. Anyone has some idea?
quadBRI CAPI!!!
The quadbri cards do not use/support CAPI. If you don't have another
CAPI capable device in your system you can't/shouldn't use CAPI (I guess
you could use CAPI via mISDN, but what is the point?)
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PGP Key ID: 0x83E1C2EA
, als Exchange 2000 Server nebst CALs.
Hast Du schon einmal Bynari angeschaut? http://www.bynari.com
Vielleicht ist das ja etwas was Euch weiterhilft.
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PGP Key ID: 0x83E1C2EA
Hi,
some people report good success with the zaphfc cards, others, incl.
myself have mixed results.
I am using the debian stock kernel 2.4.27 with mixed results. Anyone
care to tell what kernel(s) you on successful zaphfc integrations?
Thanks.
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PGP Key ID
and calelrid works
What version asterisk and bristuff do you use?
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so far.
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) on Primary D-channel of span 1
Maybe this is helpful to find where the problem is. I will go and unload
the drivers (and hope not to crash the box).
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have good results with an AVM C4 card using the CAPI drivers. I
started out with an old ISA AVM B1 card which had echo problems, which
got fixed with some later chan_capi driver releases.
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Kapejod or Florz are able to shed some light on the issue.
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Altus Snyman wrote:
Where do you get this new version of bristuff,I had a look on the
webpage and there's only RC3
My first action every morning is to look at the top of this page:
http://www.junghanns.net/asterisk/downloads/?C=M;O=D
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Ken Jones wrote:
astfax allows you to create an email to fax gateway.
Are we going to see some integration of astfax with Courier-MTA/IMAP?
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: The question whether the calls are getting really dropped or
just silenced for a couple of seconds still stands.
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Hi,
we want to provide our users with a Click To Login interface for the
AgentCallbackLogin. Any sample.call or AGI anyone has developed out there?
Any and all help is greatly appreciated.
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Hi,
the new Grandstream release for the ATAs allows the setting of the FXS
impedence, the Onhook Voltage and the Polarity Reversal.
Anyone know how these should be set in Germany?
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for zaphfc to provide
a ZAP timing source? So, if you have zaphfc card together with the
bristuff from http://www.junghanns.net you don't need ztdummy, do you?
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. Nice
to know that upgrading won't help :-(
Do you get calls which stop in the middle of the conversation as well?
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-channel stays down. Calls come in,
but can't be answered.
Does anyone know of a fix for this, or might have some insights on how
to circumvent this problem?
Any and all help is greatly appreciated.
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Hello Stephan,
Another way is to set the callerid in your extensions.conf via exten
= 807440,2,SetCIDNum(0${CALLERIDNUM}).
Try the variable PRI_NETWORK_CID instead of CALLERIDNUM
This did the trick. I will go and update the Wiki,,,
Thanks and have a good weekend.
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User-Agent: Asterisk PBX
Date: Fri, 21 Jan 2005 17:46:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 236
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Jens, thanks for the feedback.
I've added a ZAPHFC card to my CAPI based system. Calls coming in via
ZAPHFC do not forward the caller id to the SIP phones. Calls coming in
via CAPI do forward the caller id to the SIP phones.
I think you didn't set usecallerid=yes in your zapata.conf?
Added it,
Grüßen
Peer Oliver Schmidt
the internet company
Lee Howard wrote:
On 2005.01.19 01:39 tim panton wrote:
My options include;
1) get a basic fax machine and plug it into a (iaxy/sipura?) ATA.
Configure the ATA to do alaw
so asterisk doesn't have to do any transcoding and hope it works.
2) use spandsp on my asterisk system and add
. Im Netz muss dann nur ein WebDAV Repository eingerichtet
werden, da schreibt Ihr dann in eine Datei, und alle Leute verbinden
sich auf diesen Remote Calendar.
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-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.4 (GNU/Linux
Gren
Peer Oliver Schmidt
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Stand bringen, oder aber der Newbie verbindet sich auf
http://whatismyip.org/
Die 4 mit Punktunterbrochenen Zahlen kann er Dir dann vorlesen.
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Money unter Wine zum Laufen
bringen möchte, dann braucht man IIRC IE6 vorher installiert.
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machen. Dort funktioniert das Reverse Port forwarding (oder wie auch
immer der korrekte Name ist).
Andere Möglichkeit, wenn es Dir nur um den Domänennamen geht, ist die
Anlage des Domänennamens in Deiner hosts Datei auf dem Windows Rechner.
hth
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Bill Seddon wrote:
How can Asterisk be configured to execute some number of dialplan commands
when it is started or restarted?
[..]
In the meantime I'm hoping that it is possible to use the built-in database
and be able to run some kind of autostart context. Does such a facility
exist?
Without
in the download directory.
http://www.junghanns.net/asterisk/downloads/bristuff-0.2.0-rc2b.tar.gz
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his config?
Thanks.
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in the manner you described, and if yes, could
you update the Wiki, or share your setup here?
Thanks.
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.
Is it save to assume, that to send a fax to 415-555-1234 the correct
statement would be:
TxFAX(myfax.tiff|caller=411234)
?
Thanks
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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
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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
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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
Hello,
anyone out there that is successfully using Bynari with the new IMAP ACL
offered by courier?
I do get access to the IMAP folders, I can change the rights for users
on the specific folders (like calendar), but I can't see the folders
under the other username.
i.e. user Jane shares her
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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
Hi,
I just installed 0.45.2. However, Mozilla still did not see the ability
to have shared folders.
After changing the announced CAPABILITIES of the IMAP server to include
ACL, Mozilla tells me of my rights to the folders.
Is it correct to modify the IMAP configuration file to include the
Sam Varshavchik wrote:
Is it correct to modify the IMAP configuration file to include the
capability ACL?
No. The ACL capability is announced by the server without listing it
explicitly.
Is it possible to disable the announcement of the capability?
rgds
pos
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
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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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[EMAIL PROTECTED]
-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/)
--
Best regards
Peer Oliver Schmidt
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?
--
Best regards
Peer Oliver Schmidt
the internet company
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?!
--
Best regards
Peer Oliver Schmidt
the internet company
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.
--
Best regards
Peer Oliver Schmidt
the internet company
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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
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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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