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Hi, I am using caretNWS on a RHEL x86_64 system and I am getting an error message that is nearly identical to the one occuring in http://www.r-project.org/nosvn/R.check/r-release-macosx-ix86/caretNWS-00check.txt Error in socketConnection(serverHost, port = port, open = "a+b", blocking = TRUE) : unable to open connection Calls: system.time ... .local -> tryCatch -> tryCatchList

Hi all, Been reading about chan_motif / chan_xmpp in the wiki's for 1.8, 10 and 11 version of asterisk. In each example i got the impression that the asterisk server is registering on a XMPP server as a single user with the credentials as specified in jabber.conf. Instead of a single xmpp-user, could that also be multiple users? For instance, for each sip-user an xmpp-user? When i skim

I'm having a strange problem with selinux and the mounting of a nfs directory. I'm specifying the security context as part of the mount command, yet the security context still shows nfs. The mount shows what the security context should be: [root at clienthost ~]# mount serverhost:/usr/local on /usr/local type nfs4

Hi Everybody, I know that R has snow package which can be used for Parallel Computing. However, every time I try making a cluster, the only type of cluster I'm able to make is the "SOCK" (that too because I disabled the firewalls). For the rest (i.e. MPI, NWS, and PVM), I get error every time I try making one. I get the following errors

To connect to google voice with xmpp, I've had to turn on the "less secure apps" switch. > You recently changed your security settings so that your Google Account xxxxxxx at gmail.com is no longer protected by modern security standards. > > Please be aware that it is now easier for an attacker to break into your account. My xmpp.conf : type=client

Hello, I have a Google Voice phone number and want to connect it to my asterisk box to have calls handled to my SIP account. When I call the number I receive the correct INCOMING request on Jabber portion of asterisk, but the call is not connected to the gtalk part. JABBER: asterisk INCOMING: <iq from="+ 17174695631 at voice.google.com/srvres-MTAuMTc2LjEwOC4xNjo5ODQ4" to="

Hi, I am running tinc on alpine linux 2.7.8 in 2 seperate environments. The first environment is running for about a month without any problems. The second environment causes some trouble. It looks like a memory leak on the client side. tincd.conf: ConnectTo=ServerHost Device=/dev/net/tun Mode=switch Name=ClientHost PMTUDiscovery = yes DeviceType=tap PriorityInheritance = yes

I've included my jabber.conf below. I'm betting the following errors: [Jul 18 21:05:22] ERROR[28166]: res_jabber.c:609 aji_act_hook: JABBER: Node Error [Jul 18 21:05:22] WARNING[28166]: res_jabber.c:1537 aji_recv_loop: JABBER: Got hook event. jabber test [Jul 18 21:04:16] WARNING[32691]: res_jabber.c:1421 ast_aji_send: JABBER: Not connected can't send User: bferrell at gtalk.com

Hi, I've observed this strange behaviour for a while in my test environment. It looks like that all problems gone away when I switch to "hub-mode" instead of switch mode. Does tinc still work properly in switch mode when I transport vlan tagged traffic within that tunnel? In my environment the side, which is receiving arp requests from the wired interface, is running out of

Reimplementation of my earlier patch - more proper aliasing - at the suggestion of Mike -Paul -------------- next part -------------- diff -ur icecast/CVS/Entries IceCast/CVS/Entries --- icecast/CVS/Entries 2003-04-18 11:00:19.000000000 -0400 +++ IceCast/CVS/Entries 2003-04-17 22:14:16.000000000 -0400 @@ -1,4 +1,3 @@ -/.cvsignore/1.3/Wed Jan 15 05:36:15 2003// /AUTHORS/1.2/Fri Aug 9 15:55:01

...ith no response WARNING[1868]: chan_sip.c:4124 retrans_pkt: Timeout on 5YpLDUSIs6l3xbDXsurYTu.. on non-critical invite transaction. Looking up the ip addresses : whois 185.107.94.10 ............. inetnum: 185.107.94.0 - 185.107.94.255 netname: NFORCE_ENTERTAINMENT descr: Serverhosting .................. organisation: ORG-NE3-RIPE org-name: NForce Entertainment B.V. org-type: LIR address: Postbus 1142 address: 4700BC address: Roosendaal address: NETHERLANDS phone: +31206919299 ................... whois 215.45.145.211 ..............

My port forwarding changes require an authorization (authentication) context in channel_connect_to(). I'd like to change the dispatch_* functions so that they accept an Authctxt * instead of a void * (this parameter is already used this way). In addition, I'd have to pass the authctxt all the way down to channel_connect_to(). As a side effect, it's possible to get rid of the global

Hi, Sorry for my english. My problem: *************************************************** dspam-3.9.0 (dspam-3.10.2 all the time segmentation fault) dspam.conf .................... # DeliveryHost /var/run/dovecot/lmtp # same error as IP DeliveryHost 127.0.0.33 DeliveryPort 24 DeliveryProto LMTP .................... ServerHost 192.168.1.34 ServerPort

Hi, has anybody a running setup with 2 or more tinc daemons in switch mode which transport 8021q tagged traffic? I am trying to connect two segments with about 4 x 1000 mac addresses (distributed on different vlans). I am always running out of memory on one side. This happens only on the side where the arp requests come from. Currently there is no unicast traffic between the sides; only

Hi list, I'm trying to get channel gtalk working in asterisk 1.4.5 I have it built and configured as follows: *jabber.conf:* [general] debug=yes autoprune=no autoregister=no [myaccount] type=client serverhost=talk.google.com username=myaccount at gmail.com/Talk secret=mypassword port=5222 usetls=yes usesasl=yes statusmessage="Talk to me" timeout=100 *gtalk.conf:* [general]

OK, im pulling my hair out, everything looks configured right, deleted, and started over, etc, etc. but can't seem to get this to work Gtalk.conf [general] context=google-in allowguest=yes bindaddr=192.168.xxx.xxx extenip=96.254.xxx.xxx [guest] context=google-in disallow=all allow=ulaw allow=g729 connection=jp_jabber jabber.conf [general] debug=yes

I have installed gnutls and gnutls-devel from RedHat repositories [root at asterisk asterisk]# yum install gnutls gnutls-devel I have installed iksemel with gnutls support : [root at asterisk asterisk]# cd /usr/src/iksemel-1.3/ [root at asterisk asterisk]# ./configure --with-gnutls --prefix=/usr [root at asterisk asterisk]# make [root at asterisk asterisk]# make check [root at asterisk

When a caller calls my google voice phone number, I must answer, wait and press one to accept. Sometimes even that does not work. I have tried a few different things to get asterisk to place the call in an answered state and send the DTMF 1 with the Dial macro. I found Malcom Davenports wiki page regarding Google calling which has been very helpful in troubleshooting the issue.

Hi all, Iam using asterik 1.4.8 and connected to google talk. When iam calling from my google talk account to sip phone i can hear the voice (2 way). (this happens only within the LAN). when my friend tries to call my asterisk server (connects to the public ip) using his googletalk client it comes to my sip phone but either party cant hear a voice. I have fully allowd both tcp,udp on my

Hello everyone! I'm just trying to set up my new asterisk (version 1.6.2.8-rc1). I'd be very grateful, if someone could help me here. I'd be very glad, if one of you could test googletalk with me. Last time I tried (in 1.6.0.x times) it wouldn't work in the end. But here are my gtalk and jabber.conf files. Could you please take a look and tell me, if the settings same sane?