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• Where can I find WebSphere configuration files?

  - by Nicholas Key

  Hi there, I would like to know where are the WebSphere configuration details saved? Specifically, configuration details that are shown in the Administrative Console (from the web) or from the console using wsadmin. Some of the examples would be: Java and Process Management: Class loader, Process definition, Process execution Container Settings: Session management, SIP Container Settings, Web Container Settings, Portlet Container Settings Are there XML files that persist these configuration details? Nicholas

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• DNS entries for OCS 2007 R2 basic deploy

  - by Anero

  I'm doing a test deploy on a Lab with 3 VMs: TEST-DC: DC / DHCP / DNS / Root CA (Joined to TEST.AD Domain) TEST-CS: OCS Front End (Joined to TEST.AD Domain - IP: 10.0.128.1) TEST-EDGES: OCS Edge Server (Joined to Workgroup: EDGE-WKG - Internal IP: 10.0.128.3, External IPs: 192.168.129.12 - Access Edge Server, 192.168.129.13 - Web Conferencing, 192.168.129.14 - A/V) I can login with the Communicator Client from within computers in the domain (using [email protected]) and even the Automatic Sign-In works as expected. Nevertheless, I cannot login neither from within machines in the domain nor from outside the domain using [email protected]. I'm pretty sure it is a DNS related issue, so I'm including below a list of the entries. DNS Entries on TEST-DC: Forward Lookup Zones TEST.AD sip.test.ad (Host A). IP Address: 10.0.128.1 sipinternal.test.ad (Host A). IP Address: 10.0.128.1 sipexternal.test.ad (Host A). IP Address: 10.0.128.3 _sipinternaltls._tcp.test.ad (Service Location SRV). Port: 5061. Host: sipinternal.test.ad _sipinternal._tcp.test.ad (Service Location SRV). Port: 5061. Host: sipinternal.test.ad _sip._tcp.test.ad (Service Location SRV). Port: 5061. Host: sipexternal.test.ad _sipfederationtls._tcp.test.ad (Service Location SRV). Port: 5061. Host: sipexternal.test.ad _sip._tls.test.ad (Service Location SRV). Port: 443. Host: sipexternal.test.ad TEST.COM sip.test.com (Host A). IP Address: 10.0.128.1 sipinternal.test.com (Host A). IP Address: 10.0.128.1 sipexternal.test.com (Host A). IP Address: 10.0.128.3 _sipinternaltls._tcp.test.com (Service Location SRV). Port: 5061. Host: sipinternal.test.com _sipinternal._tcp.test.com (Service Location SRV). Port: 5061. Host: sipinternal.test.com _sip._tcp.test.com (Service Location SRV). Port: 5061. Host: sipexternal.test.com _sip._tls.test.ad (Service Location SRV). Port: 443. Host: sipexternal.test.ad Validation Errors OCS Front End Edge Server I ran the OCS 2007 Automatic Sign-In Troubleshooting and all DNS entries for both TEST.AD and TEST.COM are reported to be OK. What am I missing?

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• Asterisk Outgoing CDR Logging To Mysql

  - by user3295551

  Trying to utilize the cdr logging (to mysql) using custom fields. The problem I am facing is only when an outbound call is placed, during inbound calls the custom field I am able to log no problem. The reason I am having an issue is because the custom cdr field I need is a unique value for each user on the system. sip.conf ... ... [sales_department](!) type=friend host=dynamic context=SalesAgents disallow=all allow=ulaw allow=alaw qualify=yes qualifyfreq=30 ;; company sales agents: [11](sales_agent) secret=xxxxxx callerid="<...>" [12](sales_agent) secret=xxxxxx callerid="<...>" [13](sales_agent) secret=xxxxxx callerid="<...>" [14](sales_agent) secret=xxxxxx callerid="<...>" extensions.conf [SalesAgents] include => Services ; Outbound calls exten=>_1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@myprovider) ; Inbound calls exten=>100,1,NoOp() same => n,Set(CDR(agent_id)=11) same => n,CELGenUserEvent(Custom Event) same => n,Dial(${11_1},25) same => n,GotoIf($["${DIALSTATUS}" = "BUSY"]?busy:unavail) same => n(unavail),VoiceMail(11@asterisk) same => n,Hangup() same => n(busy),VoiceMail(11@asterisk) same => n,Hangup() exten=>101,1,NoOp() same => n,Set(CDR(agent_id)=12) same => n,CELGenUserEvent(Custom Event) same => n,Dial(${12_1},25) same => n,GotoIf($["${DIALSTATUS}" = "BUSY"]?busy:unavail) same => n(unavail),VoiceMail(12@asterisk) same => n,Hangup() same => n(busy),VoiceMail(12@asterisk) same => n,Hangup() ... ... For the inbound section of the dialplan in the above example I am able to insert the custom cdr field (agent_id). But above it you can see for the Oubound section of the dialplan I have been stumped on how I would be able to tell the dialplan which agent_id is making the outbound call. My Question: how to take the agent_id=[11] & agent_id=[12] and agent_id=[13] and agent_id=[14] etc and use that as a custom field for cdr on outbound calls?

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• NBX v3001 new phone account: all incoming calls directly diverted to voicemail

  - by Pitto

  That's my problem: I've added a new user in our NBX v3001 3com (now hp) and all the incoming calls are diverted, without ringing, to the voicemail. The phone is a Sip Endpoint Terminal and I really don't have a clue why this is happening since I've installed other identical phones without troubles. I've also tried deleting and re-adding the user and changing the extension: nothing. Any hints to solve?

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• Do something by operator dial specified number in Asterisk?

  - by Rev

  I want to make ability for Asterisk phone Operators to able do something like this: While operator talking to caller, if Operator dial specified number like 244 (or something like that but not Sip-Userid's), do something (like play sound for caller or etc) for that call. So, Is this possible? Is need to change dialplan? ¦¦¦¦¦ I found this. in first paragraph it's say someething like: if operator dial exten go voiceMail.

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• SuperSocket

  - by csharp-source.net

  SuperSocket is a light weight extensible socket application framework. You can use it to build a command based server side socket application (like FTP server, SMTP/POP3/IMAP4 server, SIP server, etc) easily without thinking about how to use socket, how to maintain the socket connections and how socket works(synchronize/asynchronize). It is a pure C# project which is designed to be extended, so it is easy to be integrated to your existing system. As long as your systems (like forum/CRM/MIS/HRM/ERP) are developed in .NET language, you must be able to use SuperSocket to build your socket application as a part of your current system perfectly.

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• JSR updates

  - by heathervc

  There were many JSR updates over the last week.  See below. JSR 258, Mobile User Interface Customization API, has published a Maintenance Release. JSR 257, Contactless Communication API, has published a Maintenance Release 2. JSR 180, SIP API for J2ME, has published a Final Release 5. JSR 269, Pluggable Annotation Processing API, has published a Maintenance Release. JSR 344, JavaServerTM Faces 2.2, has published an Early Draft Review.  The review closes on 8 December.

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• Asterisk not playing custom sounds on Ubuntu Server 11.04

  - by jochy2525

  I've installed Asterisk on my Ubuntu Server, all works fine, excepts playing the custom sounds. Asterisk sounds work, but this file I've uploaded does not play (on other servers it works, it is a .WAV PCM 16bit 8000). Here is some log output: [Feb 6 22:55:45] WARNING[11045] file.c: File custom/sohoitsoluciones does not exist in any format [Feb 6 22:55:45] WARNING[11045] file.c: Unable to open custom/sohoitsoluciones (format 0x4 (ulaw)): No such file or directory [Feb 6 22:55:45] WARNING[11045] app_playback.c: ast_streamfile failed on SIP/Out4903-0000001d for custom/sohoitsoluciones How can I get Asterisk to play a custom sound?

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• UIAlertView close the App

  - by Marco

  Hello Users, in my project my app first tries to connect to the internet, but now i have to check if there is an connection available! so i made an if, else within an UIAlertView in the else part! but how can i close the whole app on a click on the following button? UIAlertView *alert = [[UIAlertView alloc] initWithTitle:@"Keine Internetverbindung" message:@"Es konnte keine Verbindung zu www.sip.de hergestellt werden!" delegate:nil cancelButtonTitle:@"Schliessen" otherButtonTitles:nil]; thank you all for helping beforehand greets Marco

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• Where to find Java/Groovy talent in Atlanta

  - by Craig winnerman

  I am on a retained search for a company seeking a Java Developer with Groovy experiencs. They offer a web hosted PBX phone system to small business. This is a permanent position and haveing a tough time finding good talent. Other pluses are JUnit, Scrum,SIP,VXML CCXMLand Junit. Any suggestions, please contact me directly at 727-789-2297 or [email protected]

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• Py2exe, PyQt4 and Postgre Driver (QPSQL)

  - by Marshall

  Hi, I`m trying to freeze my application using Py2exe. My app uses PyQt4 and it apparently works fine with py2exe. But once I`ve uninstalled PyQt, it shows the following error: QSqlDatabase: QPSQL driver not loaded QSqlDatabase: available driver: QPSQL7 QPSQL Which doesn't make sense at all. If PyQt4 is still installed, it works just fine. This is my py2exe parameters: data_files = [ ('sqldrivers', [ 'C:\Python26\Lib\site-packages\PyQt4\plugins\sqldrivers\qsqlpsql4.dll' ]) ] setup(console=["delivery.py"], options={"py2exe" : {"includes" : ["sip", "PyQt4.QtSql", "PyQt4.QtWebKit", "PyQt4.QtNetwork"]}}, data_files=data_files)

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• Where can I find WebSphere configuration files?

  - by Nicholas Key

  Hello Stackoverflow'ers, I would like to know where are the WebSphere configuration details saved? Specifically, configuration details that are shown in the Administrative Console (from the web) or from the console using wsadmin. Some of the examples would be: Java and Process Management: Class loader, Process definition, Process execution Container Settings: Session management, SIP Container Settings, Web Container Settings, Portlet Container Settings Are there XML files that persist these configuration details? Nicholas

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• More efficient way of updating UI from Service than intents?

  - by Donal Rafferty

  I currently have a Service in Android that is a sample VOIP client so it listens out for SIP messages and if it recieves one it starts up an Activity screen with UI components. Then the following SIP messages determine what the Activity is to display on the screen. For example if its an incoming call it will display Answer or Reject or an outgoing call it will show a dialling screen. At the minute I use Intents to let the Activity know what state it should display. An example is as follows: Intent i = new Intent(); i.setAction(SIPEngine.SIP_TRYING_INTENT); i.putExtra("com.net.INCOMING", true); sendBroadcast(i); Intent x = new Intent(); x.setAction(CallManager.SIP_INCOMING_CALL_INTENT); sendBroadcast(x); Log.d("INTENT SENT", "INTENT SENT INCOMING CALL AFTER PROCESSINVITE"); So the activity will have a broadcast reciever registered for these intents and will switch its state according to the last intent it received. Sample code as follows: SipCallListener = new BroadcastReceiver(){ @Override public void onReceive(Context context, Intent intent) { String action = intent.getAction(); if(SIPEngine.SIP_RINGING_INTENT.equals(action)){ Log.d("cda ", "Got RINGING action SIPENGINE"); ringingSetup(); } if(CallManager.SIP_INCOMING_CALL_INTENT.equals(action)){ Log.d("cda ", "Got PHONE RINGING action"); incomingCallSetup(); } } }; IntentFilter filter = new IntentFilter(CallManager.SIP_INCOMING_CALL_INTENT); filter.addAction(CallManager.SIP_RINGING_CALL_INTENT); registerReceiver(SipCallListener, filter); This works however it seems like it is not very efficient, the Intents will get broadcast system wide and Intents having to fire for different states seems like it could become inefficient the more I have to include as well as adding complexity. So I was wondering if there is a different more efficient and cleaner way to do this? Is there a way to keep Intents broadcasting only inside an application? Would callbacks be a better idea? If so why and in what way should they be implemented?

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• Asterisk auto Call recording

  - by Manjoor

  We are running asterisk with 8 port FXO. FXO connects to our old PBX (Samsung Office Serv 100). Now we want to record all calls routed through FXO (if it was dialed to outside or comming from outside). Here is the diagram |------|--------------------------------- | |--------------24 Lines ---------- Other clasic Phones PRI------ | PBX |--------------------------------- | | | | | |-----------|---------| | |--8 lines--| |--------- | |-----------|Asterisk |---------- 50 SIP phone |------| | |---------- |---------|---------- Is there a simple way to do this?

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• Does anyone know of a good .Net VoIP library.

  - by Vaelen

  I have been searching for ages now for a good .Net based VoIP library. After having tried conaito and SIP.Net I have still haven't found anything that truly fits my needs. Basically all of these are constructed using ActiveX. Unfortunately ActiveX and WPF don't play well together. And ClickOnce Deployment doesn't register Interop COM components. I need to find a good, pure .Net managed code implementation.

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• i cant ping to my DMZ zone from the local inside PC

  - by Big Denzel

  HI everybody. Can anyone please help me on the following issue. I got a Cisco Asa 5520 configured at my network. I cant ping to my DMZ interface from a local inside network PC. so the only way a ping the DMZ is right from the Cisco ASA firewall, there i can pint to all 3 interfaces, Inside, Outside and DMZ,,,, But no PC from the Inside Network can access the DMZ. Can please any one help? I thank you all in advance Bellow is my Cisco ASA 5520 Firewall show run; ASA-FW# sh run : Saved : ASA Version 7.0(8) ! hostname ASA-FW enable password encrypted passwd encrypted names dns-guard ! interface GigabitEthernet0/0 description "Link-To-GW-Router" nameif outside security-level 0 ip address 41.223.156.109 255.255.255.248 ! interface GigabitEthernet0/1 description "Link-To-Local-LAN" nameif inside security-level 100 ip address 10.1.4.1 255.255.252.0 ! interface GigabitEthernet0/2 description "Link-To-DMZ" nameif dmz security-level 50 ip address 172.16.16.1 255.255.255.0 ! interface GigabitEthernet0/3 shutdown no nameif no security-level no ip address ! interface Management0/0 description "Local-Management-Interface" no nameif no security-level ip address 192.168.192.1 255.255.255.0 ! ftp mode passive access-list OUT-TO-DMZ extended permit tcp any host 41.223.156.107 eq smtp access-list OUT-TO-DMZ extended permit tcp any host 41.223.156.106 eq www access-list OUT-TO-DMZ extended permit icmp any any log access-list OUT-TO-DMZ extended deny ip any any access-list inside extended permit tcp any any eq pop3 access-list inside extended permit tcp any any eq smtp access-list inside extended permit tcp any any eq ssh access-list inside extended permit tcp any any eq telnet access-list inside extended permit tcp any any eq https access-list inside extended permit udp any any eq domain access-list inside extended permit tcp any any eq domain access-list inside extended permit tcp any any eq www access-list inside extended permit ip any any access-list inside extended permit icmp any any access-list dmz extended permit ip any any access-list dmz extended permit icmp any any access-list cap extended permit ip 10.1.4.0 255.255.252.0 172.16.16.0 255.255.25 5.0 access-list cap extended permit ip 172.16.16.0 255.255.255.0 10.1.4.0 255.255.25 2.0 no pager logging enable logging buffer-size 5000 logging monitor warnings logging trap warnings mtu outside 1500 mtu inside 1500 mtu dmz 1500 no failover asdm image disk0:/asdm-508.bin no asdm history enable arp timeout 14400 global (outside) 1 interface nat (inside) 1 0.0.0.0 0.0.0.0 static (dmz,outside) tcp 41.223.156.106 www 172.16.16.80 www netmask 255.255.255 .255 static (dmz,outside) tcp 41.223.156.107 smtp 172.16.16.25 smtp netmask 255.255.2 55.255 static (inside,dmz) 10.1.0.0 10.1.16.0 netmask 255.255.252.0 access-group OUT-TO-DMZ in interface outside access-group inside in interface inside access-group dmz in interface dmz route outside 0.0.0.0 0.0.0.0 41.223.156.108 1 timeout xlate 3:00:00 timeout conn 1:00:00 half-closed 0:10:00 udp 0:02:00 icmp 0:00:02 timeout sunrpc 0:10:00 h323 0:05:00 h225 1:00:00 mgcp 0:05:00 timeout mgcp-pat 0:05:00 sip 0:30:00 sip_media 0:02:00 timeout uauth 0:05:00 absolute http server enable http 10.1.4.0 255.255.252.0 inside no snmp-server location no snmp-server contact snmp-server enable traps snmp authentication linkup linkdown coldstart crypto ipsec security-association lifetime seconds 28800 crypto ipsec security-association lifetime kilobytes 4608000 telnet timeout 5 ssh timeout 5 console timeout 0 management-access inside ! ! match default-inspection-traffic ! ! policy-map global_policy class inspection_default inspect dns maximum-length 512 inspect ftp inspect h323 h225 inspect h323 ras inspect netbios inspect rsh inspect rtsp inspect skinny inspect esmtp inspect sqlnet inspect sunrpc inspect tftp inspect sip inspect xdmcp ! service-policy global_policy global Cryptochecksum: : end ASA-FW# Please Help. Big Denzel

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• Asterisk - Trying to use call files to create a conference call between two dynamic numbers

  - by Hank

  I'm trying to setup an Asterisk system that will allow me to create a conference call between two dynamic numbers. It seems I can use 'call files' to make Asterisk initiate the call without needing an incoming call - http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out This example seems to be what I'd need: Channel: SIP/mytrunk/12345678 MaxRetries: 2 RetryTime: 60 WaitTime: 30 Context: callme Extension: 800 Priority: 2 I can generate this file with some scripting language and then place it into the Asterisk Call File folder. The problem I'm having is: How do I call out to two numbers and join them in a conference call? The MeetMe plugin/extension seems to be what I need in terms of conference calling, I'm just unsure as to how I'd use the two together and join them. Also, is it possible to have multiple 2-person conference calls at the same time? Is setting this up as simple as setting aside X amount of 'channels' in the meetme.conf?

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• WRTU54G-TM T-Mobile@Home router with 3rd party firmware

  - by dlamblin

  I've been doing a lot of reading online about the Linksys WRTU54G-TM router model that I now own. It seems getting a custom firmware onto it is not a problem. But no one is talking about retaining the Voip features (yet). So far they're all disappointed that it's not a SIP machine and used GSM over IPSec. Personally I don't care about using it with non-t-mobile. If I take the original firmware, shouldn't I be able to extract it, and it's SquashFS image, and then move all of the t-mobile specific binaries for enabling the calling features over to a custom firmware installation (maybe OpenWRT)? You might ask why, and the reason is, that if I do this I could retain my calling features, which I do want, and ssh to the router and use it to run additional software, as any OpenWRT router could do. Does anyone know if this can be done, and how the firmware's binaries could be gotten at and installed correctly?

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• How to use nsupdate to create NAPTR record

  - by Jon Skarpeteig

  What's a working example for creating a NAPTR record using nsupdate against Bind9? man nsupdate sais: update add {domain-name} {ttl} [class] {type} {data...} Adds a new resource record with the specified ttl, class and data. But I can't seem to find the correct format for NAPTR My attempt: echo -e 'update add enum.example.com 60 IN NAPTR 1.1.1.1.1."u"."E2U+sip"."!^.*[email protected]!" .'"\nsend"|nsupdate results in: invalid rdata format: not a valid number syntax error

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• Asterisk - Trying to use call files to create a conference call between two dynamic numbers

  - by Hank

  I'm trying to setup an Asterisk system that will allow me to create a conference call between two dynamic numbers. It seems I can use 'call files' to make Asterisk initiate the call without needing an incoming call - http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out This example seems to be what I'd need: Channel: SIP/mytrunk/12345678 MaxRetries: 2 RetryTime: 60 WaitTime: 30 Context: callme Extension: 800 Priority: 2 I can generate this file with some scripting language and then place it into the Asterisk Call File folder. The problem I'm having is: How do I call out to two numbers and join them in a conference call? The MeetMe plugin/extension seems to be what I need in terms of conference calling, I'm just unsure as to how I'd use the two together and join them. Also, is it possible to have multiple 2-person conference calls at the same time? Is setting this up as simple as setting aside X amount of 'channels' in the meetme.conf?

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• Home network with Windows 7 as router

  - by Michael

  Background: I have tried to use routers, but so far all of them can't handle the bandwidth, number of connections eventually limited by the hardware resources, so overall the home routers are decreasing the internet speed. I went through DD-WRT and stuff like that. Question: What I want is to use my Windows7 PC as router. It has 2 LAN cards. I'm going to connect to this router another desktop 2 pcs and notebook through wireless router. The main question is what is the most efficient way to turn this Windows7 box(and I need Windows for native NTFS support) into router with NAT/Routing/Firewall functionality? Is there any routing software recommended for this purpose or I should just use windows native "Internet Sharing"? I'm going to run SIP phones in the LAN, so I need friendly NAT(Full cone perhaps). Also I'm going to have FTP server on that Windows7 "server" PC. As firewall I'm thinking about Comodo. Need to drop all incoming, unless explicitly allowed.

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• What program should I use for SSL stripping and re-encrypting

  - by Sparksis

  I'm trying to strip a HTTP over SSL connection down to SSL and then re-encrypt the channel (with a signed certificate(s) I can provide). Of course I want to be able to store captures of all the un-encrypted data. The purpose of this is to reverse engineer a HTTP handshake that is used by a SIP program on my machine. I've tried SSLstrip but it doesn't support what I need it too. Edit: I want something to this effect https://github.com/applidium/Cracking-Siri/blob/master/tcpProxy.rb only more generic and able to write to a pcap stream that wireshark will understand (I'm not sure if this does that). Edit2: upon further inspection this does not create pcap streams. I guess if need be I can write a compatible version but that is not the desired choice.

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• How to remove package from apt-get autoremove "queue"

  - by Darth

  I just installed Calibre for ebook management via apt-get on Ubuntu 10.04, however I found out that it's one major version behind the current release, so I decided to reinstall it directly from sources. When I uninstalled the packaged version, apt added bunch of dependencies to the autoremove queue, and as I installed newer version of Calibre from sources, it has no knowledge of it being dependent on those packages. Now I basically have all libraries that I want, but they are still in the autoremove queue. The following packages were automatically installed and are no longer required: libqt4-script libqt4-designer libqt4-dbus python-lxml python-cherrypy3 python-encutils libqt4-xmlpatterns libqt4-help python-qt4 python-clientform python-sip python-django python-mechanize libqt4-svg python-django-tagging libphonon4 libqt4-xml libqt4-assistant libqt4-webkit libqt4-scripttools python-beautifulsoup python-pypdf python-dateutil python-cssutils Use 'apt-get autoremove' to remove them. How do I tell apt that I want to keep these packages installed, without reinstalling them manually?

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• Setup voip with PennyTel on Nexus One [closed]

  - by Glen

  Hi, I am trying to get VOIP working with PennyTel on my Nexus One. I have created a penny tel voip account and put $10 in. I have tried two options: Sipdroid: I installed this app and followed this guide: http://seethisnowreadthis.com/2009/07/11/get-sipdroid-to-work-with-any-sip-provider-on-your-android-phone/ But I had no luck. i just get an error that it could not authenticate PennyTel test version app by seer I entered my pennytel details in and tried to make a call but it just hangs up immediately. Thanks, Glen.

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• Using an extension to block a caller

  - by Trewq

  I have a couple of SIP phones and use callcentric. I get a lot of junk calls. I'd like to implement the following feature and would like some suggestions on how to do this: Once I get a junk call, I typically hang up. I think want to dial some number (like *23 or something) and I'd like the last number that was received to be put in a database. Any future call from that number will be directed to VM or a busy tone. I'd appreciate some pointers on how I'd go about doing this.. I prefer an open source solution.

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