similar to: TE410P in Germany

Hello ml, i need some help on my zaptel configuration. My E100P only shows some YELLOW / RED alarm when I load the wct1xxp module and do a cat /proc/zaptel/1 Span 1: WCT1/0 "Digium Wildcard E100P E1/PRA Card 0" HDB3/CCS YELLOW RED ... .. . My /etc/zaptel.conf is: span=1,1,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 loadzone=nl defaultzone=nl I tried zaptel-1.0RC2 and the latest CVS

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Guenther Boelter <gboelter at gmail.com> schrieb: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA256 > > On 05/31/2015 02:31 PM, Luca Bertoncello wrote: > > Hi list! > > > > Now all works as expected, at least in the simulation I did with > > AsteriskNOW. Hopefully it will work later, when Deutsche Telekom

Hi all, * was running ... I have a WT405P and an AVM C4 with chan_capi 0.3.4a Today chan_capi stopped working, without any changings at the system. It seems, that not * is the reason, because isdn-log also shows no calls. If I try to call * from outside via capi, I only get a busy. That is the try from inside to outside: stern01*CLI> -- data = @89930:0107901723168212 -- capi

scaling<-4 xywidth<-480 resolution<-150 png(filename = "c:/r/anschluss/plots/4.png", width = xywidth*scaling, height = xywidth*scaling,pointsize = 12, bg = "white", res = resolution*scaling) ...... barplot(xrow,col = barcolors,cex.axis=scaling, ylab="mean time till attachment in sec",cex.lab=1.2*scaling) I tried to scale the barplot but there is one

Hi all, I am trying to detect DTMF keys from a mobile when asterisk make an outgoing call to the mobile. The DTMF detection on incoming calls (also FROM mobiles) is working very well. The only problem is if asterisk called the phone... Nothing is detected. I am using a digium te205p with PMX/PSTN connection. Everything that I can find in forums are problems with dtmf detection on SIP. Any

Hello, we have a PRI (E1) to a carrier and a second one to a legacy PBX: DTAG ---pri---- * ------ Hicmo (PSTN) | | Sip and more Many normal inbound calls are direcly routed to the hicom. Outbound calls from the Hicom go through LCR and then to PSTN. Inbound faxes are working, but outbound faxes from hicom to pstn are

Hi to all, i saw that in chan_sip there is the possibility to let the * to take the number from the Remote-Party-ID header field on incoming calls from gateway. What about to let the * to generate the Remote-Party-ID on outgoing calls? this is is useful for us to let the users to have their outgoing number hidden but let our switch to get the correct record for accounting. I think that If i hide

Hello David, ----- Nachricht von David Pottage via dovecot <dovecot at dovecot.org> --------- Datum: Thu, 21 Feb 2019 13:58:14 +0000 Von: David Pottage via dovecot <dovecot at dovecot.org> Antwort an: David Pottage <david at chrestomanci.org> Betreff: Re: Virus scan + removal on a mdbox mail storage An: dovecot at dovecot.org [...] >> NO! My

Hello David, ----- Nachricht von David Pottage via dovecot <dovecot at dovecot.org> --------- Datum: Wed, 20 Feb 2019 14:56:51 +0000 Von: David Pottage via dovecot <dovecot at dovecot.org> Antwort an: David Pottage <david at chrestomanci.org> Betreff: Re: Virus scan + removal on a mdbox mail storage An: dovecot at dovecot.org > On 2019-02-20

<P>Has anyone managed to get Asterisk to work with Glophone/Voiceglo since this posting.</P> <P><A href="http://lists.digium.com/pipermail/asterisk-users/2004-February/036559.html">http://lists.digium.com/pipermail/asterisk-users/2004-February/036559.html</A></P> <P>I've tried copying the config in this listing with no success. </P>

hello list, thanks for the great piece of software :) now to my problem: i setup samba long time ago with Version 3.0.14a-Debian. Now i want to put my samba server into a fax gateway. i create a printcap entry like the following: fax:\ :lp=/dev/null:\ :sd=/var/spool/lpd/faxlp:\ :if=/usr/local/bin/sambafax:\ :sh:sf:mx#0: and add a referring entry stanza into smb.conf [fax]

Hi all, i am stuck with the configuration of asterisk - modules are loaded ( zaptel and wct4xxp ) - i have zaptel.conf configure, output of ztcfg -vv --- snip -- rapid:~# ztcfg -vv Zaptel Configuration ====================== SPAN 1: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 2: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 3: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet

Hola, I am trying to isolate the memory leak I suspect in a mailman installation ? I found: http://blogs.sun.com/sanjeevb/date/200506 It gives an error: god at irt-smtp-02:~ 9:21am 65 # ./memleak.d 10312 dtrace: failed to compile script ./memleak.d: line 3: probe description pid10312:libc.so.1:malloc:entry does not match any probes I am on SunOS 5.10 Generic_127112-07 i86pc i386 i86pc Are

Yes I can hear the DTMF keys. I ve tried 2 different phones and 3 different mobile network providers. Nothing. I played with the rx/txgain values from hearing nothing to too loud... I have no more ideas. Marc -----Urspr?ngliche Nachricht----- Von: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Im Auftrag von Steve Underwood Gesendet: Donnerstag, 4.

Hi all, I am having problems to get the SeaLevel IO card to work with CentOS distro. Basically the card is being recognised and shown by lspci BUT /proc/devices file is not updated with the new devices does anyone know why. ? Could someone tell me what/where infos are needed to get /proc/devices to be updated. The reason I require the /proc/devices infos is because I want to run a mknod

Hello, i just got asterisk up and working (without alsa). Looks really promising. Now i have some questions regarding hardware I have the following setup PSTN --- serveral ------------PBX (Asterisk) ------- digital phones PRI (S2M) Ports analog phones VoIP On the PSTN side i could use Eicon/DIVA

i am using asterisk 1.6.2 with mISDN and a passive AVM Fritz!card on CentOS 5.5. The only thing, i want to do is a call-redirection from an isdn-call to my mobile via sip-account. My extension conf is: general] static=yes writeprotect=no [globals] OUT_PORT=1 [ISDN] exten => 12345,1,Dial(SIP/012346737222 at sipprovider.local) If i call to the msn 12345, the SIP-call is going out, but after

Hi *, now for a long time i am trying to set the outgoing callerid, without luck. I am here in Germany, my asterisk has a pri interface connected to a PMX installed by Telekom. All telephone calls are preselected to EcoVoice. I am using asterisk 1.2.7.1, zaptel 1.2.5 and libpri 1.2.2. A week ago we tried with a device able to simulate a telephone system so send out a callerid, and that

Hello, I'm using Asterisk 1.2 with MySQL support. I use sip_buddies table for SIP clients definition. My problem is that I can not define CLIR. Sip.conf docs says that restrictid = yes hide caller identification. The problem is that definition of sip_buddies field named restrictid is char(1). I tried to set restrictid = y, = 1 - no results. I changed definition of the filed to restrictid

Hi list! Finally I got my Asterisk works with my two phones... It was a problem on my Firewall (for the phone of my wife) and on my Dialplan (for forwarding calls). Now all works as expected, at least in the simulation I did with AsteriskNOW. Hopefully it will work later, when Deutsche Telekom changes my ISDN to VoIP... Well, now I have some time to spend with "fooling"... My phone