Hello,

I am just curius if this is normal Trixbox behaviour:

asterisk1*CLI> show channels
Channel Location State Application(Data)
Zap/1-1 s@from-pstn:1 Up Bridged Call(SIP/2137-b79169c0
SIP/2137-b79169c0 s@macro-dialout-trun Up Dial(ZAP/g0/18772344xxx|300|)
Zap/5-1 s@from-pstn:1 Up Bridged Call(SIP/2128-b720e208
SIP/2128-b720e208 s@macro-dialout-trun Up Dial(ZAP/g0/18669104xxx|300|)
4 active channels
2 active calls

Is it normal that if 2 calls is make 4 channels is occupying? I thoght one call = one channel. So if we use T1 = 23 channels only 11 calls can be made? It can’t be like that.
I assume I am wrong. Can someone explain?

Thank You,

MST

Hello, and thanks for the help!

I am trying to setup Trixbox to pass DTMF tones straight through to an analog extension port, without regenerating or touching the DTMF tones at all.

I have a an associate who tells me this is possible, but isn’t familiar enough with Trixbox specifically to tell me what file needs to be edited to make this happen.

(We are trying to use DID lines for an alarm receiver, and we think we have determined that Asterisk is fouling up the DTMF tones when regenerating them, making them unreadable to the receiver…)

Anyway, if anyone can tell me how to get Trixbox to pass DTMF tones straight through, let me know!

Thanks

Hello,

I belive I found a bug in the app_tx module but I do not know how to report it. When a fax does not go through because it called an incorrect phone number that turns out to be answered by a person or an answering machine instead of a fax machine, the result it passes back is passed when it should be failed. However, the module does indeed detect that it failed, just the result code is 0 instead of a failed indicator of a negative number. Here is a log snippet to demonstrate:

[Oct 27 08:40:51] DEBUG[28616] app_txfax.c: ==============================================================================
[Oct 27 08:40:51] DEBUG[28616] app_txfax.c: Fax send not successful - result (51) The call dropped prematurely.
[Oct 27 08:40:51] DEBUG[28616] app_txfax.c: ==============================================================================
[Oct 27 08:40:51] DEBUG[28616] app_txfax.c: execResult=0

How should I report this bug or what can I do to fix it?

Thanks!!
Ryan

Hello, could anyone help me to add or configure a X-Lite softphone in a GXE5024, i’m doing a work and i dont know how to do it, i need make calls from my X-Lite softphone with another one thought a GXE5024.. thanks in forward

Hello,

I successfully (I think) installed and configured FreeTDS using instructions found at a few websites including FreeTDS’s, unixODBC’s.

I have trixbox 2.2.4 (CentOS 4.7), and downloaded/installed freetds 0.82. I’m trying to connect to a MS SQL Sever 2000 machine (running Windows 2000 Server) via ODBC. I can connect to it using tsql, but it fails when trying to use isql, it fails.
My isql command line:

isql -v MSSQLServer myuser mypass

I get:

[IM004][unixODBC][Driver Manager]Driver’s SQLAllocHandle on SQL_HANDLE_HENV failed
[ISQL]ERROR: Could not SQLConnect

Let me know and I can provide other info.

Hello

I have 400 devices (Cisco877) in NCM And I need to run a command script at the

400 devices at the same time.

Hello

     I'm running Orion 9.1 SP3.  I'm monitoring ~200 Cisco routers.  I'm trying to figure out if there's a way I can get a report which will show me where daily traffic flows are growing among those routers.  Overall traffic flow incoming from our ISP recently grew suddenly a bit and I want to see if that appears to be from a particular site(s) with increasing traffic or if it's just an effect of traffic growing among all our sites overall.

    At first thought this seems like an obvious kind of question to ask from Orion and all it's gathered data.  However, as I start looking at the canned reports and also at the custom report writer, it seems to me there is no straightforward way to try and put something like that together, if at all.

     Either that or I am completely blowing it for due dilligence.  Which I won't swear to never suffering from :)  Thank you for any advice or pointers anyone has time to give!

-John

Hello to everyone,

i have buy 3 Grandstream BT-200, just to try it , after a new installation of Trixbox 2.2.4 ….. i create 3 new extensions an and i receive from asterisk console :

Name/username Host Dyn Nat ACL Port Status
104/104 192.168.0.94 D N 5060 OK (25 ms)
103/103 192.168.0.93 D N 5060 OK (26 ms)
102/102 192.168.0.90 D N 5060 OK (6 ms)

Example:
I make a call from 102 to 103, after pickup i want transfer this call to 104 …..
102 -> call -> 103
103 -> FLASH -> 104
after to complete i push TRANSFER
now 102 -> talk to -> 104

but i receive on display CALL ON HOLD and my BT-200 ring !!!! after pickup i see on screen 503 …. this is an server side error (503 Service unavailable)

I have another pc where i have installed asterisk 1.4 and freepbx 2.5.0 and this problem dont appear !!!!

PLS HELP ME !!!

TNX
SAT

—————————————————————————————————–

    -- Executing Macro("SIP/102-0859a568", "exten-vm|103|103") in new stack
    -- Executing Macro("SIP/102-0859a568", "user-callerid") in new stack
    -- Executing NoOp("SIP/102-0859a568", "user-callerid: device 102") in new stack
    -- Executing GotoIf("SIP/102-0859a568", "0?report") in new stack
    -- Executing GotoIf("SIP/102-0859a568", "0?start") in new stack
    -- Executing Set("SIP/102-0859a568", "REALCALLERIDNUM=102") in new stack
    -- Executing NoOp("SIP/102-0859a568", "REALCALLERIDNUM is 102") in new stack
    -- Executing Set("SIP/102-0859a568", "AMPUSER=102") in new stack
    -- Executing Set("SIP/102-0859a568", "AMPUSERCIDNAME=Commerciale") in new stack
    -- Executing GotoIf("SIP/102-0859a568", "0?report") in new stack
    -- Executing Set("SIP/102-0859a568", "CALLERID(all)=Commerciale <102>") in new stack
    -- Executing Set("SIP/102-0859a568", "REALCALLERIDNUM=102") in new stack
    -- Executing NoOp("SIP/102-0859a568", "TTL:  ARG1: 103") in new stack
    -- Executing GotoIf("SIP/102-0859a568", "0?continue") in new stack
    -- Executing Set("SIP/102-0859a568", "__TTL=64") in new stack
    -- Executing GotoIf("SIP/102-0859a568", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,21)
    -- Executing NoOp("SIP/102-0859a568", "Using CallerID "Commerciale" <102>") in new stack
    -- Executing Set("SIP/102-0859a568", "FROMCONTEXT=exten-vm") in new stack
    -- Executing Set("SIP/102-0859a568", "VMBOX=103") in new stack
    -- Executing Set("SIP/102-0859a568", "EXTTOCALL=103") in new stack
    -- Executing Set("SIP/102-0859a568", "CFUEXT=") in new stack
    -- Executing Set("SIP/102-0859a568", "CFBEXT=") in new stack
    -- Executing Set("SIP/102-0859a568", "RT=15") in new stack
    -- Executing Macro("SIP/102-0859a568", "record-enable|103|IN") in new stack
    -- Executing GotoIf("SIP/102-0859a568", "0?2:4") in new stack
    -- Goto (macro-record-enable,s,4)
    -- Executing DeadAGI("SIP/102-0859a568", "recordingcheck|20090415-124948|1239792588.12") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
  recordingcheck|20090415-124948|1239792588.12: Inbound recording not enabled
    -- AGI Script recordingcheck completed, returning 0
    -- Executing NoOp("SIP/102-0859a568", "No recording needed") in new stack
    -- Executing Macro("SIP/102-0859a568", "dial|15|tr|103") in new stack
    -- Executing DeadAGI("SIP/102-0859a568", "dialparties.agi") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
  dialparties.agi: Starting New Dialparties.agi
  == Parsing '/etc/asterisk/manager.conf': Found
  == Parsing '/etc/asterisk/manager_custom.conf': Found
  == Manager 'admin' logged on from 127.0.0.1
  dialparties.agi: priority is 1
  dialparties.agi: Caller ID name is 'Commerciale' number is '102'
  dialparties.agi: Methodology of ring is  'none'
       >  dialparties.agi: USE_CONFIRMATION:  'FALSE'
       >  dialparties.agi: RINGGROUP_INDEX:   ''
    --  dialparties.agi: Added extension 103 to extension map
    --  dialparties.agi: Extension 103 cf is disabled
    --  dialparties.agi: Extension 103 do not disturb is disabled
       >  dialparties.agi: extnum: 103
       >  dialparties.agi: exthascw: 1
       >  dialparties.agi: exthascfb: 0
       >  dialparties.agi: extcfb:
       >  dialparties.agi: exthascfu: 0
       >  dialparties.agi: extcfu:
    --  dialparties.agi: dbset CALLTRACE/103 to 102
  == Manager 'admin' logged off from 127.0.0.1
    -- AGI Script dialparties.agi completed, returning 0
    -- Executing Dial("SIP/102-0859a568", "SIP/103|15|tr") in new stack
    -- Called 103
    -- SIP/103-085a3e70 is ringing
    -- SIP/103-085a3e70 answered SIP/102-0859a568
    -- Music class default requested but no musiconhold loaded.
    -- Executing Macro("SIP/103-085b30f8", "exten-vm|104|104") in new stack
    -- Executing Macro("SIP/103-085b30f8", "user-callerid") in new stack
    -- Executing NoOp("SIP/103-085b30f8", "user-callerid: device 103") in new stack
    -- Executing GotoIf("SIP/103-085b30f8", "0?report") in new stack
    -- Executing GotoIf("SIP/103-085b30f8", "0?start") in new stack
    -- Executing Set("SIP/103-085b30f8", "REALCALLERIDNUM=103") in new stack
    -- Executing NoOp("SIP/103-085b30f8", "REALCALLERIDNUM is 103") in new stack
    -- Executing Set("SIP/103-085b30f8", "AMPUSER=103") in new stack
    -- Executing Set("SIP/103-085b30f8", "AMPUSERCIDNAME=INT Mirko") in new stack
    -- Executing GotoIf("SIP/103-085b30f8", "0?report") in new stack
    -- Executing Set("SIP/103-085b30f8", "CALLERID(all)=INT Mirko <103>") in new stack
    -- Executing Set("SIP/103-085b30f8", "REALCALLERIDNUM=103") in new stack
    -- Executing NoOp("SIP/103-085b30f8", "TTL:  ARG1: 104") in new stack
    -- Executing GotoIf("SIP/103-085b30f8", "0?continue") in new stack
    -- Executing Set("SIP/103-085b30f8", "__TTL=64") in new stack
    -- Executing GotoIf("SIP/103-085b30f8", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,21)
    -- Executing NoOp("SIP/103-085b30f8", "Using CallerID "INT Mirko" <103>") in new stack
    -- Executing Set("SIP/103-085b30f8", "FROMCONTEXT=exten-vm") in new stack
    -- Executing Set("SIP/103-085b30f8", "VMBOX=104") in new stack
    -- Executing Set("SIP/103-085b30f8", "EXTTOCALL=104") in new stack
    -- Executing Set("SIP/103-085b30f8", "CFUEXT=") in new stack
    -- Executing Set("SIP/103-085b30f8", "CFBEXT=") in new stack
    -- Executing Set("SIP/103-085b30f8", "RT=15") in new stack
    -- Executing Macro("SIP/103-085b30f8", "record-enable|104|IN") in new stack
    -- Executing GotoIf("SIP/103-085b30f8", "0?2:4") in new stack
    -- Goto (macro-record-enable,s,4)
    -- Executing DeadAGI("SIP/103-085b30f8", "recordingcheck|20090415-124957|1239792597.14") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
  recordingcheck|20090415-124957|1239792597.14: Inbound recording not enabled
    -- AGI Script recordingcheck completed, returning 0
    -- Executing NoOp("SIP/103-085b30f8", "No recording needed") in new stack
    -- Executing Macro("SIP/103-085b30f8", "dial|15|tr|104") in new stack
    -- Executing DeadAGI("SIP/103-085b30f8", "dialparties.agi") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
  dialparties.agi: Starting New Dialparties.agi
  == Parsing '/etc/asterisk/manager.conf': Found
  == Parsing '/etc/asterisk/manager_custom.conf': Found
  == Manager 'admin' logged on from 127.0.0.1
  dialparties.agi: priority is 1
  dialparties.agi: Caller ID name is 'INT Mirko' number is '103'
  dialparties.agi: Methodology of ring is  'none'
       >  dialparties.agi: USE_CONFIRMATION:  'FALSE'
       >  dialparties.agi: RINGGROUP_INDEX:   ''
    --  dialparties.agi: Added extension 104 to extension map
    --  dialparties.agi: Extension 104 cf is disabled
    --  dialparties.agi: Extension 104 do not disturb is disabled
       >  dialparties.agi: extnum: 104
       >  dialparties.agi: exthascw: 1
       >  dialparties.agi: exthascfb: 0
       >  dialparties.agi: extcfb:
       >  dialparties.agi: exthascfu: 0
       >  dialparties.agi: extcfu:
    --  dialparties.agi: dbset CALLTRACE/104 to 103
  == Manager 'admin' logged off from 127.0.0.1
    -- AGI Script dialparties.agi completed, returning 0
    -- Executing Dial("SIP/103-085b30f8", "SIP/104|15|tr") in new stack
    -- Called 104
    -- SIP/104-085bbbe8 is ringing
    -- SIP/104-085bbbe8 answered SIP/103-085b30f8
    -- Music class default requested but no musiconhold loaded.
  == Spawn extension (macro-dial, s, 10) exited non-zero on 'SIP/103-085b30f8<ZOMBIE>' in macro 'dial'
  == Spawn extension (macro-dial, s, 10) exited non-zero on 'SIP/103-085b30f8<ZOMBIE>' in macro 'exten-vm'
  == Spawn extension (macro-dial, s, 10) exited non-zero on 'SIP/103-085b30f8<ZOMBIE>'
    -- Executing Macro("SIP/103-085b30f8<ZOMBIE>", "hangupcall") in new stack
    -- Executing ResetCDR("SIP/103-085b30f8<ZOMBIE>", "w") in new stack
    -- Executing NoCDR("SIP/103-085b30f8<ZOMBIE>", "") in new stack
    -- Executing GotoIf("SIP/103-085b30f8<ZOMBIE>", "1?skiprg") in new stack
    -- Goto (macro-hangupcall,s,6)
    -- Executing GotoIf("SIP/103-085b30f8<ZOMBIE>", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,9)
    -- Executing Wait("SIP/103-085b30f8<ZOMBIE>", "5") in new stack
  == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/103-085b30f8<ZOMBIE>' in macro 'hangupcall'
  == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/103-085b30f8<ZOMBIE>'
    -- Incoming call: Got SIP response 400 "Bad Request" back from 192.168.0.93
    -- Music class default requested but no musiconhold loaded.
  == Spawn extension (macro-dial, s, 10) exited non-zero on 'SIP/102-0859a568' in macro 'dial'
  == Spawn extension (macro-dial, s, 10) exited non-zero on 'SIP/102-0859a568' in macro 'exten-vm'
  == Spawn extension (macro-dial, s, 10) exited non-zero on 'SIP/102-0859a568'
    -- Executing Macro("SIP/102-0859a568", "hangupcall") in new stack
    -- Executing ResetCDR("SIP/102-0859a568", "w") in new stack
    -- Executing NoCDR("SIP/102-0859a568", "") in new stack
    -- Executing GotoIf("SIP/102-0859a568", "1?skiprg") in new stack
    -- Goto (macro-hangupcall,s,6)
    -- Executing GotoIf("SIP/102-0859a568", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,9)
    -- Executing Wait("SIP/102-0859a568", "5") in new stack
  == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/102-0859a568' in macro 'hangupcall'
  == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/102-0859a568'

Hello,

I am just thowing it out there to see if there is a simpler solution to have a service whereby when a voice mail message is placed, it notifies my mobile phone by SMS.

I found this post here:

http://www.trixbox.org/forums/trixbox-forums/trixbox-projects/vm-…

however, it is a couple of years old, and it looks extremely difficult.

If anyone knows of a simple method, could you buzz me back.

Kind regards,
Anthony

Hello,

Just wondering if there is a way to have a incoming call to ring on 2 extensions at the same time?

I notice under “Incoming Call” –> “Set Destination”, it only allows you to select 1 extension.

Thanks for any help.

Kind regards,
Anthony

Hello,
I tried working with Kamillio (OpenSER) for this but this is way beyond my expertise. I have no clue how to work with openser and asterisk. So I thought of another idea. For failover and load balancing, I can use DNS round robin to take care of this. However, I need to ask for some help.
Can I take multiple trixbox machines (I would like to start with two) and setup a cron job to copy the SQL database from one machine to another every few minutes? If this is done, will everything in the FreePBX GUI copy over, and will asterisk automatically update the new settings?

If all this can be done without issues, this would be great. Please let me know how this can be done and if there are any sample scripts I can use
Thanks

Hello,

I’ve just installed the current trixbox-Version; calling from and to outside via sipgate (germany) works well; internal calls too.

But I have set up an ring group with an external member (mobile phone)

When calling to this ring group, the CallerID of my sipgate-Account is shown and not the origin-callers-id.

Sipgate allows me to set the caller id, but how can I set it “dynamically” for forwarded calls?

Any idea?

Regards,

Philipp

Hello,

I am just wondering how I would set up my trixbox so that I can dial the following numbers?

13 xxxx
1300 xxx xxx
1800 xxx xxx

I live in Australia, and the country cose is +61

I currently have my “Trunk” set up so that nothing is in the “Dial Rules” field.

I have my “Dial Pattern” in my “Outbound Route” set up as:

|.

Kind regards,
Anthony