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• Oracle anuncia resultados de Q3 FY10

  - by Paulo Folgado

  Oracle Reports GAAP EPS of $0.23, Non-GAAP EPS of $0.38New Software Licenses Up 13%, Applications New Licenses Up 21%Oracle Corporation today announced fiscal 2010 Q3 GAAP total revenues were up 17% to $6.4 billion, while non-GAAP total revenues were up 18% to $6.5 billion. Excluding the impact of Sun Microsystems, Inc., which Oracle acquired on January 26, 2010, GAAP total revenue grew 7%. GAAP new software license revenues were up 13% to $1.7 billion, and up 10% to $1.7 billion excluding Sun. GAAP software license updates and product support revenues were up 13% to $3.3 billion, while non-GAAP software license updates and product support revenues were up 12% to $3.3 billion. GAAP operating income was down 5% to $1.8 billion, and GAAP operating margin was 29%. Non-GAAP operating income was up 13% to $2.9 billion, and non-GAAP operating margin was 45%. GAAP net income was down 10% to $1.2 billion, while non-GAAP net income was up 9% to $1.9 billion. GAAP earnings per share were $0.23, down 11% compared to last year while non-GAAP earnings per share were up 9% to $0.38. GAAP operating cash flow on a trailing twelve-month basis was $8.2 billion. "Our solid top line growth, coupled with disciplined expense management, was key in generating $8.0 billion of free cash flow over the last twelve months," said Oracle CFO Jeff Epstein."The Sun integration is going even better than we expected," said Oracle President, Safra Catz. "We believe that Sun will make a significant contribution to our fourth quarter earnings per share as well as meet the profitability goals we set for next year.""Exadata is the fastest growing product in Oracle's history," said Oracle President, Charles Phillips. "Introduced a little over a year ago, the Exadata pipeline is now approaching $400 million with Q4 bookings forecast at nearly $100 million. This strengthens both sales growth and profitability in our Sun server and storage businesses.""Every quarter we grab huge chunks of market share from SAP," said Oracle CEO, Larry Ellison. "SAP's most recent quarter was the best quarter of their year, only down 15%, while Oracle's application sales were up 21%. But SAP is well ahead of us in the number of CEOs for this year, announcing their third and fourth, while we only had one."In addition, Oracle's Board of Directors declared a cash dividend of $0.05 per share of outstanding common stock to be paid to stockholders of record as of the close of business on April 14, 2010, with a payment date of May 5, 2010. Future declarations of quarterly dividends and the establishment of future record and payment dates are subject to the final determination of Oracle's Board of Directors.Q3 Earnings Conference Call and WebcastOracle will hold a conference call and web broadcast today to discuss these results at 2:00 p.m. Pacific. You may listen to the call by dialing (800) 214-0694 or (719) 955-1425, Passcode: 567035. To access the live Web broadcast of this event, please visit the Oracle Investor Relations Web site at http://www.oracle.com/investor.

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• Silverlight Cream for June 15, 2010 -- #882

  - by Dave Campbell

  In this Issue: Colin Eberhardt Zoltan Arvai, Marcel du Preez, Mark Tucker, John Papa, Phil Middlemiss, Andy Beaulieu, and Chad Campbell. From SilverlightCream.com: Throttling Silverlight Mouse Events to Keep the UI Responsive Colin Eberhardt sent me this link to his latest at Scott Logic... about how to throttle Silverlight -- no not that, you'd have to go to one of the *other* blogs for that :) ... this is throttling the mouse, particularly the mouse wheel to keep the UI from freezing up ... check out the demos, you'll want to read the code Data Driven Applications with MVVM Part I: The Basics Zoltan Arvai started a series of tutorials on Data-Driven Applications with MVVM at SilverlightShow... this is number 1, and it looks like it's going to be a good series to read. Red-To-Green scale using an IValueConverter Marcel du Preez has an interesting post up at SilverlightShow using an IValueConverter to do a red/yellow/green progress bar ... this is pretty cool. Infragistics XamWebOutlookBar & Caliburn With assistance from Rob Eisenburg, Mark Tucker was able to build a Caliburn sample including the Infragistics XamWebOutlookBar, and he's sharing his experience (and code) with all of us. Printing Tip – Handling User Initiated Dialogs Exceptions John Papa responded to a common printing problem by writing it up in his blog. Note this problem quite often appears during debug, so check it out... John also has a quick tip on an update to the PrintAPI in Silverlight 4. Automatic Rectangle Radius X and Y Phil Middlemiss has another great Blend post up -- this one on rounding off buttons... they look great to me, but he's looking for advice -- how about that Phil? They look great to me :) WP7 Back Button in Games Planning on selling 'stuff' in the Windows Phone Marketplace? Are you familiar with the required use of the Back Button? How about in a game? ... Andy Beaulieu discusses all this and has some code you'll want to use. Windows Phone 7 – Call Phone Number from HyperlinkButton Chad Campbell [no relation :) ] is discussing dialing a number from a hyperlink in WP7 - oh yeah, it's a phone as well :) -- I think I've only seen a number attempt to be called -- hmm... and we're not yet either because we all have emulators, but this is a good intro to the functionality for when we may actually have devices! Stay in the 'Light! Twitter SilverlightNews | Twitter WynApse | WynApse.com | Tagged Posts | SilverlightCream Join me @ SilverlightCream | Phoenix Silverlight User Group Technorati Tags: Silverlight    Silverlight 3    Silverlight 4    Windows Phone MIX10

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• AsteriskNow Migration / Shared Extension Space

  - by Aaron C. de Bruyn

  I am testing the possibility of migrating from an old Avaya phone system to AsteriskNow. The migration would cover several hundred phones--but spread out over several years. (Management wants to move buildings to the new phone system one by one as cables get cut or time permits.) Two other directive is that extensions must not change and they want a GUI that other admins (non-Linux geeks) can manage. They currently use 9XXX for all extensions. We linked the Avaya and Asterisk box via PRI card and they both are communicating. From the Avaya side, if we move (for example) extension 9001 to Asterisk, we forward the call over the PRI to the AsteriskNow box and the SIP phone rings. In AsteriskNow we have an outgoing rule '_9XXX' that routes all 4-digit extensions starting with 9 back to Avaya. Here's the trouble. Dialing 9001 (the extension moved over to AsteriskNow) causes the call to be routed out the PRI to the Avaya box, then the Avaya box routes the call back to Asterisk, and Asterisk routes it to the SIP phone. As we get more and more users switched over, it will use up more and more channels over the PRI card. Is there a way I can ask Asterisk to check it's local extensions first--then forward off to the Avaya system if it starts with '_9XXX'? (I know how I can do it when editing the raw config files, I'm just looking for a way to do it in the GUI so other admins can manage it if necessary.) As a last-ditch plan, I know I can specifically add '_9001' as an outgoing call rule and sent it directly to extension 9001--but I'd really hate to do that for several hundred phones

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• Inbound SIP calls through Cisco 881 NAT hang up after a few seconds

  - by MasterRoot24

  I've recently moved to a Cisco 881 router for my WAN link. I was previously using a Cisco Linksys WAG320N as my modem/router/WiFi AP/NAT firewall. The WAG320N is now running in bridged mode, so it's simply acting as a modem with one of it's LAN ports connected to FE4 WAN on my Cisco 881. The Cisco 881 get's a DHCP provided IP from my ISP. My LAN is part of default Vlan 1 (192.168.1.0/24). General internet connectivity is working great, I've managed to setup static NAT rules for my HTTP/HTTPS/SMTP/etc. services which are running on my LAN. I don't know whether it's worth mentioning that I've opted to use NVI NAT (ip nat enable as opposed to the traditional ip nat outside/ip nat inside) setup. My reason for this is that NVI allows NAT loopback from my LAN to the WAN IP and back in to the necessary server on the LAN. I run an Asterisk 1.8 PBX on my LAN, which connects to a SIP provider on the internet. Both inbound and outbound calls through the old setup (WAG320N providing routing/NAT) worked fine. However, since moving to the Cisco 881, inbound calls drop after around 10 seconds, whereas outbound calls work fine. The following message is logged on my Asterisk PBX: [Dec 9 15:27:45] WARNING[27734]: chan_sip.c:3641 retrans_pkt: Retransmission timeout reached on transmission [email protected] for seqno 1 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6528ms with no response [Dec 9 15:27:45] WARNING[27734]: chan_sip.c:3670 retrans_pkt: Hanging up call [email protected] - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). (I know that this is quite a common issue - I've spend the best part of 2 days solid on this, trawling Google.) I've done as I am told and checked https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions. Referring to the section "Other SIP requests" in the page linked above, I believe that the hangup to be caused by the ACK from my SIP provider not being passed back through NAT to Asterisk on my PBX. I tried to ascertain this by dumping the packets on my WAN interface on the 881. I managed to obtain a PCAP dump of packets in/out of my WAN interface. Here's an example of an ACK being reveived by the router from my provider: 689 21.219999 193.x.x.x 188.x.x.x SIP 502 Request: ACK sip:[email protected] | However a SIP trace on the Asterisk server show's that there are no ACK's received in response to the 200 OK from my PBX: http://pastebin.com/wwHpLPPz In the past, I have been strongly advised to disable any sort of SIP ALGs on routers and/or firewalls and the many posts regarding this issue on the internet seem to support this. However, I believe on Cisco IOS, the config command to disable SIP ALG is no ip nat service sip udp port 5060 however, this doesn't appear to help the situation. To confirm that config setting is set: Router1#show running-config | include sip no ip nat service sip udp port 5060 Another interesting twist: for a short period of time, I tried another provider. Luckily, my trial account with them is still available, so I reverted my Asterisk config back to the revision before I integrated with my current provider. I then dialled in to the DDI associated with the trial trunk and the call didn't get hung up and I didn't get the error above! To me, this points at the provider, however I know, like all providers do, will say "There's no issues with our SIP proxies - it's your firewall." I'm tempted to agree with this, as this issue was not apparent with the old WAG320N router when it was doing the NAT'ing. I'm sure you'll want to see my running-config too: ! ! Last configuration change at 15:55:07 UTC Sun Dec 9 2012 by xxx version 15.2 no service pad service tcp-keepalives-in service tcp-keepalives-out service timestamps debug datetime msec localtime show-timezone service timestamps log datetime msec localtime show-timezone no service password-encryption service sequence-numbers ! hostname Router1 ! boot-start-marker boot-end-marker ! ! security authentication failure rate 10 log security passwords min-length 6 logging buffered 4096 logging console critical enable secret 4 xxx ! aaa new-model ! ! aaa authentication login local_auth local ! ! ! ! ! aaa session-id common ! memory-size iomem 10 ! crypto pki trustpoint TP-self-signed-xxx enrollment selfsigned subject-name cn=IOS-Self-Signed-Certificate-xxx revocation-check none rsakeypair TP-self-signed-xxx ! ! crypto pki certificate chain TP-self-signed-xxx certificate self-signed 01 quit no ip source-route no ip gratuitous-arps ip auth-proxy max-login-attempts 5 ip admission max-login-attempts 5 ! ! ! ! ! no ip bootp server ip domain name dmz.merlin.local ip domain list dmz.merlin.local ip domain list merlin.local ip name-server x.x.x.x ip inspect audit-trail ip inspect udp idle-time 1800 ip inspect dns-timeout 7 ip inspect tcp idle-time 14400 ip inspect name autosec_inspect ftp timeout 3600 ip inspect name autosec_inspect http timeout 3600 ip inspect name autosec_inspect rcmd timeout 3600 ip inspect name autosec_inspect realaudio timeout 3600 ip inspect name autosec_inspect smtp timeout 3600 ip inspect name autosec_inspect tftp timeout 30 ip inspect name autosec_inspect udp timeout 15 ip inspect name autosec_inspect tcp timeout 3600 ip cef login block-for 3 attempts 3 within 3 no ipv6 cef ! ! multilink bundle-name authenticated license udi pid CISCO881-SEC-K9 sn ! ! username xxx privilege 15 secret 4 xxx username xxx secret 4 xxx ! ! ! ! ! ip ssh time-out 60 ! ! ! ! ! ! ! ! ! interface FastEthernet0 no ip address ! interface FastEthernet1 no ip address ! interface FastEthernet2 no ip address ! interface FastEthernet3 switchport access vlan 2 no ip address ! interface FastEthernet4 ip address dhcp no ip redirects no ip unreachables no ip proxy-arp ip nat enable duplex auto speed auto ! interface Vlan1 ip address 192.168.1.1 255.255.255.0 no ip redirects no ip unreachables no ip proxy-arp ip nat enable ! interface Vlan2 ip address 192.168.0.2 255.255.255.0 ! ip forward-protocol nd ip http server ip http access-class 1 ip http authentication local ip http secure-server ip http timeout-policy idle 60 life 86400 requests 10000 ! ! no ip nat service sip udp port 5060 ip nat source list 1 interface FastEthernet4 overload ip nat source static tcp x.x.x.x 80 interface FastEthernet4 80 ip nat source static tcp x.x.x.x 443 interface FastEthernet4 443 ip nat source static tcp x.x.x.x 25 interface FastEthernet4 25 ip nat source static tcp x.x.x.x 587 interface FastEthernet4 587 ip nat source static tcp x.x.x.x 143 interface FastEthernet4 143 ip nat source static tcp x.x.x.x 993 interface FastEthernet4 993 ip nat source static tcp x.x.x.x 1723 interface FastEthernet4 1723 ! ! logging trap debugging logging facility local2 access-list 1 permit 192.168.1.0 0.0.0.255 access-list 1 permit 192.168.0.0 0.0.0.255 no cdp run ! ! ! ! control-plane ! ! banner motd Authorized Access only ! line con 0 login authentication local_auth length 0 transport output all line aux 0 exec-timeout 15 0 login authentication local_auth transport output all line vty 0 1 access-class 1 in logging synchronous login authentication local_auth length 0 transport preferred none transport input telnet transport output all line vty 2 4 access-class 1 in login authentication local_auth length 0 transport input ssh transport output all ! ! end ...and, if it's of any use, here's my Asterisk SIP config: [general] context=default ; Default context for calls allowoverlap=no ; Disable overlap dialing support. (Default is yes) udpbindaddr=0.0.0.0 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all) ; Optionally add a port number, 192.168.1.1:5062 (default is port 5060) tcpenable=no ; Enable server for incoming TCP connections (default is no) tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces) ; Optionally add a port number, 192.168.1.1:5062 (default is port 5060) srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; Note: Asterisk only uses the first host ; in SRV records ; Disabling DNS SRV lookups disables the ; ability to place SIP calls based on domain ; names to some other SIP users on the Internet ; Specifying a port in a SIP peer definition or ; when dialing outbound calls will supress SRV ; lookups for that peer or call. directmedia=no ; Don't allow direct RTP media between extensions (doesn't work through NAT) externhost=<MY DYNDNS HOSTNAME> ; Our external hostname to resolve to IP and be used in NAT'ed packets localnet=192.168.1.0/24 ; Define our local network so we know which packets need NAT'ing qualify=yes ; Qualify peers by default dtmfmode=rfc2833 ; Set the default DTMF mode disallow=all ; Disallow all codecs by default allow=ulaw ; Allow G.711 u-law allow=alaw ; Allow G.711 a-law ; ---------------------- ; SIP Trunk Registration ; ---------------------- ; Orbtalk register => <MY SIP PROVIDER USER NAME>:[email protected]/<MY DDI> ; Main Orbtalk number ; ---------- ; Trunks ; ---------- [orbtalk] ; Main Orbtalk trunk type=peer insecure=invite host=sipgw3.orbtalk.co.uk nat=yes username=<MY SIP PROVIDER USER NAME> defaultuser=<MY SIP PROVIDER USER NAME> fromuser=<MY SIP PROVIDER USER NAME> secret=xxx context=inbound I really don't know where to go with this. If anyone can help me find out why these calls are being dropped off, I'd be grateful if you could chime in! Please let me know if any further info is required.

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• RRAS VPN on windows 2k3 AD, can access rras server only.

  - by nopsax

  I'm setting up a test lab and here is the current configuration: 192.168.86.201 - a windows 2003 machine acting as PDC with AD/DNS/DHCP/WINS. 192.168.86.62 - windows 2003 machine is the RRAS server with IAS, also a file/print server. 192.168.86.6 - gateway/router to internet 192.168.86.21 - Windows XP Workstation Everything works on the internal network, File/Print/AD etc. Whenever a user connects via vpn to the RRAS server remotely using their domain credentials, they are assigned an ip address from the 192.168.86.201 machine along with the wins server address etc. The vpn user can then ping/access resources on the RRAS server, but cannot ping/access resources of any other machines by name or ip. However, if I ping by name, it does resolve to the correct ip address, just no replies. I did notice that on the RRAS server the 'internal' interface gets an ip address of 192.168.86.75 when a remote user connects, and the remote user is assigned, for example 192.168.86.71 . The RRAS server responds on both the .62 and .75 ip addresses. The client also unchecks the 'use remote default gateway option'. Also, I tried connecting a laptop to the physical network, joining the domain, then going remote and dialing the connection before domain login, and everything seems to work, e.g. browse-able shares via network neighborhood. But I can't really join the domain remotely if I cannot access any other resources. I really need to monitor traffic to see whats happening to those packets but won't be able to until this weekend. Any help is appreciated, will provide whatever configurations are needed.

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• VoIP setup for one external PSTN line

  - by Jcl

  I'm completely new to VoIP and the likes, and I'm trying to find information about what could be the best setup for this. I need 4 (maybe more in the future, but maximum 5 or 6) wireless extensions, connected to 1 PSTN line, and maybe 2 in the future. I've been trying to gather information about the gear needed but everything I find seems too much over-the-top (and extremely expensive). The main problem is that the physical place we are on doesn't have possibilities of having a decent internet connection, so using a external VoIP "virtual PBX" is not an option. Thing is, even if small, phone is critical to this organization. I currently have an analog DECT/GAP PBX which does what I need, however the PBX is very bad and the call quality is horrible, and that's why I want to change it. The requirements would be: 4 wireless terminals (routing cable is not an option), all of them ringing on incoming PSTN calls. Ability to do internal calls (4 separate offices) and ability to pass calls between terminals. The 4 terminals should be able to access the external PSTN line without dialing any special codes. Very important: terminals should be able to issue commands on the PSTN line to the external operator in the form *nn*nnnnnnnn# . Don't know wether this could face to be a problem, but I've had problems with analog PBX which would take any * as a PBX command and wouldn't allow terminals to send it to the external lines. Not so important, but would be nice to have: call waiting music Could anyone recommend such a setup? I need to be able to do this on a EXTREMELY LIMITED budget (that is: I don't have a limit, but all should get as much to zero as possible). I have enough spare powerful computers and a 300mbps wireless network which works just fine, so that's not to include in the budget. Don't really know if this is the best place to ask, but it's the most StackExchange-related site I've found to this subject.

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• How can I use my cell phone to establish a dial-up networking connection?

  - by gWiz

  I am using Windows 7 and have a BlackBerry with T-Mobile (U.S.). I have paired the phone with my computer over Bluetooth, which automatically creates a serial port for it. I am able to open the port in PuTTY and successfully issue AT commands to the modem, including dialing. However, while using Windows to create and establish a Dial-Up Networking connection, I get an error dialog stating "Error 678. The remote computer did not respond." In my testing, I also tried setting up a connection to dial a number connected to a phone. When attempting to connect over this connection, the phone does ring but the very moment I answer the call, my computer displays the above error dialog. What must be done to successfully establish such a PPP connection? Some special AT initialization string perhaps? To clarify, I'm not referring to the well-described and popular technique known as "tethering," in which the remote host of the data link is the mobile service provider. I am interested specifically in establishing direct data links with remote hosts other than my mobile service provider. Think old-school landline connection to your friend's computer or BBS. Edit 1 As grawity pointed out in comments, the missing piece of the puzzle is the actual modulator that is compatible with v-series protocols, which I expected to be built into the cellphone. So far the best only software alternative I could find is this experimental project. Edit 2 Found this forum discussion today. The participants state that there is no old-school modem in the BlackBerry. Edit 3 When I place a call in PuTTY with ATD, immediately after the call is answered (and the callee is initiating the handshake) the cellphone returns OK. This is not the expected behavior for establishing a data connection. The phone should reciprocate the handshake, and upon success return CONNECT. (Alternatively it should return BUSY or NO CARRIER, but never simply OK.) Windows DUN must be interpreting this as the "Error 678" I was seeing.

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• PSTN Trunk TDM400P Install on Asterisk / Trixbox

  - by Jona

  Hey All, I'm trying to get a TDM400P card with FXO module to connect to our PSTN line. The card is correctly detected by Linux: [trixbox1.localdomain asterisk]# lspci 00:09.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface I've run setup-pstn which produces the following output trixbox1.localdomain ~]# setup-pstn -------------------------------------------------------------- Detecting PSTN cards and USB PSTN Devices -------------------------------------------------------------- Hardware present! STOPPING ASTERISK Asterisk Stopped STOPPING FOP SERVER FOP Server Stopped Unloading DAHDI hardware modules: done Loading DAHDI hardware modules: wct4xxp: [ OK ] wcte12xp: [ OK ] wct1xxp: [ OK ] wcte11xp: [ OK ] wctdm24xxp: [ OK ] opvxa1200: [ OK ] wcfxo: [ OK ] wctdm: [ OK ] wcb4xxp: [ OK ] wctc4xxp: [ OK ] xpp_usb: [ OK ] Running dahdi_cfg: [ OK ] SETTING FILE PERMISSIONS Permissions OK STARTING ASTERISK Asterisk Started STARTING FOP SERVER FOP Server Started Chan Extension Context Language MOH Interpret Blocked State pseudo default en default In Service 1 from-pstn en default In Service dahdi_scan returns: dahdi_scan [1] active=yes alarms=OK description=Wildcard TDM400P REV I Board 5 name=WCTDM/4 manufacturer=Digium devicetype=Wildcard TDM400P REV I location=PCI Bus 00 Slot 10 basechan=1 totchans=4 irq=209 type=analog port=1,FXO port=2,none port=3,none port=4,none And asterisk can see the channel: > trixbox1*CLI> dahdi show channel 1 > Channel: 1LI> File Descriptor: 14 > Span: 11*CLI> Extension: I> Dialing: > noI> Context: from-pstn Caller ID: I> > Calling TON: 0 Caller ID name: > Mailbox: none Destroy: 0LI> InAlarm: > 1LI> Signalling Type: FXS Kewlstart > Radio: 0*CLI> Owner: <None> Real: > <None>> Callwait: <None> Threeway: > <None> Confno: -1LI> Propagated > Conference: -1 Real in conference: 0 > DSP: no1*CLI> Busy Detection: no TDD: > no1*CLI> Relax DTMF: no > Dialing/CallwaitCAS: 0/0 Default law: > ulaw Fax Handled: no Pulse phone: no > DND: no1*CLI> Echo Cancellation: > trixbox1128 taps trixbox1(unless TDM > bridged) currently OFF Actual > Confinfo: Num/0, Mode/0x0000 Actual > Confmute: No > Hookstate (FXS only): Onhook A cat of /etc/asterisk/dahdi.conf shows: [trixbox1.localdomain ~]# cat /etc/asterisk/dahdi-channels.conf ; Autogenerated by /usr/sbin/dahdi_genconf on Tue May 25 17:45:13 2010 ; If you edit this file and execute /usr/sbin/dahdi_genconf again, ; your manual changes will be LOST. ; Dahdi Channels Configurations (chan_dahdi.conf) ; ; This is not intended to be a complete chan_dahdi.conf. Rather, it is intended ; to be #include-d by /etc/chan_dahdi.conf that will include the global settings ; ; Span 1: WCTDM/4 "Wildcard TDM400P REV I Board 5" (MASTER) ;;; line="1 WCTDM/4/0 FXSKS (SWEC: MG2)" signalling=fxs_ks callerid=asreceived group=0 context=from-pstn channel => 1 callerid= group= context=default I have configured a "ZAP Trunk (DAHDI compatibility Mode)" with the ZAP identifier 1 and an outbound route, but when ever I try to make an external call via it I get the "All Circuits are busy now, please try your call again later message". I have one outbound route which uses the dial pattern 9|. and the Trunk Zap/1 and one Zap Trunk which uses Zap Identifier (trunk name): 1 and has no Dial Rules. The FXO module is directly connected to our phone line from BT via a BT-RJ11 cable. When running tail -f /var/log/asterisk/full and placing a call I get the following output: [May 26 11:10:52] VERBOSE[2723] logger.c: == Using SIP RTP TOS bits 184 [May 26 11:10:52] VERBOSE[2723] logger.c: == Using SIP RTP CoS mark 5 [May 26 11:10:52] VERBOSE[2723] logger.c: == Using SIP VRTP TOS bits 136 [May 26 11:10:52] VERBOSE[2723] logger.c: == Using SIP VRTP CoS mark 6 [May 26 11:10:52] WARNING[2661] pbx.c: FONALITY: This thread has already held the conlock, skip locking [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [901483890915@from-internal:1] Macro("SIP/801-b7ce8c28", "user-callerid,SKIPTTL,") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-user-callerid:1] Set("SIP/801-b7ce8c28", "AMPUSER=801") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-user-callerid:2] GotoIf("SIP/801-b7ce8c28", "0?report") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-user-callerid:3] ExecIf("SIP/801-b7ce8c28", "1?Set(REALCALLERIDNUM=801)") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-user-callerid:4] Set("SIP/801-b7ce8c28", "AMPUSER=801") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-user-callerid:5] Set("SIP/801-b7ce8c28", "AMPUSERCIDNAME=Jona") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-user-callerid:6] GotoIf("SIP/801-b7ce8c28", "0?report") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-user-callerid:7] Set("SIP/801-b7ce8c28", "AMPUSERCID=801") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-user-callerid:8] Set("SIP/801-b7ce8c28", "CALLERID(all)="Jona" <801>") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-user-callerid:9] Set("SIP/801-b7ce8c28", "REALCALLERIDNUM=801") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-user-callerid:10] ExecIf("SIP/801-b7ce8c28", "0?Set(CHANNEL(language)=)") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-user-callerid:11] GotoIf("SIP/801-b7ce8c28", "1?continue") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Goto (macro-user-callerid,s,20) [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-user-callerid:20] NoOp("SIP/801-b7ce8c28", "Using CallerID "Jona" <801>") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [901483890915@from-internal:2] Set("SIP/801-b7ce8c28", "_NODEST=") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [901483890915@from-internal:3] Macro("SIP/801-b7ce8c28", "record-enable,801,OUT,") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-record-enable:1] GotoIf("SIP/801-b7ce8c28", "1?check") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Goto (macro-record-enable,s,4) [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-record-enable:4] AGI("SIP/801-b7ce8c28", "recordingcheck,20100526-111052,1274868652.1") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck [May 26 11:10:52] VERBOSE[2858] logger.c: recordingcheck,20100526-111052,1274868652.1: Outbound recording not enabled [May 26 11:10:52] VERBOSE[2858] logger.c: -- <SIP/801-b7ce8c28>AGI Script recordingcheck completed, returning 0 [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-record-enable:5] MacroExit("SIP/801-b7ce8c28", "") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [901483890915@from-internal:4] Macro("SIP/801-b7ce8c28", "dialout-trunk,1,01483890915,") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:1] Set("SIP/801-b7ce8c28", "DIAL_TRUNK=1") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/801-b7ce8c28", "0?sub-pincheck,s,1") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/801-b7ce8c28", "0?disabletrunk,1") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:4] Set("SIP/801-b7ce8c28", "DIAL_NUMBER=01483890915") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:5] Set("SIP/801-b7ce8c28", "DIAL_TRUNK_OPTIONS=tr") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:6] Set("SIP/801-b7ce8c28", "OUTBOUND_GROUP=OUT_1") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/801-b7ce8c28", "1?nomax") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Goto (macro-dialout-trunk,s,9) [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/801-b7ce8c28", "0?skipoutcid") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:10] Set("SIP/801-b7ce8c28", "DIAL_TRUNK_OPTIONS=") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:11] Macro("SIP/801-b7ce8c28", "outbound-callerid,1") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/801-b7ce8c28", "0?Set(CALLERPRES()=)") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/801-b7ce8c28", "0?Set(REALCALLERIDNUM=801)") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/801-b7ce8c28", "1?normcid") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Goto (macro-outbound-callerid,s,6) [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-outbound-callerid:6] Set("SIP/801-b7ce8c28", "USEROUTCID=") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-outbound-callerid:7] Set("SIP/801-b7ce8c28", "EMERGENCYCID=") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-outbound-callerid:8] Set("SIP/801-b7ce8c28", "TRUNKOUTCID=") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/801-b7ce8c28", "1?trunkcid") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Goto (macro-outbound-callerid,s,12) [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/801-b7ce8c28", "0?Set(CALLERID(all)=)") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-outbound-callerid:13] ExecIf("SIP/801-b7ce8c28", "0?Set(CALLERID(all)=)") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/801-b7ce8c28", "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:12] ExecIf("SIP/801-b7ce8c28", "0?AGI(fixlocalprefix)") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:13] Set("SIP/801-b7ce8c28", "OUTNUM=01483890915") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:14] Set("SIP/801-b7ce8c28", "custom=DAHDI/1") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/801-b7ce8c28", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^))") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:16] Macro("SIP/801-b7ce8c28", "dialout-trunk-predial-hook,") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/801-b7ce8c28", "") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/801-b7ce8c28", "0?bypass,1") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/801-b7ce8c28", "0?customtrunk") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:19] Dial("SIP/801-b7ce8c28", "DAHDI/1/01483890915,300,") in new stack [May 26 11:10:52] WARNING[2858] app_dial.c: Unable to create channel of type 'DAHDI' (cause 0 - Unknown) [May 26 11:10:52] VERBOSE[2858] logger.c: == Everyone is busy/congested at this time (1:0/0/1) [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:20] Goto("SIP/801-b7ce8c28", "s-CHANUNAVAIL,1") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1) [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] GotoIf("SIP/801-b7ce8c28", "1?noreport") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,3) [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:3] NoOp("SIP/801-b7ce8c28", "TRUNK Dial failed due to CHANUNAVAIL (hangupcause: 0) - failing through to other trunks") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [901483890915@from-internal:5] Macro("SIP/801-b7ce8c28", "outisbusy,") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-outisbusy:1] Playback("SIP/801-b7ce8c28", "all-circuits-busy-now,noanswer") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- <SIP/801-b7ce8c28> Playing 'all-circuits-busy-now.ulaw' (language 'en') [May 26 11:10:54] VERBOSE[2858] logger.c: -- Executing [s@macro-outisbusy:2] Playback("SIP/801-b7ce8c28", "pls-try-call-later,noanswer") in new stack [May 26 11:10:54] VERBOSE[2858] logger.c: -- <SIP/801-b7ce8c28> Playing 'pls-try-call-later.ulaw' (language 'en') [May 26 11:10:54] WARNING[2661] pbx.c: FONALITY: This thread has already held the conlock, skip locking [May 26 11:10:54] VERBOSE[2858] logger.c: == Spawn extension (macro-outisbusy, s, 2) exited non-zero on 'SIP/801-b7ce8c28' in macro 'outisbusy' [May 26 11:10:54] VERBOSE[2858] logger.c: == Spawn extension (from-internal, 901483890915, 5) exited non-zero on 'SIP/801-b7ce8c28' [May 26 11:10:54] VERBOSE[2858] logger.c: -- Executing [h@from-internal:1] Macro("SIP/801-b7ce8c28", "hangupcall") in new stack [May 26 11:10:54] VERBOSE[2858] logger.c: -- Executing [s@macro-hangupcall:1] ResetCDR("SIP/801-b7ce8c28", "vw") in new stack [May 26 11:10:54] VERBOSE[2858] logger.c: -- Executing [s@macro-hangupcall:2] NoCDR("SIP/801-b7ce8c28", "") in new stack [May 26 11:10:54] VERBOSE[2858] logger.c: -- Executing [s@macro-hangupcall:3] GotoIf("SIP/801-b7ce8c28", "1?skiprg") in new stack [May 26 11:10:54] VERBOSE[2858] logger.c: -- Goto (macro-hangupcall,s,6) [May 26 11:10:55] VERBOSE[2858] logger.c: -- Executing [s@macro-hangupcall:6] GotoIf("SIP/801-b7ce8c28", "1?skipblkvm") in new stack [May 26 11:10:55] VERBOSE[2858] logger.c: -- Goto (macro-hangupcall,s,9) [May 26 11:10:55] VERBOSE[2858] logger.c: -- Executing [s@macro-hangupcall:9] GotoIf("SIP/801-b7ce8c28", "1?theend") in new stack [May 26 11:10:55] VERBOSE[2858] logger.c: -- Goto (macro-hangupcall,s,11) [May 26 11:10:55] VERBOSE[2858] logger.c: -- Executing [s@macro-hangupcall:11] Hangup("SIP/801-b7ce8c28", "") in new stack [May 26 11:10:55] VERBOSE[2858] logger.c: == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/801-b7ce8c28' in macro 'hangupcall' [May 26 11:10:55] VERBOSE[2858] logger.c: == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/801-b7ce8c28' I'm guessing I've missed a configuration step somewhere but no idea where, any help greatly appreciated.

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• stop android emulator call

  - by Shahzad Younis

  I am working on an Android application, having functionality like voicemail. I am using BroadcastReceiver to get dialing events. I have to get the event "WHEN CALL IS UNANSWERED (not picked after few rings) FROM RECEIVER". I will do some actions on caller end against this event. I am using AVD emulator, and I do call from one instance to another instance and it calls perfectly, but the problem is: It continuously calls until I reject or accept the call. This way I cannot detect that "CALL IS UNANSWERED AFTER A NUMBER OF RINGS". So I want the Caller emulator to drop the call after a number of rings (if unanswered) like a normal phone. I can do it (drop the call after some time) by writing some code, but I need the natural functionality of phone in the emulator. Can anyone please guide me? Is there any settings in the emulator? Or something else? The code is shown below in case it helps: public class MyPhoneReceiver extends BroadcastReceiver { @Override public void onReceive(Context context, Intent intent) { Bundle extras = intent.getExtras(); if (extras != null) { String state = "my call state = " + extras.getString(TelephonyManager.EXTRA_STATE); Log.w("DEBUG", state); } }

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• smart phone UI limitations

  - by Manny

  I would like to know, what limitations there are for how far one can go in terms of replacing UI components of current touch screen smart phones, in particular iPhone, Blackberry and android based phones. What I would like to do is create a custom UI for dialing out and incoming calls. I have some experience with Blackberry development. The theme builder for it, can be used to customize certain items on the incoming call screen, but it doesn't look like that you can increase the size of answer button. I know Blackberry also gives you access to all the phone APIs, but I'm not sure that you can create your own UI that can gain preference over the Blackberry incoming call screen. And if you try to customize the incoming call screen by adding any buttons to it, they would be rendered as pictures. I could possibly design a complete UI for android, since different manufactures have different UI for android based phones. Can I do what I want to do using iPhone, Blackberry or android? Or any other phone for that matter? I am guessing may be for Nokia phones using Qt, but I prefer the 3 platforms I listed. Thanks for all your help.

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• Lync Server 2010

  - by ManojDhobale

  Microsoft Lync Server 2010 communications software and its client software, such as Microsoft Lync 2010, enable your users to connect in new ways and to stay connected, regardless of their physical location. Lync 2010 and Lync Server 2010 bring together the different ways that people communicate in a single client interface, are deployed as a unified platform, and are administered through a single management infrastructure. Workload Description IM and presence Instant messaging (IM) and presence help your users find and communicate with one another efficiently and effectively. IM provides an instant messaging platform with conversation history, and supports public IM connectivity with users of public IM networks such as MSN/Windows Live, Yahoo!, and AOL. Presence establishes and displays a user’s personal availability and willingness to communicate through the use of common states such as Available or Busy. This rich presence information enables other users to immediately make effective communication choices. Conferencing Lync Server includes support for IM conferencing, audio conferencing, web conferencing, video conferencing, and application sharing, for both scheduled and impromptu meetings. All these meeting types are supported with a single client. Lync Server also supports dial-in conferencing so that users of public switched telephone network (PSTN) phones can participate in the audio portion of conferences. Conferences can seamlessly change and grow in real time. For example, a single conference can start as just instant messages between a few users, and escalate to an audio conference with desktop sharing and a larger audience instantly, easily, and without interrupting the conversation flow. Enterprise Voice Enterprise Voice is the Voice over Internet Protocol (VoIP) offering in Lync Server 2010. It delivers a voice option to enhance or replace traditional private branch exchange (PBX) systems. In addition to the complete telephony capabilities of an IP PBX, Enterprise Voice is integrated with rich presence, IM, collaboration, and meetings. Features such as call answer, hold, resume, transfer, forward and divert are supported directly, while personalized speed dialing keys are replaced by Contacts lists, and automatic intercom is replaced with IM. Enterprise Voice supports high availability through call admission control (CAC), branch office survivability, and extended options for data resiliency. Support for remote users You can provide full Lync Server functionality for users who are currently outside your organization’s firewalls by deploying servers called Edge Servers to provide a connection for these remote users. These remote users can connect to conferences by using a personal computer with Lync 2010 installed, the phone, or a web interface. Deploying Edge Servers also enables you to federate with partner or vendor organizations. A federated relationship enables your users to put federated users on their Contacts lists, exchange presence information and instant messages with these users, and invite them to audio calls, video calls, and conferences. Integration with other products Lync Server integrates with several other products to provide additional benefits to your users and administrators. Meeting tools are integrated into Outlook 2010 to enable organizers to schedule a meeting or start an impromptu conference with a single click and make it just as easy for attendees to join. Presence information is integrated into Outlook 2010 and SharePoint 2010. Exchange Unified Messaging (UM) provides several integration features. Users can see if they have new voice mail within Lync 2010. They can click a play button in the Outlook message to hear the audio voice mail, or view a transcription of the voice mail in the notification message. Simple deployment To help you plan and deploy your servers and clients, Lync Server provides the Microsoft Lync Server 2010, Planning Tool and the Topology Builder. Lync Server 2010, Planning Tool is a wizard that interactively asks you a series of questions about your organization, the Lync Server features you want to enable, and your capacity planning needs. Then, it creates a recommended deployment topology based on your answers, and produces several forms of output to aid your planning and installation. Topology Builder is an installation component of Lync Server 2010. You use Topology Builder to create, adjust and publish your planned topology. It also validates your topology before you begin server installations. When you install Lync Server on individual servers, the installation program deploys the server as directed in the topology. Simple management After you deploy Lync Server, it offers the following powerful and streamlined management tools: Active Directory for its user information, which eliminates the need for separate user and policy databases. Microsoft Lync Server 2010 Control Panel, a new web-based graphical user interface for administrators. With this web-based UI, Lync Server administrators can manage their systems from anywhere on the corporate network, without needing specialized management software installed on their computers. Lync Server Management Shell command-line management tool, which is based on the Windows PowerShell command-line interface. It provides a rich command set for administration of all aspects of the product, and enables Lync Server administrators to automate repetitive tasks using a familiar tool. While the IM and presence features are automatically installed in every Lync Server deployment, you can choose whether to deploy conferencing, Enterprise Voice, and remote user access, to tailor your deployment to your organization’s needs.

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• ADF Partner Community News Session - Open Invitation: "ADF as a basis of Fusion Apps - the biggest ADF project ever (in English)"

  - by Frank Nimphius

  After a successful guest performance of Ted Farrell in 2011, this year's international ADF speaker to speak during an ADF News session is Chris Muir from Oracle.  ADF News Session - Friday September 14, 8:30 AM - 9.00 AM (CET) - Topic: ADF as a basis of Fusion Apps - the biggest ADF project ever (in English) +++ this webcast will be conducted in English +++ dial-in numbers conc. ADF News Session, Sep. 14 2012 You are invited to join the next ADF News Session, that is going to take place September 14 2012 speaker:  Chris Muir / Oracle time:         8:30 AM (CET) duration:  30 minutes topic:        ADF as a basis of Fusion Apps - the biggest ADF project ever (in English) dial-in webconf: https://oraclemeetings.webex.com conf ID:      595 484 157 confkey:    123456 Please enter your name and an abbreviation of you company name when dialing in (please don´t use blanks and special characters). Please notice that this information will be visible to all participants of the webcast. Thank you. dial-in telco:           +49 (0)69 2222 16 106 or +49 (0)800 66 485 15           ConfCode: 208 503 9           SecurityPasscode: 112233  Other toll-free dial in numbers for EMEA countries are listed below (information is supplied without liability): Normal 0 false false false EN-US X-NONE X-NONE MicrosoftInternetExplorer4 /* Style Definitions */ table.MsoNormalTable {mso-style-name:"Table Normal"; mso-tstyle-rowband-size:0; mso-tstyle-colband-size:0; mso-style-noshow:yes; mso-style-priority:99; mso-style-qformat:yes; mso-style-parent:""; mso-padding-alt:0in 5.4pt 0in 5.4pt; mso-para-margin-top:0in; mso-para-margin-right:0in; mso-para-margin-bottom:10.0pt; mso-para-margin-left:0in; line-height:115%; mso-pagination:widow-orphan; font-size:11.0pt; font-family:"Calibri","sans-serif"; mso-ascii-font-family:Calibri; mso-ascii-theme-font:minor-latin; mso-fareast-font-family:"Times New Roman"; mso-fareast-theme-font:minor-fareast; mso-hansi-font-family:Calibri; mso-hansi-theme-font:minor-latin; mso-bidi-font-family:"Times New Roman"; mso-bidi-theme-font:minor-bidi;} table.MsoTableGrid {mso-style-name:"Table Grid"; mso-tstyle-rowband-size:0; mso-tstyle-colband-size:0; mso-style-priority:59; mso-style-unhide:no; border:solid windowtext 1.0pt; mso-border-alt:solid windowtext .5pt; mso-padding-alt:0in 5.4pt 0in 5.4pt; mso-border-insideh:.5pt solid windowtext; mso-border-insidev:.5pt solid windowtext; mso-para-margin:0in; mso-para-margin-bottom:.0001pt; mso-pagination:widow-orphan; font-size:11.0pt; font-family:"Calibri","sans-serif"; mso-ascii-font-family:Calibri; mso-ascii-theme-font:minor-latin; mso-hansi-font-family:Calibri; mso-hansi-theme-font:minor-latin; mso-bidi-font-family:"Times New Roman"; mso-bidi-theme-font:minor-bidi;} Austria 0800005967 Belgium 080048331 Croatia 0800222323 Czech Republic 800701080 Denmark  80889099 Estonia 8000111325 Egypt 08000000213 Finland 0800112073 France 0805632866 Greece 00800127897 Hungary 0680011201 Iceland 8008779 Ireland 1800932479 Israel 1809452571 Italy 800897629 Latvia 80002397 Luxembourg 80026598 Netherlands 08000235028 Norway 80010796 Poland 8001213557 Portugal 800814990 Romania 0800895563 Russia 81080029351012 Saudi Arabia 8008444320 Slovak Republic 0800001586 Slovenia 080080466 South Africa 0800980961 Spain 800098600 Sweden 856619465 Switzerland 0800650026 Turkey 00800 44632129 Ukraine 0800500166 United Arab Emirates 8000440344 United Kingdom 08006948154  

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• PPTP connection fails with errors 800/806

  - by Mark S. Rasmussen

  I've got a client (Server 2008 R2) that won't connect to our production environment PPTP VPN server (Server 2003, running RRAS). The server is behind a firewall that has TCP1723 open as well as GRE. Other clients at our office are able to connect just fine. Our office is behind a Juniper SSG5-Serial firewall, but all outgoing traffic is allowed, and multiple other clients are able to connect to VPN servers without issues. I've also setup a completely different VPN server on another network outside of our office. The functioning clients connect just fine - the Server 2008 R2 machine doesn't. Thus it's definitely a problem with this machine in particular. I've rebooted it. I've disabled the firewall, no dice on either. I've run PPTPSRV and PPTPCLNT on the server/client and they're able to communicate perfectly - indicating there's no problem using neither TCP1723 nor GRE. The Server 2008 R2 machine is also running as a VPN server itself (incoming connection) and that's working perfectly. We have the issues no matter if there are active incoming connections or not. I'm not sure what my next debugging step would be; any suggestions? EDIT: The event log on the server has the following warning from RasMan: A connection between the VPN server and the VPN client xxx.xxx.xxx.xxx has been established, but the VPN connection cannot be completed. The most common cause for this is that a firewall or router between the VPN server and the VPN client is not configured to allow Generic Routing Encapsulation (GRE) packets (protocol 47). Verify that the firewalls and routers between your VPN server and the Internet allow GRE packets. Make sure the firewalls and routers on the user's network are also configured to allow GRE packets. If the problem persists, have the user contact the Internet service provider (ISP) to determine whether the ISP might be blocking GRE packets. Obviously this points to GRE being a potential problem. But seeing as I have other clients connectiong without problems, as well as PPTPSRV and PPTPCLNT being able to communicate, I'm suspecting this might be a red herring. EDIT: Here are the anonymized events logged by the client in chronological order: CoId={742CB15C-A7E0-47B7-8240-0EFA1139CBD9}: The user XXX\YYY has started dialing a VPN connection using a per-user connection profile named ZZZ. The connection settings are: Dial-in User = XXX\YYY VpnStrategy = PPTP DataEncryption = Require PrerequisiteEntry = AutoLogon = No UseRasCredentials = Yes Authentication Type = CHAP/MS-CHAPv2 Ipv4DefaultGateway = No Ipv4AddressAssignment = By Server Ipv4DNSServerAssignment = By Server Ipv6DefaultGateway = Yes Ipv6AddressAssignment = By Server Ipv6DNSServerAssignment = By Server IpDnsFlags = Register primary domain suffix IpNBTEnabled = Yes UseFlags = Private Connection ConnectOnWinlogon = No. CoId={742CB15C-A7E0-47B7-8240-0EFA1139CBD9}: The user XXX\YYY is trying to establish a link to the Remote Access Server for the connection named ZZZ using the following device: Server address/Phone Number = XXX.YYY.ZZZ.KKK Device = WAN Miniport (PPTP) Port = VPN3-4 MediaType = VPN. CoId={742CB15C-A7E0-47B7-8240-0EFA1139CBD9}: The user XXX\YYY has successfully established a link to the Remote Access Server using the following device: Server address/Phone Number = XXX.YYY.ZZZ.KKK Device = WAN Miniport (PPTP) Port = VPN3-4 MediaType = VPN. CoId={742CB15C-A7E0-47B7-8240-0EFA1139CBD9}: The link to the Remote Access Server has been established by user XXX\YYY. CoId={742CB15C-A7E0-47B7-8240-0EFA1139CBD9}: The user XXX\YYY dialed a connection named ZZZ which has failed. The error code returned on failure is 806. Running Wireshark on the client shows it trying and retrying to send a "71 Configuration Request" While the server shows the incoming client requests, but apparently without replying: Given that this is GRE traffic, I think rules out the GRE traffic being blocked. Question is, why doesn't the server reply? This is the Configuration Request the server receives from the non functioning client (meaning no response is sent to the client request): And this is the Configuration Request the server receives from the working client: To me they seem identical, except for differing keys and magic numbers, and the fact that one client receives a response while the other doesn't.

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• Site-to-Site PPTP VPN connection between two Windows Server 2008 R2 servers

  - by steve_eyre

  We have two Windows Server 2008 R2 machines, one in our main office and one in a new office which we have just moved offsite. The main office has previously been handling client-to-server PPTP VPN connections. Now that we have moved our second server out of office, we want to set up a demand-dial or persistent VPN connection from the second server to the primary. Using a custom setting RRAS profile, we have successfully managed to set up a site-to-site VPN connection so that from the second server itself, it can access any of the devices in the main office and communicate back. However, any connected machines in the second office cannot use this connection, even when using the second server as gateway. The demand-dial interface is setup from the Second Server dialing into Main Server and a static route set up on RRAS for 192.168.0.0 with subnet mask 255.255.0.0 pointing down this network interface. The main office has the network of 192.168.0.0/16 (subnet mask 255.255.0.0). The second office has the network of 172.16.100.0/24 (subnet mask 255.255.255.0). What steps do we need to take to ensure traffic from the second office PCs going towards 192.168.x.x addresses use the VPN route? Many Thanks in advance for any help the community can offer. Debug Information Here is the route print output from the second server: =========================================================================== Interface List 23...........................Main Office 22...........................RAS (Dial In) Interface 16...e0 db 55 12 fa 02 ......Local Area Connection - Virtual Network 1...........................Software Loopback Interface 1 12...00 00 00 00 00 00 00 e0 Microsoft ISATAP Adapter 14...00 00 00 00 00 00 00 e0 Microsoft ISATAP Adapter #2 24...00 00 00 00 00 00 00 e0 Microsoft ISATAP Adapter #3 =========================================================================== IPv4 Route Table =========================================================================== Active Routes: Network Destination Netmask Gateway Interface Metric 0.0.0.0 0.0.0.0 172.16.100.250 172.16.100.222 261 127.0.0.0 255.0.0.0 On-link 127.0.0.1 306 127.0.0.1 255.255.255.255 On-link 127.0.0.1 306 127.255.255.255 255.255.255.255 On-link 127.0.0.1 306 <MAIN OFFICE IP> 255.255.255.255 172.16.100.250 172.16.100.222 6 172.16.100.0 255.255.255.0 On-link 172.16.100.222 261 172.16.100.113 255.255.255.255 On-link 172.16.100.113 306 172.16.100.222 255.255.255.255 On-link 172.16.100.222 261 172.16.100.223 255.255.255.255 On-link 172.16.100.222 261 172.16.100.224 255.255.255.255 On-link 172.16.100.222 261 172.16.100.225 255.255.255.255 On-link 172.16.100.222 261 172.16.100.226 255.255.255.255 On-link 172.16.100.222 261 172.16.100.227 255.255.255.255 On-link 172.16.100.222 261 172.16.100.228 255.255.255.255 On-link 172.16.100.222 261 172.16.100.229 255.255.255.255 On-link 172.16.100.222 261 172.16.100.230 255.255.255.255 On-link 172.16.100.222 261 172.16.100.255 255.255.255.255 On-link 172.16.100.222 261 192.168.0.0 255.255.0.0 192.168.101.87 192.168.101.17 266 192.168.101.17 255.255.255.255 On-link 192.168.101.17 266 224.0.0.0 240.0.0.0 On-link 127.0.0.1 306 224.0.0.0 240.0.0.0 On-link 172.16.100.222 261 224.0.0.0 240.0.0.0 On-link 172.16.100.113 306 224.0.0.0 240.0.0.0 On-link 192.168.101.17 266 255.255.255.255 255.255.255.255 On-link 127.0.0.1 306 255.255.255.255 255.255.255.255 On-link 172.16.100.222 261 255.255.255.255 255.255.255.255 On-link 172.16.100.113 306 255.255.255.255 255.255.255.255 On-link 192.168.101.17 266 =========================================================================== Persistent Routes: Network Address Netmask Gateway Address Metric 0.0.0.0 0.0.0.0 192.168.0.200 Default 0.0.0.0 0.0.0.0 172.16.100.250 Default =========================================================================== IPv6 Route Table =========================================================================== Active Routes: If Metric Network Destination Gateway 1 306 ::1/128 On-link 16 261 fe80::/64 On-link 16 261 fe80::edf4:85c6:3c15:dcbe/128 On-link 1 306 ff00::/8 On-link 16 261 ff00::/8 On-link 22 306 ff00::/8 On-link =========================================================================== Persistent Routes: None And here is the route print from one of the second office PCs: =========================================================================== Interface List 11...10 78 d2 32 53 27 ......Atheros AR8151 PCI-E Gigabit Ethernet Controller 1...........................Software Loopback Interface 1 12...00 00 00 00 00 00 00 e0 Microsoft ISATAP Adapter 13...00 00 00 00 00 00 00 e0 Teredo Tunneling Pseudo-Interface =========================================================================== IPv4 Route Table =========================================================================== Active Routes: Network Destination Netmask Gateway Interface Metric 0.0.0.0 0.0.0.0 172.16.100.250 172.16.100.103 10 127.0.0.0 255.0.0.0 On-link 127.0.0.1 306 127.0.0.1 255.255.255.255 On-link 127.0.0.1 306 127.255.255.255 255.255.255.255 On-link 127.0.0.1 306 172.16.100.0 255.255.255.0 On-link 172.16.100.103 266 172.16.100.103 255.255.255.255 On-link 172.16.100.103 266 172.16.100.255 255.255.255.255 On-link 172.16.100.103 266 224.0.0.0 240.0.0.0 On-link 127.0.0.1 306 224.0.0.0 240.0.0.0 On-link 172.16.100.103 266 255.255.255.255 255.255.255.255 On-link 127.0.0.1 306 255.255.255.255 255.255.255.255 On-link 172.16.100.103 266 =========================================================================== Persistent Routes: None IPv6 Route Table =========================================================================== Active Routes: If Metric Network Destination Gateway 1 306 ::1/128 On-link 11 266 fe80::/64 On-link 11 266 fe80::e973:de17:a045:aa78/128 On-link 1 306 ff00::/8 On-link 11 266 ff00::/8 On-link =========================================================================== Persistent Routes: None

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