Hello All once again...
Has anyone got any experience with calling to Turkey?
Voipjet seems to have good quality and rates, but I was wondering if
there are any termination providers over there, or providers that can
supply a DID, even in a home-user scenario.
Thanks,
Greg
__
Hello All,
I am trying broadvoice's europe plus calling plan for unlimited to
Poland. My first attempts though, were not that good. I could hear the
other side, but they could not clearly hear me.
Is this because broadvoice's connection just is not up to par? Has
anyone else been using this pla
Hello All,
I was wondering. What are the primary advantages to using a PRI over a
T1? As I understand it, the PRI terminates very fast, meaning you can
do immediate answer and dial... This is very handy on the BRI line I
have on the asterisk.
Can T1 signalling also do immediate answer, or does
A) AGI prefers the CLI version.
B) Use VERBOSE, write to stderr or dump any debug messages in your own
log file
C) Of course not, thanks to you. Include scripts and debug output, and
maybe we'll get closer.
Just tell me we're not doing your homework for you.
> -Original Message-
> From
Maybe I am to sensetive, but what is an FXO?
I have a device in my hand, it says it has an FXS and FXO port (besides
WAN and LAN port)
The SIP settings are only effecting the FXS.
The FXO is connected to the phone company but can only be reached from
the phone connected to FXS by prepending a
Hi
Digium cards are compatible with indian telephony..
I am using it.
But there is problem I am facing to configure caller ID.
What cidsignalling is used in india?
Regards
Gurminder
On 8/8/05, Ankit <[EMAIL PROTECTED]> wrote:
> Hi everybody,
>
> I need a little clarification regardin
Hello list,
There seem to be some problem with the ACD of asterisk
where when we use this parameter in queues.conf .
We could not get any announcement as expected.
Iam useing the latest CVS-head
Even weight also doesnot seem to work properly
I tried like this where we have two queues one with
hi dave,
any suggestions on myencoutrered problem below?
thnks so much.
chris
- Original Message -
From: "chris" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Monday, August 08, 2005 4:51 PM
Subject: Re: [Asterisk-Users] function declaration isn
hi rajesh,
thx for the info., also did u purchase the card directly frm digium or
there are distributors of digium cards in india.
-ankit
On 8/9/05, rajeshkumar nayak <[EMAIL PROTECTED]> wrote:
> hi
>
> Digium card is compatible with the indian telephone line.I am currently
> using TDM400P
Hi,
I've experienced excessive delay when called from one extension number
to another...
This happened unstable, as the delay range between 2 - 20 seconds...
I'm using Duron 950 MHz with memory 256 MB as asterisk server and my
asterisk currently serves 30-40 accounts..
Concurrent calls vary
Robert Christian wrote:
Does anyone know of a way to make a standard analog phone plugged into an
FXS port do something other than get a dialtone when you pick it up? For
example, if the phone should automatically ring someone or play a greeting
when picked up without having to enter an extensio
All of a sudden, my account doesn't appear to work, or even perhaps
exist with SIPPhone. Is anyone else having trouble?
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My experience with an MP-108 was similar. Incredibly complex to setup,
and very little help from MFR, or even ABPTECH, the main US reseller.
We just couldn't get it working properly.
Ended up with a TE110P with an Adit 600 channel bank, which ROCKS.
Unbelieveably easy to setup. No echo whatsoeve
I'll give it a shot.. Do you know if they have any plans to merge this in?
On 8/8/05, Gary Reuter <[EMAIL PROTECTED]> wrote:
> On 8/8/05, Jason DiCioccio <[EMAIL PROTECTED]> wrote:
> > I guess the problem is with SIPPhone then. I opened a ticket with
> > them. I'll post their response when I hav
I'll bet a slab
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Craig
Sent: Tuesday, August 09, 2005 11:25 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] X100P with Caller-ID in Australia,
You require no changes to detect caller id with a
> > Do you use this? Are you happy with it's performance? My experience a
> > year ago with other small FXOs was very dissappointing. I tried the
> > very earliest SPA-3000, TDM-400 and X101p. None were satisfactory so
> > resorted to call forwarding my two POTS lines to a lines from an ITSP.
> No
Leo Burd wrote:
Hello everyone,
I'm having all sorts of problems with my PHP AGI scripts... Basically,
my scripts run fine from the command line and don't do anything well
called from Asterisk. Here are my questions:
Probably because it Asterisk is trying to give you info.
a) Does Asteris
On 8/8/05, Jason DiCioccio <[EMAIL PROTECTED]> wrote:
> I guess the problem is with SIPPhone then. I opened a ticket with
> them. I'll post their response when I have one.
>
I wouldn't bet money on that yet...
I've seen identical DTMF problems (doubled and mangled) digits and
I've never used SIP
Chris Mason wrote:
JP Carballo wrote:
Chris Mason (Lists) wrote:
Overall, I'm happy. It has sturdy construction, standard features,
and most of all works just fine with *.
Did you find any noise, hum or gating on the FXO ports on incoming
calls?
Nope. Crystal clear calls.
That's
Has anyone written a php/perl or a hack to the 'hint' function in
Asterisk that will let you monitor a MeetMe conference?
So if anyone was in a conference, I could have a button light up on my
Snom 360?
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Robert Christian wrote:
Does anyone know of a way to make a standard analog phone plugged into
an FXS port do something other than get a dialtone when you pick it
up? For example, if the phone should automatically ring someone or
play a greeting when picked up without having to enter an exten
I guess the problem is with SIPPhone then. I opened a ticket with
them. I'll post their response when I have one.
Thanks!
-JD-
On 8/8/05, Tarpo, Louie <[EMAIL PROTECTED]> wrote:
> I find a verbosity of 10 (asterisk -rvv) gets me adequate logging for
> my purposes. I've been really pou
Steve Underwood wrote:
JP Carballo wrote:
I'll post the VG-400 settings here and on voip-info.org as soon as I
get my notes together.
Overall, I'm happy. It has sturdy construction, standard features,
and most of all works just fine with *.
I haven't used the VG-400, but I have used their
Was using this monitor line to get soxmix to mix test-in.wav and test-
out.wav into test.wav.
exten => 1200,1,Monitor(wav|/tmp/test|m)
When I start the conference, the * console shows this:
monitor executing ( nice -n 19 soxmix "//tmp/test-in.wav" "//tmp/test-
out.wav" "//tmp/test.wav" && rm
On Monday 08 August 2005 21:19, Robert Christian wrote:
> Does anyone know of a way to make a standard analog phone plugged into an
> FXS port do something other than get a dialtone when you pick it up? For
> example, if the phone should automatically ring someone or play a greeting
> when picked
You require no changes to detect caller id with a x100p card in Au,
however most carriers only provide cid as an extra cost option. Your
most likely problem is CID is not being sent to you. If you believe you
have the caller id being sent to you, Find a caller id box/phone and
confirm it is there.
Does anyone know of a way to make a standard analog phone
plugged into an FXS port do something other than get a dialtone when you pick
it up? For example, if the phone should automatically ring someone or
play a greeting when picked up without having to enter an extension?
- Robert
JP Carballo wrote:
Chris Mason (Lists) wrote:
Overall, I'm happy. It has sturdy construction, standard features,
and most of all works just fine with *.
Did you find any noise, hum or gating on the FXO ports on incoming
calls?
Nope. Crystal clear calls.
That's PSTN -> VG-400 -> Ph
On 8/8/05, Jon Whitear <[EMAIL PROTECTED]> wrote:
>
> >Most likely your current card will work in Australia, but you need to
> >patch the Asterisk Source to support the Australian Caller ID standard.
> >
> Yes, I've done that (and just to make sure, I've just upgraded to 1.0.9,
> and applied that
Colin E. McDonald wrote:
The new update seems to have cured my issue with calls intersecting and
Zap lines not being hung up after the user terminates the session but
now I am having sound issues with all of my phones. The sounds seems to
be very low on all of them and there is a definite change
JP Carballo wrote:
I'll post the VG-400 settings here and on voip-info.org as soon as I
get my notes together.
Overall, I'm happy. It has sturdy construction, standard features, and
most of all works just fine with *.
I haven't used the VG-400, but I have used their 1 port unit. The only
h
> Are there any Asterisk interfaces with .NET?
There is a port of the Manager API implementation of Asterisk-Java
available for .NET from Chad Kitching.
You can download it from http://www3.mb.sympatico.ca/~chadk/
=Stefan
signature.asc
Description: This is a digitally signed message part
__
I find a verbosity of 10 (asterisk -rvv) gets me adequate logging for
my purposes. I've been really pounding on our sipphone number the past half
hour or so and I'm seeing the same issues you are. Sometimes it hits
correctly, sometimes it doesn't. IE, Dialing 5954, some of the times
Hi,
Are there any Asterisk interfaces with .NET?
Thanks,
Alvin
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I have implemented a simple findme solution based on DID's.
In the findme context, after trying each respective number (at s,5 and
s,6), I would like a voice saying "The person was not available, press
pound to try the next number." Otherwise, it hangs up after 20 seconds
without dialing the nex
Morning,
I've installed asterisk on FC4 and had a few problems with zaptel stuff.
Having installed it on SuSE I was able to check a few things that were
different.
When installed with the RPMS, the udev stuff gets put into the normal 50-udev
file under /etc/udev. On SuSE, it worked when the n
The setting for Australia looks correct so that means I'm not setting
the country code correctly, despite what I thought.
How can I determine if Asterisk has the correct country code? (I did
try to check this again on the Wiki but it's down at the moment :(
tony
David Phelan wrote:
Have a
The new update seems to have cured my issue with calls intersecting and
Zap lines not being hung up after the user terminates the session but
now I am having sound issues with all of my phones. The sounds seems to
be very low on all of them and there is a definite change from the same
set when it w
Could someone please look at this information and help me decipher what it
should actually mean to me? I've found a bit of information here and there
but I'd like to know what I'm supposed to be reading into this information:
pbx*CLI> iax2 show channels
Channel Peer Usern
jonny hashem wrote:
my customers complain that when they make a call they
hear the another side very well but the another side
hears the first side well but in low sound.what is the
ptoblem here and i have to change?
Could you be more vague?
Try giving us hardware, relevant
What are the advantages of using woomera IAX2 instead of native IAX2?Put woomera aside right now, This is something that brings a cross platform IAX2 stack that can for example be used in Gnomemeeting or anything else that uses OPAL, using a closed and open familiar API. This can be used on window
On 8/8/05, Tarpo, Louie <[EMAIL PROTECTED]> wrote:
> RFC2833 is sent out of band. What's the output on your asterisk console?
I don't see any output during this time on my asterisk console.
Unless there's additional logging I'd need to enable?
Thanks for the help!
-JD-
_
As I recall, should channels start as channel=>2 and not channel=2?
I have all mine config'ed channel => 2 and it works fine...
Greg
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Time
Bandit
Sent: Saturday, August 06, 2005 3:39 PM
To: Asterisk Users M
Hello All,
Right now I have several providers. Voipjet, Teliax, and more recently
Broadvoice.
Broadvoice gives me unlimited to europe, but what I want to do is
determine the best way to setup a dialplan so for example, certain
countries will go through the cheapest route.
I am really only inter
Good day,
I would recommend using an LCR engine to do this. There is at least one
listed in the wiki. I am also nearing completion of an lcr engine that
integrates with ASTPP, asterisk billing software. It will be easy to
setup once I get it working. :-)
Darren Wiebe
[EMAIL PROTECTED]
[E
I've posted about this before, but it's been so long I thought I'd see
if there is a new solution (can't find anything on google or wiki)
I use the Record() app on my incoming zap calls to record a persons name
if their caller id is not in the db. After the name is recorded, the
call is parked an
> As I recall, should channels start as channel=>2 and not channel=2?
>
> I have all mine config'ed channel => 2 and it works fine...
>
> Greg
Yes, thanks for the correction.
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We start asterisk under a detached screen with GNU screen. We use the
'L' flag to log to a text file if we want. Then we can "screen -r" to
the actual asterisk CLI any time we want.
/usr/bin/screen -L -d -m -S asterisk /usr/sbin/asterisk -vgc
http://www.gnu.org/software/scree
On Monday August 08 2005 7:06 pm, Carlos Chavez wrote:
> On Mon, 2005-08-08 at 18:42 -0400, John Millican wrote:
> > Hello all,
> > I know this has been covered on list but can not find the answer I need,
> > lots of references to no authority found, but none with an answer.
> > I have two * server
RFC2833 is sent out of band. What's the output on your asterisk console?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Jason
DiCioccio
Sent: Monday, August 08, 2005 5:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asteris
Since Thursday (Aug/4/2005) my Vonage line that is fowrwarded (not
even using their ATA) to a number that rings to a PRI that is
connected to asterisk stop passing on DTMF, anybody having this issue?
When calling directly the asterisk box, or forwarding any other lines,
there are no problems. Just
So the way I understand this is with rfc2833, DTMF is sent out of
band. So does this mean that SIPPhone is interpreting the tones
incorrectly? Asterisk shouldn't be doing any actual tone detection
with this method, right?
On 8/8/05, Tarpo, Louie <[EMAIL PROTECTED]> wrote:
> Yes we are. I just d
Hi:
In our existing call center, we defined agents in different tasks, some of them
are assigned as primary for a given task, for other tasks as overflow, which we
want agents to work with some projects flexibly. Does Asterisk queuing can
handle this kind of routing mechanism?
Thanks,
Tielin
On Mon, 2005-08-08 at 18:42 -0400, John Millican wrote:
> Hello all,
> I know this has been covered on list but can not find the answer I need, lots
> of references to no authority found, but none with an answer.
> I have two * servers, one behind firewall with nat the other on a dmz with
> nat.
Hello all,
I know this has been covered on list but can not find the answer I need, lots
of references to no authority found, but none with an answer.
I have two * servers, one behind firewall with nat the other on a dmz with
nat. Both servers register with each other successfully.
home is tod
Have a look at the indications.conf file
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tony Davidson
Sent: Tuesday, 9 August 2005 8:17 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Detecting hangup - TDM
Chris Mason (Lists) wrote:
Overall, I'm happy. It has sturdy construction, standard features,
and most of all works just fine with *.
Did you find any noise, hum or gating on the FXO ports on incoming calls?
Nope. Crystal clear calls.
That's PSTN -> VG-400 -> Phone and PSTN -> VG-400
I've searched the Wiki and this forum with little success. I have a
TDM400P in my server which functions fine. Except it will continue
ringing about 3 times after hangup. I.e. it's failing to detect the
hangup tone.
I was previously running a Sipura 3000 and had the same issue. After
resea
Hi,
i am using a HFC-4S Board with the bristuff patches from Junghans.
3 of 4 ISDN-NTs working well but the first one of this group making troubles.
(all in TE Mode.)
every 10 minutes all active lines getting disconnected from this NT :-(
asterisk showing up this information:
== Primary D-Cha
On Mon, 2005-08-08 at 16:33 -0500, Eric Wieling aka ManxPower wrote:
> Bryce Chidester wrote:
> > On Mon, 2005-08-08 at 17:03 -0400, Craig Bruenderman wrote:
> >
> >>I'm attempting to set up call recording with Asterisk. Using
> >>
> >>automon => *1 ; One Touch Record
> >>
> >>i
Hi,
my zaphfc is flooding my syslog with two messages (even without asterisk
running). Is this "normal"?:
--
zaphfc: bchan rx fifo not enough bytes to receive! (z1=1360, z2=1353,
wanted 8 got 7), probably a buffer overrun.
zaphfc: dropped audio
Really nothing new. They've done this with wired carriers for years.
--
Tom
On 8/8/05, Adam Megacz <[EMAIL PROTECTED]> wrote:
>
> Scary.
>
> http://www.eff.org/news/archives/2005_08.php#003876
>
> --
> PGP/GPG: 5C9F F366 C9CF 2145 E770 B1B8 EFB1 462D A146 C380
>
> _
History page is back.
I think it got too big for the software to deal
with.
I changed it to show only the last 100 versions.
There is an 'undo' option on the history page, but its never
worked correctly, and so I have not enabled it.
I'm working on a software upgrade that will hopefully addr
Scary.
http://www.eff.org/news/archives/2005_08.php#003876
--
PGP/GPG: 5C9F F366 C9CF 2145 E770 B1B8 EFB1 462D A146 C380
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Chris Mason (Lists) wrote:
Has anyone found a suitable but not exorbitant 4-6 port FXO => sip
gateway? I need something more compact than a channel bank and using
many Sipura 3000s is a bit cumbersome. Suggestions?
Yoda VG-400 325 USD or 183 QUID
--
JP Carballo
http://www.netfone2x.com
Bri
Bryce Chidester wrote:
On Mon, 2005-08-08 at 17:03 -0400, Craig Bruenderman wrote:
I'm attempting to set up call recording with Asterisk. Using
automon => *1 ; One Touch Record
in features.conf does not appear to be working. I'm using Polycom 501's
but when someone dials *1
On Mon, 2005-08-08 at 17:03 -0400, Craig Bruenderman wrote:
> I'm attempting to set up call recording with Asterisk. Using
>
> automon => *1 ; One Touch Record
>
> in features.conf does not appear to be working. I'm using Polycom 501's
> but when someone dials *1 while in a cal
Is anyone using services like Voicepulse, Nufone or Sixtel with IAX
Realtime? I simply cannot get those services to work no matter what I do. I
already have voicemail and sip running from realtime, I also have a second
Asterisk server that connects using IAX2 and that one works perfectly wit
Overall, I'm happy. It has sturdy construction, standard features, and
most of all works just fine with *.
Did you find any noise, hum or gating on the FXO ports on incoming calls?
--
Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
Int: (305) 704-7249 Fax: (815)301-9759
Cell:
Rosario Pingaro wrote:
We need to configure asterisk to authenticate two sip ATAs, but the
stream must go directly from one to another ata without tuching asterisk.
Is this possible adding canreinvite=yes into sip.conf?
is it true laso if asterisk doesn't recognize the spd (t38)?
thanks
is there some one tha has bee able to passthrough
t38 into asterisk?
thanks
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Tzafrir Cohen wrote:
On Mon, Aug 08, 2005 at 03:50:02PM -0500, Eric Wieling aka ManxPower wrote:
Anish Basu wrote:
Hi,
For some reason, my AGI perl scripts cannot write to the CLI console using
standard error. I ran the agi-test.agi test script that came with asterisk
and verified that the
You dont run applications using the Manager API as AGI scripts but as
standalone Java applications.
So in you case proably via
java -cp asterisk-java-0.1.jar:. ManagerAPI
=Stefan
On Mon, 2005-08-08 at 16:31 +0530, Bharat M. Sarvan wrote:
> Ok Mr. Stefan,
> The contents of the file f
I'm attempting to set up call recording with Asterisk. Using
automon => *1 ; One Touch Record
in features.conf does not appear to be working. I'm using Polycom 501's
but when someone dials *1 while in a call, nothing happens.
I'm wondering if the phone or Asterisk is even det
On Mon, Aug 08, 2005 at 03:50:02PM -0500, Eric Wieling aka ManxPower wrote:
> Anish Basu wrote:
> >Hi,
> >
> >For some reason, my AGI perl scripts cannot write to the CLI console using
> >standard error. I ran the agi-test.agi test script that came with asterisk
> >and verified that the problem wa
Anish Basu wrote:
Hi,
For some reason, my AGI perl scripts cannot write to the CLI console using
standard error. I ran the agi-test.agi test script that came with asterisk
and verified that the problem was not with the code. Asterisk is always
started with 4 or more v's, yet this the CLI outpu
I am using Asterisk CVS-D2005.06.24.04 and I am trying to figure out where
the DB information such as DND and CFIM are stored. I checked
/var/lib/asterisk/astdb, but the file seems to be the same size even after
adding information to the DB. I would like to able to share this database
across mult
We need to configure asterisk to authenticate two
sip ATAs, but the stream must go directly from one to another ata without
tuching asterisk.
Is this possible adding canreinvite=yes into
sip.conf?
is it true laso if asterisk doesn't recognize the
spd (t38)?
thanks
Rosario
_
I'm sorry, I didn't take a close look at your conf files, so I dont
know if this is your problem, (and anyway, I dont know enough about
[EMAIL PROTECTED] to know if the same things is a problem there)
however, it might be useful for others, sipgates documentation is
"weak" over the issu
Arik Funke wrote:
can anybody tell me how to create an extension that starts with a *?
The expression matching works well if * is embedded in numbers but if
the extension starts with *, it is not executed but extension s
instead. Is there another way besides using a lot of if statements in
th
Hi,
For some reason, my AGI perl scripts cannot write to the CLI console using
standard error. I ran the agi-test.agi test script that came with asterisk
and verified that the problem was not with the code. Asterisk is always
started with 4 or more v's, yet this the CLI output does not show up.
Today the front page of http://www.voip-info.org/ was taken out by a
spammer. It also seem the history page for http://www.voip-info.org/ was
also nuked. I've restored the best I could using google cache, but still
missing some information.
Who is an admin on http://www.voip-info.org/ and
Hello,
I am interested in offering the voicemail web access to faculty & staff as a
part of my new Asterisk
deployment, and was just wondiering about the security implications.
Have there been exploits for it? Is it ready for primetime use? What kind of
things should
I be wary of about this
With the lack of info on Yoda Communications in Taiwan and their
hardware, I thought I'd post my experience.
I got my hands on a few H.323 VG-400's and VG-100TA's.
http://www.yoda.com.tw/model.php?type=VoIP_Solution&pname=VG400
2 of the VG-400's were 2FXO/2FXS models.
A couple were deployed to
Bill Wesson wrote:
Chris,
I'm testing a Grandstream GXP-2000. It supports multiple MWI.
Very nice! I didn't know about thatis there anything specific you
have to do to associate a softbutton with a particular extension's
voicemail so the MWI works? I didn't see anything about this in
Do you use this? Are you happy with it's performance?
I am using the SPA-3000 with good results, I just want a more
professional solution.
--
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NetConcepts
(264) 497-5670 Fax: (264) 497-8463
Int: (305) 704-7249 Fax: (815)301-9759
Cell: 264-235-5670
Yahoo IM: [EMAIL PROTECTED]
Hello everyone,
I'm having all sorts of problems with my PHP AGI scripts... Basically,
my scripts run fine from the command line and don't do anything well
called from Asterisk. Here are my questions:
a) Does Asterisk require PHP CLI or CGI? From the command line, my
script seems to work
Johan Nordström wrote:
Try send an email to [EMAIL PROTECTED] I did regarding an other
issue and recieved an answer from [EMAIL PROTECTED] .
Johan
There are two major products that come out of Berkeley: LSD and UNIX.
We don't believe this to be a coincidence. -- Jeremy S. Anderson
Paul Bel
Bastian Schern wrote:
Hello everybody,
now I'm using MySQL for SIP/IAX friends and CDR. Is it also possible to
use PostreSQL instead of MySQL?
Regards
Bastian
If you want to go thru the hassle of installing ODBC and all related
stuff to run PSQL, sure you can.
-Matthew
P.S. stick with
res_mysql.conf controls the RealTime interface driver to MySQL.
cdr_mysql.conf controls the MySQL CDR Addon.
Are you running CVS-HEAD? Have you installed res_config_mysql.so?
What happens when you type "realtime mysql status" ?
Did you look in the debug log for errors?
-Matthew
Wei Kun wrote:
Hi:
I unseccessfuly tried to place a call from one IAX
behind a nat through another IAX with a real IP. I got
the following error message on Asterisk Console:
Executing Dial("OSS/dsp", "IAX2/wassim/01)") in new
stack
-- Called wassim/01)
Call rejected by 195.112.214.98: No such
context/exten
hi
Digium card is compatible with the indian telephone line.I am currently using TDM400P(TDM11B) and this is working fine.It has one FXS and one FXO interface .
rajeshAnkit <[EMAIL PROTECTED]> wrote:
Hi everybody,I need a little clarification regarding the hardware to be used with asterisk. I w
Wei Kun wrote:
Hi;
In case your * and mysql are running on the same machine, and you get error
"Failed to connect to mysql database server ..." when using Asterisk with
Mysql database, check the location of mysql.sock
not /tmp/mysql.sock, but /var/lib/mysql/mysql.sock
Regards
Kun
The locatio
Yes we are. I just double checked our line, and oddly, the dtmf tones aren't
getting sent to our asterisk server. Switched it back to rfc2833, and it
works. It was the other way around when I first connected us. Some informal
testing just now doesn't show the DTMF tone problem in rfc2833 mod
Andrew Kohlsmith wrote:
On Monday 08 August 2005 10:56, Kib Eki wrote:
Misunderstanding: I need to change the calleridnum because there is missing
the 0 before the number.
SetCIDNum(0${CALLERIDNUM}) or something?
Not to be nitpicky, but * will complain that SetCIDNum is deprecated
On Mon, 8 Aug 2005, Kib Eki wrote:
> >>2. A call made from a SIP client to the outside lacks the extension in the
> >>number: Ex: PSTN number is 6789-0. The extension 234 is not added to the
> >>PSTN number like 6789-234 when dialing out over the PSTN.
> >
> >
> > Again, trivial dialplan stuff.
On Mon, 8 Aug 2005, Eric Wieling aka ManxPower wrote:
> Peter Svensson wrote:
> > See internationalprefix, nationalprefix etc in the file zapata.conf.
>
> Those options are only available in BRIStuff.
They have been in HEAD for quite some time. The 1.0.x-releaes are note
really usable in a lot
I am working with some Polycom IP600's and for outgoing calls from the
phones, the presence features work fine. I am utilizing the
100,hint,SIP/100
for these calls. The problem that I am seeing is that any inbound
calls that originate from a IVR in which the extension is dialed, does
not show
I am having quality problems on SIP bound calls made over the Zap channels.
All Sip only calls (Cisco phone through Asterisk to another Sip device sound
fine).
Our setup looks like this:
User --> Executone PBX --> Asterisk Server --> Router --> Internet
The user is using a legacy handset that wo
Louie,
On 8/8/05, Tarpo, Louie <[EMAIL PROTECTED]> wrote:
> We encountered the same issue. change dtmfmode=rfc2833 to dtmfmode=inband
> and make sure you're using a ulaw connection. If you use a lossy codec, it
> will scramble the DTMF tones.
Are you using SIPPhone? When I use dtmfmode=inban
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