Re: [Asterisk-Users] AGI Diad number
Hi Ben,
in my dialplan I use:
exten =
_0.,1,AGI,lcr.agi|${EXTEN:${TRUNKMSD}}
then you can get this argv in your AGI
with:
$number = $ARGV[0];
should give a you a point where to start... If you
have questions please ask!
bye...
Thorsten
- Original Message -
From:
Ben Merrills
To: [EMAIL PROTECTED]
Sent: Wednesday, June 30, 2004 12:46
PM
Subject: [Asterisk-Users] AGI Diad
number
Is there a way of getting the
dialled number from an AGI? Is it passed in the initial variables, or can it
be pulled out or passed across from the dial
plan?
Cheers,
Ben
Merrills
Griffin
Internet
[Asterisk-Users] problem with zaphfc
hi everybody, I'm running my asterisk with a HFC-S card in NT-mode with a modded NTBA (NT1) (=simply crossed cable) and two ISDN-phone behind it. Now, when ever I user both phones at the same time, the sound is very, very crappy, as if it is played at a slower speed (like playing a 7'' single at 33 speed - in those old venyl days). I have not modified my NTBA with a second ohm resistors - can this be the problem? thank you! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] problem with zaphfc
the problem not only occures when I use both phones - when I'm using phone 1 and annother calls knocks on for example - the sound is also not ok. any hints? I'm using a VIA C3 600 MHz with a dual-riser from VIA (make 2 PCI-slots out of one). maybe this is the problem ?! - Original Message - From: FastJack [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, June 26, 2004 12:52 PM Subject: [Asterisk-Users] problem with zaphfc hi everybody, I'm running my asterisk with a HFC-S card in NT-mode with a modded NTBA (NT1) (=simply crossed cable) and two ISDN-phone behind it. Now, when ever I user both phones at the same time, the sound is very, very crappy, as if it is played at a slower speed (like playing a 7'' single at 33 speed - in those old venyl days). I have not modified my NTBA with a second ohm resistors - can this be the problem? thank you! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] transfer bug (#701 - remote party hears alison, not me)
Hi, this is no bug. When you want to park a call just hit #700. Alison will then tell you on which extension the call was parked. To pick up this call just dial the announced extension (e.g. 701). When you press #700 while in a call you connect this call to the call parked at this extension and, if no call is parked, he will talk to alison. But he would hear that there is no call parked and not that it is a invalid extension. I hope I got it rigth, if not I'm sorry. Bye - Original Message - From: Andrew Kohlsmith [EMAIL PROTECTED] I hit #701 instead of #700 though -- after a pause, I got a fast busy and the call was gone. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] grandstream and timeservers
hi everybody, suddenly my budgedtone doesn't display the right time anymore.I tried several timeservers and even installed my own one but... my grandstream displays 1900-01-02 as the time.I also tried several images - currently I'm using 1.0.4.63. can anybody help me with that? thanks!
Re: [Asterisk-Users] problems with bri-stuff.0.0.2rc20a
hi klaus-peter, yepp... with overlapdial=yes (almost) everything works great, again. one problem is left... touchtones are not working anymore so I can't use voicemail-system, parking and stuff. thank you for your help. ...bye thorsten - Original Message - From: Klaus-Peter Junghanns [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, May 03, 2004 12:35 AM Subject: Re: [Asterisk-Users] problems with bri-stuff.0.0.2rc20a Hola, if you have overlapdial=no in zapata.conf then * will jump into the s extension on a NT span (this way you can use DigitTimeOut and ResponseTimeOut to make patterns like _X. work as expected.). So, either you create an s extension, e.g.: exten = s,1,DigitTimeOut(3) or you set overlapdial=yes in zapata.conf. best regards Klaus -- Klaus-Peter Junghanns CEO, CTO Junghanns.NET GmbH Breite Strasse 13a - 12167 Berlin - Germany fon: (de) +49 30 79705390 fon: (uk) +44 870 1244692 fax: (de) +49 30 79705391 iaxtel: 1-700-157-8753 http://www.Junghanns.NET/asterisk/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] problems with bri-stuff.0.0.2rc20a
forget it... seems to work - no idea what was/is wrong. - Original Message - From: FastJack [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, May 03, 2004 11:38 AM Subject: Re: [Asterisk-Users] problems with bri-stuff.0.0.2rc20a hi klaus-peter, yepp... with overlapdial=yes (almost) everything works great, again. one problem is left... touchtones are not working anymore so I can't use voicemail-system, parking and stuff. thank you for your help. ...bye thorsten ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] problems with bri-stuff.0.0.2rc20a
Hi klaus-peter, I thought I fixed this error... but when ever I pickup the phone before I dial the number (the sitution I got the former descibed problem fixed with overlapdial=yes) I can dial an extension but I cannot send any furhter digits so voicemail and early b3-connects with chan_capi do now work. I hope you can help me again. ... bye thorsten - Original Message - From: FastJack [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, May 03, 2004 11:38 AM Subject: Re: [Asterisk-Users] problems with bri-stuff.0.0.2rc20a hi klaus-peter, yepp... with overlapdial=yes (almost) everything works great, again. one problem is left... touchtones are not working anymore so I can't use voicemail-system, parking and stuff. thank you for your help. ...bye thorsten ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] problems with bri-stuff.0.0.2rc20a
hi everybody, just upgraded my bri-stuff driver to 0.0.2rc20a. now i have a strange problem :-( i have immediate = no but when i pickup the phone i get : *CLI == D-Channel on span 1 up -- Extension 's' in context 'default' from '6294094' does not exist. Rejecting call on channel 2, span 1 i have started asterisk with -vvc so there should be a debug message if immediate mode was on. maybe anyone (klaus-peter) can help. i'm using a hfc-card in nt-mode. i'm not 100% shure but i think that my phone is using uk-tones (ring ...) since the update but all language-settings are nl. looking forward to get some help ;) thorsten ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] calling between two zap points with zaphfc
hi everybody, just went into some trouble (again!!) while I was trying to make a call between two (isdn)phones connected to my hfc-s card. I am running junghanns.net's hfc-bri-driver. the call is terminated after a few seconds. anyone else got this to work? btw: I am using a NTBA as powersource for the two phone. the first phone is an old teles phone. the other one i a siemens cordless phone (with own powersupply). I have not modified my NTBA to have 50 ohm!! making calls to the outside world from these two phones (even two at the same time) via my avm-fritz and chan_capi works perfektly. ony thoughts? bye
Re: [Asterisk-Users] OT: SNOM and TAPI
hi christian, have a look at http://www.julmar.com/. TSP++ version 2 is a opensource, GPLed library for creating a tapi service provider. I think this is a good point to start. I was just dreaming of having such a baby for use with asterisk* via it's manager function. bye thorsten - Original Message - From: Christian Stredicke [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, February 23, 2004 10:58 PM Subject: RE: [Asterisk-Users] OT: SNOM and TAPI Sorry we are not so good in implementing Windows-stuff... Maybe has someone out there a template for TAPI? Something for someone who never did something with COM or DCOM or .net or whatever... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] *8# and zaphfc in NT-mode
hi everybody, does the zaphfc driver support the *8#, *78#, *72#, ... functions when running in NT-mode? thanks... bye thorsten
[Asterisk-Users] SEGFAULT (capi amd hfc-s NT)
hi everybody, just ran into trouble... I place an outgoing call from my zap (hfc in NT-mode) via chan_capi. the I transfer the call to a SIP-phone (x-lite orgs budgetone). if the called person now presses any key on his phone my asterisk segfaults :( any ideas? anyone??!
Re: [Asterisk-Users] Call Redirection
hi, I think it should be even great, to have an ack-password so if the phone is answered by someone unexpected (e.g. your wife!!) the person CAN_NOT answer this call! any thoughts? - Original Message - From: AstGrp [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, February 21, 2004 7:09 AM Subject: [Asterisk-Users] Call Redirection I have a question regarding call redirection. Example call comes in to a extension. No one answers then call gets redirected out to cell phone. I need to implement something like for our tech support line. Call rings multiple extensions then if no one answers it gets forwarded out to a cell. I have tried the following : [FWD] exten = s,1,Dial(Zap/g2/7041234567) ;Tech Support exten = 4200,1,Dial,SIP/gclarkSIP/kelworth|15 exten = 4200,2,Goto(FWD,s,1) But everytime I try this - the phone that is generating the call receives another call from the pbx. It appears that the call is going out but, calls back the same user who is making the call. Thanks, -gcc ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] tapi for asterisk*
hi everybody, wouldn't it be great to have a tapi provider that works with asterisk? all we would need is a custum tapi service provider that uses asterisk's manager functionsto play a call between a configured extension (e.g. your iax/sip/zap phone and anumber that you provide using e.g. microsoft outlook). I found the folling in the net that could give some help: http://www.julmar.com/here you can find a gpl licensed lib to create a tsp. http://www.voip-info.org/wiki-Asterisk+manager+dialout again: here is a draw of how it could be done. - in your properties you have your default extension configured. - then you select a contact in MS outlook and want to place a call using tapi and the asterisk tapi service provider. - the tsp then uses asterisk manager function to place a call between your extension and the person you want to call. (e.g. Channel: Capi/@12345:0815 to Exten: 4711) or we can somehow drop a .call-file in asterisk outgoing directory if this is easier / more secure. who would be interested in helping develloping such a project. My programming skills are not that good but I will try to do my best. bye
Re: [Asterisk-Users] Re: HFC-S cards?
hallo reinhard, ich werde mir gleich das modell von conrad kaufen. du schreibst du hast sie mit zaphfc betrieben. hast du sie im NT oder TA moduls laufen lassen? ich moechte hier gerne ein ISDN-telefon an meine asterisk-box anschliessen. vielen dank fuer deine hilfe bis denn ... thorsten - Original Message - From: Reinhard Max [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, February 17, 2004 11:44 AM Subject: [Asterisk-Users] Re: HFC-S cards? Hi, On Mon, 16 Feb 2004 at 17:22, FastJack wrote: anyone knows where to get one of theses cards (or any other based on the HFC-S chipset) in germany? I bought a Longshine LCS-8051 Card recently. http://www.longshine.de/produkt-ger/modem/8051.htm It works with i4l and zaphfc. K+K Computer sells them for EUR 28.59 + shipping. http://www.kkcomputer.de cu Reinhard ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zaptel BRI and HFC-S cards
hi everybody, is there any documentation available how to run asterisk with a hfc-s based card in NT mode? does it need a special kind of wire, is a t1 (ntba) needed for termination / powersupply between the card and a phone? thanks in advance ... thorsten ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ISDN LCR USING ASTERISK
hi everybody,
here is what I've done to make my asterisk* act as a LCR.
first, you'll have to install isdnrate (part of isdnutils) and get a recent
rate-??.dat (check rates4linux.sourceforge.net for that.)
to test isdnrate just try the following command:
lcr -o -b3 -l60 *any_number_you_want_to_test*
the -o tells isdnrate to only use provides activated in /etc/isdn/rate.conf
(e.g. if you have some preselection providers or tisdn-xxl)
-b3 is for the best 3 providers
-l60 says call duration 60 seconds (the default-value I also used in my AGI
is 153 secs. so if you want to use annother duration please change the
commandline in the agi)
then I wrote a very little (and simple) AGI.
-
/lcr.agi ---
---
#!/usr/bin/perl
$|=1;
while(STDIN) {
chomp;
last unless length($_);
if (/^agi_(\w+)\:\s+(.*)$/) {
$AGI{$1} = $2;
}
}
$number = $ARGV[0];
$length = 60;
$raw = `/usr/bin/isdnrate -o -b1 -L -l$length $number`;
$raw =~ /([0-9]*)_.;(.*?);/;
$prefix = $1;
$provider = $2;
print VERBOSE \Using LCR Provider $provider - $prefix!\\n;
$result = STDIN;
print SET VARIABLE LCR $prefix\n;
$result = STDIN;
as you can see, my AGI just sets a variable called LCR.
here is how I use it in my dialplan:
exten = _0.,1,Answer
exten = _0.,2,DigitTimeout,5 ; Set Digit Timeout to 5 seconds
exten = _0.,3,ResponseTimeout,10 ; Set Response Timeout to 10
seconds
exten = _0.,4,agi,/lcr.agi|${EXTEN:${TRUNKMSD}}
exten = _0.,5,Dial,CAPI/@6294096:b${LCR}${EXTEN:${TRUNKMSD}}|60|T
I know that this is all very simple - and maybe there are some errors in my
setup but I just wanted to share my expirence with you.
bye
... thorsten
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[Asterisk-Users] HFC-S cards?
hi everybody, Acer ISDN-Surf, Billion Bipac ISDN, Trust PCI ISDN Modem, D-LINK DMI-128+ to name a few ;-) anyone knows where to get one of theses cards (or any other based on the HFC-S chipset) in germany? my computer-trader maybe can get d-link's card but he don't know how long it could take. does anyone (hello kapejod ;)) ) know wich one should be my first choise, just in case I find more than one of these babys. thanks! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Firefly and asterisk*
Hi Adam, I just got it to work ;)) I added an entry at the bottom of my iax.conf : register = *MY_FIREFLY_NUMBER*:[EMAIL PROTECTED] [firefly] type=friend host=firefly.virbiage.com context=incoming then, when a firefly user calls me, he is taken to incoming/s. I'm not sure if type=friend is right and if any other options should be set but IT WORKS!!! The only firefly related problem I'm still having it is that firefly erases the leadig 00 from every number in my (externel) contacts-list. bye and thanks - Original Message - From: Adam Hart [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, January 31, 2004 2:32 AM Subject: Re: [Asterisk-Users] Firefly and asterisk* - Original Message - From: FastJack [EMAIL PROTECTED] GREAT!!! Just got my asterisk* calling firefly users. Setup was really easy: snip Anyone knows how to receive calls on my asterisk*-box from the firefly-network? I'll fix this soon, then you should be able to connect to firefly network just like a normal iax2 connection. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Introducing Firefly
Hi, just installed Firefly. Looks great, sound is also great. I just got the following problem. I'm using Firefly with my asterisk*-box. When I enter a contact with the number +00233612345 Firefly just erases the 00 when I restart it. Am I missing something? Thanks! Great software!!! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Firefly and asterisk*
GREAT!!! Just got my asterisk* calling firefly users. Setup was really easy:
just add an extention
exten = _8XXX,1,Answer
exten = _8XXX,2,DigitTimeout,5
exten = _8XXX,3,ResponseTimeout,10
exten =
_8XXX,4,Dial,IAX2/*YOUR_FIREFLY_NUMBER*:[EMAIL PROTECTED]
.com/${EXTEN}|60|T
now I can call users in the firefly-network from every phone that is
connected to my asterisk*-box.
I also added a register = ... entry to my iax.conf but this doesn't seam to
work.
Anyone knows how to receive calls on my asterisk*-box from the
firefly-network?
thanks!
- Original Message -
From: Andy Powell [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, January 30, 2004 10:12 AM
Subject: Re: [Asterisk-Users] Introducing Firefly
Hi,
I downloaded this the other day and finally got it to stop crashing. It
appears that any response from asterisk
that implies an error (for example dialing a non-existant number, using the
wrong password, selecting a codec
that you've configured a local * not to use etc) resulted in a crash. I've
only tested the IAX proto not sip or your
own network. running XP with uptodate patches on a local lan.
When it works it works really well, although I don;t particularly like in
initial beep and end beep when i make
a call (I haven't played with all the options so it may be that I can turn
this off).. sound quality is good. All in all
a nice little app. Are you planning on allowing other people to run your
server side (like Jabber does) in their
environments?
If you need any further debugging info on the crashes, let me know...
HTH
Andy
*** REPLY SEPARATOR ***
On 28/01/2004 at 12:11 Adam Hart wrote:
After many months of development, I'm pleased to announced Firefly - an
IAX soft phone and network.
The firefly softphone - free, runs under windows, allows connection to
multiple networks, skinable interface, connection to firefly network, IAX2
protocol, (SIP in next release), codecs supported - iLBC, G.711 ulaw/alaw,
GSM. - contact lists, selectable ringtones.
download from here - http://www.virbiage.com/firefly/
The firefly network - also free, runs on an enhanced version of IAX2
(simply uses IAX2 text messages for customised part), voicemail, text
messaging, online presence, ability to indicate status (available, away,
NA). I believe you can connect using a standard asterisk box but you'll
miss out on the extended features. The network runs on iLBC so
unforunately it won't work with most IAX2 clients (unless you get * to
translate)
Thousands of people have used it but it's still regarded in beta, as we
are still in heavy development (so expect a few bugs). It doesn't use
iaxcomm as we needed our own framework to support sip, including our own
jitterbuffer. If you don't wish to connect to the firefly network, click
cancel when it asks you.
Coming soon features
SIP - in alpha, few bugs outstanding
music onhold - playing mp3s while the other party is onhold
fast audio - will reduce the latency by 40-50ms
speex - (if anyone wants it?)
Feel free to contact me on or off the list to report bugs and suggestions.
I'll post everytime we release a new version (probably every week),
including fixed bugs and new features
Our website is http://www.virbiage.com/
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Re: [Asterisk-Users] Debian Packages and Mirrors
hi everybody... have you checked the asterisk backports from www.backports.org? I'm currently building my asterisk system and i think i will use these debs as I've successfully used alot of debs from backports.org in almost every production-server we have. don't know the quality of the asterisk packages from backports.org but I'm almost sure they are great ;)) bye thorsten - Original Message - From: William Waites [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, January 23, 2004 5:08 PM Subject: [Asterisk-Users] Debian Packages and Mirrors FYI and to whom it may concern, I have made Debian packages of Asterisk et. al. You still need to build a new kernel and the zaptel modules from source, but Asterisk and libpri are manageable with dpkg. The debs as well as mirrors of the source distribution are here: http://www.ntgos.com/Projects/Asterisk/Download http://parc.styx.org/asterisk I would also like to mirror the CVS repository as well as set up a cvsweb... -w -- /~\ The ASCII Ribbon Campaign \ /No HTML/RTF in email X No Word docs in email / \ Respect for open standards ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Debian Packages and Mirrors
Title: RE: [Asterisk-Users] Debian Packages and Mirrors hi everybody... http://www.backports.org has asterisk 0.7.1 for woody ;)) bye - Original Message - From: Kostur, Andre To: '[EMAIL PROTECTED]' Sent: Friday, January 23, 2004 5:45 PM Subject: RE: [Asterisk-Users] Debian Packages and Mirrors Note that there are also asterisk packages in the standard Debian repositories http://packages.debian.org/cgi-bin/search_packages.pl?keywords=asterisksearchon=namessubword=1case=insensitiveversion=allrelease=all v0.1.11 in stable, v0.5.0 in testing, v0.7.1 in unstable (unless you're not on an i386) The source for the zaptel interface is there too (package name: zaptel). Haven't looked for libpri... we don't have a PRI service...) What do you have different in your packages?
Re: [Asterisk-Users] New sounds also now in CVS
great!! but when will asterisk use some of these new babies?;)) it would be really great to have app_queue saying you are currently caller number 7 in the queue (=you-are-curr-call-num.gsm + 7.gsm + in-the-queue.gsm) that would be really really great. when speaking of app_queue. i think it would also make sense to have some announces during the music-on-hold, maybe even different stages (like please hold the line, we are sorry that we weren't able to connect you by now, ...). music-on-hold plays for a given time and then the first announcement comes, than music-on-hold again, after that the second announcement (or, if only one is defined, the first announcement again). app_queue is the most important application for everybody who has to handle support-calls. sorry for my bad english ;)) bye ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] newbie ISDN question
hello klaus-peter this sounds great ; will the phones that are connected to a bri in nt-mode still allow all isdn-functions (in special : caller id-display)? thanks... - Original Message - From: Klaus-Peter Junghanns [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, January 14, 2004 12:46 PM Subject: Re: [Asterisk-Users] newbie ISDN question Thorsten, theoretically you can connect 8 phones per port, but only 2 can be used at the same time. We advise to use 2 per port and in some scenarios 3 might be an option. So you can connect 8 ISDN phones to the quadBRI card. The drivers are still released as experimental and have some bugs. We are planning to be stable in about 2 weeks. The cards are in stock, so delivery will be fast. We ship with worldwide with UPS. best regards kapejod -- Klaus-Peter Junghanns CEO, CTO Junghanns.NET GmbH Breite Straße 13 - 12167 Berlin - Germany fon: (de) +49 30 79705390 fon: (uk) +44 870 1244692 fax: (de) +49 30 79705391 iaxtel: 1-700-157-8753 http://www.Junghanns.NET/asterisk/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] best SIP-softphone?
hi everybody. i'm currently looking for a good SIP-softphone. i tried x-lite but i'm not happy with it's soundquality as there is a very high noiselevel. do you know a better softphone? and would a hardware SIP-phone offer (almost) the same audioquality as my isdn-phone? if so what phone would you recommend (and where can i get it in germany)? thanks!! bye thorsten
[Asterisk-Users] newbie ISDN question
hi everybody, sorry for posting such a stupid question ;) i've managed to run asterisk* with my AVM fritz2.0 card and a some VOIP-softphones (SIP, H323). the functions of asterisk* really satisfied me ;))) now i want to run asterisk* istead of our old PBX. but it would be great to connect some phones directly to my box. how does a E100P from digium work. can i connect it to my ISDN-line and my internal phones (ISDN)? it would look like this: [PHONE2] / [PC]-[E100P] - [PHONE1] \ [ISDN-LINE] thank you for your help!!! thorsten ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] newbie ISDN question
hi klaus-peter, thank you for your replay. btw: i am using you chan_capi already ;)) it works great!!! how many internel phones could be connected to this card? how stable is the driver (can i use it for a production-system)? sorry for all that stupid questions - i know linux and ip and pc-hardware but telephone-technics are all new for me. how long would delivery of that card take? thanks (oder besser gesagt: VIELEN DANK ;) ) thorsten - Original Message - From: Klaus-Peter Junghanns [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, January 14, 2004 11:54 AM Subject: Re: [Asterisk-Users] newbie ISDN question Hi Thorsten, the E100P is a PRI ISDN Card (S2M in Germany). You cannot connect phones to that card. 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 best regards kapejod -- Klaus-Peter Junghanns CEO, CTO Junghanns.NET GmbH Breite Straße 13 - 12167 Berlin - Germany fon: (de) +49 30 79705390 fon: (uk) +44 870 1244692 fax: (de) +49 30 79705391 iaxtel: 1-700-157-8753 http://www.Junghanns.NET/asterisk/ hi everybody, sorry for posting such a stupid question ;) i've managed to run asterisk* with my AVM fritz2.0 card and a some VOIP-softphones (SIP, H323). the functions of asterisk* really satisfied me ;))) now i want to run asterisk* istead of our old PBX. but it would be great to connect some phones directly to my box. how does a E100P from digium work. can i connect it to my ISDN-line and my internal phones (ISDN)? it would look like this: [PHONE2] / [PC]-[E100P] - [PHONE1] \ [ISDN-LINE] thank you for your help!!! thorsten ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
