience, so always a
few fixes are needed.
On 10/30/06, Christian Jensen [EMAIL PROTECTED] wrote:
I have looked on this list but may have missed it. I am having
problems
This is a problem of the codec you are attempting to use. Wich codec
is?, it seems to Asterisk cannot identify the codec you are using.
Regards
On 10/28/06, Giedrius Augys [EMAIL PROTECTED] wrote:
Hi,
I have just installed fresh FreeBSD 6.1 and asterisk. But I get warning
about power. I have
what about
exten = h,n,System(mycommand /some/file /some/other/dir/)
Where mycommand is your custom shell script to sleep before moving the file.
On 10/27/06, Alexander Burke [EMAIL PROTECTED] wrote:
Hello, all!
I'm having a problem with the following snippet that executes upon hangup:
AFAIK, you will need to do the first. ARA-odbc-sqlite
On 10/26/06, Erick Perez [EMAIL PROTECTED] wrote:
Can I safely assume that SQLite can be used to code something for
Asterisk Realtime instead of the much used mysql database?
I have read several old posts, but nothing point me to an answer.
Hi Alvaro, MAGI did not make it into Asterisk, and AFAIK, is almost
not used at all in the Asterisk community (I would like to know if
somebody else is using it around here). Right now im in the middle of
a development for the company I work, based on MAGI patch. See more
details on
Jan. Unfortunately, Originate is somehow a mess. Answering your
question, the reason is not really a reason, at least not in release
1.2.12.1, that is the one im using. The reason is set according to the
last communication frame read from the originated channel. Let me
explain. When you originate
DTA0204 loop e
Incomplete or incorrect ANI was received by CDTI2/CSDTI2 FW (outgoing
trunk) which reports this fact to the main CPU by SSD messin 9 (NI
problem report).
As the error message says, incomplete or incorrect ANI was received,
that is not configured on zaptel hardware, but in MFC/R2
of spand
but by the version of kernel of linux, I proved it in two equipment with
fedora4 and works, but I need to compile it in fedora5 and when I do it it
marks the error to me that mentions before. Somebody knows where encounter
an updated version of these archives?
On 10/11/06, Moises Silva
,
asterisk-svn-21231-DTMF_event.patch.
Would that be the one?
Frank
On 10/12/06, Moises Silva [EMAIL PROTECTED] wrote:
Hi Frank, I sent a patch updated here:
http://bugs.digium.com/view.php?id=6082
But that was some months ago, I havent seen a bugmarshall for a while
there, so I keep
on bug6082 it is hard to tell whether
the DTMF event patch is still in there when I last compiled that
branch the DTMF event was not coming up.
Is the DTMF patch incorporated into 1.2.12.1, or will I need to apply
the original patch to 1.2.12.1?
On 10/10/06, Moises Silva [EMAIL PROTECTED
Has somebody installed this configuration: Asterisk + E1 with MFC/R2
(Telefónica Argentina) in Argentina before? I need to know if it´s
possible with MFC/R2 argentine variation.
I have not tested in Argentina, but support is included in the code,
so I suppose it should work.
Regards
--
Su
Diego, this is an english mailing list, there is no need to post in
spanish the same message.
Your error is due to missmatching versions between libmfcr2 and
spandsp. Downgrading spandsp will fix the problem.
Regards
On 10/11/06, DiegoF [EMAIL PROTECTED] wrote:
hola a todos de nuevo, tengo el
be at highest level to be able
to see the problem clearly. If you have more that 1 port in your PCI
cards, try using a loop, like explained in the document, to discard
problems on your side.
Regards
On 10/9/06, Carlos Chavez [EMAIL PROTECTED] wrote:
On Mon, 2006-10-09 at 19:39 -0500, Moises Silva wrote
Jan, im sorry to get back to you so late, ive been busy. It seems i
sent you an incorrect patch I was testing, but I have found the
correct patch in mantis:
http://bugs.digium.com/view.php?id=6682
Please be aware that the patch I sent you initially used a funciton
that received 1 or more DTMF
Steve is their a CLI command you can make from the console that will
tell you the answer? LOL or are we expected to count?
asterisk -rx 'show dialplan' | grep '\[ Context' | wc -l
--
Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org;
You are using wrong matching versions between libmfcr2 and spandsp.
Old versions of spandsp does not receive any argument to init R2 tx,
the firm was like this:
void r2_mf_tx_init(void)
newer spandsp versions use a firm like this:
r2_mf_tx_state_t *r2_mf_tx_init(r2_mf_tx_state_t *s)
So, your
Same problem as your other post. dtmf_put is no longer available in
newer spandsp versions, the solutions is the same as with libmfcr2,
downgrade spandsp, or upgrade chan_unicall (not always a matching
version exists, sometimes you need to fix it)
Regards
On 10/9/06, Carlos Chavez [EMAIL
Where can I get the source to apply the patch, is it difficult to apply it?
I dont remember the exact place where I get it, was in
bugs.digium.com, but I dont remember the number/name of the bug. I
made a fix to the patch so it can apply to 1.2.12.1
You can get it here
Could you describe better the call path?
are you doing this?
Asterisk 2 IAX2 --- Asterisk 1 - FXO/FXS - NBX
Or something else? If so, it would be nice to post relevant parts of
the Asterisk CLI with all the log levels activated in logger.conf
Regards
On 10/5/06,
You are just not loading the module. Connect to Asterisk terminal
# asterisk -vr
and load the module
CLI load app_senddtmf.so
Best Regards.
On 10/4/06, Jan du Toit [EMAIL PROTECTED] wrote:
Hi all.
I'm confused. On http://www.voip-info.org/wiki-Asterisk+manager+API it
says that the
yep,
# modprobe ztdummy
You need some special routines compiled in the kernel, google around a
bit to find wich ones.
Other solution may be use app_conference, is not included in asterisk
sources, that app does not require zaptel timing.
Regards
On 10/4/06, omar parihuana [EMAIL PROTECTED]
I could be wrong here, but I think that you're looking for SendDTMF and
not PlayDTMF. getting it confuddled with PlayTones?
He is not confused. PlayDTMF is a manager command, not an dial plan
application, but included in the same module that SendDTMF
(app_senddtmf.so). I dont think is available
Are you sure you deleted all the old asterisk modules when upgrading?
On 9/18/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
I upgraded to 1.2.12.1 - the problem is much, MUCH worse. Before, it was
once or twice per week, now EVERY TIME someone uses chanspy it crashes the
machine.
Anyone
Hi, dont expect too much help providing only the Asterisk version you
are using. You need to tell us the call path, DTMF mode (inband,
outband, SIP INFO etc) used and call technologies involved (SIP, ZAP,
IAX2 etc).
Regards
On 9/15/06, Nitesh Divecha [EMAIL PROTECTED] wrote:
Hello All,
Can
trying something like that in a dial plan you create yourself will
reproduce the problem.
Thanks
Frank
On 9/15/06, Moises Silva [EMAIL PROTECTED] wrote:
could you post the output of the asterisk console in verbose mode?
In logger.conf
[logfiles]
console = notice,warning,error,verbose,debug
If you want to have a safe asterisk I would recommend using svscan
from daemontools package, more wonderfull software of D.J. Bernstein.
http://cr.yp.to/daemontools/svscan.html
Regards
On 9/15/06, Giorgio Incantalupo [EMAIL PROTECTED] wrote:
Hi Julian,
I know I have two process.the
No such variable exists, why dont you set your own variable before
calling goto ?
Regards
On 9/15/06, Mike [EMAIL PROTECTED] wrote:
Is it possible to know, when an extension is reached through a Goto command,
what the context of the Goto command was?
Useless but representive example:
number I get the busy
voice prompt okay.
Injecting the DTMF tones on the call channel is a really a kludge to
simulate actual key presses in the hope that the dead air problem will
go away and it .
On 9/15/06, Moises Silva [EMAIL PROTECTED] wrote:
Frank. PlayDTMF and SendDTMF is the same
Why oh why do so many people do all this modprobe stuff manually or in
rc.local etc.?
If you are running a RedHat / Fedora / CentOS distribution, just do
make config in the zaptel directory, and it will create a proper
startup script in init.d and set up the rc.d links for invocation at
boot
http://www.voip-info.org/wiki-Asterisk+cmd+SendDTMF
Regards
On 9/14/06, Frank Church [EMAIL PROTECTED] wrote:
How can DTMF be sent down a channel?
I am thinking of method where say a channel id can be grabbed from
Asterisk Manager events and a DTMF signal sent down that channel,
through AGI,
be?
On 9/14/06, Moises Silva [EMAIL PROTECTED] wrote:
http://www.voip-info.org/wiki-Asterisk+cmd+SendDTMF
Regards
On 9/14/06, Frank Church [EMAIL PROTECTED] wrote:
How can DTMF be sent down a channel?
I am thinking of method where say a channel id can be grabbed from
Asterisk
So what should I do to build zaptel for the new kernel?
As Steve Totaro said, when running the newer kernel, go to zaptel sources and:
make clean
make make install
I remembered that i had to make linux26 make install, but not
sure if this is still necessary for newer zaptel drivers.
No
Hi Ivan. As you see in this page:
http://www.neobits.com/do/dtls?pid=9583
This card is a bundle, wich means supports both, FXO, and FXS. FXO
ports should be used to connect your 3 telco lines, and FXS port to
connect some phones.
Regards
On 9/8/06, Iván Vega R. [EMAIL PROTECTED] wrote:
Hi
Yep, that should help, and the short answer to you question is NO.
Regards
On 9/9/06, Doug Lytle [EMAIL PROTECTED] wrote:
Rene wrote:
Hi all,
I am trying to understand contexts a bit better. The problem I have is
This should help:
Sangoma has excellent support, why dont you ask them?
On 8/31/06, Angelito Manansala [EMAIL PROTECTED] wrote:
Hello Guys,
We have a problem in configuring Sangoma A104. We want the 2 ports to be
configured as E1 and the 2 ports as T1.
We already run wancfg and configure the 2 ports as T1 and
100ms
On 8/28/06, Roi Stork [EMAIL PROTECTED] wrote:
The version that I'm using is 1.2.7.1.
What is the default value of writetimeout in manager.conf?
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asterisk-users mailing list
To
I'm trying to evaluate my path to several voip providers, so I set
qualify=400 in iax.conf. But, I'm not seeing any
REACHABLE/UNREACHABLE or LAG messages in the logs. Is there a logging
option to set so these will show up?
these messages are logged with verbosity LOG_NOTICE, so, in the
We use our own CDR, but as I understand, the C option resets the CDR,
that does not means is not going to save cdr, but is going to restart
the CDR. So, a simple NoCDR() before dialing should work, or ForkCDR()
and then NoCDR() if you want to save previous data.
Regards
On 8/27/06, Master Abi
WARNING[4670]: cdr.c:443 ast_cdr_free: CDR on channel
'SIP/7002-081ac898' not posted
Aug 28 15:27:18 WARNING[4670]: cdr.c:445 ast_cdr_free: CDR on channel
'SIP/7002-081ac898' lacks end
-- Executing Dial(SIP/7002-081ac898, SIP/7003|20|tr) in new stack
I am using 1.2.11
Regards
Moises Silva
just ignore the warning, no CDR will be saved
On 8/28/06, Master Abi [EMAIL PROTECTED] wrote:
That is what I thought, but then how do I STOP recording CDR's. If I use
it in the h extension, it also gives a warning.
Moises Silva wrote:
Normal behaviour since the call record before executing
Sure is possible. Look into google 'asterisk agi fastagi'.
Regards
On 8/23/06, Javier Lara Sanchez [EMAIL PROTECTED] wrote:
Dear All,
I need to buid an IVR that could make a request to a data base (oracle) in a
remote host.
The idea is that an user dial a extension with 2 options
I also want to add that I saw a great improvement from versions 1.0.x to
versions 1.2.x. Let's see what 1.4 will bring, but I hope a 2.0 version
with a complete rearchitecturing could finally make Asterisk the Apache
of telephony as I read somewhere. (Or wait for the OpenPBX guys to
awake from
Hi Mario. Have you tried to enable AGI debug?
CLI agi debug
That will show what Asterisk is receiving from your script.
Also enable all the debug messages in the logger.conf file for the console
Go and try that and post what you see here, and we may be able to help you
On 8/17/06, Mario
Once in a while, the same questions are asked in this mailing list
about Unicall and MFCR2. I wrote a document in spanis about 2 months
ago about MFCR2 signaling and how to debug it with testcall. I have
translated the document into english per users request, and made some
other improvements.
Douglas. Please take this as a constructive comment. I have followed
your questions in asterisk-dev and users lists, and you always seem to
make non constructive comments about the people giving code/work for
Free. And you focus in the negative part, never giving importance to
the positive
On 8/15/06, John Novack [EMAIL PROTECTED] wrote:
I, for one, didn't take his comment as anything other than constructive
Yes, I agree is possible just lack of niceness. But may be you dont
have idea of the bunch of peyorative comments about IAX2 protocol he
made in asterisk-dev list without
I guess you didnt even write in google asterisk java before sending a question to the list right?http://asterisk-java.sourceforge.net/
On 8/14/06, Lennie De Villiers [EMAIL PROTECTED] wrote:
Hi,
Is there a API or framework available to write solution in the Java programming
We (Intervoice Solutions Company, http://www.ivsol.net/) are about to release as free open source, a PHP router daemon that does just that, but requires a patch to asterisk called MAGI. Contact me off-list if you are interested.
I call it free open source since i havent had the details about the
Lennie: Tomorrow in the morning I will be talking with the people that will decide when to make the release and how handle this. I have your email, you will have news from me soon :)Regards
On 8/14/06, Lennie De Villiers [EMAIL PROTECTED] wrote:
Hi,
What would you suggest I use?
I'm
features.conf
On 8/11/06, Douglas Garstang [EMAIL PROTECTED] wrote:
Does anyone know how I can set/increase the inter digit timeout on Asterisk
assisted transfers?
Doug.
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asterisk-users
I had a similar problem with a Siemens, most probably you are
specifying the wrong number of expected ANI digits. Try with mx,0,4
as protocolvariant, that will tell Unicall to expect 0 ANI digits, but
of course, in Asterisk environment you wont be able to get callerid.
Play around incrementing
Hi Carlos, are you new in asterisk-users? My recommendation would be
to post more information about what is Asterisk showing in the verbose
console. Posting only configurations with a brief description of the
problem from a end user perspective is something that I dont see
much usefull.
Regards
If I understand your question properly, it is possible, since AGI is
just an interface that uses STDIN, STDOUT and STDERR to communicate.
So Rails has nothing to do, you only need a programming language, not
necessarily a framework.
Regards
On 8/7/06, shawn bright [EMAIL PROTECTED] wrote:
Hey
Well, since you are not providing technical information like
protocols, I only can tell you that if you are using inband DTMF, yes,
is possible that poor quality in the link makes DTMF go wrong.
Regards
On 8/7/06, Kohler, Jeffrey [EMAIL PROTECTED] wrote:
I'm having problems where touch
quality, are there any
suggestions for improvement?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Moises
Silva
Sent: Monday, August 07, 2006 10:09 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DTMF problems
Well
I have done something similar with Avantel, but not sharing channels
in the same link. I received 1 E1 line for voice, and other E1 line
for Internet, but in theory sharing channels should not be a problem.
I could not make HDLC work with the kernel HDLC generic software
driver and Digiums
Please feel free to contact me off-list for more details. I have 2
years of experience designing PHP asterisk applications for routing
and other stuff. You can find my CV at:
http://phpmexic.u33.0web-hosting.com/wordpress/?page_id=8
May be the asterisk-biz list would be more appropiate for this
:
On Mon, 2006-08-07 at 14:18 -0500, Moises Silva wrote:
I have done something similar with Avantel, but not sharing channels
in the same link. I received 1 E1 line for voice, and other E1 line
for Internet, but in theory sharing channels should not be a problem.
I could not make HDLC work
Not sure if it can help you, but check this patch:
http://bugs.digium.com/view.php?id=5841
Is for a new application called Bridge meant to bridge 2 channels.
On 8/7/06, Leon Sun [EMAIL PROTECTED] wrote:
Hi,
I searched web for few hours and couldn't find any solution about linking 2
calls
do you have notransfer=no in iax.conf iaxy entry?
On 8/5/06, Jeff Iddings [EMAIL PROTECTED] wrote:
IAX Provider-- Asterisk/NAT Firewall-- IAXy (192.168.0.x)
When a call comes into the Asterisk box, it then rings the IAXy.
When the IAXy is answered, I get..
-- IAX2/homeiaxy-6 answered
Is possible that you are missing the XML file with the supertones
definitions. Usually is located at /usr/share/spandsp/global-tones.xml
, but it depends on how you configured the spandsp package
(./configure --prefix=/usr/blah). Notice that spandsp and
libsupertone should be configured with
configuration files, or some other config engine.
What do you mean with this?
Regards.
Moises Silva
--
Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org;
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asterisk-users
may be you are looking for asterisk application ForkCDR(), more info
in voip-info.org
Regards
On 7/31/06, Daniel Salama [EMAIL PROTECTED] wrote:
I have an Perl AGI script which accepts inbound calls and offers an
IVR service. Depending on certain options that are selected on the
IVR, the
may be im missing something, but i think the pseudo channel you are
looking for is called Local and you can call some extension that you
know the only thing it does is play the message you need. So you can
originate a call to that Local channel and bridge it to the Meetme
conference where your
I think is a problem in the reload routine of unicall. Note that I
have not the newest version, and im not able to reload, it does not
give me the same message, but still i cannot reload and the unicall
channels are no longer available after executing reload. I think you
should avoid using
There must be several ways, however one that comes to my mind is use dnotify
http://oskarsapps.mine.nu/dnotify.html
and execute a command that create a .call file everytime a new file is
created in the mailboxes
http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out
To call the
Asterisk does not have softphone interfaces. You can write a softphone
to support some VoIP protocol supported by Asterisk, and voila, you
can connect to Asterisk. Supported and common protocols are IAX2, SIP
and H323. For IAX you have a library called iaxclient, so you are not
required to make
from the comments in mfcr2.c
/*
There also appear to be R2 variants for at least the following:
Australia
Belgium
Costa Rica
Eastern Europe
Ecuador (ITU)
Ecuador (IME)
Finland
Greece
Guatemala
Israel
New Zealand
Paraguay
Peru
South Africa
Uruguay
*/
Chavez [EMAIL PROTECTED] wrote:
On Wed, 2006-07-19 at 10:29 -0500, Moises Silva wrote:
Carlos. Unblocking the remote side is NOT your responsibility, unless
you own the 2 end points :). I suppose you are getting connected to
some telco (avantel, telmex, etc), if so, is telco's responsibility
1) Why do the zaptel and librpi drivers and libraries pretend to
handle E1 cards, but apparently know nothing about the MFCR2 protocols?
Is there any other normal way to use the E1's (with respect to telephony)
Zaptel is the driver code. Does not need to know anything about higher
level
On 7/20/06, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Thu, Jul 20, 2006 at 03:51:25PM -0500, Moises Silva wrote:
1) Why do the zaptel and librpi drivers and libraries pretend to
handle E1 cards, but apparently know nothing about the MFCR2 protocols?
Is there any other normal way to use
Carlos. Unblocking the remote side is NOT your responsibility, unless
you own the 2 end points :). I suppose you are getting connected to
some telco (avantel, telmex, etc), if so, is telco's responsibility to
unlock their side.
To discard any problem with Asterisk, try using testcall utility
From where did you downloaded the snapshot? could you post a link to
the sources?
I think this is a problem of missmatch version of old libunicall an
newer libmfcr.
Those undefined macros should be part of the libunicall headers, so
when compiling the new libmfcr2, it does not find the newer
Are you sure the other end is configured properly?
What does zttool says?
Have you turned on all the asterisk debug messages to look further?
Regards
On 7/19/06, Lincoln Zuljewic Silva [EMAIL PROTECTED] wrote:
Hello all. I have a Digitum TE110P board configured and working (I think
that it's
IAX2 does not need such a thing, since always send the DTMF out of band.
On 7/19/06, Peter Beckman [EMAIL PROTECTED] wrote:
I know you can use dmtfmode in sip.conf, but does it do anything in an
iax.conf context?
ie.
iax.conf:
...
[super]
auth=md5
type=friend
username=super
secret=man
between the two machines.
Thanks
Lincoln
Moises Silva wrote:
Are you sure the other end is configured properly?
What does zttool says?
Have you turned on all the asterisk debug messages to look further?
Regards
On 7/19/06, Lincoln Zuljewic Silva [EMAIL PROTECTED] wrote:
Hello all. I have
The show channels output is always truncated.
On 7/17/06, marek cervenka [EMAIL PROTECTED] wrote:
hi,
i have problem with showing actual channels
asteriskshow chanels
SIP/123456789-b6c4b2 [EMAIL PROTECTED] Up Busy()
(last 2 chars are NOT showed)
but the name of channel is longer
Callme stupid, but im not understanding your problem. Suggestions that
may help others to answer:
1. A little bit more clear in your examples? :)
2. Try describing the Asterisk behaviour under every circumstance.
Regards
On 7/17/06, Sebastian Reitenbach [EMAIL PROTECTED] wrote:
Hi,
I have
If the SIP or IAX peer are registered as extension 37, the generated
channels would be
SIP/37- or IAX2/37-
The last 4 digits are for making a difference in case that the same
peer is active in more than 1 call.
Regards
On 7/13/06, Reynaldo Baquerizo [EMAIL PROTECTED] wrote:
Hi
I've
AFAIK operation now in progress is a common status when you open a
socket connection. When you use blocking sockets usually you dont see
this because the connect call does not return until the connection
is done. But when using non-blocking sockets, the connect call returns
immediatly and if you
hiring some one to do it :)
sorry, i couldnt avoid to tell it, but your question is so generic
that the response will be generic, unless some kind sould takes
several minutes of their time to explain it to you.
First i would recommend you this document:
I think you are really confused. I dont see a reason why dialing
555666 the call should go to client SIP/test. What you are doing is
dialing to Zap channel 1 (whatever it is) the number 5556662, so, what
do you have connected at the other end of the Zap/1 channel?
On 7/7/06, Ralph Liebessohn
Oops, i missed the crossover cable part. I have used crossover cable,
so it should work, but the DNID must be complete. Wich signaling are
you using?
Regards
On 7/7/06, James Hawks [EMAIL PROTECTED] wrote:
When you dial directly you are bypassing the zap and just dialing an
internal
One of the ends must be configured as pri_net and the other as
pri_cpe. By the error I think the problem is with your configuration,
does zttool says no alarms in spans?
Post your configuration files zapata.conf and zaptel.conf
Regards
On 7/7/06, Marco Mouta [EMAIL PROTECTED] wrote:
by Ports
please type in google.com:
STUN server ALG
The fourth result is a good and small explanation.
On 6/26/06, Martin Joseph [EMAIL PROTECTED] wrote:
On Jun 26, 2006, at 9:32 AM, Raymond Tant wrote:
Hi all,
Could someone point at resources for running Asterisk behind a
firewall.
STUN keeps
what do you mean by could not print out message to stderr???
Try being more descriptive about your problem. Error messages, how
have you tried etc.
On 6/26/06, Zichao Wu [EMAIL PROTECTED] wrote:
Hi, guys, I used /usr/src/asterisk/agi/eagi-test.c script to test AGI API,
but that script could
Piece of cake Julian:
http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+Action+Redirect
Regards
On 6/19/06, Julian Lyndon-Smith [EMAIL PROTECTED] wrote:
At the moment when one of our users wants to transfer a call, they press
the transfer button on the phone, enter the
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