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  • What backup utilities are available for Trixbox CE?

    - by Tim Post
    I'm running Trixbox CE (v2.6.2.3) and I need to make a full system backup of all configurations, recordings, databases, etc in a manner that can be easily restored on a brand new installation of the same version of Trixbox CE. I started to write something myself using rsync, however I feel like I'm probably missing some things and needlessly copying other things. What are my options?

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  • Lync 2010, Kamailio, & Trixbox 2.6.23 (Asterisk 1.4)

    - by slashp
    I'm having an issue trying to connect Lync 2010 phone calls with our trixbox PBX. I've gotten to the point where Kamailio seems to be functioning properly and acting as a bridge between TCP traffic (from Lync) & UDP traffic (to the trixbox, as Asterisk 1.4 does not support SIP over TCP). Our Lync box IP: 10.100.10.41 Our Kamailio box IP: 10.100.10.44 Our trixbox IP: 10.100.10.2 The issue I'm running into is as follows when enabling SIP debugging for the Kamailio box: <--- SIP read from 10.100.10.44:5060 ---> PRACK sip:TNECLTSLY01.contoso.com:5068;transport=Tcp;maddr=10.100.10.41 SIP/2.0 FROM: <sip:9121;[email protected];user=phone>;epid=CF2380792B;tag=4852bab430 TO: <sip:[email protected];user=phone>;epid=CF2380792B;tag=3684a6a24e CSEQ: 24 PRACK CALL-ID: 192daae6-00e1-4140-bddd-0394b35d475b MAX-FORWARDS: 70 Via: SIP/2.0/UDP 10.100.10.44;branch=z9hG4bKcydzigwkX;i=d VIA: SIP/2.0/TCP 10.100.10.41:51677;branch=z9hG4bK159fc989 CONTACT: <sip:TNECLTSLY01.contoso.com:5068;transport=Tcp;maddr=10.100.10.41> CONTENT-LENGTH: 0 USER-AGENT: RTCC/4.0.0.0 MediationServer RAck: 1 23 INVITE <-------------> --- (12 headers 0 lines) --- Sending to 10.100.10.44 : 5060 (NAT) <--- Transmitting (NAT) to 10.100.10.44:5060 ---> SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 10.100.10.44;branch=z9hG4bKcydzigwkX;i=d;received=10.100.10.44 Via: SIP/2.0/TCP 10.100.10.41:51677;branch=z9hG4bK159fc989 From: <sip:9121;[email protected];user=phone>;epid=CF2380792B;tag=4852bab430 To: <sip:[email protected];user=phone>;epid=CF2380792B;tag=3684a6a24e Call-ID: 192daae6-00e1-4140-bddd-0394b35d475b CSeq: 24 PRACK User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> trixbox1*CLI> <--- SIP read from 10.100.10.44:5060 ---> ACK sip:[email protected];user=phone SIP/2.0 FROM: "John Jones"<sip:9121;[email protected];user=phone>;tag=4852bab430;epid=CF2380792B TO: <sip:[email protected];user=phone>;tag=3684a6a24e;epid=CF2380792B CSEQ: 23 ACK CALL-ID: 192daae6-00e1-4140-bddd-0394b35d475b MAX-FORWARDS: 70 Via: SIP/2.0/UDP 10.100.10.44;branch=z9hG4bKcydzigwkX;i=d VIA: SIP/2.0/TCP 10.100.10.41:51677;branch=z9hG4bK79a21c CONTENT-LENGTH: 0 My SIP trunk on the trixbox looks like this: [from-lync] exten => _+4XXX!,1,Noop(Stripping + from start of number) exten => _+4XXX!,n,Goto(from-internal,${EXTEN:1}) Though I am still having no luck getting the + stripped or the call to go through. Any ideas would be greatly appreciated. Thank you! -slashp

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  • Trixbox: external SIP with no sound

    - by Leandro Vidal
    I have a trixbox server and every works find except the external SIPs. Inside net all sound goes fine, but if I use a SIP phone outside the net, I can connect, I can receive calls but I there is no sound. I have this text in the sip_nat.conf: nat=yes externhost=xxxxx.dyndns.org localnet=192.168.1.0/255.255.255.0 localhost=192.168.1.210 externrefresh=10 qualify=yes And I have the ports from 5036 to 5082, 4569 and from 10000 to 20000 redirected to 192.168.1.210 on TCP and UDP. What's wrong? Thank you very much in advance

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  • Trixbox CentOS Default GW Problem (Multi-homed server)

    - by slashp
    I'm having an issue with a CentOS trixbox server which is dual-homed (one private facing NIC [eth1], one internet-facing NIC [eth0]). I can't seem to get the default gateway to set properly to our ISP's GW via eth0. I've modified the /etc/sysconfig/network to contain both a GATEWAY & GATEWAYDEV line and removed the GATEWAY line from /etc/sysconfig/network-scripts/ifcfg-eth1 (as well as /etc/sysconfig/network-scripts/ifcfg-eth0). No default GW shows up in the routing table unless it's specified in the ifcfg-eth1 file (which both the wrong interface and wrong gateway IP), otherwise, the routing table simply does not contain a default gateway..any ideas would be greatly appreciated! Thanks! EDIT Just realized when attempting to add the default gateway manually using the route add command, I receive an error stating: SIOCADDRT: Network is unreachable I know this error can occur when your default gateway and interface IP address are not on the same subnet..in this case, my public IP address of eth0 is a /29.

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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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  • How do I configure additional phone lines asterisk/trixbox?

    - by Matt
    I have a 4 port Digium card in there, and have 4 lines running smoothly. Now, we added ANOTHER 4 port card and have 4 more analog lines coming into the Trixbox server. It still runs the 4 fine, but what do I need to do to add the additional 4 phone numbers/lines? I want it to act exactly as before, there's nothing special about the new lines. We just need more lines so that when we have 4 out of state customers call, we can have 4 more call and not get the busy signal. Trixbox CE 2.8

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  • Tracing spoofed mobile phone numbers

    - by RaDeuX
    I am being harassed by some prank caller that is spoofing his/her number neither T-Mobile nor the police can do anything about it. I have been told from one of my friends that if I set up an Asterisk server, I can accomplish the tracing of the prank caller. I am hardly knowledgeable in terms of networking, so a lot of what they told me was filled with jargon I couldn't really understand. But first things first, I downloaded Asterisk 1.5.0 and was finally able to install it (had issues with partitioning... In the end I just had Asterisk hog the entire HDD space). I tried out Asterisk, and it was slightly complicated for me so I decided to install trixbox 2.8.0.4 instead. It looks very similar to Asterisk... I'm not entirely sure what to do from here. I know I have to get the server up and running, but do I need a PBX card or something to accomplish what I'm trying to do? I'm running trixbox on a laptop to minimize electricity usage. Also, will I have to open any ports for the server? I have limited administrative permissions because of my father who is very uncomfortable with opening ports.

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  • Sonicwall settings for Polycom TFTP

    - by Michael Glenn
    I'm switching our VoIP phones (Polycom 301s and 501s) to our data network. They were previously segmented to their own network. This means disabling the DHCP on the Trixbox (Asterisk) server and configuring the Sonicwall TZ 210 DHCP to indicate that Trixbox is the TFTP server. The Polycom phones are stating "could not contact boot server". All phones are configured to TFTP and were confirmed working when previously using the Trixbox server for DHCP. Trixbox DHCP is now turned off. I've configured options 66(as String), 128(as IP) and 150(as IP) in DHCP and added them to a TFTP Option Group. I've enabled "Allow BOOTP Clients to use Range" for the Dynamic IP range and assigned the Option Group TFTP as the DHCP Generic Option Group. Any idea what I'm missing? Is there a separate tool to inspect the DHCP response to compare Trixbox to the Sonicwall?

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  • IP telephony open source systems

    - by danke
    I'm trying to pick an IP telephony technology to learn. I heard of Asterisk, trixbox, freePBX, and my head was already spinning being not sure what to learn. Then I came across this article listing some more like Kamailio, Yate, CallWeaver, FreeSWITCH, SipXecs and now my head REALLY is spinning http://www.cio.com.au/article/323016/five_open_source_ip_telephony_projects_watch . Can someone give me a run down of how all these technologies tie together? What is the trend now, because I'd like to start learning. Note: Anyone please re-tag this question if you know better, because I'm new to this field and not sure about the best tags.

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  • Provisioning SIP Phones over the internet

    - by Jorge Fernandez
    I have a few SIP Phones that are located of site and connect to my PBX over the internet to make calls. For some reason one of these phones has become unprovisioned. In my office phones get provisioned by the server via TFTP. The ones that I have off site I pre-provisioned manually before I sent them off-site (I'm in Florida the phone is in New Jersey). Whats the best way to provision these over the internet? TFTP is very insecure. Sending the plain text profiles with the SIP Account and Password over the internet is out of the question. The phones have been off-site for about 6 months without any issues. Im using Trixbox and Cisco 7940 Phones.

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