Already set.
Andrew
_
Andrew Yager
Real World Technology Solutions
Real People, Real SolUtions (tm)
ph: (02) 9945 2567 fax: (02) 9945 2566
mob: 0405 15 2568
http://www.rwts.com.au/
_
On 25/06/2004, at 3:38 PM, Peter Boot wrote:
I had the same problem
I had the same problem when using a Grandstream 486 I solved it by using the
nat=yes config option
>>-Original Message-
>>From: [EMAIL PROTECTED]
>>[mailto:[EMAIL PROTECTED] On Behalf Of
>>Andrew Yager
>>Sent: Friday, June 25, 2004 3:31 PM
>>To: [EMAIL PROTECTED]
>>Subject: [Asterisk-Use
Hi,
I'm confused as anything by this bug. I'm hoping that it is just
something screwy in my config.
I have 1 Cisco 7960 and several Grandstream BT101 & 102's, and a Digium
TDM31B.
I'm running the latest CVS (CVS-HEAD-05/27/04-17:22:40) of both
Asterisk and the Zaptel driver on Fedora Cora 1.
try tcpdump -i lo port 5432 or icmp
(or tethereal if you have it)
Prehaps it's trying a UNIX socket connection?
also, please change your database password as you've now supplied
ip,user,pass to the mailing list :) Hopefully, you've got it restricted
to localhost
Caleb Kow wrote:
Here we go:
[EMA
Here we go:
[EMAIL PROTECTED] root]# netstat -ap
Active Internet connections (servers and established)
Proto Recv-Q Send-Q Local Address Foreign Address
State PID/Program name
tcp0 0 *:32768 *:*
LISTEN 3221/
tcp0
On Thu, Jun 24, 2004 at 09:52:45AM -0700, Chris A. Icide wrote:
> It sounds like what you are looking for is an Asterisk-wide (or perhaps
> channel-specific) preserve_codec option. Where preserve_codec=1 means that
> asterisk tries to preserve the originating codec if at all possible, and
> pre
Steve Underwood wrote:
There is not simple answer to that. If you configuration ensures zero
timing hiccups then it should work. If any timing mismatches are handled
by the system, a fax in progress ay that moment will fail. A retry might
get it through. It depends a lot on the rate of timing hi
On 2004.06.24 18:28 Kevin P. Fleming wrote:
Lee Howard wrote:
I stand corrected.
After a little bit of work with the fax application to adjust the
timings (increasing all of the pauses), all is well with V.17 also.
I assume you're using no compression (G711u) between the X100P and
the SPA-2000, t
Bonzo Armstrong wrote:
On Wed, Jun 23, 2004 at 08:24:03PM -0700, Bonzo Armstrong wrote:
On Mon, Jun 21, 2004 at 03:04:52PM -0500, Nik Martin wrote:
Try this if possible. Connect the channel bank to * via the 400' cable, but
in the same room as the * box, with all the cable coiled on the floor.
N
Hi all,
I am trying to setup music on hold.
In extensions.conf did put the following:
exten => 6601,1,WaitMusicOnHold(30)
Using a sip phone, x-lite, after I dialed 6601 I get the following:
-- Reloading module 'cdr_csv.so' (Comma Separated Values CDR Backend)
-- Executing WaitMusicOnHold
Hi all,
I am having problems with some X100P equivalent cards.
They worked fine under RH9 and Fcore1, but under Fidora core 2 they are
detected as
In /etc/sysconfig/hwconf
"Individual Computers - Jens Schoenfeld|Intel 537"
Which is NOT what they are under the previous verions...
And ztcfg cannot
Kevin P. Fleming wrote:
Lee Howard wrote:
I stand corrected.
After a little bit of work with the fax application to adjust the
timings (increasing all of the pauses), all is well with V.17 also.
I assume you're using no compression (G711u) between the X100P and the
SPA-2000, then. Are you findin
Lee Howard wrote:
I stand corrected.
After a little bit of work with the fax application to adjust the
timings (increasing all of the pauses), all is well with V.17 also.
I assume you're using no compression (G711u) between the X100P and the
SPA-2000, then. Are you finding this to be a reliable c
My IP 600 gives me a "call-waiting" tone when another call comes in.
I'm quite sure there's a setting for that in the xml. As for the
feature buttons, I'll look at that this weekend, but it seems to me that
using SIP with these phones precluded alot of the key programming stuff
i.e., there was
On Thu, 24 Jun 2004 18:04:47 -0500, Eric Wieling <[EMAIL PROTECTED]> wrote:
>
> How is this different from the way standard call waiting works when
> provided from your telco?
I think what Brian wants is a way to disconnect the current call and
take the new call, rather than putting the current c
Mark Elkins [EMAIL PROTECTED] wrote:
> Um - If my secretary transfer's a call from her BT101 to her
> own number
> - she looses the call. What can I do to stop this from
> happening - apart from dyeing her hair from blond to brunette ???
>
> Shouldn't Asterisk refuse to do this?
Why on earth woul
Probably an impedance problem. PCM line signals are designed to be
transmitted over a *twisted telephone cable* having 120 ohms at 1 MHz.
I'm not sure that cat6 cable fulfil this requirement.
Maybe cat3.
Jorge
Bonzo Armstrong wrote:
On Wed, Jun 23, 2004 at 08:24:03PM -0700, Bonzo Armstrong wrote
>The last char on the line needs to be a ) not a }
>
>"w" only works on ANALOG PSTN ports.
>
>Recent CVS -head has an option to send digits after the remote side
answers. See "show application dial"
I'm running Asterisk CVS-HEAD-06/15/04-17:27:40
I now have this in my dialplan:
exten => _91NX
Basically outgoing calls through zap channels doesn't detect that the other
end answered. In my cdr I see hang-up no answer, plus the console shows that
the channel is ringing..while I am actually talking to someone. Incoming
calls seems to be fine.
Wojtek
- Original Message -
From: "Wojci
Have recompiled everything for Zaptel without
any errors on FC2 2.6 and very happy!
Anyway when I load it up at start I get
zaptel framework [ok]loading zaptel hardware modules:running ztcfg:
Zaptel ZT_CHANCONFIG failed on channel 1: No such device or address
(6)
I have tried everything, th
Use the call file, and set the channel to something like:
Zap/1/160w
then set the extension/context/etc to point to something like this:
exten => 100,1,playback("Now call will be recorded")
exten => 100,2,Record("some file")
exten => 100,3,playback("beep")
Now, to stop recording you have two cho
Title: Message
I am interested in
possibly building a few Tormenta 2 Rev B cards for myself. Before I
get much further, though, I would like to determine the
difference between the T2B card and the Digium Quad-span T1
cards.
I get the general
impression that they are similar, but the D
I have similar problem with outbound calls...
Wojtek
- Original Message -
From: "Brent Franks" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Thursday, June 24, 2004 7:16 PM
Subject: RE: [Asterisk-Users] tdm (and x100p?) echo - fix is coming!
>
> > -Original Message-
> > From
Tobi,
By the looks of it your problem has nothing to do with chan_capi.
Jun 24 22:19:49 WARNING[1086696368]: pbx.c:1819 ast_pbx_run: Channel
'CAPI[contr1/**some_number**]/0' sent into invalid extension 's' in
context 'default', but no invalid handler
Have a look at the error message - it tells yo
I'm new to asterisk but I've been managing an Artisoft Televantage
server for about 2 years.
My question is, does Asterisk has the capability to 'classify calls'
based on the dialed number?
This is called 'Call Classifier' in Televantage jargon wherein it lets
call center agents identify callers, t
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Rich Adamson
> Sent: Thursday, June 24, 2004 5:01 PM
> Be careful with that thought... here's the three lines that were
> manually changed for testing purposes only (these would have bee
Hi,
I have an X100P card and Asterisk installed on a system running Redhat
8. I have a normal phone line plugged into a wall jack on one end, and
plugged into the "Line" port on the X100P card on the other end. I am
trying to make an outgoing call to a local phone number using a *.call
file b
I believe the proper way to do this is use defaultip=000.000.000.000
rather than host=dynamic. Learned this one the hard way, just like
you.
Brian
On Thu, 24 Jun 2004 17:23:58 -0400, Jeremy McNamara <[EMAIL PROTECTED]> wrote:
>
> Matt wrote:
> > NOTICE[-1147675728]: Peer '004' is trying to
How is this different from the way standard call waiting works when
provided from your telco?
On Thu, 2004-06-24 at 15:47, Brian Capouch wrote:
> I have mailed the list on this topic, asked on the IRC channel, and even
> called Digium. I have two customers who are pretty upset about this,
> and
Please ignore my problem, I just added faxdetection to zapata.conf and
everything is back to normal.
Thanks,
W
- Original Message -
From: "Wojciech Tryc" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Thursday, June 24, 2004 6:37 PM
Subject: [Asterisk-Users] fax detection
> Everything
Hi,
Speaking of programming the IP 600, does anybody know how to programm
any of the "feature" buttons to send a combination of digits while a
call is in progress? The most obvious use would be to send "#700" while
a call is in progress and label the button "Park". If I could do that, I
would b
Everything but fax detection seems to be fixed in the latest CVS.
Anyincoming fax on Zap channel does not get detected. Anyone?
Thanks,
Wojtek
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To UN
On Tue, 2004-06-22 at 21:44, Aaron J. Angel wrote:
> After doing some quick research, it appears HANGUPCAUSE is only implemented
> in chan_zap and chan_sip. What about the other channels?
They are out of luck until someone creates a patch to add that feature
to the other channels. I belive chan_
The last char on the line needs to be a ) not a }
"w" only works on ANALOG PSTN ports.
Recent CVS -head has an option to send digits after the remote side
answers. See "show application dial"
On Thu, 2004-06-24 at 15:11, Josh Reineke wrote:
> Here is an example of my dialplan, where 1234 is t
That is totally and completely wrong. Asterisk supports both alphanum
extensions and sip peers and iax peers.
On Thu, 2004-06-24 at 14:51, aaron wrote:
> asterisk does not support alphabet extensions.
>
> On Thu, 24 Jun 2004 05:53:53 -0700 (PDT), Pete Rose <[EMAIL PROTECTED]> wrote:
> >
> > Gre
Chris,
Do you get echo issues? If not could you let us have your config and
which echo canceller you use.
Thanks
Chris
On Thu, 2004-06-24 at 20:40, Chris Lee wrote:
> Chris Stenton wrote:
> > I am finding that I have to increase the txgain in zapata.conf to 8 when my
> > X101P is connected to m
Caleb Kow wrote:
Results of netstat -ap
You seem to be missing the top part of the output which looks like this:
[EMAIL PROTECTED] build]# netstat -ap
Active Internet connections (servers and established)
Proto Recv-Q Send-Q Local Address Foreign Address State
PID/Program nam
>
> Sorry guys... These are all great tips, but also this doesn't work: the
>gateway is not under my control, it is actually a real phone switch,
>which
>isn't owned by us. Unfortunately I can't tell them to add a second IP ...
>:-)
As I could understand so far, you wanna do G729 passthu from a S
Action "Command", which I just discovered, is now my friend:
> Action: Command
> Command: Dial 123456
>
seems to fit well for a softphone application
with the local soundcard.
At least it has solved my problem and works well for me.
Roger.
___
Asterisk-Us
On Thu, 24 Jun 2004, Matt wrote:
> NOTICE[-1147675728]: Peer '004' is trying to register, but not
> configured as host=dynamic
>
> from all the phones I've set as host=xxx.xxx.xxx.xxx
It sounds like * is letting you know that it got a registration attempt
where none was expected.. Have you
Matt wrote:
NOTICE[-1147675728]: Peer '004' is trying to register, but not
configured as host=dynamic
How much clearer do you need to be? Asterisk is telling you exactly what
the problem is. Have you tried simply doing host=dynamic into your iax.conf?
If you still want IP based access co
Sorry for thinking out loud here. The bracket at the end of the first line
should have been a parantheses.
-- Executing Dial("SIP/4299-cf70", "Zap/g1/190747496061234") in
new stack
-- Called g1/190747496061234
-- Hungup 'Zap/1-1'
Now I get the prompt for the access code,
Hi!
> NOTICE[-1147675728]: Peer '004' is trying to register, but not
> configured as host=dynamic
>
> from all the phones I've set as host=xxx.xxx.xxx.xxx
The explanation is simply: "Registration" only makes sense for
host=dynamic settings. The only purpose of registering is to tell the
Hi!
> Our telco has setup toll access account codes for outgoing calls. I would
> like to include these account codes in the dialplan for certain extensions
> (fax lines, modems) so that they are not prompted for the 4 digit code when
> making a toll call.
Look at Dial() with option D:
D(digits
Stefan de Konink [EMAIL PROTECTED] wrote:
>
> Since 21 june skype is available to be used on Linux, with a static
> binary, which includes QT, of 8 meg its big.
>
> I presume, with some hacking, there could be a possibility to use the
> Skype program as a Channel. (Eq. Skype is started, and with
Oops! stupid error. I forgot to add the mailbox context
I have some new messages in mailbox 824, but the manager command
reports this:
Response: Success
Message: Mailbox Message Count
Mailbox: 824
NewMessages: 0
OldMessages: 0
The MailboxStatus command also reports 0 for message status.
__
Simon Brown [EMAIL PROTECTED] wrote:
> There is a very good and working WinCE IP phone available from SJPhone.
>
Marvellous. Microsoft will bring their legendary stability, security
and reliability to the VoIP world.
Oops - there goes my lunch.
--
_/ _/ _/_/_/_/ _/_/ _/_/_/ _/
I have mailed the list on this topic, asked on the IRC channel, and even
called Digium. I have two customers who are pretty upset about this,
and it hasn't helped my standing with them that I have so far in several
months not been able to help them.
It seems a simple request: I have a system w
Hi,
I installed Asterisk with CAPI support. Everything works fine while
starting Asterisk, but when a call comes in Asterisk hangsup the call
after two times of ringing.
The output is like:
Jun 24 22:19:49 NOTICE[1082178480]: chan_capi.c:1931 capi_handle_msg:
CONNECT_IND ID=002 #0x011d LEN=0048
Caleb,
postgresql is usually run under user postgres and is very sensitive to
config files permissions. Both postgres.conf and pg_hba.conf should be
owned by postgres:postgres and have 0600 permissions. I often get the
same error after editing these files and leaving them owned by root :)
Ivan
Thanks very much.
That fixed it.
On Thu, 2004-06-24 at 14:31, Brancaleoni Matteo wrote:
> checkout the libpri and compile & install
> *before* asterisk
>
> Matteo.
>
> Il gio, 2004-06-24 alle 19:01, Joseph ha scritto:
> > Just did a new cvs download and then tried to compile.
> >
> > I get th
Freddy Setiawan [EMAIL PROTECTED] wrote:
> Hi there, linux got so many distro, but which one that have more
> compability with the Asterisk?
>
My Asterisk server is running on Gentoo, with the 2.6.7-gentoo-r5
kernel. The Zaptel drivers work nicely too. I should think that any
of the GNU/Linux d
Here is an example of my dialplan, where 1234 is the account code required
by my telco for a toll access call.
exten => _91NXXNXX,1,Dial(Zap/g1/${EXTEN:1}1234}
exten => _91NXXNXX,2,Congestion
When I include the 1234, with any variation of www before the code, I
get a "Reorder" err
> > > I don't think that I'd play around with other values though. There were
> > > three lines of code changed in chan_zap.c and two of those lines were
> > > dependent on values from each other. Example: changing from 400
> (default)
> > > to 800 required another statement change from "w" to "ww"
On Thu, Jun 24, 2004 at 09:34:02AM -0400, Timothy R. McKee wrote:
> Looking back, I see you are running B8ZS/ESF. I ran into similar problems
> with a 100' run to a CAC AB-II. As soon as I switched to AMI/D4(SF) all my
> problems went away.
Did I say that? I've actually been running AMI/D4 to
On Wed, Jun 23, 2004 at 08:24:03PM -0700, Bonzo Armstrong wrote:
> On Mon, Jun 21, 2004 at 03:04:52PM -0500, Nik Martin wrote:
> >
> > Try this if possible. Connect the channel bank to * via the 400' cable, but
> > in the same room as the * box, with all the cable coiled on the floor.
Next best
Quoting Brian Weaver <[EMAIL PROTECTED]>:
> What sucks is there is no way to contact this company if you're not a
> subscriber.. Zip, notta.. No email address, phone number, nothing.
I receive an E-mail from them when we wanted to sign up for multiple lines of service.
It came
from: "[EMAIL P
Hi,
I was thinking of raising a feature request, but thought I ask here first.
Basically I am looking for options to allow more than one type on incomming
dtmf types for incomming voice calls.
I am using purely SIP on my system and to the best of my knowledge there is
only option to specify one t
asterisk does not support alphabet extensions.
On Thu, 24 Jun 2004 05:53:53 -0700 (PDT), Pete Rose <[EMAIL PROTECTED]> wrote:
>
> Greets,
> I have asteisk up and running, on asterisk console
> when i dial by extension (1) I see all
> transacitons of the call, but when I dial by name
> it seem
On Thu, 2004-06-24 at 13:01, Joseph wrote:
> Just did a new cvs download and then tried to compile.
>
> I get this error message:
> chan_zap.c:59:2: #error "You need newer libpri"
> Then there are some more chan_zap.c errors.
>
> Here is the cvs command:
> export CVSROOT=:pserver:[EMAIL PROTECTED
Chris Stenton wrote:
I am finding that I have to increase the txgain in zapata.conf to 8 when my
X101P is connected to my BT phone line, otherwise people can hardly hear me.
This then gives echo issues.
Do other people have the same problem on BT lines. I was wondering if I
should give BT a call an
> On Thursday 24 June 2004 11:01, Rich Adamson wrote:
> > Per the doc in the configs samples, you have to implement
> > echotraining=800 (instead of "yes")
> > to take advantage of the new code from yesterday.
>
> 800ms -- that's quite a delay between pickup and being able to speak; was that
> t
Hi all,
This is probably a really stupid question so I apologise in advance; I've
been looking at this all day and after 12hours I've got nowhere fast.
The situation is:
I've got a couple of 7960's on wireless adapters, the wireless network can
be, shall we say, a little flakey. The phones that
Ours works fine with the WRT54G with 128bit WEP and g729.
Timothy R. McKee
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jonathan Moore
Sent: Thursday, June 24, 2004 12:19
To: [EMA
Ah nice,
Let me know what SIP version you get, if it's any more recent than the one I
have, I'd love to get a copy ;-)
(There are some issues in my version that make the phone rather useless)
Florian
> -Original Message-
> Yep I stayed and was able to get through to their ip-phone
> su
Hi,
I am working on a project to record agent calls when completing specific
transactions with customers.
Since these calls do not go through the asterisk box (They go through a
lucent G3), we're thinking that service observe would be the easiest way to
accomplish our goal.
Here's what I need
Try building the kernel and the build the zaptel drivers. That worked
for me.
On Jun 24, 2004, at 1:20 PM, Tony Nichols wrote:
Still no go I have asked Digium tech support to look into it. I
need
the later cvs to get around a bug with the latest tdm400 card (load
driver - unload driver - lo
On Jun 24, 2004, at 9:56 AM, Asterisk wrote:
I did try that. Any help would be gratefully received.
Make sure you have the kernel source and try building the kernel. Or
at least start building it. Before I did that, I couldn't compile the
driver. After getting into the build proper of the ke
We have it working fine with g729 off of the WAP54g, but did try G711
under the WAP11 with no success. I don't think what I was running
into was a codec problem, because it would affect network layer
communications with a phone which did not register. I.e. pinging the
phone would yield about 7% r
checkout the libpri and compile & install
*before* asterisk
Matteo.
Il gio, 2004-06-24 alle 19:01, Joseph ha scritto:
> Just did a new cvs download and then tried to compile.
>
> I get this error message:
> chan_zap.c:59:2: #error "You need newer libpri"
> Then there are some more chan_zap.c err
I wanted to sign up for the pay as you go plan from iconnect
anyway, and see they have the Cisco ATA for $99 and the Grandstream
phone for $39.00
Anyone know if they ship these devices "locked"? I know iconnect
seems pretty friendly about letting any sip device connect.
What sucks is there is n
Hi Sam,
I have tried adding in the -i flag into the postgresql startup command
line but it displays the following upon starting up of Asterisk:
Jun 25 02:00:02 ERROR[1074494336]: cdr_pgsql.c:298 my_load_module:
cdr_pgsql: Unable to connect to database server localhost. Calls will
not be logged!
Our telco has setup toll access account codes for outgoing calls. I would
like to include these account codes in the dialplan for certain extensions
(fax lines, modems) so that they are not prompted for the 4 digit code when
making a toll call. I have played around with the 'w' command with ZAP
c
I'm running Asterisk CVS-HEAD-06/07/04, and I'm having problems with
the MailboxCount and MailboxStatus commands in the manager interface.
I have some new messages in mailbox 824, but the manager command
reports this:
Response: Success
Message: Mailbox Message Count
Mailbox: 824
NewMessages: 0
OldM
Results of netstat -ap
Active UNIX domain sockets (servers and established)
Proto RefCnt Flags Type State I-Node PID/Program
namePath
unix 2 [ ACC ] STREAM LISTENING 5881 3623/
/tmp/.iroha_unix/IROHA
unix 2 [ ACC ] STREAM LIST
Yep I stayed and was able to get through to their ip-phone support in
france. And with me only knowing english and the guy on the other end
speaking "broken" english we kinda hashed out that it was a bad stick of
flash ram in the phone. Communitech the USA provider for the phone is
overnighting me
Hi!After flashing the rom to the new version and reconfiguring the unit, many problems were solved...Unfortunatly today something strange seems to happen.The 3804 stopped sending registration request to the asterisk proxy.Here it is the most significant part of its configuration:proxy modeproxy ip
What type of configuration do you have setup with the phone? We found the units
to be pretty unusable unless you set them up to use g729. With the g729 on 128
bit wep doesn't seem to quite have enough horse power, but 64 bit wep and no wep
seem pretty workable. Before we tried the g729 we thought t
Still no go I have asked Digium tech support to look into it. I need
the later cvs to get around a bug with the latest tdm400 card (load
driver - unload driver - load driver again to make it work.
t o n y
On Thu, 2004-06-24 at 08:15, Tony Nichols wrote:
> On Wed, 2004-06-23 at 14:32, asterisk w
I am having the same, some people can just about hear me while others do
not say a thing or it is fine.
I can hear them fine.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris
Stenton
Sent: 24 June 2004 17:11
To: [EMAIL PROTECTED]
Subject: [Asterisk-Us
> I am finding that I have to increase the txgain in zapata.conf to 8 when
> my X101P is connected to my BT phone line, otherwise people can hardly
> hear me. This then gives echo issues.
Im having the same issue so far im on rxgain=2.0 and txgain=6.0. Seems to
work perfectly apart from the echo
Just did a new cvs download and then tried to compile.
I get this error message:
chan_zap.c:59:2: #error "You need newer libpri"
Then there are some more chan_zap.c errors.
Here is the cvs command:
export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot
cvs login
cvs checkout zaptel asterisk libpr
Inband dtmf does not work with speex(only ulaw). Switch your dtmf mode to
rfc2833. :)
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sola 2000
Sent: Thursday, June 24, 2004 11:16 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Speex
I a
> > I don't think that I'd play around with other values though. There were
> > three lines of code changed in chan_zap.c and two of those lines were
> > dependent on values from each other. Example: changing from 400
(default)
> > to 800 required another statement change from "w" to "ww". I'd have
On 07:01 AM 6/24/2004, Rich Adamson wrote:
>Now I better understand what you're trying to do.
>
>I'm not a programmer, but I'm fairly certain that you can't dynamically
>change codec preference within asterisk on a per call basis. However,
>just as soon as this gets posted, someone will likely jump
When setting up the que's do you have to add the que to the context?
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yes that worked, also if I want that to be sent to an agi
> exten => 1234,2,agi,myagi.agi|${testvar}
thanks
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Philipp von
> Klitzing
> Sent: Wednesday, June 23, 2004 6:04 PM
> To: [EMAIL PROTECTED]
> Subje
On Thursday 24 June 2004 11:01, Rich Adamson wrote:
> Per the doc in the configs samples, you have to implement
> echotraining=800 (instead of "yes")
> to take advantage of the new code from yesterday.
800ms -- that's quite a delay between pickup and being able to speak; was that
the default whe
On 2004.06.23 12:19 Lee Howard wrote:
On 2004.06.18 23:15 Seth Mattinen wrote:
I've been trying to get fax reception to work using an SPA-2000 to
ring the fax machine or modem that's taking fax calls. I was curious
if anyone else had tried something similar, and if so, had any luck
getting it to
Hi,
> -Original Message-
> Thanks! well after doing some other .cfg file changes I
> hardlocked the phone durring startup! Any ideas? (pushing
> 1,4,7 on powerup isn't helping)
Ouch! Can you check if it is still fetching any config files from your
FTP-server at boot ? Might be your conf
Hi,
> -Original Message-
> Is there any software based solution to establish a video
> connection with * and sip protocol?
MSN messenger 4.7 with any windows capturing device should work. Make sure
you force the codecs properly, because MSN tries to negotiate some form of
MJPEG which Ast
---snip---
> I don't think that I'd play around with other values though. There were
> three lines of code changed in chan_zap.c and two of those lines were
> dependent on values from each other. Example: changing from 400 (default)
> to 800 required another statement change from "w" to "ww". I'd
I am finding that I have to increase the txgain in zapata.conf to 8 when my
X101P is connected to my BT phone line, otherwise people can hardly hear me.
This then gives echo issues.
Do other people have the same problem on BT lines. I was wondering if I
should give BT a call and get them to increa
I recently got a Pulver Innovations WiSIP wireless SIP phone to
determine if we want to use them in our organization. Since the WiSIP
phone arrived last week, I have had nothing but headaches. I do think
I now have the problem narrowed down.
I have spent a bulk of my time trying to get the WiSIP
Thanks Eric. That works.
Kannaiyan
- Original Message -
From: "Eric Wieling" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Thursday, June 24, 2004 3:18 PM
Subject: Re: [Asterisk-Users] Delay in Zap Calls?
> On Thu, 2004-06-24 at 03:35, Kannaiyan Natesan wrote:
> > I have this line i
> Use two separate entries with type=peer and type=user instead of one
> entry with type=friend.
Tried that as well. This triggers yet another misbehaviour...
I tried to define 2 peers (for the outgoing calls), one called [gateway-g729] and one
called [gateway-ulaw], each allowing only the codec
On 25/06/2004, at 12:57 AM, Andrew Yager wrote:
I'll see what setting echotraining=800 does for me...
It still sounds good. There was no noticeable echo on the three calls
we tried. Difficult to say whether it is a greater improvement or not,
but I'm sure I'll have a feel for it after tomorrow.
Looks like PostgreSQL is running in UNIX local socket mode (which is
default) and does not allow incoming TCP/IP connections even for
localhost. Did you check for "tcpip_socket = true" line in your
postgresql.conf file (it is /var/lib/pgsql/data/ directory on my
system)? You can also check perm
> On Thursday 24 June 2004 09:01, Rich Adamson wrote:
>
> > Per the doc in the configs samples, you have to implement
> > echotraining=800 (instead of "yes")
> > to take advantage of the new code from yesterday.
>
> Just to be clear, to take advantage of the enhancement, do we need to pull the
> > > I am curious did you play with the echotraining flag
> > > echotraining=yes or use the delay values for echotraining=some ms ?
> >
> > Per the doc in the configs samples, you have to implement
> > echotraining=800 (instead of "yes") to take advantage of the new code
> from >yesterday.
> I'd
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