Yes, you can:
asterisk -vvvgcn|tee /tmp/log
regards
Martin
On Thu, 27 Feb 2003, Roy Sigurd Karlsbakk wrote:
> hi
>
> can I log all console output while having console access as with
>
> asterisk -vvvgc
>
> ?
> --
> Roy Sigurd Karlsbakk, Datavaktmester
> ProntoTV AS - http://www.pronto.tv/
> Tel:
asterisk -r -x 'put the command here'
regards
Martin
On Thu, 27 Feb 2003, Rattana BIV wrote:
> Hi,
>
> Does anyone know how to execute an asterisk command in shell ?
>
> I wanted to make a script who put extension in asterisk.
>
> Regards
>
> Rattana
>
> _
asterisk -r -x "this should work"
On Thu, 27 Feb 2003, Rattana BIV wrote:
> it doesn't work with me =(
>
>
> - Message d'origine -
> De : "Martin Pycko" <[EMAIL PROTECTED]>
> À : <[EMAIL PROTECTED]>
> Envoyé : jeudi 27 fév
You can also do that using Background application:
[transfer]
exten => s,1,Background,some-file ;it can be silence
exten => _XXX,1,Flash ;collecting the digits
exten => _XXX,2,SendDTMF,${EXTEN}
exten => _XXX,3,Hangup
[called_context]
exten => 1000,1,Goto,tr
gt; Put It just connect to asterisk and provide CLI command line. But the help
> doesn't print.
>
> what do you think ?
>
>
> Regards
> Rattana
> - Message d'origine -
> De : "Martin Pycko" <[EMAIL PROTECTED]>
> À : <[EMAIL PROTE
If you want to have a system capable of making one call
at a time X100P is a good start.
regards
Martin
On Fri, 28 Feb 2003, Tyrone Mills wrote:
> Hello All,
>
> I'm new to Asterisk/IVR and this list. I know this is the 'users' list, but
> the 'dev' list seems dead and this list seems to have a
The only problem with outgoing call mechanism when used
with X100P is that if you're not in the US then
'callprogress' might not work for you and X100P
right after you dial the number will assume that
the remote side picked up the phone. With analog lines
it's dificult to know when the call is bein
Use modprobe instead of insmod. If you use insmod
then you have to first insmod zaptel.
regards
Martin
On Fri, 28 Feb 2003, Victor Sanchez wrote:
> i have error in install module of tor2.
>
> but it compile good.
>
>
> what happen ?
>
> ivr2:/usr/src/zaptel # make clean; make install
> rm -f to
ines, but if this is the case,
> then perhaps digital is the way to go.
>
> Thanks,
>
> Tyrone
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Martin Pycko
> Sent: February 28, 2003 9:39 AM
> To: [EMAIL PROTECTED]
> S
Well you need:
/usr/sbin/asterisk
/usr/lib/asterisk/modules (but not all of them)
/etc/asterisk/ (but not all of them)
/etc/zaptel.conf (if you use zaptel devies)
/lib/modules/linux_version/misc/zaptel.o (if aplicable)
/lib/modules-linux_version/misc/zapte_driver_that_you_need.o (if
aplicable)
lib
Sure, you just add to [general] section
context=default_context_for_everybody_that_is_not_authenticated
SIP-PSTN is already there.
regards
Martin
On Sat, 1 Mar 2003, Bill Jordan wrote:
> I wanted to know if there was any way to setup an asterisk server as a
> PSTN gateway?
>
> That is, I wanted
Make sure that you have in zapata.conf
musiconhold=random
regards
Martin
On Sun, 2 Mar 2003, Lenny Tropiano / asterisk.org Mailing list wrote:
>
>
> I have the following in extensions.conf:
>
> [global]
> MP3ROOT=/var/lib/asterisk/mohmp3
>
> [default]
> exten => ,1,Answer ;
DO you run a recent CVS ?
regards
Martin
On Sun, 2 Mar 2003, Art O'Dea wrote:
> I have zaptel and zapata compiled and installed for the X100P in my
> system (the * box has the x100p, and the ATA-186 and * are on a class-c
> network, all behind a firewall. Calling in to voicemail works fine, but
Yes, asterisk does SIP to IAX/H323/PSTN/MGCP etc
and any other combination.
regards
Martin
On Sun, 2 Mar 2003, Thomas Jalsovsky wrote:
>
> Hello,
>
> Does asterisk do transcoding when the call goes
> through the system, codecs are the same but signaling protocol is changed.
> example:
> SI
let's say you have one T1 span configured like this
in zapata.conf
context=incoming
group = 1
channel => 1-23
then in extensions.conf
[incoming]
exten => fax,1,Dial,Zap/25 #FXS port that fax is plugged to
exten => _X,1,... (the rest)
when asterisk detects fax tones on incoming call
> I need a system that can:
>
> - Accept phone calls and give a greeting
That's basically playing back a voice file
>
> - Allow users to select to either leave a message for callback or browse
> FAQs
IVR menu
>
> - Present a multi-level voice menu a user can use to drill down to find
> their answ
> how to match callerid from 001... ?
> and if don't know how many numbers ?
You can do it the same way you match extensions:
exten => s/_001.,1,blabla
exten => s/_00[2-90].,1,bleble
regards
Martin
>
> exten => s/0_,Answer don't work-
> anything else ?
>
> tnx
> Thomas
>
>
>
>
>
But you can connect several asterisk boxes as one system.
regards
Martin
On Tue, 4 Mar 2003, Sphyrna wrote:
> NO, THE ASTERISK HAS A PRATICAL LIMIT OF 8 E1S CURRENTLY. THE I/O ERRORS
> STOP EVERYTHING
> - Original Message -
> From: "Florian Overkamp" <[EMAIL PROTECTED]>
> To: <[EMAIL PRO
pbx/pbx_spool.c
On Wed, 5 Mar 2003, Rattana BIV wrote:
> Hi,
>
> I wanted to know in which code source the file sample.call is proceeded when
> we put it in /var/spool/asterisk/outgoing/
>
> I try to make an application to asterisk who check when an user in H323
> (netmeeting) is connect or not.
qualify=1000 in sip.conf in the phone config entry
regards
Martin
On Wed, 5 Mar 2003, Mark Spencer wrote:
> > But if I close my sip phone and a call goes through it will still wait
> > the 25 seconds before it goes to voice mail even though my Sip phone is
> > not even on. If I restart Asterisk
Can you use MSGSM codec with MSN Windows Messenger 4.x ?
regards
Martin
On Thu, 6 Mar 2003, Wade Weppler wrote:
> I'll try to be a little more specific:
>
> http://www.microsoft.com/exchange/downloads/2000/IMClient47.asp
>
> This one works for me.
>
> -wade
>
> > -Original Message-
> > F
Do you have tos=lowdelay in iax.conf ?
You may also try to turn off the jitterbuffer (jitterbuffer=no).
Also make sure that asterisk is really using gsm codec.
WHen you do "iax show channels" in the format column it should
show number '2' = GSM.
Also when you look at Makefiles make sure that optim
out of asterisk is far away from the limits I would let to
go out to customers:-( )
M
On 8 Mar 2003, William X Walsh wrote:
>
> He's using H.323 not iax
>
> On Sat, 2003-03-08 at 09:58, Martin Pycko wrote:
> > Do you have tos=lowdelay in iax.conf ?
> > You may also try
asterisk -vvvcg
when the segfault happens
gdb ./asterisk core.[pid]
regards
Martin
On Sat, 8 Mar 2003, Ben Clark wrote:
> I am also getting a seg fault when asterisk tries to load
> chan_oh323.so. What should I try to get it to work?
>
> [chan_oh323.so] => (OpenH323 Channel Driver)
>== Par
how about "sip debug" ?
regards
Martin
On Sat, 8 Mar 2003, T Aksoy wrote:
> Hi,
>
> On the release of asterisk I was using before this one, I used to issue a "set
> verbose 100" command and I would see all the sip registrations taking place. Now
> that doesn't seem to work.
>
> Could someone c
It's because you're not using lookupblacklist
application on your callflow.
regards
Martin
On Sun, 9 Mar 2003, Jim Archer wrote:
> What does blacklisting a call with *80 do? I tried it by dialing from my
> cell, which presents caller id. I then blacklisted and the console debug
> reported that
you put some definitions in
[globals] section in extensions.conf
later you just use SetGlobalVar variable to change the
values of global variables and then you just take
the value of a variable like this: ${variable} or
like this ${${variable}} or like this ${extension_${EXTEN}} etc.
a trivial ex
put transfer=yes in the begining of zapata.conf
after [channels]
regards
Martin
On Mon, 10 Mar 2003, Mike Reiling wrote:
> Did that... Doesn't seem to help
>
>
> On Monday, March 10, 2003, at 09:49 AM, James Sharp wrote:
>
> >
> >> parkext => #700; What ext. to
Do you have a proper zaptel.conf and zapata.conf ?
When you modprobe do you have anny errors ?
What does "ztcfg -vv" says ?
regards
Martin
On Mon, 10 Mar 2003, Brian J. Schrock wrote:
> Howdy,
>
> I just added a second USB converter from Digium and I am having a
> problem. When I modprobe the dr
We have some feedback from our customers that sometimes
they are able to run two S100U's on a signle machine.
regards
Martin
On Mon, 10 Mar 2003, Ray Dzek wrote:
> I was told specifically by Digium that only one USB FXS device was supported
> per system.
>
>
> - Original Message -
> From
I think UHCI
Martin
On Mon, 10 Mar 2003, Brian J. Schrock wrote:
> UHCI or OHCI?
>
> On Monday, March 10, 2003, at 04:23 PM, Martin Pycko wrote:
>
> > We have some feedback from our customers that sometimes
> > they are able to run two S100U's on a signle machin
>From now on (taking about asterisk's CVS) you can access environmental
variables using ${ENV(VARENV)}
regards
Martin
On Tue, 11 Mar 2003, Rattana BIV wrote:
> I try to detect if an user who use Netmeeting is connected or not.
> I think in order to do that, Netmeeting-user open a web page (in PH
It's received in a SIP header.
regards
Martin
On Tue, 11 Mar 2003, T Aksoy wrote:
> Hi Mark,
>
> Not familiar with "received=". What does it do? Has it got any application
> within the nat domain?
>
> Thanks
> Tan
>
>
> - Original Message -
> From: "Mark Spencer" <[EMAIL PROTECTED]>
> To
It's dynamically changed to "Yes" when the fax gets detected on this
channel.
regards
Martin
On Wed, 12 Mar 2003, Darrell Eldridge wrote:
> I still haven't been able to get fax detection going,
> but I came across something: when I execute "zap show
> channel 47" one of the parameters shown is
You may try to add
relaxdtmf=yes
just before channel => 4 in zapata.conf
regards
Martin
On Wed, 12 Mar 2003, Brian J. Schrock wrote:
> I am using background, the pbx-invalid stuff should (if DTMF
> recognition is working correctly) not get played.
>
> On Wednesday, March 12, 2003, at 01:30 PM,
IAX is short and I like it. Besides if that additional '2' irritates you
then anyways in the near future when IAX2 is working fine ppl will switch
eventually to IAX2 and then we'll refer to IAX2 as IAX
Martin
On Thu, 13 Mar 2003, Mark Spencer wrote:
> What do you all think of renaming IAX2
You can playback a second or two of silence ...
regards
Martin
On Thu, 13 Mar 2003, T Aksoy wrote:
> Hi,
>
> We are testing a number of sip phones from different manufacturers. With one phone
> in particular, when I dial the asterisk voicemail, it misses around half a second
> from the beginni
[deeper]
exten => s,1,Playback,you-re-in-the-deepest-menu
exten => s,2,Goto,options|s|1
[options]
exten => s,1,Background,prompt-1-deeper-2-back
exten => 1,1,Goto,deeper|s|1
exten => 2,1,Goto,sales|s|1
[sales]
exten =>
s,1,Background,prompt-1-information-2-connect-or-stay-on-the-line-0-operator-*-
or Packet Telephony (Simple) Protocol
On 13 Mar 2003, Karl Putland wrote:
> What about ITP
>
> Internet/IP
> Telephony
> Protocol
>
> On Thu, 2003-03-13 at 09:40, Mark Spencer wrote:
> > > LIghtweight
> > > Voice over IP
> > > Exchange
> >
> > Or:
> >
> > Lightweight
> > Internet
> > Voice
> > E
Of courese:
exten => 9998,1,Dial,SIP/9998|30|tTm
Notice when you don't use the timeout you do have to use the options
separator "|" like this:
exten => 9998,1,Dial,SIP/9998||tTm
but I think that T is not yet implemented
regards
Martin
On Fri, 14 Mar 2003, WipeOut . wrote:
> Thanks the 'show app
The formats that asterisk uses are #define'd in
asterisk/include/asterisk/frame.h
RTP formats are #define'd in asterisk/rtp.c
regards
Martin
On Fri, 14 Mar 2003, John Vozza wrote:
> I've been trying to find a list of codec "format numbers" so I can more
> clearly understand the following messag
Note that the message comes from chan_iax2.c that is under developement.
It uses iax.conf as well as chan_iax.c
regards
Martin
On Sat, 15 Mar 2003, John Vozza wrote:
> Thanks to all who set me straight on the codec format stuff...
>
> I have a "remote" asterisk system running on my laptop which
You have to add
immediate=yes
to zapata.conf to the declaration of this channel.
Then right after someone picks up the phone asterisk will just
right to 's' extension of the specified context.
regards
Martin
On Tue, 18 Mar 2003, Don Pobanz wrote:
> We have need of a ringdown circuit in an elevat
It's fixed now.
> what's the intended behaviour of ${variable:a:b}?
it's the same as substring application
> given that ${exten} = 501234
> until yesterday ${exten:2} would give '1234'
and it does now
>
> with current CVS ${exten:2} is '50' while ${exten:2:4} is '1234'
>
> how do I just strip ch
Sure. You configure it as HDLC or CISCO HDLC and you have
hdlc0 interface to send data.
regards
Martin
On Thu, 20 Mar 2003, David Luyens wrote:
> Hi,
>
> I would like to use * as a compression box.
> Between 2 sites I have an E1 leased line.
>
> So would it be possible to use 1 port of an E400P
HDLC/CISCO HDLC.
regards
Martin
On Thu, 20 Mar 2003, David Luyens wrote:
> Thanks Martin, could you point me into the direction on how to do this?
>
> David
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Martin Pycko
> Sent
> Greetings Asterisk users.
>
> When I launch Asterisk, I get the following
>
> Asterisk CVS-03/20/03-16:56:24, Copyright (C) 1999-2001 Linux Support
> Services, Inc.
> Written by Mark Spencer <[EMAIL PROTECTED]>
> =
> [ Bootin
You could use a global variable like this:
[globals]
max_channels_1=10
max_channels_2=13
appl_1=0
[incoming1]
exten => _.,1,GotoIf,$[${appl_1} > ${max_channels_1}]?hangup|1:2
exten => _.,2,SetGlobalVar,appl_1=$[${appl_1} + 1]
exten => _.,3,do_what_you_want_to_do
exten => h,1,SetGlobalVar,appl_1=
I think it's all about SNOM 100 phone with IAX image.
Martin
On Mon, 24 Mar 2003, WipeOut . wrote:
> Its still there..
>
> http://www.asterisk.org/main/index.html
>
> Click "Hardware" on the left menu..
>
> - Original Message -
> From: Steve Kann <[EMAIL PROTECTED]>
> Date: 24 Mar 2003
It's there
On Mon, 24 Mar 2003, Darrell Eldridge wrote:
> Is the macro functionality (described in the draft
> handbook Version 2, Section 4.3.11 Using Macros)
> already available? I'm having trouble making it work
> and wonder if I need to keep trying or wait until it's
> in the code.
>
>
how about libpri ?
On Tue, 25 Mar 2003, Lenny Tropiano / asterisk.org Mailing list wrote:
> gcc -shared -Xlinker -x -o chan_phone.so chan_phone.o
> gcc -c -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations
> -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -ma
You can try to do that using authenticate application
regards
Martin
On Thu, 27 Mar 2003, James Golovich wrote:
>
>
>
> On 27 Mar 2003, Matthew Farley wrote:
>
> > My asterisk system is now working wonderfully (thanks to all of you for
> > your invaluable contribution to the software world and y
You have to call record like this
exten => 8000,1,Record,intro:gsm
(read show application record)
Martin
On Thu, 27 Mar 2003, Michael K. Rodriguez wrote:
> This the error I receive when I try to record a GSM file
>
>
>
>
>
> -- Executing Record("SIP/67.98.37.220:5060", "intro|gsm") in new
Do also "frame 0"
Martin
On Thu, 27 Mar 2003, Eric Wieling wrote:
> I'm getting occasional seg faults when a call ends. Here is the
> backtrace. Calls are going to a SIP device.
>
> (gdb) bt
> #0 0x08057bae in ast_queue_frame (chan=0x8125240, fin=0xbf5fea0c, lock=1)
> at channel.c:344
> #
You can use "debug channel " on asterisk console
to find out which channel sends disconnect/hangup
You have to trace all the channels on both of your systems.
regards
Martin
On Thu, 27 Mar 2003, diana wrote:
> Hello,
>
> I have a very complicated system which contains 2 *'s.
>
> PSTN --- CI
> Will to ports on this card be able to act as FXO as well, or just as FXS?
Maybe later.
But there was some posting about "FXS to FXO converter" a few weeks before
???
> If the answer is yes, can we control which ports do which in any
> combination?
Why not ?
> Finally, can this card coexist
The same as you go over the number of PRI channels ?
regards
Martin
On Thu, 27 Mar 2003, James O. Sizemore III wrote:
> Quick question what happens if you go over
> your channel licenses?
>
> Mark Spencer wrote:
>
> >>So it looks like the best codec is the GSM codec as far and badwidth
> >>vs vo
You must be using some old code. Try to use code from CVS. Instructions
are on www.digium.com
regards
Martin
On Fri, 28 Mar 2003, [ISO-8859-7] ÓôáìÜôçò ÊåêÝò wrote:
> Hello everybody.
>
> I have a test box with asterisk and till now I have successfuly made it
> work with iax.
> I'm trying to loa
I'm afraid you have to do "debug channel" also on [EMAIL PROTECTED]/3
(since that info doesn't tell me much)
The same on the other box.
regards
Martin
On Fri, 28 Mar 2003, diana wrote:
> << [ TYPE: Null Frame (4) SUBCLASS: N/A (5) ] [Zap/1-1]
> -- Hungup 'Zap/1-1'
> == Spawn extension (flu
> I would like to known if these "T2" links are related to the E1
> stuff that everybody talk about on this forum. In other words, can I
If you're in Europe than your T2 are 99.9% E1's (30 voice channels + 1
signalling)
> link the free "T2" card of the Bosch to a Linux box with an E100P
> interf
You shouldn't have problems in a recent CVS with that.
regards
Martin
On Fri, 28 Mar 2003, Brian Capouch wrote:
> Eric Wieling wrote:> Try commeting out the username= and secret= and set the
> > host=dynamic. If that works you can try adding them back in.
> >
>
> That did work!!
>
> But (sigh.
What signalling are you going to use ?
regards
Martin
On Mon, 31 Mar 2003, Eduardo Goncalves wrote:
> Hi,
>
> I'm abaut to install asterisk and I want to know if buying an E400P (Quad Span E-1
> Interface) from digium my linux box will be ready (of course, after configure it) to
> work with PS
Read www.digium.com at Documentation->FAQ
Martin
On Mon, 31 Mar 2003, Allan Wang wrote:
> Steven,
>
>
> Could you please give me your config files for X100P
> and S100U? I just got mine, but I have difficulty to
> make it work.
> Thank you.
>
>
>
>
> Allan
>
>
>
> --- Steven Critchfield <[EMAI
cvs update your libpri
regards
Martin
On Mon, 31 Mar 2003, Alex Zarubin wrote:
> Hi,
>
> There are several channels on the PRI span with the periodic warning:
>
> WARNING[9226]: File chan_zap.c, Line 5437 (pri_dchannel): Ring requested on
> channel 21 already in use on span 1. Hanging up owner.
ith Qsig. Also with R2, but here in Brasil I prefer the first.
>
> regards
> Eduardo
>
>
> On Mon, 31 Mar 2003 15:17:45 -0600 (CST)
> Martin Pycko <[EMAIL PROTECTED]> wrote:
>
> > What signalling are you going to use ?
> >
> > regards
> >
then download the sources and compile it ...
On Wed, 2 Apr 2003, it wrote:
> I installed the cygwin yesterday. But it seems that the cygwin does not have
> the cvs command.
>
> $ cvs
> bash: cvs: command not found
>
>
> Regards
> john
>
>
> - Original Message -
> From: "Michael Bielicki"
Yes, we can do if's (GotoIf), +,_,*,/,%, etc ... like this
exten => t,2,SetVar,looptest=$[${looptest} + 1]
regards
Martin
On Tue, 1 Apr 2003, John Harragin wrote:
> Hey do we have the ability to incriment a variable?
> exten => t,2,SetVar,looptest=$((looptest + 1))
> I was thinking of doing a li
asterisk -vvvcg (use g option to generate the coredump file)
than gdb asterisk core.pid
bt
Also you might send a log of "pri intense debug span "
regards
Martin
On Wed, 2 Apr 2003, Alex Zarubin wrote:
> Configuration:
> Linux wpbx 2.4.9-13 #1 Tue Oct 30 20:11:04 EST 2001 i686 unknown
> P4 2.5 G
cvs update -r 1.x channels/chan_sip.c
make install
where 'x' is from 1 to 30
version 1.30 is dated 2003-04-02
if not sure check "rcs2log -v |more"
regards
Martin
On Tue, 1 Apr 2003, Russ Beaupre, P.E. wrote:
> A while ago SIP transfer via the # key on a call to a cell phone via
> iconnect was
You configure them as usual
zaptel.conf
fxoks=1-n #(n - how many cards you have)
Then you can just plug a single phone line to each of them
and then in zttool which one will go into OK from RED state.
regards
Martin
On Thu, 3 Apr 2003, Jim Archer wrote:
> Hi All...
>
> If I have more than 1 X1
Can you use Playback instead ?
Playback doesn't use mpg123.
regards
Martin
On Thu, 3 Apr 2003, Tamas Levente wrote:
> Hey,
> I've installed 0.59r mpg123 on a redhat box. I set the extension up for the
> mp3player. I called and it was playing the file back,but it was full of drops. like
> sound
I woudln't write that if it wouldn't support mp3.
On Thu, 3 Apr 2003, Tamas Levente wrote:
> And does playback support mp3?
> - Original Message -
> From: "Martin Pycko" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Thursday, April
codecs/codec_mp3_d.c -> xing ...
On Thu, 3 Apr 2003, Michael Bielicki wrote:
> what does playback use ?
>
> On Thursday 03 Apr 2003 16:18, Martin Pycko shaped the electrons to say:
> > Can you use Playback instead ?
> > Playback doesn't use mpg123.
> >
>
Some people run fax over IAX using ulaw codec on the local LAN.
Martin
On Thu, 3 Apr 2003, Brian J. Schrock wrote:
> From what I have heard packetizing fax does not work well, does not
> matter if it is IAX or SIP. I think that was straight from digium tech
> support.
>
> On Wednesday, April 2,
You could configure the channels of port 1 as
clear=1-24
in zaptel.conf
This way you'll have one big pipe on /dev/zap/1 accessible for you.
Unfortunatelly you cannot do clear=1-96 and have it all 4 spans
on /dev/zap/1.
regards
Martin
On Thu, 3 Apr 2003, Chris Albertson wrote:
>
> Is there a blo
Do you really have the channels in asterisk ?
"zap show channels"
Is the alarm on the E1 circuit ?
Martin
On Thu, 5 Jun 2003, Jay Banda wrote:
> Hello All.
>
> Does anyone have experience with the Valiant Comms vcl30 channel
> and the Digium E100P in asterisk ? We have the vcl30 channel bank,
>
It does use sendmail. Which app are you using ?
voicemail or voicemail2 ?
Martin
On Thu, 5 Jun 2003, Derek Beaumont wrote:
> I used to have email notification working with my voicemail services but
> it stopped working when I installed the new version of asterisk.
>
> I have not changed my voic
Yes, it is. Sip supports callpickup's with *8 same as zaptel hardware.
You just need to configure it in sip.conf.
regards
Martin
On Thu, 5 Jun 2003, John Todd wrote:
>
> I haven't tried this SIP features, but in the latest sip.conf.sample,
> this is included:
>
> ;[pingtel]
> ;type=friend
> ;use
It should be exten=>sip,1,Dial(SIP/sipphone)||t
Martin
On 6 Jun 2003, Dave Wolven wrote:
> Hi Don't know if someone answered this yet...
>
> when calling the dialapp append the |t to it
>
> exten=>sip,1,Dial(SIP/sipphone)|t
>
> This will allow you to hit # and then the callparking extension.
>
>
You need to have
disallow=all
allow=g723.1
and the other remote phone has to use also the G723 codec. Otherwise
asterisk will try to transcode but it doesn't have the G.723 code itself.
regards
Martin
On Fri, 6 Jun 2003, Dave Alan Caruana wrote:
> I added the line "allow G723.1" in my sip.conf
What network card are you using ? (model and vendor)
Martin
On Tue, 27 May 2003, Nick Eggleston wrote:
> We are using the zaptel driver to deliver a combined voice/data T1 circuit.
>
> The data channel-group is using the "cisco" hdlc protocol (on the linux side)
> and connects with a cisco route
1) you need two accounts in iconnecthere
2) you need to register with two accounts
3) then simply receive the call using one and send it over another account
Martin
On Wed, 28 May 2003, pradeep kumar wrote:
> Hi All.
>
> I am trying to setup asterisk so that I can place two outbound calls via
>
If you don't have any hardware for conferencing than you could use the
ztdummy from zaptel package. Check the archives. look for ztdummy
Martin
On Tue, 27 May 2003, Rahul Gupta wrote:
> Hello ,
>I am a newbie to * and have just been able to call
> a sip User Agent on a different machine thru
I didn't know that one can run Ethernet over T100P.
Now what NIC card are you using ?
Martin
On Wed, 28 May 2003, Nick Eggleston wrote:
> Digium T100P
>
> On Wed, 28 May 2003, Martin Pycko wrote:
>
> > What network card are you using ? (model and vendor)
> >
>
It should detect the incoming call on a 2nd or 3rd ring.
I don't know about the limit though.
regards
Martin
On Wed, 28 May 2003, WipeOut . wrote:
> Hi Mark,
>
> I have tried a value of 10 and still had the problem.. How high can this value be
> set and what would the effect be of setting it to
FXO ports don't get DID numbers usually so they'll always go to 's'
Martin
On Wed, 28 May 2003, Jon Pounder wrote:
> When immediate is set on a port that is an fxo, what is the meaning of this ?
>
> Will it go immediately to the "s" extension of the context when the line
> first rings, or someth
Check if you can enable "remote disconnection supervision" with your PBX.
If not you may try using the software detection when you specify
callprogress=yes
or
busydetect=yes
before the definition of your channel in zapata.conf
Martin
On Wed, 28 May 2003, Manuel Marin Garcia wrote:
> Aste
Nope.
usecallerid=no should work for it.
If not you might try to modify the code in chan_zap.c
Martin
On Wed, 28 May 2003, Jon Pounder wrote:
>
> my question was -> will immediate put an end to the extra 2 rings before
> pickup ?
> (I know they go to "s" eventually.)
>
> At 10:40 AM 5/28/2003 -
Make sure that you don't have a R2 signalling. Since then you'll have
problems EuroISDN PRI is all right.
Martin
On Wed, 28 May 2003, Ricardo Saar Gemignani wrote:
> Hello
>
>I'm starting to learn about Asterisk and trying to install the first one.
> I've a doubt. Here in Brazil the Tel
Try to add this line to zapata.conf
relaxdtmf=yes
before the "channel => " definition
Martin
On Wed, 28 May 2003, Nick Eggleston wrote:
> We've got an asterisk system hooked up to a number of telephones via a channel
> bank.
>
> [*]T100P---CAC(access bank)---Phones...
>
> What we are seeing is
You _CAN_ use a wildcard on the callerid matching. It goes through the
same code.
regards
Martin
On Thu, 29 May 2003, Jamie Carl wrote:
> I was just thinking that. Shouldn't this be a feature?
> I'm sure coding it would be a cut and past job. :)
>
> Another one for the TO-DO list Mark.
> :)
>
Try running asterisk like this:
screen -d -m asterisk -vvvc
or
screen -d -m asterisk -c
or
screen -d -m asterisk -f
Martin
On Thu, 29 May 2003, Tjardick van der Kraan wrote:
> When we have the G.729 codec (ordered from digium) active in * we have the
> following problem:
>
> running * in standa
Do you have your zap channel in asterisk when you type "zap show channels"
?
If not than make sure you have a proper config files (zaptel.conf &
zapata.conf)
Martin
On Thu, 29 May 2003, Eduardo Goncalves wrote:
> Hi list,
>
> I have the follow configuration:
> ===
> extension.conf:
Lets say that your E1 channels are assinged to
context=incoming
channel => 1-15,17-31
Then in extensions.conf in context
[incoming]
exten => fax,1,Dial,Zap/1 ;if your Zap/1 port is FXS where the fax
;is attached
(all other extensions)
regards
Martin
On Thu,
Then propably your board stoped taking interrupts. Try changing the PCI
slot or IRQ. Make sure you don't run X-windows.
regards
Martin
On Thu, 29 May 2003, Eduardo Goncalves wrote:
> On Thu, 29 May 2003 11:41:01 -0500 (CDT)
> Martin Pycko <[EMAIL PROTECTED]> wrote:
>
>
What bandwidth do you have available for you connection (upsteram and
downstream)? Do you have any CIR for VSAT connection ?
Martin
On Thu, 29 May 2003, Jim Ockers wrote:
> Hi all,
>
> For some reason VSAT or Satellite Internet services are not mentioned
> (or searchable) in this list's archives
01 -0500 (CDT)
> Martin Pycko <[EMAIL PROTECTED]> wrote:
>
> > Then propably your board stoped taking interrupts. Try changing the PCI
> > slot or IRQ. Make sure you don't run X-windows.
>
>
> My box has only one slot. I changed
4:32:37 -0500 (CDT)
> Martin Pycko <[EMAIL PROTECTED]> wrote:
>
> > Check whether "strace -xx cat /dev/zap/1" gives you any output
> > If it stops and waits than your board is not taking interrupts.
> > Is the board running on the separate IRQ ?(/proc/inter
Didn't you just write a post before that it was running ?
The EBUSY means that you propably have asterisk running and the port is
busy or you have strace line on some other console
Martin
On Thu, 29 May 2003, Eduardo Goncalves wrote:
> On Thu, 29 May 2003 14:32:37 -0500 (CDT)
> M
So it means that the board is working all right but there is problem with
the telco or you're using diffrent signalling for your circuit.
Martin
On Thu, 29 May 2003, Eduardo Goncalves wrote:
> On Thu, 29 May 2003 15:06:12 -0500 (CDT)
> Martin Pycko <[EMAIL PROTECTED]> wrote:
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