similar to: Merging Data Sets with Full Outer Join

Hi all, I would like to use a loop for tasks that occurs repeatedly: # Groups # Umsatz <= 0: 1 (NICHT kaufend) # Umsatz > 0: 2 (kaufend) for (year in c("2011", "2012", "2013", "2014", "2015")) { paste0("Kunden$Kunde_real_", year) <- (paste0("Kunden$Umsatz_", year) <= 0) * 1 +

Hello all, i changed today from my local repository to a repository on a server. The repository now lies on the same server where my application is going to be deployed. Everything went fine create my repository on the server and checked out my working copy. After some changes on the code i wanted to deploy my application but it failed. The error is: "svn: Network connection closed

Hello all I have two Linux boxes. The first one is samba servers for the Windows clients. The second one is backup storage. So every few hours rsnc synchronises the data directories on the second machine. They are Suse Linux 9 Linux max 2.6.11.4-21.7-smp #1 SMP Thu Jun 2 14:23:14 UTC 2005 i686 i686 i386 GNU/Linux rsync version 2.6.3 protocol version 28 From time to time I get errors reported,

I'm trying to figure out if a file has changed since the last rsync call. I use the following command line: rsync -cvv /mnt/xxx/vol1/dbase/100/kunden.dbf /mnt/label | grep "^total: " | sed -e 's/.* data=//' This gives a 0 if the file is unchanged and the file size if the file has changed. Adding the "dry-run" option "n" to the command line always

Dear fellow list readers This is the situation: ISDN Devices => Patton ISDN to SIP GW => Asterisk PJSIP The Patton GW resides on a dynamic IP address, so I cannot really use match=ip in the identify section. The Patton does not send a line parameter. The ISDN Devices behind the patton have different MSN and should be able to send them in the From: Header, so the default endpoint

Hi Gang Server, two interfaces, routing to two different networks. Two transports defined, each bound to the corresponding ip assigned to the interface. But still, especially when an 183 message is sent, the Contact header does contain the wrong IP Address. Is this a known issue 13.18.3? Or is there a way to make absolutely sure the IP addresses within the Contact header is corresponding to

Hi Gang To increase security against phished passwords and similar attacks, we consider offering customers to define IP ranges (or GeoIP locations) from which their dynamic registrations are being accepted. I can already look at the source IP in the dial plan, so no issue with validate an INVITE against a source IP. But I would also like to prevent registrations from outside of this

Hi Joshua Thank you for your reply. Indeed, Ubuntu only ships with this old version. Upgraded to 16.2. via PPA. Problem persisted. Well, I already mentioned that this is a machine with two physical interfaces with different routes which on the 'external' side handles SIP customer registrations and has an 'internal' IC Trunk to a commercial Voice Switch via private IP Range. I

Hi Gang I gave up on running asterisk with two interfaces without it mixing up the ip addresses. So I have removed one transport definition from pjsip.conf Now * keeps complaining: res_pjsip.c:3461 ast_sip_set_tpselector_from_transport_name: Unable to retrieve PJSIP transport 'transport-name' I did a grep on /etc/asterisk for that transport name. It's in any file anymore.

Hi Richard > That could be possible and would be a bug in chan_sip. Ok, so I switched to PJSIP to see if this behaves differently So ip do a Transfer(PJSIP/${DESTNUMBER}@trunk) And this results in: Failed to parse destination URI '[destnumber scrubber]' for channel PJSIP/trunk-00000011 Do I have to specify the destination number differently when using Transfer with pjsip that I

Hi Joshua > The "rtp_keepalive" option can be used to have the RTP stack send an > RTP packet out. Try that and see what happens. Once again 'bullseye' that fixed the problem. Thank you! Mit freundlichen Gr?ssen -Beno?t Panizzon- -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29

Hi Well it is well some time that my last DUNDI peer has become unreachable. I guess too many issues with spoofed numbers etc. But I am wondering, do people, especially larger entities like telcos, still use DUNDI? I know that in some Hamradio communities, DUNDI is used to interconnect PBXes, but that is with private phone number ranges, not connected to the public. Want some DUNDI peering?

Hi Joshua I had a shot at your suggestion, bug still no success. I fear the 181 is sent before the macro is called. I want to change the Diversion Header in the 181 message sent back to the caller to put the number it contains in the correct e164 format (stripping the 0 and adding +41 for Switzerland) but just any 'dialplan set' value would do for an example :-) Could you please make

Hi Don, Hi to all readers, many thanks for all your answers and all your help. I adapted Don's code to my data and Don's code does the trick: str(Kunden01) for (year in 2011:2015) { Reeller_Kunde <- paste0("Reeller_Kunde_", year) Umsatz <- paste0("Umsatz_", year) cat('Creating', Reeller_Kunde,'from', Umsatz,'\n') Kunden01[[

Hello List I work at an SIP Provider and we have added and SBC in front of our Voice Switch to protect it. This requires all our SIP Trunk customers to register via a 'proxy'. I struggle with Asterisk to work over a proxy. This is what I have done so far. register => username at sip.example.com:password at sbc.example.com This works fine, asterisk is sending registrations via the

Hi List Implementing screening and routing I have stumbled over this issue: [pbx-router] exten => s,1,NoOp(ROUTER FROM: ${CALLERID(Number)} TO: ${DESTINATION}) same => n,Set(SOURCE=${CHANNEL(name)}) same => n,Set(PAI=${PJSIP_HEADER(read,P-Asserted-Identity)}) same => n,Set(FROM=${CALLERID(Number)}) same => n,Set(TO=${DESTINATION}) same

Hi Gang According to: https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Configuration_res_pjsip#Asterisk12Configuration_res_pjsip-endpoint_device_state_busy_at And endpoint should return busy if this number is reached. We have PBX Trunks registering to the Asterisk. So we want to limit the number of concurrent calls to a PBX and return busy, if more than the configured number of channels

Dear List I fear I stumbled over a bug in asterisk 13.14.1. My 'phones' are roaming around, sometimes some are connecting from ipv6 enabled networks, another time they are not. If a connection is ipv6 I would prefer to use ipv6 to avoid ipv4-nat problems. I have not specified a transport in the endpoint section, so that the appropriate transport which corresponds to the registration

Hello List Next question where google did not spit out an unsable answer. When redirecting a call with Transfer, I would like to correctly indicate the reason. I did try this: exten => XX,1,NoOp(Call to ${EXTEN} from ${CALLERID(all)}) exten => XX,n,Dial(SIP/ZZ) exten => XX,n,set(REDIRECTING(reason)=cfb) exten => XX,n,Transfer(SIP/YY) I did try with 'reason'

Hi, Version: prototype 1.6.0 I''m using the $F function to get the value of a field. It works fine in FireFox, but IE 6 and 7 are reporting "object does not support his property or method" (translated from german). In IE debugger it hightlights the line 3485: --- return Form.Element.Serializers[method](element); --- I searched the web, but this problem seems to be quite unique