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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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  • 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 [[email protected]:1] Macro("SIP/801-b7ce8c28", "user-callerid,SKIPTTL,") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]:1] Set("SIP/801-b7ce8c28", "AMPUSER=801") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]:2] GotoIf("SIP/801-b7ce8c28", "0?report") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]:3] ExecIf("SIP/801-b7ce8c28", "1?Set(REALCALLERIDNUM=801)") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]:4] Set("SIP/801-b7ce8c28", "AMPUSER=801") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]:5] Set("SIP/801-b7ce8c28", "AMPUSERCIDNAME=Jona") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]:6] GotoIf("SIP/801-b7ce8c28", "0?report") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]:7] Set("SIP/801-b7ce8c28", "AMPUSERCID=801") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]:8] Set("SIP/801-b7ce8c28", "CALLERID(all)="Jona" <801>") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]:9] Set("SIP/801-b7ce8c28", "REALCALLERIDNUM=801") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]:10] ExecIf("SIP/801-b7ce8c28", "0?Set(CHANNEL(language)=)") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]: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 [[email protected]:20] NoOp("SIP/801-b7ce8c28", "Using CallerID "Jona" <801>") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]:2] Set("SIP/801-b7ce8c28", "_NODEST=") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]:3] Macro("SIP/801-b7ce8c28", "record-enable,801,OUT,") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]: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 [[email protected]: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 [[email protected]:5] MacroExit("SIP/801-b7ce8c28", "") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]:4] Macro("SIP/801-b7ce8c28", "dialout-trunk,1,01483890915,") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]:1] Set("SIP/801-b7ce8c28", "DIAL_TRUNK=1") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]:2] GosubIf("SIP/801-b7ce8c28", "0?sub-pincheck,s,1") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]:3] GotoIf("SIP/801-b7ce8c28", "0?disabletrunk,1") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]:4] Set("SIP/801-b7ce8c28", "DIAL_NUMBER=01483890915") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]:5] Set("SIP/801-b7ce8c28", "DIAL_TRUNK_OPTIONS=tr") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]:6] Set("SIP/801-b7ce8c28", "OUTBOUND_GROUP=OUT_1") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]: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 [[email protected]:9] GotoIf("SIP/801-b7ce8c28", "0?skipoutcid") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]:10] Set("SIP/801-b7ce8c28", "DIAL_TRUNK_OPTIONS=") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]:11] Macro("SIP/801-b7ce8c28", "outbound-callerid,1") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]:1] ExecIf("SIP/801-b7ce8c28", "0?Set(CALLERPRES()=)") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]:2] ExecIf("SIP/801-b7ce8c28", "0?Set(REALCALLERIDNUM=801)") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]: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 [[email protected]:6] Set("SIP/801-b7ce8c28", "USEROUTCID=") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]:7] Set("SIP/801-b7ce8c28", "EMERGENCYCID=") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]:8] Set("SIP/801-b7ce8c28", "TRUNKOUTCID=") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]: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 [[email protected]:12] ExecIf("SIP/801-b7ce8c28", "0?Set(CALLERID(all)=)") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]:13] ExecIf("SIP/801-b7ce8c28", "0?Set(CALLERID(all)=)") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]:14] ExecIf("SIP/801-b7ce8c28", "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]:12] ExecIf("SIP/801-b7ce8c28", "0?AGI(fixlocalprefix)") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]:13] Set("SIP/801-b7ce8c28", "OUTNUM=01483890915") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]:14] Set("SIP/801-b7ce8c28", "custom=DAHDI/1") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]: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 [[email protected]:16] Macro("SIP/801-b7ce8c28", "dialout-trunk-predial-hook,") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]:1] MacroExit("SIP/801-b7ce8c28", "") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]:17] GotoIf("SIP/801-b7ce8c28", "0?bypass,1") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]:18] GotoIf("SIP/801-b7ce8c28", "0?customtrunk") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]: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 [[email protected]: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 [[email protected]: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 [[email protected]: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 [[email protected]:5] Macro("SIP/801-b7ce8c28", "outisbusy,") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]: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 [[email protected]: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 [[email protected]:1] Macro("SIP/801-b7ce8c28", "hangupcall") in new stack [May 26 11:10:54] VERBOSE[2858] logger.c: -- Executing [[email protected]:1] ResetCDR("SIP/801-b7ce8c28", "vw") in new stack [May 26 11:10:54] VERBOSE[2858] logger.c: -- Executing [[email protected]:2] NoCDR("SIP/801-b7ce8c28", "") in new stack [May 26 11:10:54] VERBOSE[2858] logger.c: -- Executing [[email protected]: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 [[email protected]: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 [[email protected]: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 [[email protected]: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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  • 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 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 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". The FXO module is directly connected to our phone line from BT via a BT-RJ11 cable. I'm guessing I've missed a configuration step somewhere but no idea where, any help greatly appreciated.

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  • Home PBX to answer/take external calls via PSTN

    - by ageis23
    I have a Thomson 585V6 router which has built in voip support. I want to be able to use a softphone to make calls. for example phone my dad's mobile. Any incomming calls to my normal bt number should be taken via my pc as well. What I have done so far: I've wired the pstn port on the router to the telephone jack. The router is connected to my pc. I have installed asterisk on the pc I want to take calls on. The sip client authenticates to the sip server. output from twinkle: Sun 22:46:45 home, registration succeeded (expires = 3600 seconds) how do I take external calls/ answer incoming calls from pstn?

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  • Considerations for a business looking to transition from PSTN to IP Telephony

    - by Bryce Thomas
    Full disclosure - This is related to a homework assignment question. I am not asking you to do my work for me, I am merely looking for some pointers and considerations to direct me in my further research. I have an assignment I'm working on where I've been given a scenario where a business wants to look into transitioning to using "Internet Telephone" as opposed to a traditional PSTN/PBX system and I need to write a report on it. I'm after some high level pointers from people, especially anyone that has been involved in a real life transition of this nature, on what some of the most important considerations are. These can be financial considerations, initial setup considerations, ongoing administrative considerations, quality of service considerations or anything else that is pertinent to performing such a transition.

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  • McAfee Secure Messaging Service / Postini: false positive?

    - by Martin
    Hello, I'm puzzled by this email message that gets quarantined by McAfee Secure Messaging Service (it's based on Postini) for no reason that I can think of. Here are the Postini headers: X-pstn-2strike: clear X-pstn-neptune: 0/0/0.00/0 X-pstn-levels: (S: 0.02932/98.63596 CV:99.9000 FC:95.5390 LC:95.5390 R:95.9108 P:95.9108 M:97.0282 C:98.6951 ) X-pstn-settings: 3 (1.0000:1.0000) s cv gt3 gt2 gt1 r p m c X-pstn-addresses: from [db-null] X-pstn-disposition: quarantine I read the docs (http://www.mcafee-sms.com/webdocs/admin%5Fee%5Fmcafee/wwhelp/wwhimpl/common/html/wwhelp.htm?context=MACAFFHelp&file=header%5Foverview.html#951634) and in short, the x-pstn-settings header tells me that NONE of the filters was triggered, but the x-pstn-levels header tells me that the final score (0.02932) is low enough to classify the email as bulk/spam. Can anyone explain to me why the final score is so low when none of the filters were triggered? Does anyone have any suggestions on how to prevent this from happening? Regards, Martin

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  • Cheap Solution for Routing a Toll Free Number to a Standard POTS Number

    - by VxJasonxV
    I do some technical work for an Internet Radio Show/Podcast, and need to fix something that has been broken for a while. The hosts have a Skype-In number to take listener calls, and for convenience sake, I bought and paid for a toll free number for a period of time. I used to use Asterlink for routing calls, but they folded and sent my number to OneBox, but they're ridiculously expensive by comparison. I'm looking for a cheap solution for this one simple task. Forward toll free calls to a skype-in number. The definition of cheap is as cheap or cheaper than Asterlink was. I paid something like $2 a month, and then the termination/call rate, which was a fraction of a sent for termination, and only whole cents after some serious time on the call. A $20 preload lasted me months at a time. I don't want to be upsold too, I want a simple web based management screen (CDR/stats are fun!), and obviously, it needs to be reliable. What vendors out there are you a fan of that solves this need?

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  • Cheap Solution for Routing a Toll Free Number to a Standard POTS Number

    - by VxJasonxV
    I do some technical work for an Internet Radio Show/Podcast, and need to fix something that has been broken for a while. The hosts have a Skype-In number to take listener calls, and for convenience sake, I bought and paid for a toll free number for a period of time. I used to use Asterlink for routing calls, but they folded and sent my number to OneBox, but they're ridiculously expensive by comparison. I'm looking for a cheap solution for this one simple task. Forward toll free calls to a skype-in number. The definition of cheap is as cheap or cheaper than Asterlink was. I paid something like $2 a month, and then the termination/call rate, which was a fraction of a sent for termination, and only whole cents after some serious time on the call. A $20 preload lasted me months at a time. I don't want to be upsold too, I want a simple web based management screen (CDR/stats are fun!), and obviously, it needs to be reliable. What vendors out there are you a fan of that solves this need?

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  • Is there a cheaper alternative to Skype for VoIP to PSTN calls to Vietnam?

    - by Nick Bolton
    My dad uses Skype to make calls to Vietnam PSTN, but finds that the rates are a little on the pricey side. It's probably not relevant, but he's living in Thailand right now. Is there an application similar to Skype, which is cheaper for calling Vietnam? The answer may well be no, since maybe the international telecom peers in Vietnam are just generally expensive... Who knows? While I'm asking, maybe it's worth someone mentioning if there's a cheaper alternative to Skype in general? I'm thinking that maybe not as Skype's pretty cheap anyway, but it's worth mentioning.

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  • asterisk extensions.conf & sip.conf

    - by Josh
    I'm trying to get my Dialplan to work. When I call, the only thing I get is a dial tone to enter extension "no Background(thanks-calling) is played". When extension 123 is dialed, busy signal is triggered and asterisk CLI get frozen. Any help will be appreciate it. Conf files below. ; PSTN on sip.conf [pstn] type=friend host=dynamic context=pstn username=pstn secret=password nat=yes canreinvite=no dtmfmode=rfc2833 qualify=yes insecure=port,invite disallow=all allow=ulaw ; PSTN on extensions.conf [pstn] exten => s,1,Answer exten => s,2,Wait,2 exten => s,4,DigitTimeout,5 exten => s,5,ResponseTimeout,10 exten => s,6,Background(thanks-calling) exten => 0,1,Goto(incoming,123,1) ; (Member Services) [incoming] exten => 123,1,NoOP(${CALLERID}) ; show the caller ID info in the console exten => 123,n,Ringing() exten => 123,n,Answer() exten => 123,n,Playback(silence/1) exten => 123,n,Playback(connecting1) exten => 123,n,Wait(3) exten => 123,n,Dial(SIP/line1,60) exten => 123,n,Congestion

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  • asterisk extensions.conf & sip.conf

    - by Josh
    I'm trying to get my Dialplan to work. When I call, the only thing I get is a dial tone to enter extension "no Background(thanks-calling) is played". When extension 123 is dialed, busy signal is triggered and asterisk CLI get frozen. Any help will be appreciate it. Conf files below. ; PSTN on sip.conf [pstn] type=friend host=dynamic context=pstn username=pstn secret=password nat=yes canreinvite=no dtmfmode=rfc2833 qualify=yes insecure=port,invite disallow=all allow=ulaw ; PSTN on extensions.conf [pstn] exten => s,1,Answer exten => s,2,Wait,2 exten => s,4,DigitTimeout,5 exten => s,5,ResponseTimeout,10 exten => s,6,Background(thanks-calling) exten => 0,1,Goto(incoming,123,1) ; (Member Services) [incoming] exten => 123,1,NoOP(${CALLERID}) ; show the caller ID info in the console exten => 123,n,Ringing() exten => 123,n,Answer() exten => 123,n,Playback(silence/1) exten => 123,n,Playback(connecting1) exten => 123,n,Wait(3) exten => 123,n,Dial(SIP/line1,60) exten => 123,n,Congestion

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  • Email postfix marked as spam by google

    - by Rodrigo Ferrari
    Hello friends, I searched about this question, found some few answers but no idea how to fix, the problem is that I realy dumb with all this! I configured the postfix and done everything how the install how to told. It send the email, but get marked as spam! The header is this one: Delivered-To: [email protected] Received: by 10.223.86.203 with SMTP id t11cs837410fal; Wed, 12 Jan 2011 04:02:21 -0800 (PST) X-pstn-nxpr: disp=neutral, [email protected] X-pstn-nxp: bodyHash=9c6d0c64fa3a4d663c9968e9545c47d77ae0242e, headerHash=1ab8726bd17a23218309165bd20fe6e0911627cd, keyName=4, rcptHash=178929be6ed8451d98a4df01a463784e6c59b3b4, sourceip=174.121.4.154, version=1 Received: by 10.100.190.13 with SMTP id n13mr537609anf.76.1294833740396; Wed, 12 Jan 2011 04:02:20 -0800 (PST) Return-Path: <[email protected]> Received: from psmtp.com ([74.125.245.168]) by mx.google.com with SMTP id w2si1297960anw.132.2011.01.12.04.02.19; Wed, 12 Jan 2011 04:02:20 -0800 (PST) Received-SPF: pass (google.com: domain of [email protected] designates 174.121.4.154 as permitted sender) client-ip=174.121.4.154; Authentication-Results: mx.google.com; spf=pass (google.com: domain of [email protected] designates 174.121.4.154 as permitted sender) [email protected] Received: from source ([174.121.4.154]) by na3sys010amx168.postini.com ([74.125.244.10]) with SMTP; Wed, 12 Jan 2011 12:02:19 GMT Received: from localhost (server [127.0.0.1]) (using TLSv1 with cipher DHE-RSA-AES256-SHA (256/256 bits)) (No client certificate requested) by brasilyacht.com.br (Postfix) with ESMTP id 87C121290142; Wed, 12 Jan 2011 09:50:29 -0200 (BRST) From: YachtBrasil <[email protected]> Reply-To: Vendas <[email protected]> Cc: YachtBrasil <[email protected]> To: [email protected] Subject: teste Date: Wed, 12 Jan 2011 09:50:29 -0200 Content-Type: text/html; charset=UTF-8 Content-Transfer-Encoding: quoted-printable Content-Disposition: inline MIME-Version: 1.0 Message-Id: <[email protected]> X-pstn-2strike: clear X-pstn-neptune: 0/0/0.00/0 X-pstn-levels: (S: 1.96218/99.81787 CV:99.9000 FC:95.5390 LC:95.5390 R:95.9108 P:95.9108 M:97.0282 C:98.6951 ) X-pstn-settings: 3 (1.0000:1.0000) s cv gt3 gt2 gt1 r p m c X-pstn-addresses: from <[email protected]> [db-null] I'm out of ideas on how to fix this, I think it's dns issue, but I have marked the spf inside my tinydns =( Is there anything I can check to know why this email is marked as spam? Any help will be appreciated! Thanks and sorry for my bad english.

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  • SIP and NAT routers?

    - by OverTheRainbow
    Hello SIP was not built with NAT routers in mind, and I'd like to get to the bottom of this issue to check what needs to be done on all devices so it works with NAT routers, and understand in what context it just can't be used and I should check more NAT-friendly alternatives like IAX. A picture being worth a thousand words, here's the layout I need to use: http://img62.imageshack.us/img62/4077/sipandnatrouters.jpg The PBX server is located in the private LAN behind a NAT router connected to the Internet (I know it'd be easier if it were located in the public network, but this router doesn't support DMZ's so the server has to be in the private network) A couple of (soft|hard)phones are located on the same LAN and connected to the PBX server, along with a PSTN gateway (Linksys 3102 or a Digium PCI card) Remote users using (soft|hard)phones are located somewhere on the Net with dynamic IP's and are also located behind NAT routers I may or may not have control over the local NAT router where the PBX server is located, but I have no control over the remote NAT routers, either because the users don't have the computer knowledge to map ports or because the routers are off-limit (eg. web cafés, hotel LAN's, etc.) Is it possible to configure the PBX server, the (soft|hard)phones, and the PSTN gateway so that the all conversations work fine, no matter the endpoints (POTS caller/local phone, POTS caller/remote phone, local phones, remote phone/local phone)? In which cases may I expect problems, and are there solutions? FWIW, I'm leaning toward using Freeswitch, but I could end up using Asterisk if there are technical advantages to it in this context. Thank you for any info.

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  • Free Software Simulators for SS7, ISDN, SIP, etc., Telecom Protocols.

    - by RBA
    Hi, I am learning Protocols where I have major use of Media Gateway Controllers, Media Gateway, PSTN N/w, VOIP N/w. Calls getting gatewayed from one node to another. Kindly help me in finding out some related software simulators where I can view pictorially the messages being exchanged between the various nodes in telecom architecture. Thanks

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  • Exposing the ipPhone attribute to Communicator and the OCS address book service

    - by Doug Luxem
    I am in the process of integrating OCS with our Cisco phone system using CUCIMOC. After some fiddling with the phone normalization rules, it appears that I can get PSTN numbers to be dialed though the CUCIMOC interface (yay!). However, during this process I came to realize that the ipPhone attribute in Active Directory does not appear to be exposed to Communicator (and CUCIMOC). What is strange though, is that I can see from the OCS address book service "Invalid_AD_Phone_Numbers.txt" that the attribute is processed by the address book service. My question is, how do I expose the ipPhone field in Office Communicator? Currently, Communicator maps like this - Work = telephoneNumber Mobile = mobile Home = homePhone Attributes such as otherHomePhone, ipPhone, otherMobile, otherTelephone, otherIpPhone are ignored.

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  • Annex-A vs Annex-B ADSL splitters

    - by Odys
    While trying to improve our Adsl connection, I found that we use a splitter that is from an unknown ventor and have no clue whether is for AnnexA (PSTN) or AnnexB (ISDN) line. Is it possible that somebody used the wrong splitter and just because it connected nobody ever cared? We have an ISDN line but the splitter used is blank. I am not really sure if it is AnnexB. Would it be possible to use an AnnexA filter over an ISDN line (even with that much signal loss)? (I am posting here because this is about our connection to the rack, I'm not sure if there is a more proper SExchange site)

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  • Asterisk doesn't start properly at system startup. DNS lookup fails.

    - by leiflundgren
    When I start my Ubuntu system it attempts two DNS lookups. One to find out what my internet-routers external ip is. And one to find the IP of my PSTN-SIP-provider. Both fails. [Apr 7 22:14:54] WARNING[1675] chan_sip.c: Invalid address for externhost keyword: sip.mydomain.com ... [Apr 7 22:14:54] WARNING[1675] acl.c: Unable to lookup 'sip.myprovider.com' And since the DNS fails it cannot register properly a cannot make outgoing or incoming calls. If I later, after bootup, restart asterisk everything works excelent. Any idea how I should setup things so that either: Delay Asterisk startup so that DNS is up and healthy first. Somehow get Asterisk to re-try the DNS thing later. Regards Leif

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  • Asterisk doesn't start properly at system startup. DNS lookup fails.

    - by leiflundgren
    When I start my Ubuntu system it attempts two DNS lookups. One to find out what my internet-routers external ip is. And one to find the IP of my PSTN-SIP-provider. Both fails. [Apr 7 22:14:54] WARNING[1675] chan_sip.c: Invalid address for externhost keyword: sip.mydomain.com ... [Apr 7 22:14:54] WARNING[1675] acl.c: Unable to lookup 'sip.myprovider.com' And since the DNS fails it cannot register properly a cannot make outgoing or incoming calls. If I later, after bootup, restart asterisk everything works excelent. Any idea how I should setup things so that either: Delay Asterisk startup so that DNS is up and healthy first. Somehow get Asterisk to re-try the DNS thing later. Regards Leif

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  • Best VoIP VPS Location?

    - by ToiletOverflow
    Our office currently runs a few phones over a VoIP line. Through our VoIP provider, we have a virtual private server. We chose them as the VPS provider because the VPS came pre-loaded with all of the software that was necessary. However, I've discovered that I would rather manage the software myself and would prefer to work on a different platform. The primary reason that we have stayed with them is because as our VoIP provider, I presume that there is some advantage in call quality because they have "direct access to the PSTN", which I would presume is an advantage when it comes to call termination and overall call quality. My question boils down to: What is better from a call quality perspective? 1) A server located 20ms closer to us (60ms), offered by a different company. 2) The current server at the VoIP/SIP provider (80ms)

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  • Test-service on Internet for testing incoming INVITE

    - by leiflundgren
    I am trying to set up Asterisk at home. I think I'm having trouble configuring my firewall, so that inbound traffic is accepted, but I am not sure. I got the idea that, perhaps, there is a service out on the Internet, where I can, though a web-browser, initiate an incoming call, an INVITE. And then see the SIP-trace that the remote-part experience. Anyone know of such a service? Note. I have a SIP-PSTN provider so I can generate inbound calls. But I cannot see the SIP-logs from my provider...

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  • Unofficial Prep guide for TS: Microsoft Lync Server 2010, Configuring (70-664)

    - by Enrique Lima
    Managing Users and Client Access (20 percent)   Objective Materials Configure user accounts http://technet.microsoft.com/en-us/library/gg182543.aspx Deploy and maintain clients http://technet.microsoft.com/en-us/library/gg412773.aspx Configure conferencing policies http://technet.microsoft.com/en-us/library/gg182561.aspx Configure IM policies http://technet.microsoft.com/en-us/library/gg182558.aspx Deploy and maintain Lync Server 2010 devices http://technet.microsoft.com/en-us/library/gg412773.aspx Resolve client access issues http://technet.microsoft.com/en-us/library/gg398307.aspx   Configuring a Lync Server 2010 Topology (21 percent)   Objective Materials Prepare to deploy a topology http://technet.microsoft.com/en-us/library/gg398630.aspx Configure Lync Server 2010 by using Topology Builder http://technet.microsoft.com/en-us/library/gg398420.aspx Configure role-based access control in Lync Server 2010 http://technet.microsoft.com/en-us/library/gg412794.aspx http://technet.microsoft.com/en-us/library/gg425917.aspx Configure a location information server http://technet.microsoft.com/en-us/library/gg398390.aspx Configure server pools for load balancing http://technet.microsoft.com/en-us/library/gg398827.aspx   Configuring Enterprise Voice (19 percent)   Objective Materials Configure voice policies http://technet.microsoft.com/en-us/library/gg398450.aspx Configure dial plans http://technet.microsoft.com/en-us/library/gg398922.aspx Manage routing http://technet.microsoft.com/en-us/library/gg425890.aspx http://technet.microsoft.com/en-us/library/gg182596.aspx Configure Microsoft Exchange Unified Messaging integration http://technet.microsoft.com/en-us/library/gg398768.aspx Configure dial-in conferencing http://technet.microsoft.com/en-us/library/gg398600.aspx Configure call admission control http://technet.microsoft.com/en-us/library/gg520942.aspx Configure Response Group Services (RGS) http://technet.microsoft.com/en-us/library/gg398584.aspx Configure Call Park and Unassigned Number http://technet.microsoft.com/en-us/library/gg399014.aspx http://technet.microsoft.com/en-us/library/gg425944.aspx Manage a Mediation Server pool and PSTN Gateway http://technet.microsoft.com/en-us/library/gg412780.aspx   Configuring Lync Server 2010 for External Access (19 percent)   Objective Materials Configure Edge Services http://technet.microsoft.com/en-us/library/gg398918.aspx Configure a firewall http://technet.microsoft.com/en-us/library/gg425882.aspx Configure a reverse proxy http://technet.microsoft.com/en-us/library/gg425779.aspx   Monitoring and Maintaining Lync Server 2010 (21 percent)   Objective Materials Back up and restore Lync Server 2010 http://technet.microsoft.com/en-us/library/gg412771.aspx Configure monitoring and archiving http://technet.microsoft.com/en-us/library/gg398199.aspx http://technet.microsoft.com/en-us/library/gg398507.aspx http://technet.microsoft.com/en-us/library/gg520950.aspx http://technet.microsoft.com/en-us/library/gg520990.aspx Implement troubleshooting tools http://technet.microsoft.com/en-us/library/gg425800.aspx Use PowerShell to test Lync Server 2010 http://technet.microsoft.com/en-us/library/gg398474.aspx

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  • Regex for continues sentence

    - by SmartestVEGA
    I have the following text, I need to extract the exception name and the continuing sentence from the file, but the file has continuous sentences without a space. ??????>????????????????????????????????????????????????????????????????????????????????????????????????????????????????? ???????????????????????????????????????????????????????????????????????????????????????????????????????????????????????? ???????????????????????????????????????????????????????????????????????????????????????????????????????????????????????? ?????????????????????????????????????????????????????????????????????????????????????????B????!48#$%&'+-/0123????5679<[email protected] >?CA???????????????????????????????????????????????????????????????????????????????????????????????????????????????????? ???????????????????????????????????????????????????????????????????????????????????????????????????????????????????????? ????????????????RootEntry?????????F?|?`???__nameid_version10?????????|?`???|?`??__substg10_00020102????????????__substg1 0_00030102????????????????????????????????????????????????????????????????????????????????????????????????!????#$??????? 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[email protected]??0 I need to extract the following keywords: ConnectException, javanetConnectException and any exception as well as its exception type. Please help me in writing a regex for this?

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  • String manipulation in C#

    - by SmartestVEGA
    I have the following text, I need to extract the exception name and the continuing sentence from the file, but the file has continuous sentences without a space. ??????>????????????????????????????????????????????????????????????????????????????????????????????????????????????????? ???????????????????????????????????????????????????????????????????????????????????????????????????????????????????????? ???????????????????????????????????????????????????????????????????????????????????????????????????????????????????????? ?????????????????????????????????????????????????????????????????????????????????????????B????!48#$%&'+-/0123????5679<[email protected] >?CA???????????????????????????????????????????????????????????????????????????????????????????????????????????????????? ???????????????????????????????????????????????????????????????????????????????????????????????????????????????????????? ????????????????RootEntry?????????F?|?`???__nameid_version10?????????|?`???|?`??__substg10_00020102????????????__substg1 0_00030102????????????????????????????????????????????????????????????????????????????????????????????????!????#$??????? 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MESSAGEServerconnectionfailureduringtransactionDuetounderlyingexception'javanetSocketExceptionjavanetConnectExceptionCon nectiontimedout'BEGINNESTEDEXCEPTIONjavanetSocketExceptionMESSAGEjavanetConnectExceptionConnectiontimedoutSTACKTRACEjava netSocketExceptionjavanetConnectExceptionConnectiontimedoutatcommysqljdbcStandardSocketFactoryconnectStandardSocketFacto ryjava156atcommysqljdbcMysqlIO<init>MysqlIOjava284atcommysqljdbcConnectioncreateNewIOConnectionjava2672atcommysqljdbcCon nection<init>Connectionjava1474atcommysqljdbcNonRegisteringDriverconnectNonRegisteringDriverjava266atorgapachecommonsdbc pDriverConnectionFactorycreateConnectionDriverConnectionFactoryjava37atorgapachecommonsdbcpPoolableConnectionFactorymake ObjectPoolableConnectionFactoryjava291atorgapachecommonspoolimplGenericObjectPoolborrowObjectGenericObjectPooljava771ato rgapachecommonsdbcpPoolingDataSourcegetConnectionPoolingDataSourcejava95atorgapachecommonsdbcpBasicDataSourcegetConnecti onBasicDataSourcejava548atsunreflectGeneratedMethodAccessor530invokeUnknownSourceatsunreflectDelegatingMethodAccessorImp linvokeDelegatingMethodAccessorImpljava25atjavalangreflectMethodinvokeMethodjava597atorgspringframeworkaopsupportAopUtil sinvokeJoinpointUsingReflectionAopUtilsjava310atorgspringframeworkaopframeworkReflectiveMethodInvocationinvokeJoinpointR eflectiveMethodInvocationjava182atorgspringframeworkaopframeworkReflectiveMethodInvocationproceedReflectiveMethodInvocat ionjava149atorgspringframeworkaopframeworkadapterThrowsAdviceInterceptorinvokeThrowsAdviceInterceptorjava126atorgspringf rameworkaopframeworkReflectiveMethodInvocationproceedReflectiveMethodInvocationjava171atorgspringframeworkaopframeworkJd kDynamicAopProxyinvokeJdkDynamicAopProxyjava204at$Proxy20getConnectionUnknownSourceatorgspringframeworkormhibernate3Loca lDataSourceConnectionProvidergetConnectionLocalDataSourceConnectionProviderjava82atorghibernatejdbcConnectionManageropen ConnectionConnectionManagerjava417atorghibernatejdbcConnectionManagergetConnectionConnectionManagerjava144atorghibernate jdbcAbstractBatcherprepareQueryStatementAbstractBatcherjava105atorghibernateloaderLoaderprepareQueryStatementLoaderjava1 561atorghibernateloaderLoaderdoQueryLoaderjava661atorghibernateloaderLoaderdoQueryAndInitializeNonLazyCollectionsLoaderj ava224atorghibernateloaderLoaderdoListLoaderjava2145atorghibernateloaderLoaderlistIgnoreQueryCacheLoaderjava2029atorghib ernateloaderLoaderlistLoaderjava2024atorghibernateloadercriteriaCriteriaLoaderlistCriteriaLoaderjava94atorghibernateimpl SessionImpllistSessionImpljava1533atorghibernateimplCriteriaImpllistCriteriaImpljava283atorgspringframeworkormhibernate3 HibernateTemplate$36doInHibernateHibernateTemplatejava1061atorgspringframeworkormhibernate3HibernateTemplatedoExecuteHib ernateTemplatejava419atorgspringframeworkormhibernate3HibernateTemplateexecuteWithNativeSessionHibernateTemplatejava374a torgspringframeworkormhibernate3HibernateTemplatefindByCriteriaHibernateTemplatejava1051atorgspringframeworkormhibernate 3HibernateTemplatefindByCriteriaHibernateTemplatejava1044atcommazarvoiceccadaohibernateModelDAOHibernatefindModelDAOHibe rnatejava189atcommazarvoiceccadaohibernateAbstractCriteriaDAOHibernatefindAbstractCriteriaDAOHibernatejava113atcommazarv oiceccaloggingserviceimplLoggingConfigurationServiceImplupdateFiltersLoggingConfigurationServiceImpljava104atcommazarvoi ceccaloggingserviceimplLoggingConfigurationServiceImplupdateLoggingConfigurationLoggingConfigurationServiceImpljava95atc ommazarvoiceccaloggingserviceimplLoggingConfigurationServiceImpl$1runLoggingConfigurationServiceImpljava69ENDNESTEDEXCEP TIONAttemptedreconnect3timesGivingupP&uU??$???En??g=%0f-8"!?ESSAGE?!`av?+?'[email protected]?q-?DP1??oun?ly1?''!`'java?+?tSock?% ?!`4wP+?5i6?'??%@?`t'"?"?BEGIEN/0TED!X?%?PTIO9?9?4%?"?/'67/89??ACKTRB?/??<[email protected]?9%[email protected] ?<?it>P?M?284N/[email protected]?Qt6??Q?STP?UN14?7Q?I?N"g??aD/?K?\?]?U??N$Ra??%Iq`??!p]?GXKG?T?GXb?M37`O?a_bcPo??ld??KF?D?Obj Li??f<291g?hO`?fpPG?qc?k?k??!!wk?p??M?77noiH]R$??aD?p%?gd?sv?M?95toubcB>a?wOd?|?M?54?8Z?s3flP?p?$?dMhp?Ac%??5?ppnvoD??kn ?r0?}d???HDelP?g$?a?|Ip????????Qtz^D`p?W?t7???%?9g/$asp?1rPa?w?k{??ob??pbA?0?U%[email protected]??Jo?`[email protected]??e???f?P?????????/??tI?? c?!2Q?}?????M?8V~?o????????D??d?/???49?o???aL?0]?Thr!sAdv??e???%[email protected]????????????]tO?O?_Jdk?Dy$?m???P`^xK?????mv 0Z?$????w??????I??$?3L??l}GX??????w?????M??O?$\S?Mpaw?]??0??d???M?4????/?Ow??Z&[email protected]?b?-?|?t{?]???%??eQPK?!??????M?0z_ $k????]???????????M????????do?3??`??{????A3??P?1i?zT?LazyP?l??s??22???????L]??Z????????w?Ig????C{???????????????]?]"D0C? 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[email protected]??0 I need to extract the following keywords: ConnectException, javanetConnectException and any exception as well as its exception type.

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  • String manipulation in C# ( Brainstroming ;-) )

    - by SmartestVEGA
    I have the following text , i need to extract the exception name and the continuing sentence from the file... but the file has continues sentence without a space. 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coding7bit------=_Part_51_8002724931266681608155--__substg10_0C1E001F!????'__substg10_0C1F001F????????????<__substg10_0E[email protected][email protected][mazarvoiceSMTPAppender]Applicationmessage00000002BLREXCH/O=ad/OU=ad/cn=Recipients/cn=_OperationsSupportMicrosoftExchangeServer__substg10_0E04001F#%????4__substg10_0E1D001F?????????????__substg10_0E28001F"????-?__substg10_0E29001F????????????0?00000002BLREXCH/O=ad/OU=ad/cn=Recipients/cn=_OperationsSupportMicrosoftExchangeServerZxLZFu#O?rcpg125?2CtexA???????PV?U?%Qch??set2?%?3F?03???05"[email protected]???SQLHErr???S??!a8?????ERROR??OZ?%?rpd?%?URL="jdbcmysql//[email protected]/bv2?a%[email protected]?nn?t=t?r__substg10_1000001F'????"?#__substg10_10090102????????????3^__substg10_1035001F????Q?__substg10_10F3001F????????????T?02-20100008WARNorghibernateutilJDBCExceptionReporterSQLError0SQLState0800102-20100008ERRORorghibernateutilJDBCExceptionReporterSQLexceptionraisedforJDBCURL="jdbcmysql//stg-c5-dbmst13306/bv2?autoReconnect=true&useUnicode=true&characterEncoding=utf-8"!MESSAGEServerconnectionfailureduringtransactionDuetounderlyingexception'javanetSocketExceptionjavanetConnectExceptionConnectiontimedout'BEGINNESTEDEXCEPTIONjavanetSocketExceptionMESSAGEjavanetConnectExceptionConnectiontimedoutSTACKTRACEjavanetSocketExceptionjavanetConnectExceptionConnectiontimedoutatcommysqljdbcStandardSocketFactoryconnectStandardSocketFactoryjava156atcommysqljdbcMysqlIO<init>MysqlIOjava284atcommysqljdbcConnectioncreateNewIOConnectionjava2672atcommysqljdbcConnection<init>Connectionjava1474atcommysqljdbcNonRegisteringDriverconnectNonRegisteringDriverjava266atorgapachecommonsdbcpDriverConnectionFactorycreateConnectionDriverConnectionFactoryjava37atorgapachecommonsdbcpPoolableConnectionFactorymakeObjectPoolableConnectionFactoryjava291atorgapachecommonspoolimplGenericObjectPoolborrowObjectGenericObjectPooljava771atorgapachecommonsdbcpPoolingDataSourcegetConnectionPoolingDataSourcejava95atorgapachecommonsdbcpBasicDataSourcegetConnectionBasicDataSourcejava548atsunreflectGeneratedMethodAccessor530invokeUnknownSourceatsunreflectDelegatingMethodAccessorImplinvokeDelegatingMethodAccessorImpljava25atjavalangreflectMethodinvokeMethodjava597atorgspringframeworkaopsupportAopUtilsinvokeJoinpointUsingReflectionAopUtilsjava310atorgspringframeworkaopframeworkReflectiveMethodInvocationinvokeJoinpointReflectiveMethodInvocationjava182atorgspringframeworkaopframeworkReflectiveMethodInvocationproceedReflectiveMethodInvocationjava149atorgspringframeworkaopframeworkadapterThrowsAdviceInterceptorinvokeThrowsAdviceInterceptorjava126atorgspringframeworkaopframeworkReflectiveMethodInvocationproceedReflectiveMethodInvocationjava171atorgspringframeworkaopframeworkJdkDynamicAopProxyinvokeJdkDynamicAopProxyjava204at$Proxy20getConnectionUnknownSourceatorgspringframeworkormhibernate3LocalDataSourceConnectionProvidergetConnectionLocalDataSourceConnectionProviderjava82atorghibernatejdbcConnectionManageropenConnectionConnectionManagerjava417atorghibernatejdbcConnectionManagergetConnectionConnectionManagerjava144atorghibernatejdbcAbstractBatcherprepareQueryStatementAbstractBatcherjava105atorghibernateloaderLoaderprepareQueryStatementLoaderjava1561atorghibernateloaderLoaderdoQueryLoaderjava661atorghibernateloaderLoaderdoQueryAndInitializeNonLazyCollectionsLoaderjava224atorghibernateloaderLoaderdoListLoaderjava2145atorghibernateloaderLoaderlistIgnoreQueryCacheLoaderjava2029atorghibernateloaderLoaderlistLoaderjava2024atorghibernateloadercriteriaCriteriaLoaderlistCriteriaLoaderjava94atorghibernateimplSessionImpllistSessionImpljava1533atorghibernateimplCriteriaImpllistCriteriaImpljava283atorgspringframeworkormhibernate3HibernateTemplate$36doInHibernateHibernateTemplatejava1061atorgspringframeworkormhibernate3HibernateTemplatedoExecuteHibernateTemplatejava419atorgspringframeworkormhibernate3HibernateTemplateexecuteWithNativeSessionHibernateTemplatejava374atorgspringframeworkormhibernate3HibernateTemplatefindByCriteriaHibernateTemplatejava1051atorgspringframeworkormhibernate3HibernateTemplatefindByCriteriaHibernateTemplatejava1044atcommazarvoiceccadaohibernateModelDAOHibernatefindModelDAOHibernatejava189atcommazarvoiceccadaohibernateAbstractCriteriaDAOHibernatefindAbstractCriteriaDAOHibernatejava113atcommazarvoiceccaloggingserviceimplLoggingConfigurationServiceImplupdateFiltersLoggingConfigurationServiceImpljava104atcommazarvoiceccaloggingserviceimplLoggingConfigurationServiceImplupdateLoggingConfigurationLoggingConfigurationServiceImpljava95atcommazarvoiceccaloggingserviceimplLoggingConfigurationServiceImpl$1runLoggingConfigurationServiceImpljava69ENDNESTEDEXCEPTIONAttemptedreconnect3timesGivingupP&uU??$???En??g=%0f-8"!?ESSAGE?!av?+?'[email protected]?q-?DP1??oun?ly1?''!'java?+?tSock?%?!4wP+?5i6?'??%@?t'"?"?BEGIEN/0TED!X?%?PTIO9?9?4%?"?/'67/89??ACKTRB?/??<[email protected]?9%[email protected]?<?it>P?M?284N/[email protected]?Qt6??Q?STP?UN14?7Q?I?N"g??aD/?K?\?]?U??N$Ra??%Iq??!p]?GXKG?T?GXb?M37O?a_bcPo??ld??KF?D?ObjLi??f<2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For eg: the following text has so many exception keywords but i need to extract those keywords like ConnectException javanetConnectException so on ... like whatever exception in the text and the next continuing sentence.. ConnectException: Connection timed out javanetConnectException : Connection timed out

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