similar to: e100p in norway?

hi I recently spoke to mr McNamara on IRC, and he mentioned there was a "far better way to do realtime-stuff than the usual realtime in asterisk, and that this was GPL". He failed, however, to ever mention how this could be done, so I just wonder if someone else might know... ? roy -- Roy Sigurd Karlsbakk roy@karlsbakk.net --- In space, loud sounds, like explosions, are even

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Hi, Has anyone connected * to a Colt E1 line in Europe? If so could you send me the zaptel.conf and zapata.conf. Thanks, Aaron __________________________________ Do you Yahoo!? New and Improved Yahoo! Mail - Send 10MB messages! http://promotions.yahoo.com/new_mail

I've suddenly started seeing a consistent problem with read.xls. No matter what xls file I try I always get an error message of this type: Error in xls2sep(xls, sheet, verbose = verbose, ..., method = method, : Intermediate file '/var/folders/cb/vvshkpm90lx_y2n69qlyw4z40000gn/T//RtmpK50r4g/file546a2722.csv' missing! In addition: Warning message: running command

I have a Cisco 7940 when you call in from outside and dial the Cisco phone extension I get this Read_channel ## vpb/1-3: Setting record mode, bridge = 0 WARNING[18451]: File chan_sip.c, Line 1111 (sip_write): Asked to transmit frame type 8, while native formats is 4 (read/write = 4/4) == Spawn extension (default, 1004, 1) exited non-zero on 'vpb/1-3' -- hangup on vpb (vpb/1-3)

Hi, I have been trying to get a Wildcard TE122 card running here the last couple of days. libpri and zaptel are all installed and configured to E1 specs. The jumper on the card is on, so configured for E1 (I'm in Norway). When running zttool, I get 'Alarms: RED' on the single card installed. Other information in zttool: Current alarms: Red Alarm Sync source: Internally clocked

Hi, I switched back to 2.2.27 with the same config that I am using for 2.2.32 and it work fine. Sep 11 11:49:37 imap-login: Info: Login: user=<viboge>, method=PLAIN, rip=88.89.118.45, lip=148.123.160.116, mpid=18709, TLS, session=<v7o22OZYrsdYWXYt> Sep 11 11:49:40 imap(mailuser) Session-ID v7o22OZYrsdYWXYt RemoteIP 88.89.118.45 Maildir /var/nextmail/nfs2.flex14/49/79/841 Info:

Hi, I am using Asterisk for a call center on a Dual Xeon machine.. I currently have 109 active channels 53 active calls Every body is complaining about quality and cpu is around 80% idle. Is there any tuning I can do??? Besides that, Asterisk normally goes down once or twice per day... Thank you Dov -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, Here is the gdb output. Arvid GNU gdb (GDB) Red Hat Enterprise Linux 7.6.1-94.el7 Copyright (C) 2013 Free Software Foundation, Inc. License GPLv3+: GNU GPL version 3 or later <http://gnu.org/licenses/gpl.html> This is free software: you are free to change and redistribute it. There is NO WARRANTY, to the extent permitted by law. Type "show copying" and "show

Hello all. I just received a TE110P and am trying to hook it to my Mitel 200SX has anyone successfully done this? My configuration is as follows. Asterisk -> TE110P ->Kentrox (csu/dsu) -> Mitel T1 Card. All I get is a blinking yellow on my TE110P card and an alarm on my Mitel. T1 card. Any advice would be great. Zaptel.conf span=1,0,1,d4,ami e&m=1-23 dchan=24

Hello, ?can i implement this as 10% censored data where t gives me failure and x censored. Thank you p=2;b=120 n=50 set.seed(132); r<-sample(1:50,45) t<-rweibull(r,shape=p,scale=b) t set.seed(123);? cens <- sample(1:50, 5)? x<-runif(cens,shape=p,scale=b)? x Chris Guure Researcher, Institute for Mathematical Research UPM

Hi, I enabled rawlog. Everything seems to be OK but the dovecot-log shows a fatal error and core dump. (created a core file) Arvid Rawlog in <<< STAT >>> <<< LIST >>> <<< UIDL 1 >>> <<< UIDL >>> <<< QUIT >>> Rawlog out <<< +OK Logged in. >>> <<< +OK 22 17589388

Hey.. Has anyone played around with Asterisk on the Itanium2, Opteron or Athlon64?? Can Asterisk (or Linux for that matter) actually make good use of a 64bit system?? Later..

Hi *, Does anyone have a lead on a Nagios plugin that speaks IAX or a small app to do so? I'm trying to set up remote monitoring for my Asterisk server and only IAX2 traffic is allowed through the firewall. Simply using check_udp to port 4569 yields no usable answer and Asterisk complains about receiving a midget packet or something like that :) jens

Hello, I've search through all of the lists and cannot find any descriptions of live monitoring (monitoring a phone call going on between an extension and a zaptel channel live from another extension while the monitoring phone is muted). I am aware of the monitor function which is actually a call recorder, but I'm looking for live monitoring from a muted extension. is this easily

Lets say I have an * at my business, with 7960 SIP phones. All the sip phones are registered using their extension number (like 305), but I would also like to put my SIP URI on my business card and in a name format, not an extension number (like lee.goodman), so that the SIP URI would read lee.goodman@asterisk.company.com. How would I set this up in extensions.conf? I got

We've changed E1 providers and I'm trying to reconfigure an E400P to make it work with the new lines. They're supposedly "standard" EuroISDN lines (in the UK). I'm initially just trying to get a single line up. I have the following in /etc/zaptel.conf: span=1,0,0,ccs,hdb3 bchan=1-15 dchan=16 bchan=17-31 loadzone=uk defaultzone=uk The LED on the back

Hi, It is quite easy. I use a client (thunderbird or windows live mail) set it up to use starttls on standard port 110. Logon and just do a list on the mailbox. The fatal error occurred when I logoff. Arvid -----Original Message----- From: dovecot [mailto:dovecot-bounces at dovecot.org] On Behalf Of dovecot-request at dovecot.org Sent: 11. september 2017 14:57 To: dovecot at dovecot.org

Hi, I have * set up as a PSTN->VoIP gateway (with an E1 with multiple numbers pointing to it). I'd really like to be able to dial out to a SIP server like so: exten => _X.,1,Dial(SIP/${DNID}@hostname) I.e. the remote SIP server receives a SIP INVITE with a "To:" header containing the dialed number (e.g. 02085555555@computer.company.com). This is equivalent to having a