Re: [asterisk-users] centos asterisk 1.8 rpms: chan_gtalk and res_jabber missing?

Hi, I'm also interested in rpm packages including chan_gtalk and res_jabber because I do not want to have a build environment on my productive server. Does anybody knows the reason why this is not available via rpm? best regards Christoph Am 28.11.2011 05:30, schrieb Vladimir Mikhelso

Re: [asterisk-users] Update problem | CLI commands missing

Hi List, is there somebody how is able to help me here? Or at least to get more details why this occurs? best regards Christoph Am 08.06.2011 18:00, schrieb Christoph Timm: Hi List, I'm running into trouble, if I perform a 'yum update' on my AsteriskNOW. Currently I

[asterisk-users] Update problem | CLI commands missing

: func_version.so => (Get Asterisk Version/Build Info)". Does anyone know something about this problem? best regards Christoph -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asteri

[asterisk-users] IPv6 and IPv4 NAT not working

" and/or "externaddr" settings. I think that is a bug because the externaddr is used correct during the outbound calls. My Asterisk version is 1.8.3.3! Is somebody able to help me with that issue? Should I write a

[asterisk-users] qsigchannelmapping parameter

Hi, I've connected Asterisk with 4 PRI to a Siemens HiPath 4000. For CALLERID(name) feature I wanna use Q.SIG as switchtype. Cause Siemens PBX orders Channels logical I need the parameter qsigchannelmapping=logical. Here is my chan_dahdi.conf trunkgroups] [channels] language=de context=default

Re: [asterisk-users] Exit Dial Application

s not a Zap problem. I tried it with two SIP phones, same behavior. Bit odd : / > > Regards, > Atis chris... - -- commpany dialog solutions gmbh Dipl.-Ing.(FH) Christoph Fürstaller IP-Communications Ischlerbahnstraße 14, 5301 Eugendorf Tel: +43 662 879512 Fax: +43 662 875960 IP-Tel

Re: [asterisk-users] Exit Dial Application

Non-Commercial Discussion > Subject: Re: [asterisk-users] Exit Dial Application > > On Wed, Apr 15, 2009 at 4:47 PM, Christoph Fuerstaller > wrote: >> -BEGIN PGP SIGNED MESSAGE- >> Hash: SHA1 >> >> Hi Danny, >> >> Danny Nicholas schrieb: &

Re: [asterisk-users] Exit Dial Application

> > -Original Message- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Christoph > Fürstaller > Sent: Tuesday, April 14, 2009 11:50 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re

Re: [asterisk-users] Exit Dial Application

the macro. And voila it's working. Do I have to answer the channel before Dial option 'd' is working? It's a bit odd, cause the dial duration starts counting and I hear a 'beep'. That's not ideal : / I've attached a full.log. > > Regards, > Atis > chr

Re: [asterisk-users] Exit Dial Application

t not helpful for me : / Is it possible to deactivate the 'd' option? Or what else could cause my problem? > > > Regards, > Atis thanks for your help, chris... - -- commpany dialog solutions gmbh Dipl.-Ing.(FH) Christoph Fürstaller IP-Communications Ischlerbahnstraße

Re: [asterisk-users] Exit Dial Application

sed that part :) > > Try enabling "full" log in logger.conf, set verbosity to 3 and debug > to 1, and see what goes in it. > > Regards, > Atis > - -- commpany dialog solutions gmbh Dipl.-Ing.(FH) Christoph Fürstaller IP-Communications Ischlerbahnstraße 14, 530

Re: [asterisk-users] Exit Dial Application

n the >current context, or the context defined in the EXITCONTEXT > variable, >if it exists. > > Regards, > Atis > > On Tue, Apr 14, 2009 at 7:49 PM, Christoph Fürstaller > wrote: > Hi, > > Thanks for your replay. But this can only be

Re: [asterisk-users] Exit Dial Application

t 5 seconds for user to press 5, then hangup if > they don't. > > -Original Message- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Christoph > Fuerstaller > Sent: Tuesday, April 14, 2009 5:04 A

[asterisk-users] Exit Dial Application

DTMF's are transmitted correctly, I can enter the voicmail menue. I'm using Asterisk 1.4.21.1. Any successions are very appreciated. Chris... - -- commpany dialog solutions gmbh Dipl.-Ing.(FH) Christoph Fürstaller IP-Communications Ischlerbahnstraße 14, 5301 Eugendorf Tel: +43 662 87

[asterisk-users] Kirk 600v3 Server with sip secret

3] nat=yes secret=4333 login=4333 callerid=Christoph Fuerstaller<4333> call-limit=10 setvar=intern=1 callgroup=2 pickupgroup=2 [EMAIL PROTECTED] language=de disallow=all allow=g729 allow=gsm allow=ulaw allow=alaw type=friend host=dynamic dtmfmode=RFC2833 canreinvite=no qualify=yes context=intern

Re: [asterisk-users] Dial-Chain interrupted by Operator "Called Party not reachable" Messages

Hi Eric, thanks for your hint. Unfortunately it doesn't work, I just tested it. It seems that at least in german T-D1 Mobile Network a mobile call is "answered" even if that mobile is switched off. On Thu, Oct 04, 2007 at 05:30:11PM -0500, Eric ManxPower Wieling wrote: >

[asterisk-users] Dial-Chain interrupted by Operator "Called Party not reachable" Messages

Hi, I have the following problem: I want asterisk to dial a chain of n-numbers until somebody picks up the line. I am using Digium E1 Hardware (zaptel) for dialing out. Dialing a Chain is basically no problem, I use somwthing like: dial(no1,50) dial(no2,50) dial(no3,50) However, If no1 is not re

Re: [asterisk-users] Problems sending more than 2 SMS with asterisk / smsq

seems no dialplan invoked when I send sms. I use: smsq -d 017xxx -m TEST1 --motx-channel=Zap/g1/0193010 (Germen Telekom Message Center) How could I invoke smsq differently to use an own context of the dialplan ? Thanks Christoph On Mon, Jul 09, 2007 at 05:21:42PM +0200, Matthias Huber wrote

[asterisk-users] Authenticaion on incoming calls

isk. Hope someone can help me with that. Christoph -BEGIN PGP SIGNATURE- Version: GnuPG v2.0.4 (GNU/Linux) iD8DBQFGiKv9R0exH8dhr/YRAgDMAJ9gI6bDTZl8eER0xrCQTMTxq4L/EwCdGZ/t KCUHLkCQzFIto5bk5O8p2Og= =Ypco -END PGP SIGNATURE- ___ --Bandwi

Re: [asterisk-users] Transfercapability DIGITAL

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi robb, Have you just seen the bearer capability in asterisk or is the call nat working? I've seen that a digital call shows up as speech. You are using Zap? Or are you using mISDN? Cause there you have to set an extra parameter in the dial statemen

Re: [asterisk-users] Destar web interface problem

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi Alejandro, SSH to your box like this: ssh [EMAIL PROTECTED] -L 8080:127.0.0.1:8080 (in Putty it's something called portforwarding) This will bind 127.0.0.1:8080 from your asterisk-box to 127.0.0.1:8080 of your local box. So you can access the websi

Re: [asterisk-users] misdn and debian

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, Have you got zaptel installed on your box? And loading the zaphfc module during bootup? I discovered the problem you reported when I switched to mISDN and having installed/loading zaphfc during bootup. Than asterisk doesn't start and the system ha

[asterisk-users] Asterisk Feature attended transfer

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, I'm using the biult in feature attended transfer. If someone calls me, I hit the #, dial another extension and connect these two extensions. When hitting # and dialing the nr, asterisk only diales the new nr for 15 seconds. Is it possible to incre

Re: [asterisk-users] Set(CALLERID(all) not working with 'unknown' call?

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi Jan, Is this call from PSTN? Probably the Nr is prohibited in PSTN, then asterisk doesn't set the CALLERID. Try this: exten => _3072,1,Answer exten => _3072,n,SetCallerPres(allowed) exten => _3072,n,Set(CALLERID(all)=DIRECT <0850553072>) Look here

Re: [asterisk-users] Using server side phonebook directory with SPA941

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, I've deploayed such an centraliced phonebook. The phonebook is stored in an MySQL DB. If a call comes in to asterisk, an agi script is started which queries the database with that nr. If a result is found, I set the CALLERID(name) and send it to t

Re: [asterisk-users] Choppy sound with chan_capi + Fritz Card USB

/showthread.php?t=117614) Christoph Hi everybody, I have a problem which I cannot eliminate on my own. Has anybody any idea for the following: I am using the asterisk-version from Debian-Testing (1.2.13) with the latest chan_capi (also tried an older version). When using the Capi-Channel, everything

[asterisk-users] Autoprovisioning ST2030S

and Outband (RFC2833) but I cant get it to set to SIP INFO. Parameter is the first one on the advanced_advanced.html webpage. Anyone knows how I can do that? christoph -BEGIN PGP SIGNATURE- Version: GnuPG v2.0.3 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org

Re: [asterisk-users] strange things on call transfer

ltel 4400 > and Asterisk ?) : what hardware ? what config files ? > > Thank you > > Minh VO > MIVOC Systemes SAS > FRANCE > - Original Message - > From: "Christoph Fürstaller" <[EMAIL PROTECTED]> > To: > Sent: Wednesday, March 14, 2007 9:16

[asterisk-users] strange things on call transfer

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, I'm setting up an Asterisk system which is connected to an Alcatel 4400 PBX. On the * I permit g729 and gsm as codecs. If I try to transfer a call by hitting the # key, I get this messages and nothing happens on the phone: WARNING[30110]: codec_i

Re: [asterisk-users] Best phone for easy provisioning

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, I'm using a Thomason ST2030. Had difficulties in the beginning, but after a firmware upgrade it works fine. And autoprovisioning works good. Most of the parameters are described in their official (marked as confidential) admin documentation from t

Re: [asterisk-users] mISDN

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi Patrck schrieb: > On Mon, 2007-02-05 at 11:06 +0100, Tomislav Parčina wrote: >> Hi list! >> >> How to make outgoing call thru other mISDN channel group of all channels on >> first group are busy? >> >> I believe I'll need to >> - Check of there i

[asterisk-users] pridialplan/prilocaldialplan

meone can explain that to me. Thanks in advance, Christoph - -- Dipl.-Ing. Kurt Krenn - IT-Beratung Franz-Josef-Strasse 33/4/43, 5020 Salzburg Tel: +43 662 879512 Fax: +43 662 875960 IP-Tel: +43 780 kkrenn (557366) Email: [EMAIL PROTECTED] sip: [EMAIL PROTECTED] -BEGIN PGP SIGNATURE- Ve

Re: [asterisk-users] Asterisk very slow when internet down

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, For that I've set up a local DNS Cache (on the asterisk) - maradns. And entered 127.0.0.1 as the first DNS Server in d/etc/resolv.conf. To decrease the time asterisk is trying to do a dns lookup, I've added this options to /etc/resolv.conf: optio

Re: [asterisk-users] Adding 4000 Lines to asteriskdb via asterisk -rx ?

> Option A: Use the manager interface. > Tzafrir , Thanks, the idea to use the manager interface is wonderful. It is really fast and no data gets lost. I don't think 4000 Rows are a noticeable amaount of data for a db1 database. I coded this: #!/usr/bin/perl use Asterisk::Manager; my $astman

[asterisk-users] Adding 4000 Lines to asteriskdb via asterisk -rx ?

sterisk ? I managed to Build a Perl DB_File Module for db1.8.5 but I do not have the know how how to use DB_file and db1 databases. Are there some external utilities to lock and update the asteriskdb ? Is there a better way ? Thanks Christoph -- Two hours of trial and error can save ten minut

Re: [asterisk-users] E1 crossover system

Dunno about your country but I have the same setup here in Germany. I have 1:1 Cables (I think). I have in zapata.conf: [channels] progzone=nl callgroup=1 pickupgroup=1 loadzone=nl,us defaultzone=nl context=zap-in priindication=inband switchtype=euroisdn ... group=1 context=telekom signalling=pr

Re: [Asterisk-Users] Change in audio file while listening to it

. Probably a thing which can be fixed in your dialplan by re-reading the file if it is empty. HTH, Christoph Marco Trucchi schrieb: Hello everybody, does anybody know how to handle the following problem? I update some gsm audio files every 10 minutes, by rewriting directly on them. I've no

[Asterisk-Users] access to caller/pickupgroup in extension.conf

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, Is it possible to get the callergroup or pickupgroup of a phone in the dialplan? So I can make decisions depending on the caller/pickupgroup. chris... -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (MingW32) Comment: Using GnuPG with Mozill

[Asterisk-Users] Help Getting Local Exchange Dialtone on PRI

t; _9149.,103,Playtones(busy) exten=> _9149.,104,Busy [alcatel] exten=> _X.,1,NoOp(Call from ${CALLERID} to ${EXTEN}) exten=> _X.,2,Dial(Zap/g1/${EXTEN}) exten=> _X.,3,Hangup exten=> _X.,103,Playtones(busy) exten=> _X.,104,Busy Thanks Christoph _

Re: [Asterisk-Users] Outgoing calls via Sipgate

ount... What's different here is that it's _0X., (see the dot)? That should make a little difference. Hope it helps! Christoph > > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list &g

Re: [Asterisk-Users] Cannot boot machine up after working on zaptel....

oad/kernel-2.6 and then uncomment the line that loads the module. You should then be able to boot "normally" and do what you have to do in order to get it to work. Does this also happen when you load the driver using modprobe? Christoph - -- GPG Key ID: 33D6AA8C AIM: zeitgeist2600 ICQ:

Re: [Asterisk-Users] 1.2 in production w/100+ phones?

On Wednesday 18 January 2006 18:22, Peder @ NetworkOblivion wrote: > Is anybody using 1.2 (or 1.2.1) in a production network using Realtime > (voicemail, sip or extensions) with 100+ SIP phones? If so, what are > your experiences? We've been running 1.0.3 for about a year and it's > been rock-sol

[Asterisk-Users] Where do I find *asterisk-capi*

Pls, where do I find asterisk-capi I am using now asterisk 1.2.1 with a SuSE 9.3 in SuSE 9.3 there was the old version for 1.0.6 ... can I use that old asterisk-capi for the current and on my system installed version 1.2.1 ??? thnx chris ___ --Bandwidt

[Asterisk-Users] Major Problems UTStarcom F1000 registering -- pls help

Hi there, I am trying desperatly to register my WiFi Phone UTStarcomm F1000 with my asterisk server. I already changed the name of the user to "anonymous" since it looks like the phone sends that name. The WiFi Phone's IP is 192.168.1.217, the asterisk server's IP is 192.168.1.200 What is it t

Re: [Asterisk-Users] calling to asterisk and listening to music (GSM) -->>Anyone, please?????

for me in the same: The music is disturbed. So I am sure it has to do with the mobile GSM-standard and not with asterisk. Christoph ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digiu

Re: [Asterisk-Users] digits won't play

It was indeed the problem with the "language 'de'" setting, setting the SIP client to US gives me the numbers. On Thursday 29 September 2005 12:00, Christoph Eicke wrote: > Hi! > > I have a strange problem. In an AGI I tell Asterisk to playback a number, > for ex

[Asterisk-Users] digits won't play

have looked inside of /var/lib/asterisk/sounds/digits and all files are present... does it have to do anything with the "language 'de'"? Where do I change that? Thanks, Christoph ___ --Bandwidth and Colocation sponsored by Easynews.

Re: [Asterisk-Users] Asterisk on windows

On Wednesday 28 September 2005 14:14, Kanishka Somaratne wrote: > why can't we compile the asterisk coading in windows, it's done in c++ so > it should work in windows as well oh, and did you try google? how about this: http://www.digium.com/index.php?menu=astwind it's a bit of a cheat though 'ca

Re: [Asterisk-Users] Asterisk on windows

On Wednesday 28 September 2005 14:14, Kanishka Somaratne wrote: > why can't we compile the asterisk coading in windows, it's done in c++ so it's written in C... have you bothered to look at the source code? ___ --Bandwidth and Colocation sponsored by Eas

Re: [Asterisk-Users] analogue phone with asterisk

An interesting read if you still have an analogue phone that does not speak DTMF tones (yet): http://www.geisterstunde.org/drupal/?q=w48_asterisk On Tuesday 27 September 2005 19:10, Rajesh Bhairampally wrote: > I am a newbee to asterisk. I recently installed [EMAIL PROTECTED] Everything > went we

Re: [Asterisk-Users] Which codec?

On Friday 23 September 2005 11:19, Dan Journo wrote: > Is there a guy somewhere on how much bandwidth each codec uses, along with > the advantages and disadvantages of each one? > Dan Journo calculate it yourself: http://www.cisco.com/en/US/tech/tk652/tk698/technologies_tech_note09186a0080094ae2.

[Asterisk-Users] IAX2 registration

ion and be able to make phone calls over that one? Thanks, Christoph ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNS

Re: [Asterisk-Users] Asterisk don't start

it looks to me like you haven't loaded the zaptel.o or ztdummy.o kernel modules... On Thursday 15 September 2005 16:10, [EMAIL PROTECTED] wrote: > Asterisk don't running, because show this message > > WARNING[6949]: chan_sip.c:8865 reload_config: Section 'authentication' > lacks type > > WARNING

Re: [Asterisk-Users] (no subject)

unless you show us some config files, I doubt that anybody can help you... On Wednesday 14 September 2005 16:46, Pablo Allietti wrote: > hi all, i have a box with a te110p and a pbx siemens... connect both > with a e1. > with a xten soft i can call extensions numbers in my office example 100 > 102

Re: [Asterisk-Users] call restrictions

you really should read about the concept of a "context" in extension.conf, that will answer your question and is also a basic key to understanding Asterisk. http://www.voip-info.org is your friend. Christoph On Wednesday 14 September 2005 10:47, Erdem HAKİ wrote: > Hello, > >

Re: [Asterisk-Users] 2 box single Asterisk

Here's my suggestion. Do a dialplan thing where when all trunks on boxA are busy, they are sent via IAX to boxB which sends them out via the ISDN trunks... this way boxA will be your primary box and boxB is your "spare" box that takes over if everything else is busy... On Tuesday 13 September 2

Re: [Asterisk-Users] Ethernet / TcpIp phones

try google for VoIP Phone ;-) or here: http://www.voip-info.org/tiki-index.php?page=Asterisk+phones On Wednesday 07 September 2005 11:19, Alex wrote: > Is there any VoIP phones available which can be plugged directly to the > Ethernet network? > > ___ >

Re: [Asterisk-Users] PHP and ASterisk Manager

the Asterisk source directory and it will give you the file that the string you are looking for is in. Then simply open the file, search for the string and look at the printf() statement. Christoph On Tuesday 06 September 2005 21:16, Anton Krall wrote: > I was able to do and if and while lo

[Asterisk-Users] strange problem

ot loading chan_alsa.so, only chan_oss.so as I think this might have something to do with the problem. Any help would be great, or any hints into a possible direction. Thanks, Christoph ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asteri

Re: [Asterisk-Users] unresolved symbol when loading ztdummy

On Wednesday 31 August 2005 11:11, Sergio Serrano wrote: > This option is under Library routines in your kernel configuration. > ah, yes. In kernel 2.6.* it is. not in 2.4.26 ;-) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users

Re: [Asterisk-Users] unresolved symbol when loading ztdummy

On Tuesday 30 August 2005 17:01, Braz wrote: > Your kernel has to be compile with CONFIG_CRC_CCITT=y or m. > I couldn't find that option in the kernel, but inserting the zaptel module before ztdummy works of course. ___ --Bandwidth and Colocation sponso

[Asterisk-Users] unresolved symbol when loading ztdummy

ardware. What is the problem that I'm having and how can I fix it? Thanks a lot, Christoph ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/l

Re: [Asterisk-Users] 1-800 number

On Friday 19 August 2005 09:33, Tzafrir Cohen wrote: > On Fri, Aug 19, 2005 at 09:15:02AM +0200, Christoph Eicke wrote: > > On Thursday 18 August 2005 22:27, Matt Hess wrote: > > > Just call a milliwatt..? > > > > you have a number? > > In your dialplan: &g

Re: [Asterisk-Users] 1-800 number

On Thursday 18 August 2005 22:27, Matt Hess wrote: > Just call a milliwatt..? you have a number? I'm also willing to pay my regular fees to my provider for those 3-4 minutes of testing. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http

Re: [Asterisk-Users] 1-800 number

t problem are you trying to solve with this? Just stepping out on a > limb but it sounds like you are trying to swat a fly with an F-16. > > -Jonathan > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Christoph > Eicke > Sent:

Re: [Asterisk-Users] 1-800 number

On Wednesday 17 August 2005 10:45, Michael K. Rodriguez wrote: > More info I don't quiet understand your mail ;-) Do you want more info from me? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/aste

Re: [Asterisk-Users] Voicemail Retrival

ets directed to their voicemail where they have some options. I hope this helps, Christoph ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options vi

[Asterisk-Users] 1-800 number

Hi! I'm searching for a 1-800 number that simply plays music for a long time (>3mins) and no one picks up. I've bothered the AT&T lines so far when trying out my SIP->PSTN connection but then always someone answered :-) Anyone have

[Asterisk-Users] Nikotel issues

low=all allow=alaw allow=ulaw allow=gsm extension.conf: [nikotel-incoming] exten => 3740525,1,NoOp(Invoming call via nikotel-us) exten => 3740525,2,Dial(IAX2/christoph&SIP/30&${CONSOLE},30) exten => 3740525,3,VoiceMail(u30) exten

Re: [Asterisk-Users] problem with sound device

ions like KDE or XMMS block the sound card, even after these are turned off. It then takes a while for the soundbard to become available again. Christoph ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman

Re: [Asterisk-Users] permission denied when monitoring channel OSS/dsp

On Monday 15 August 2005 11:11, Christoph Eicke wrote: > Hi! > > When I want to monitor the OSS/dsp channel through the Asterisk management > interface, I get a "permission denied" error: > > Action: Monitor > Monitor: OSS/dsp > File: 1124096949 > Mix: 1 &g

[Asterisk-Users] permission denied when monitoring channel OSS/dsp

3 10:08 vm -> /var/spool/asterisk/voicemail/default drwxr-xr-x 3 asterisk asterisk 4096 Jul 4 16:53 voicemail So, there shouldn't be any problems writing to disk. Anything else that I need to take into account, especially something special about monitoring the OSS/dsp channel?

Re: [Asterisk-Users] OT: Sendmail question

al host, don't start sendmail as a server daemon (shouldn't bind to port 25) and do a "nc -lt -p 25" and look what it's trying to do when sending mail via the command line. Christoph ___ Asterisk-Users mailing list Asterisk-Users@lis

Re: [Asterisk-Users] Billion BRI PCI card

On Friday 12 August 2005 09:43, John Fawcett wrote: > when using in NT mode does the card require additional power or is it > able to supply enough power by itself to the S0 bus? I don't know the exact specifics about the Billion card, but I have a setup where I have an extra NTBA connected to th

[Asterisk-Users] txgain for SIP?

g and also with different model headsets, so my last hope is txgain for SIP. Thanks Christoph ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options

Re: [Asterisk-Users] CLI and Dial

On Wednesday 10 August 2005 16:43, Esben Stien wrote: > Moises Silva <[EMAIL PROTECTED]> writes: > > make sure you have the next line in /etc/asterisk/modules.conf > > load => app_dial.so > > Not only that, but be sure to have a sound system loaded in the > modules.conf files. there were actually

[Asterisk-Users] CLI and Dial

istuff specific thing or is my 1.08 installation simply lacking features? Thanks, Christoph ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options

Re: Abwesenheitsnotiz: [Asterisk-Users] Nortel Option 11 and TE110P o f Digium

On Friday 05 August 2005 19:29, Alvaro Parres wrote: > ??? i dont understand. > it's an out-of-office reply until august 9th > On 8/5/05, Siegel, Joerg <[EMAIL PROTECTED]> wrote: > > Ich bin am 9.8. wieder im Hause! > > > > Mit freundlichen Grüßen, > > > > Jörg Siegel. > >

[Asterisk-Users] USB ISDN devices

Has anyone had any luck getting USB ISDN devices to work with Asterisk? I have bought a DayTrek miniVigor 128 and would like to get it to work with CAPI or mISDN. Has anyone every successfully done something like this? Thanks, Christoph ___ Asterisk

Re: [Asterisk-Users] Is there a right place for a include_once statement in a PHP AGI script?

t the place where it's necessary for that code to be executed. Christoph ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] announce to caller in queues (asterisk for art!)

I have created a phone system for a rather unorthodox purpose. We are making a 30 minute show that consists of callers calling in and giving 20-30 second reports from around their city. The calls are automatically answered by FireFly from an asterisk queue and the sound goes straight out

[Asterisk-Users] ABI manager - redirect

quot; tone and then connects back to it's original extension which starts over again. Both, the extension that the SIP/8080 channel is connected to and where it should be redirected to are in the same context ("local"). Any hints would be great, Thanks, Christoph _

Re: [Asterisk-Users] Asterisk Configuration

On Monday 25 July 2005 14:07, Afzaal Mirza wrote: > I am new to this mailing list. Can someone send me a guide or steps to > configure Asterisk on Linux box? I will highly appreciate. > http://www.voip-info.org ___ Asterisk-Users mailing list Asterisk-Us

Re: [Asterisk-Users] VoiceMailMain issue..

WARNING[1117207472]: pbx.c:1274 pbx_extension_helper: No > application 'VoiceMailMain ' for extension (home, 22999, 1) > > Anybody knows why? Have you checked /usr/lib/asterisk/modules/ and made sure that app_voicemail.so is there? Christoph __

Re: [Asterisk-Users] Problems installing asterisk-addons

n and the Asterisk version coming with the bristuff package has no support for the realtime extension yet. Christoph ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE

Re: [Asterisk-Users] a ne pas voir

On Thursday 21 July 2005 12:41, ali kia wrote: > hi all > i suggest to create a goup in hotmail in order to discuss any problem on > line in msn i think it's more practical than e-mail group > also I would prefer not to switch to something M$ based... __

Re: [Asterisk-Users] Mahler's Book - New Project

ble to get a paper copy *quickly*? > I saw earlier today that Paul Mahler's own company sells the book on their website as an ebook... so you could just print it out. BUT... I also found the book very bad. No clear structure, concepts are not really explained and it lacks in d

Re: [Asterisk-Users] bristuff patches and realtime mysql

n the wiki for RealTime states that HEAD is > required. There is no RealTime in STABLE. I'm confused now... you're saying I should make sure that I should use asterisk-addons from 1.0.8 but then at the same time there is no RealTime support

Re: [Asterisk-Users] bristuff patches and realtime mysql

pp_addon_sql_mysql.c:164: error: for each function it appears in.) make: *** [app_addon_sql_mysql.o] Error 1 It would be really great if you could get it to work as I think it's an excellent addition for Asterisk. If you need more info just write me a message, Christoph __

[Asterisk-Users] bristuff patches and realtime mysql

Asterisk crashes. Anyone using res_config_mysql.so with the bristuff package and can help me? Thanks, Christoph ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or

Re: [Asterisk-Users] res_config_mysql.so in CVS asterisk-addons broken?

ubject line ;-) After thinking about it at home I realized that maybe I should not only get the asterisk-addons from CVS but also the main Asterisk program (I only used a binary before). After recompiling Asterisk, compiling Asterisk-addons now also worked :-) Thanks, Christoph > __

[Asterisk-Users] res_config_mysql.so in CVS asterisk-addons broken?

Hi! I would like to use the realtime extension of Asterisk and got the latest asterisk-addons from CVS. Upon compiling things, I got a couple of error messages from app_addon_mysql... is it me, or are the files in the CVS broken? Thanks, Christoph

[Asterisk-Users] Incoming SIP calls with no extension

to "play" with the configuration, and I am afraid that a caller has to use some extension to call in from the BRI interface when I replace the pbx, just like now with the software sip phone. Thanks, Christoph ___ Asterisk-Users mailing list A

[Asterisk-Users] X100P Clone any hints for recognizing RINGing ?

different. It works in 1 of 10 cases that Asterisk recognizes the ringing on the phone :-) Where could I change which parameters to make Asterisk a little more "sensible" ? Please help and do not tell me to buy another card - I do understand what I bought, but now I already ha

[Asterisk-Users] Username containing an "@"

Hi, I have a problem to register with my provider, because my username is @. Thus my registry line contains a double @ sign and everything is parsed incorrectly. How can I quote the username to ignore the first @ ? cu chrisb. ___ Asterisk-Users mailing

[Asterisk-Users] ISDN to SIP

27; -- CAPI Hangingup -- removed pipe for PLCI = 0x101 Here is my sip.conf: [general] bindaddr = 0.0.0.0 port = 5060 context = default maxexpirey = 3600 defaultexpirey = 120 srvlookup = yes tos = 0x18 disallow = all allow = gsm allow = alaw allow = ulaw allow = g729 register => christoph:[EMAIL PROT

Re: [Asterisk-Users] HP ProLiant server for Asterisk

80s. You can even resize Logical Drives of the RAID (e.g. when putting in some new harddisc) without shutting down the server! Good Luck! Christoph ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listi

Re: [Asterisk-Users] Asterisk calls without soft phones

i Chawki, yes you can. You will habe to create the appropriate Extension to Dial out and you will have to have a look at the oss.conf or alsa.conf (and that the oss- or alsa-Module is loaded) After enabling the local Soundcard this way you can simply type "dial " on your local console an

RE: [Asterisk-Users] please Can some bady help me ???

o linux ? Try typing which make (without "-") and the command you should have to type to compile is make; make install (not ":") But I am not sure if there are development tools on Novell Linux Desktop ready to install... Christoph ___

Re: [Asterisk-Users] SS7 for *

e SS7 for ? I know that it is the protocol used by telecommunication companies for data exchange between their switches etc., but what do you need it for ? Christoph ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman

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