or at max in weeks).
Using the new firmware is there still the issue with needing to patch
chan_sip.c, or does it work out of the box? Do you have details on how
it should be implemented within *?
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camper, once intercom is working
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regards
Peer Oliver Schmidt
th einternet
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important, as I
have another interested party to be deployed during the June/July time
frame, which needs intercom functionality.
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Adi Linden wrote:
I am looking for a German language softphone. Is there such a thing?
DIAX has german language support.
rgds
pos
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To
completing the call successfully.
Any and all help is greatly appreciated.
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Hi,
does the hint extension work together with the Snom phones in stable? I
don't get an error in the dialplan, but it does not work either.
On SIP/26 I want to monitor SIP/22. This is what I do right now:
extension.conf
[incoming]
exten = 955,hint,SIP/22
exten = 955,1,Dial(SIP/22)
sip.conf
[26]
Bob Goddard wrote:
On Tuesday 21 December 2004 20:03, Peer Oliver Schmidt wrote:
does the hint extension work together with the Snom phones in stable? I
don't get an error in the dialplan, but it does not work either.
On SIP/26 I want to monitor SIP/22. This is what I do right now:
extension.conf
David Ishmael wrote:
Im not sure if this is possible, but I was hoping to find an address
book that runs on Windows XP that will allow me to select a phone number
and send that to my Asterisk. The Asterisk system would make the call
and connect the call to a SIP phone (Grandstream Budge
Jon Lawrence wrote:
I can receive incoming calls. However, I can't call out.
When ever i initiate an outgoing call, I get the following on the console:
Executing Dial(SIP/2014-8817, CAPI/*msn|bdialednumber) in new stack
Dec 9 23:10:24 WARNING[1390]: chan_capi.c:653 capi_call: Destination *msn*
Jon Lawrence schrieb:
On Friday 10 December 2004 09:50, Peer Oliver Schmidt wrote:
Jon Lawrence wrote:
I can receive incoming calls. However, I can't call out.
When ever i initiate an outgoing call, I get the following on the
console:
Executing Dial(SIP/2014-8817, CAPI/*msn|bdialednumber) in new
Jon Lawrence schrieb:
On Friday 10 December 2004 10:41, Peer Oliver Schmidt wrote:
My msn is 1234, the called number is 0123-45678. This is my log entry
Executing Dial(SIP/26-dd65, CAPI/1234:b012345678|60|T) in new stack
In extenstions.conf I have
exten = _.,1,dial(CAPI/1234,b${EXTEN},60,T
Marco Parmeggiani schrieb:
I'm trying to use an hfc based pci card with asterisk but every call fails
falling in the congestion extension.
exten = _0.,1,Dial(${TRUNK}:${EXTEN:${TRUNKMSD}}||tr)
exten = _0.,2,Congestion
Looking in the syslog i can see:
isdn: HiSax,ch0 cause: E001B
isdn card: HFC
Maros RAJNOCH wrote:
exten = h,1,system(/var/lib/asterisk/bin/mailfax ${FAXFILE} ${EMAILADDR}
${CALLERIDNUM} ${CALLERIDNAME})
mailfax binary will be executed after any hang-up, also after calls, not
only faxes. I know I can use some variable and if statement to run
mailfax only if that variable
Steve Totaro schrieb:
You might want to look into fli4l (http://www.fli4l.de). It is a
router/whatever plus there is a module add-on with asterisk. Might be
worth a try.
Is there a good site to check this out that is in English?
For fli4l itself, yes. For the opt_modul, no. After reading the
Alan Ingleby wrote:
I also wanted to set up this machine to be our network
firewall/nat Our existing firewall runs linux on a p90, and runs
fine, but I figured it's time to upgrade.. Will this cause any
problems for *?
You might want to look into fli4l (http://www.fli4l.de). It is a
Hi,
I want to run a queue with CallBacklogin which works fine. However, I
want the system to directly connect without the user having to press #
Ideas anyone?!
TIA
rgds
pos
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Alan Ingleby wrote:
You might want to look into fli4l (http://www.fli4l.de). It is a
router/whatever plus there is a module add-on with asterisk. Might be
worth a try.
Erm.. My new PC doesn't have a floppy drive :-)
It works of a hdd/cd (maybe usb) as well.
rgds
pos
Hi,
the app HasNewVoiceMail can't find my voicemail. This is the errormessage:
Dec 3 14:24:01 NOTICE[1481]: app_hasnewvoicemail.c:104
hasvoicemail_exec: Voice mailbox 25 at
/var/spool/asterisk/voicemail/default/25/(null) does not exist
however this is the output of lspbx:~# ls -l
Mike Dent schrieb:
On Fri, 03 Dec 2004 14:20:35 +0100, Peer Oliver Schmidt
the app HasNewVoiceMail can't find my voicemail. This is the
errormessage:
Those file permissions could be wrong?
Mine are liked:-
-rw-r--r-- 1 root root 9339 Nov 17 09:47 busy.gsm
-rw-r--r-- 1 root root 6765 Nov 17 09
[EMAIL PROTECTED] schrieb:
After calling the number and no response of our client the voice-box
gives response. Thats ok... but after the voice-box, which ist self-
configured by our client the server respondes with the notivication to
leave your message please speak after... blablabla
Does
Wengrzik, Andreas schrieb:
hello
isdn4linux is one solution. another is to use an HFC PCI card with
bristuff from http://www.junghanns.net/asterisk/. I'd recommend the
latter
my problem is that i in first step i have to use asterisk only for sending an
error message as wave file
to a
Eric Hall wrote:
When back to the top-level and did a make
I get this
make[1]: *** [app_rxfax.o] Error 1
make[1]: Leaving directory `/usr/src/asterisk/apps'
make: *** [subdirs] Error 1
[EMAIL PROTECTED] asterisk]#
I just fought a battle with spandsp/rxfax and won.
My winning strategy can be
by
capi4hylafax and asterisk, which works fine and dandy. But CAPI is more
expensive than the HFC version.
Any and all information is greatly appreciated.
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Julian wrote:
We are going to have people in our office who do not sit at the same desk
throughout the day (or week), and have Cisco 7940 phones using the SIP
image.
[..]
I really want to find the extension
Isn't this a case for Queues with callback login?
Just a thought
rgds
pos
and any other module that gets complained about.
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It seems to be made by KIRK. Here is a link I found:
http://www.kirktelecom.com/company/suk110.asp
No pricing found so far.
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usb-uhci 23344 0 (unused)
usbcore62924 1 [usb-uhci]
pbx:/usr/src/zaptel# lsmod|grep ppp
ppp_generic20388 0 (unused)
slhc4784 0 [ppp_generic isdn]
Any and all help is greatly appreciated.
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Holger Schurig wrote:
Hi all !
We have 3 NTBAs which are all going to our existing PBX. Our areacode is
06003 and our DDI enabled number 9141.
I want to exchange that PBX with Asterisk, but still struggle to get it
working.
My CAPI.CONF is currently like this:
[general]
nationalprefix=0
Stephen R. Besch wrote:
Any other features you've empirical found out but that?
I note the tones with 1.0.5.0 are all files 64Kb.
the 1.0.5.0 version anyway. It hasn't fixed any of the outstanding
issues (at least those related to use with *, or added any really
useful functionality.
Two
Philipp von Klitzing wrote:
Did you try out the new ring tones? One of them contains a regular ring,
followed by a voice announcing the caller id of the calling party. VERY
neat. It seems the ring tones can contain not only sound, but also
either code to be executed, or a flag to announce the
[EMAIL PROTECTED] wrote:
msn=072,0725
[..]
== found capi with omsn =072
May 28 10:36:56 NOTICE[180241]: app_dial.c:655 dial_exec: Unable to
create channel of type 'CAPI'
== Everyone is busy at this time
Are you sure, that your format for the msn definition is correct for
Italy?
Rob wrote:
I can't make outgoing calls with CAPI (passive ISDN Fritz card). See
Asterisk error below.
Incoming calls and SIP to SIP calls do work. It looks like a msn
mismatch in extensions.conf
and capi.conf, but I can't find it.
I had the same problem. A reboot of the system solved it.
hth
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this?!
canreinvite is set to no
TIA
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Hello Brian,
It might be helpful for us all if the author could let us know more
about the environment in which this application was built. .
I'm getting all kinds of errors when I try to run it, and I suspect that
either my Postgres or PHP installations are incompatible with yours.
I am not
Brian Capouch wrote:
It might be helpful for us all if the author could let us know more
about the environment in which this application was built. .
I'm getting all kinds of errors when I try to run it, and I suspect that
either my Postgres or PHP installations are incompatible with yours.
Wipeout wrote:
Another thing I had to do was changing the defines.php file to reflect
my environment. After that, things went smooth.
On my server the links dont even work in the menu on the left.. Not sure
what is going on with the code and dont have the time to look right
now.. I will just
Adam Goryachev wrote:
Following Zyxel phone is VERY nice and you may attach a headset to it
and walk around to your hearts contend, as look as you are
anywhere near a WIFI AP.
http://www.zyxel.com/product/model.php?indexcate=1075688089indexFlagvalue=1075687935
Sounds interesting... just need to
Greg Boehnlein wrote:
On Tue, 24 Feb 2004, Greg Boehnlein wrote:
Hello all,
I have an application where I am attempting to use Agents and
CallQueues to distribute inbound calls to remote users on cell phones. The
system works quite well, except for one annoying thing that I cannot
figure
Good day,
I am in the middle of getting my self some hard phones. Anyone care to
comment on the *voice* quality of the following phones:
Cisco 7960
Siptone II
SNOM
Budgetone
I have seen a few reviews, but none go to deep into the voice quality
issue.
Thanks.
rgds
pos
Andy Powell wrote:
Snom TAPI integration is a joke...
Would you mind elaborating a bit on this? Is the future implemented,
but does not work, or is it not implemented at all? Or something
else?
The feature isn't really implemented.. you can install the 'driver'
but you only get the ability to
Andy Powell wrote:
Snom TAPI integration is a joke...
Would you mind elaborating a bit on this? Is the future implemented, but
does not work, or is it not implemented at all? Or something else?
Thanks
rgds
pos
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Hi,
anyone here running SNOM phones with TAPI integration with Outlook?
Any other hardware phone with some TAPI integration?
rgds
pos
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To
Johannes von Drachenfels wrote:
Hi,
i'm here in germany still fighting against my problems ...
We have a e100p which is sending out his callerid as 78107-0. But what i
need is to send out the extension of the inside callers to, for example:
78107-14
[..]
But i still can see only the 78107-0 when
Would the following be a doable solution:
1. An Asterisk-box on site with FXS
2. Plug Fax into FXS
3. User uses facsimile machine to call a number - Asterisk answers
4. Stores called number into variable ${FAXDESTINATION}
5. Use RcfFax of * to store fax within asterisk
6. mail stored fax together
Greg,
my Linux iptables firewall, on a private network. Both boxes cann register
iax2 to asterisk, and dial, but as soon as asterisk tries to do the native
a private network -- as in a NATed network? Maybe canreinvite=no or
nat=yes will do the magic you need.
I think he is using the IAX2
Hi Dan,
iax2 to asterisk, and dial, but as soon as asterisk tries to do the
native
BTW: I have the same problem.
I have 2 DIAX phones behind two different NAT firewalls and the * box on one
of the phones network.
It works for me.
Cool. I am sure it has nothing to do with DIAX, but might be the
Peter,
[Full quote deleted]
Suggestion for name SwIAX based on Sokol W (windows) IAX
I would not use that name, as there is a VoIP company called SWYX. You
don't want to risk any problems there, do you.
rgds
pos
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Steven,
- Integrated with the Eutectics IPP200 USB handset
integration with handsets is a great. Do you support onhook/offhook for
the IPP200? Do you plan on supporting other Eutectics phones as well,
like the IPP5xx (with dial support) or the IPP210?
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Sascha Knific wrote:
I never had the time to try out CLIR. Now I did and it doesn´t work for
me as well.
Make sure you have CLIR enabled by your telekom provider (Fallweise
Unterdrückung der Rufnummer). It was not enabled on my MSNs, so @ didn't
work. Now my provider has enabled CLIR and
Is the SIP bin same for IAX as well?
There is no special IAX image. Just use SIP and it should work with Asterisk
as well.
I want to deploy some remote SNOM, but can't use SIP. Does it use IAX or
SIP as the protocol?
TIA
rgds
Peer Oliver Schmidt
Christian,
There are a couple of images at http://snom.com/download/share. We are not
really happy with the latest image yet; hopefully we can fix the remaining
issues in a couple of days. Input appreciated (but no new feature requests
until we have this stuff stable!).
I could not find any image
Walter Doerr schrieb:
Hello,
I am trying to use * to handle anonymous ISDN callers.
Something like
exten = 5150/0,1,Congestion
should work but doesn't. Apparently because the ISDN CAPI doesn't
use 0 for callers who don't send their number.
Is there a way to make * identify ISDN callers who
Jose,
Mozilla 1.5 on Gentoo Linux 1.4 has trouble displaying the Asterisk
pages of the Wiki. (The irony!) The text is pushed off the right
margin of the page.
The problem is not related to Mozilla 1.5 on Gentoo Linux 1.4, but has
to do with Mozilla 1.5 on _any_ system. It is a known bug,
Hello kapejod,
The quadBRI card has 4 BRI ports that can individually be configured
for TE mode (to connect ISDN lines) or NT mode (to connect ISDN phones).
Please find the details at:
http://www.junghanns.net/asterisk/page17.html
when are you going to release some pricing on the card? It just
[EMAIL PROTECTED] wrote:
The performance is better with an active ISDN card or CAPI compatible driver?
Yes, you should go and get a CAPI supported card and use the CHAN_CAPI
driver.
rgds
pos
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-over-bluetooth link automagically ?
The P900 offers to be a VoiceGateway via Bluetooth. So, it looks as if
it should be able to work the other way round, only.
BTW: Nicolas, are you thinking of finishing up your SyncML tool
(http://nicolas.bougues.net/syncml/)
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Good morning,
does anyone know of a (PCI-)card to allow asterisk to have an internal
ISDN bus, ie. being able to utilize ISDN phones as extensions to
Asterisk, like FXS for analog?
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Good day,
I want to have Asterisk as my gateway to the outside world and use
another PBX to connect my existing phones.
How do I specify the dial string to forward the original Caller ID to
over the ISDN to the second PBX?
Right now, my extensions.conf looks like this:
exten =
Hello,
anyone from northern germany planning to go to
http://www.guug.de/veranstaltungen/telephony-summit-2004/
If yes, could you contact me off list. Maybe we can save some money by
car-pooling?!
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for pointing out your workaround. It is a feasible solution for
times when the computer is near the phone, most of the time, the phone
is away.
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Craig Waddington wrote:
anyone from northern germany planning to go to
http://www.guug.de/veranstaltungen/telephony-summit-2004/
Thanks for the info. I would like to go.
Is it in German or English?
According to the site mostly english.
rgds
pos
___
and present the call
information to the user.
Ideas anyone? I guess, I won't be able to get this done without some
client specific programming, will I?
All the best for 2004.
Best regards
Peer Oliver Schmidt
the internet company
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Philipp von Klitzing schrieb:
Hi!
I have an installation utilizing * with an AVM C4 (ISDN card). Using
softphones (SIP and IAX) I have sound problems, like echo.
Did you test if a good soundcard and especially a really good headset
make a difference?
I've done tests with a Labtec headset on
Hello Matteo,
I use ISDN with AVM C2, also fritz pci with tdm fxs sip phone without
any echo.
what isdn channel driver are you using?
I suggest using the avm with capi+chan_capi-0.3.0 and turn
on echosquelch in capi.conf
That is my configuration (0.3.0, echosquelch=1 in the general section
Dan,
A first basic version of DIAX as an ActiveX can be downloaded from:
http://www.laser.com/dante/diax/activediax.zip
[..]
P.S. Unfortunatelly it works only on IBM PC compatible computers (not Mac's
or Pocket PCs)
is there a technical reason for not supplying a version running under
the other
Oliver Schmidt
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Dan schrieb:
Hi,
- Original Message -
From: Steven Critchfield [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, December 11, 2003 12:10 PM
Subject: Re: [Asterisk-Users] * CVS checkout does not work on one
box -solved now
..
It doesn't have to do with the directory, just the
it right now. This is not a
priority for me.
you can't run DIAX in TS mode, as under the Win2k Terminal Server
normally there is no sound card available.
http://www.winnetmag.com/Article/ArticleID/7493/7493.html
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a manufacturer. I will
need quite a few phones next year for at least two installation.
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.
Either AVM or EICON. I have /heard/ the EICON cards are preferable
because of the on board echo cancellation
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Hello,
today I tried a DIAX - * - DIAX connection over the internet (768/128
ADSL connection on both sides).
The sound quality was great. However, we had some latency problems, and
also, if both sides where not talking the first words had some problems
getting thru.
Is this expected, is there
Michael,
I have the same problem with running iaxcomm. Did the following:
* Extract to c:\cd
* Open command prompt
* c:
* cd \cd
* iaxcomm
What happens:
A cursor change for a couple of seconds from an arrow to an arrow with a
clock. iaxcomm never shows up.
BUT, iaxcomm makes the computer nearly
not very satisfied by the sound
quality.
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to add more
lines...
you might want to re-read the results:
1000 queries = 2.302s
For me this looks like 2ms per query.
Maybe WipeOut can confirm the information (one way or another)
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Hi,
and need 'callto:' support :-)
Why you need this?
[..]
callto: support is needed if you want to imbed an HREF link in a webpage
to automatically call someone over a (soft)phone. It is an URI, just
like mailto:, http:, ftp:.
What does this mean for DIAX?
DIAX should be able to be called from
command line phone, you can use the one from Steve's iaxclient
project on Sourceforge.
You do not need a full featured phone for that..
True, but to complement the features already found in DIAX the
eforementioned request sounds feasible.
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browser.
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or maybe to a colleague
callto:[EMAIL PROTECTED]
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Anyone got any pointers on where to find USB handsets or headsets that
can be used as the audio device on a softphone?
Have look at http://www.voipvoice.com. A UK company offering
a) USB handset with on/off hook
b) USB handset with dialing pad.
They have an API for Windows, which is needed to
Hello Olle,
Olle, I can't reach the faq page, and haven't been able to for the
last four days.
I'm getting 504 gateway timeout errors.
Gateway timeout indicates something with your web proxy ...or?
I've been able to reach the Wiki all weekend, I've updated and created
several pages...
I also now
Hi Dan,
Another problem I am seeing is I cannot delete any
phone book entrys.
This is very strange...
Someone else with this issue?
I cannot reproduce it here
Just tried to delete Entry 12. Same problem here. And afterwards I can't
start DIAX anymore, except by manually editing diax.cfg and
. There are rough edges, but the start looks promising.
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Christopher Stephens schrieb:
Is there SIP client which work with Asterisk and can be embedded in a
HTML page ?
It may not be *exactly* what you're looking for, but try:
http://fwd.pulver.com/callme.php?userid=411
[..]
Unfortunately this seem to work with Internet Explorer, only.
rgds
pos
After
apt-get update apt-get upgrade -y
wget http://www.junghanns.net/asterisk/downloads/chan_capi.0.2.5c.tar.gz
tar xfvz chan_capi.02.5c.tar.gz
cd chan_capi-0.2.5c
make make install
shutdown -r now
asterisk seg faults upon calling in via ISDN.
Any ideas are greatly appreciated.
rgds
pos
Peer Oliver schmidt wrote:
After
apt-get update apt-get upgrade -y
wget http://www.junghanns.net/asterisk/downloads/chan_capi.0.2.5c.tar.gz
tar xfvz chan_capi.02.5c.tar.gz
cd chan_capi-0.2.5c
make make install
shutdown -r now
asterisk seg faults upon calling in via ISDN.
After another
apt-get
Roy Sigurd Karlsbakk wrote:
After another
apt-get install asterisk-dev
another make cycle for chan_capi
apt-get install asterisk
...
Selecting previously deselected package asterisk.
(Reading database ... 41473 files and directories currently installed.)
Unpacking asterisk (from
Josh,
I downlaoded it and tried it, SIPPS. Nice featureful sip client, however, I
haven't been able to get it to pass dtmf to *.
I have played with SIPPS as well. DTMF worked for me, but only when
using the mouse to click on the keys of the virtual keypad :(
Reported it back to Ahead.
I don't
Hi Tilghman,
I am trying to use the Windows iax client.
My iax.conf looks like this:
-snip-
[pos|
^
I'm hoping this is a typo in the email. If it isn't, that might
explain everything.
Thanks eagle eye. That was the error.
Thanks again.
rgds
pos
Hi,
in a HTML page I can write href=mailto:[EMAIL PROTECTED] and clicking on the
link will open the default mail application.
Is anything like that possible with any of the soft phones (SIP or IAX
[Windows])?
Any and all information is greatly appreciated.
rgds
pos
Dan wrote:
add in the [pos] section :
username=pos
and then if it still doesn't work try to comment the lines:
;deny=0.0.0.0/0.0.0.0
;permit=10.1.3.0/255.255.255.0
;defaultip=10.1.3.2
Did all that, even restarted asterisk (instead of reload via CLI), still
no go. Any other idea?
rgds
pos
I am
Good morning,
I am trying to use the Windows iax client.
My iax.conf looks like this:
[general]
port=5036
bindaddr=10.1.3.111
bandwidth=high
allow=gsm ; Always allow GSM, it's cool :)
tos=lowdelay
[pos|
type=friend
context=default
auth=plaintext
secret=pos
Hi,
we are a small office and would like * to answer the phone after 5
seconds and play a greeting. In parallel a phone should continue to
ring. When the phone gets picked up, the call should be transfered.
Is this possible?
Any ideas/pointers are greatly appreciated.
rgds
pos
Mark,
thank you for your information.
I am in the process of recording voicemail prompts in german. How do I
specify the language for the voice mail messages? I want to offer both
language files, based on the calling party.
Use setlanguage. Then organize the language files by directory e.g.
Hi Andreas,
I have asterisk behind my primary PBX connected via ISDN (chan_capi).
Calling out and calling in works just fine, however I can't connect to
my primary pbxs' extensions.
at my site it is working exactly as you wrote in your 1st example. How is
your PBX setup? I remember that there is
Reini Urban wrote:
Mark Spencer wrote:
Use setlanguage. Then organize the language files by directory e.g.
/var/lib/asterisk/sounds/de
/var/lib/asterisk/sounds/digits/de
Also, say.c will have to be modified to support German style number
handling.
Mark
On Sun, 27 Jul 2003, Peer Oliver schmidt
Peer Oliver schmidt wrote:
Hi Andreas,
I have asterisk behind my primary PBX connected via ISDN (chan_capi).
Calling out and calling in works just fine, however I can't connect to
my primary pbxs' extensions.
at my site it is working exactly as you wrote in your 1st example. How is
your PBX
Armand A. Verstappen wrote:
On Mon, 2003-07-28 at 08:36, Peer Oliver schmidt wrote:
Ok, the first three things I did. Unfortunately, I am no c coder. But
the logic to say german numbers is identical to the english logic, ie.
21 = twenty one
11 = eleven
210 = two hundred ten ('and' between
101 - 200 of 210 matches
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