Bug en CentOS 5.2 y zaptel

Al intentar compilar los módulos de zaptel en CentOS 5.2 se produce un error. La descripción del mismo y su solución pueden verla en este enlace.

Asterisk Detailed Variable List

Asterisk standard channel variables

There are a number of variables that are defined or read by Asterisk. Here is a list of them. More information is available in each application's help text. All these variables are in UPPER CASE only. Variables marked with a * are builtin functions and can't be set, only read in the dialplan. Writes to such variables are silently ignored.
${ACCOUNTCODE} * Account code (if specified) (Deprecated; use ${CDR(accountcode)})

${BLINDTRANSFER} The name of the channel on the other side of a blind transfer
${BRIDGEPEER} Bridged peer
${CALLERANI} * Caller ANI (PRI channels) (Deprecated; use ${CALLERID(ani)})
${CALLERID} * Caller ID (Deprecated; use ${CALLERID(all)})
${CALLERIDNAME} * Caller ID Name only (Deprecated; use ${CALLERID(name)}) ${CALLERIDNUM} * Caller ID Number only (Deprecated; use ${CALLERID(num)})
${CALLINGANI2} * Caller ANI2 (PRI channels)
${CALLINGPRES} * Caller ID presentation for incoming calls (PRI channels)
${CALLINGTNS} * Transit Network Selector (PRI channels)
${CALLINGTON} * Caller Type of Number (PRI channels)
${CHANNEL} * Current channel name
${CONTEXT} * Current context
${DATETIME} * Current date time in the format: DDMMYYYY-HH:MM:SS (Deprecated; use ${STRFTIME(${EPOCH},,%d%mNaVH:NaVS)})

${DB_RESULT} Result value of DB_EXISTS() dial plan function
${DNID} * Dialed Number Identifier (Deprecated; use ${CALLERID(dnid)})
${EPOCH} * Current unix style epoch
${EXTEN} * Current extension
${ENV(VAR)} Environmental variable VAR
${GOTO_ON_BLINDXFR} Transfer to the specified context/extension/priority after a blind transfer (use ^ characters in place of to separate context/extension/priority when setting this variable from the dialplan)
${HANGUPCAUSE} * Asterisk cause of hangup (inbound/outbound)
${HINT} * Channel hints for this extension
${HINTNAME} * Suggested Caller*ID name for this extension
${INVALID_EXTEN} The invalid called extension (used in the "i" extension)
${LANGUAGE} * Current language (Deprecated; use ${LANGUAGE()})
${LEN(VAR)} * String length of VAR (integer)
${PRIORITY} * Current priority in the dialplan
${PRIREDIRECTREASON} Reason for redirect on PRI, if a call was directed
${RDNIS} * Redirected Dial Number ID Service (Deprecated; use ${CALLERID(rdnis)}) ${TIMESTAMP} * Current date time in the format: YYYYMMDD-HHMMSS (Deprecated; use ${STRFTIME(${EPOCH},,%Y%mNaVH%M%S)})
${TRANSFER_CONTEXT} Context for transferred calls
${FORWARD_CONTEXT} Context for forwarded calls
${UNIQUEID} * Current call unique identifier
${SYSTEMNAME} * value of the systemname option of asterisk.conf Application return values
In Asterisk 1.2, many applications return the result in a variable instead of, as in Asterisk 1.0, changing the dial plan priority (+101). For the various status values, see each application's help text. ${AGISTATUS} * agi()
${AQMSTATUS} * addqueuemember()
${AVAILSTATUS} * chanisavail()
${CHECKGROUPSTATUS} * checkgroup()
${CHECKMD5STATUS} * checkmd5()
${CPLAYBACKSTATUS} * controlplayback()
${DIALSTATUS} * dial() - see also ${HANGUPCAUSE}
${DBGETSTATUS} * dbget()
${ENUMSTATUS} * enumlookup()
${HASVMSTATUS} * hasnewvoicemail()
${LOOKUPBLSTATUS} * lookupblacklist()
${OSPAUTHSTATUS} * ospauth()
${OSPLOOKUPSTATUS} * osplookup()
${OSPNEXTSTATUS} * ospnext()
${OSPFINISHSTATUS} * ospfinish()
${PARKEDAT} * parkandannounce()
${PLAYBACKSTATUS} * playback()
${PQMSTATUS} * pausequeuemember()
${PRIVACYMGRSTATUS} * privacymanager()
${QUEUESTATUS} * queue()
${RQMSTATUS} * removequeuemember()
${SENDIMAGESTATUS} * sendimage()
${SENDTEXTSTATUS} * sendtext()
${SENDURLSTATUS} * sendurl()
${SYSTEMSTATUS} * system()
${TRANSFERSTATUS} * transfer()
${TXTCIDNAMESTATUS} * txtcidname()
${UPQMSTATUS} * unpausequeuemember()
${VMSTATUS} * voicmail()
${VMBOXEXISTSSTATUS} * vmboxexists()
${WAITSTATUS} * waitforsilence() Various application variables
${CURL} * Resulting page content for curl()
${ENUM} * Result of application EnumLookup
${EXITCONTEXT} Context to exit to in IVR menu (app background()) or in the RetryDial() application ${MONITOR} * Set to "TRUE" if the channel is/has been monitored (app monitor()) ${MONITOR_EXEC} Application to execute after monitoring a call
${MONITOR_EXEC_ARGS} Arguments to application
${MONITOR_FILENAME} File for monitoring (recording) calls in queue
${QUEUE_PRIO} Queue priority
${QUEUE_MAX_PENALTY} Maximum member penalty allowed to answer caller
${QUEUESTATUS} Status of the call, one of: (TIMEOUT FULL JOINEMPTY LEAVEEMPTY JOINUNAVAIL LEAVEUNAVAIL)
${RECORDED_FILE} * Recorded file in record()
${TALK_DETECTED} * Result from talkdetect()
${TOUCH_MONITOR} The filename base to use with Touch Monitor (auto record) ${TOUCH_MONITOR_FORMAT} The audio format to use with Touch Monitor (auto record) ${TOUCH_MONITOR_OUTPUT} * Recorded file from Touch Monitor (auto record)
${TXTCIDNAME} * Result of application TXTCIDName
${VPB_GETDTMF} chan_vpb The MeetMe Conference Bridge uses the following variables:
${MEETME_RECORDINGFILE} Name of file for recording a conference with the "r" option ${MEETME_RECORDINGFORMAT} Format of file to be recorded
${MEETME_EXIT_CONTEXT} Context for exit out of meetme meeting ${MEETME_AGI_BACKGROUND} AGI script for Meetme (zap only)
${MEETMESECS} * Number of seconds a user participated in a MeetMe conference The VoiceMail() application uses the following variables:
${VM_CATEGORY} Sets voicemail category
${VM_NAME} * Full name in voicemail
${VM_DUR} * Voicemail duration
${VM_MSGNUM} * Number of voicemail message in mailbox
${VM_CALLERID} * Voicemail Caller ID (Person leaving vm)
${VM_CIDNAME} * Voicemail Caller ID Name
${VM_CIDNUM} * Voicemail Caller ID Number
${VM_DATE} * Voicemail Date
${VM_MESSAGEFILE} * Path to message left by caller The VMAuthenticate() application uses the following variables:
${AUTH_MAILBOX} * Authenticated mailbox
${AUTH_CONTEXT} * Authenticated mailbox context DUNDiLookup() uses the following variables
${DUNDTECH} * The Technology of the result from a call to DUNDiLookup()
${DUNDDEST} * The Destination of the result from a call to DUNDiLookup() The Zaptel channel sets the following variables:
${ANI2} * The ANI2 Code provided by the network on the incoming call. (ie, Code 29 identifies call as a Prison/Inmate Call) See also: NANPA ANI II Digits Assignments
${CALLTYPE} * Type of call (Speech, Digital, etc)
${CALLEDTON} * Type of number for incoming PRI extension i.e. 0=unknown, 1=international, 2=domestic, 3=net_specific, 4=subscriber, 6=abbreviated, 7=reserved
${CALLINGSUBADDR} * Called PRI Subaddress
${FAXEXTEN} * The extension called before being redirected to "fax"
${PRIREDIRECTREASON} * Reason for redirect, if a call was directed
${SMDI_VM_TYPE} * When an call is received with an SMDI message, the 'type' of message 'b' or 'u' The SIP channel uses the following variables:
${SIPCALLID} * SIP Call-ID: header verbatim (for logging or CDR matching)
${SIPDOMAIN} * SIP destination domain of an inbound call (if appropriate)
${SIPUSERAGENT} * SIP user agent
${SIPURI} * SIP uri
${SIP_CODEC} Set the SIP codec for a call
${SIP_URI_OPTIONS} * additional options to add to the URI for an outgoing call
${RTPAUDIOQOS} RTCP QoS report for the audio of this call
${RTPVIDEOQOS} RTCP QoS report for the video of this call The Agent channel uses the following variables:
${AGENTMAXLOGINTRIES} Set the maximum number of failed logins
${AGENTUPDATECDR} Whether to update the CDR record with Agent channel data ${AGENTGOODBYE} Sound file to use for "Good Bye" when agent logs out
${AGENTACKCALL} Whether the agent should acknowledge the incoming call ${AGENTAUTOLOGOFF} Auto logging off for an agent
${AGENTWRAPUPTIME} Setting the time for wrapup between incoming calls
${AGENTNUMBER} * Agent number (username) set at login
${AGENTSTATUS} * Status of login ( fail on off )
${AGENTEXTEN} * Extension for logged in agent The Dial() application uses the following variables:
${DIALEDPEERNAME} * Dialed peer name
${DIALEDPEERNUMBER} * Dialed peer number
${DIALEDTIME} * Time for the call (seconds)
${ANSWEREDTIME} * Time from dial to answer (seconds)
${DIALSTATUS} * Status of the call, one of: (CHANUNAVAIL CONGESTION BUSY NOANSWER ANSWER CANCEL DONTCALL TORTURE)
${DYNAMIC_FEATURES} * The list of features (from the applicationmap section of features.conf) to activate during the call, with feature names separated by '#' characters ${LIMIT_PLAYAUDIO_CALLER} Soundfile for call limits
${LIMIT_PLAYAUDIO_CALLEE} Soundfile for call limits
${LIMIT_WARNING_FILE} Soundfile for call limits
${LIMIT_TIMEOUT_FILE} Soundfile for call limits
${LIMIT_CONNECT_FILE} Soundfile for call limits
${OUTBOUND_GROUP} Default groups for peer channels (as in SetGroup)
See "show application dial" for more information The chanisavail() application sets the following variables:
${AVAILCHAN} * the name of the available channel if one was found
${AVAILORIGCHAN} * the canonical channel name that was used to create the channel ${AVAILSTATUS} * Status of requested channel When using macros in the dialplan, these variables are available
${MACRO_EXTEN} * The calling extensions
${MACRO_CONTEXT} * The calling context
${MACRO_PRIORITY} * The calling priority
${MACRO_OFFSET} Offset to add to priority at return from macro The ChanSpy() application uses the following variables:
${SPYGROUP} * A ':' (colon) separated list of group names. (To be set on spied on channel and matched against the g(grp) option) If you compile with OSP support, these variables are used:
${OSPINHANDLE} OSP handle of in_bound call
${OSPINTIMELIMIT} Duration limit for in_bound call
${OSPOUTHANDLE} OSP handle of out_bound call
${OSPTECH} OSP technology
${OSPDEST} OSP destination
${OSPCALLING} OSP calling number
${OSPOUTTOKEN} OSP token to use for out_bound call
${OSPOUTTIMELIMIT} Duration limit for out_bound call
${OSPRESULTS} Number of remained destinations CDR Variables

If the channel has a cdr, that cdr record has it's own set of variables which can be accessed just like channel variables. The following builtin variables are available and, unless specified, read-only. ${CDR(clid)} Caller ID
${CDR(src)} Source
${CDR(dst)} Destination
${CDR(dcontext)} Destination context
${CDR(channel)} Channel name
${CDR(dstchannel)} Destination channel
${CDR(lastapp)} Last app executed
${CDR(lastdata)} Last app's arguments
${CDR(start)} Time the call started.
${CDR(answer)} Time the call was answered.
${CDR(end)} Time the call ended.
${CDR(duration)} Duration of the call.
${CDR(billsec)} Duration of the call once it was answered.
${CDR(disposition)} ANSWERED, NO ANSWER, BUSY
${CDR(amaflags)} DOCUMENTATION, BILL, IGNORE etc
${CDR(accountcode)} The channel's account code (read-write).
${CDR(uniqueid)} The channel's unique id.
${CDR(userfield)} The channels uses specified field (read-write).
In addition, you can set your own extra variables with a traditional Set(CDR(var)=val) to anything you want. NOTE Some CDR values (eg: duration & billsec) can't be accessed until the call has terminated. As of 91617, those values will be calculated on-demand if requested. Until that makes it into a stable release, you can set endbeforehexten=yes in cdr.conf, and then use the "hangup" context to wrap up your call. Certain functional variables may be accessed with ${foo()}. A list of these functional variables may be found by typing "show functions" at the Asterisk CLI.

DUNDi Enterprise Configuration SIP

This is all you should need to setup DUNDi between two boxes with SIP.
You can add more boxes to your network in a similar way.

(Based on configurations in the configs directory of the Asterisk source code and documents published by Brian K. West aka bkw_)
DUNDi Enterprise Configuration IAX

Note: This configuration is currently non-functional, as chan_sip does not support "dbsecret" at this time. You can create a configuration where the passwords are actually included in the DUNDi mappings or in the peer definitions. A patch to support dbsecret in chan_sip is forthcoming (for CVS HEAD only).

Things to keep in mind:

• Unless you have specifically listed a host in your sip.conf the call will come in on the context defined in the [general] section by context. You may want to include the [dundi-priv-local] in this context.

On both boxes in your extensions.conf:

; Private DUNDi network
[dundi-priv-canonical]
; Direct numbers

[dundi-priv-customers]
; If you are an ITSP or Reseller, list your customers here.

[dundi-priv-via-pstn]
; If you are freely delivering calls to the PSTN, list them here

[dundi-priv-local]
include => dundi-priv-canonical
include => dundi-priv-customers
include => dundi-priv-via-pstn

[dundi-priv-switch]
; Just a wrapper for the switch
switch => DUNDi/priv

[dundi-priv-lookup]
include => dundi-priv-local
include => dundi-priv-switch

[macro-dundi-priv]
exten => s,1,Goto(${ARG1},1)
include => dundi-priv-lookup

---

sip.conf on both boxes:

[priv]
type=user
dbsecret=dundi/secret
context=dundi-priv-local

---

dundi.conf on both boxes under mappings:
In many cases you will need to replace ${IPADDR} with your local IP address

priv => dundi-priv-canonical,0,SIP,${IPADDR}/${NUMBER},nopartial
priv => dundi-priv-customers,100,SIP,${IPADDR}/${NUMBER},nopartial
priv => dundi-priv-via-pstn,400,SIP,${IPADDR}/${NUMBER},nopartial

---

now on each box cd /var/lib/asterisk/keys

astgenkey -n BOXNAMEHERE

Press enter do not put a password on the keys unless you want to init
keys every time you start asterisk.

Now put exchange public keys between the boxes.



Box A dundi.conf:

[DE:AD:BE:EF:DE:AD]   <-- EID/MAC from BOX B model = symmetric host = boxb.domain.com inkey = BOXB   <- BOX B's public key outkey = BOXA  <- BOX A's private key include = priv permit = priv qualify = yes order = primary 

Box B dundi.conf:

[BE:EF:DE:AD:BE:EF]  <-- EID/MAC from BOX A model = symmetric host = boxa.domain.com inkey = BOXA   <- BOX A's public key outkey = BOXB  <- BOX B's private key include = priv permit = priv qualify = yes order = primary 

Now you can do this in the context which your devices dialout:

exten => _91NXXNXXXXXX,1,Macro(dundi-priv,${EXTEN:1})
exten => _91NXXNXXXXXX,2,Dial(Zap/g1/${EXTEN:1}) ; This is fall through example to a PSTN such a as PRI

And find numbers in your enterprise DUNDi network.

These same things apply to the e164 DUNDi network too.

http://www.voip-info.org/wiki/view/DUNDi+Enterprise+Configuration+SIP

DUNDi Enterprise Configuration SIP with no passwords

http://www.voip-info.org/wiki/view/DUNDi+Enterprise+Configuration+SIP+with+no+passwords

DUNDi Enterprise Configuration IAX

http://www.voip-info.org/wiki/view/DUNDi+Enterprise+Configuration+IAX

Callback for Asterisk

Callback script made to do callback if called channel is busy,

How it works

• If called channel or phone is busy, it will ask to press 5 for callback or any other key to leave voicemail,
• Upon pressing 5, it will create a callfile in /var/spool/asterisk/outgoing with a special channel to be called
• Channel provided in callfile will periodically check if the extension set to callback is busy or available. as soon as it finds the channel available it will try to establish the call.

Php Agi script

#!/usr/bin/php -q
1;

$err=fopen("php://stderr","w");
$in = fopen("php://stdin","r");
while (!feof($in)) {
$temp = str_replace("\n","",fgets($in,4096));
$s = split(":",$temp);
$agi[str_replace("agi_","",$s0)] = trim($s1);
if (($temp == "") || ($temp == "\n")) {
break;
}
}
$cf = fopen("/var/spool/asterisk/outgoing/cb" . $agi"callerid" . $to,"w+");
fputs($cf,"Channel: LOCAL/cb".$agi"callerid".$to."\n");
fputs($cf,"Context: default\n");
fputs($cf,"Extension: ".$to."\n");
fputs($cf,"CallerID: CB ".$agi"callerid"."<".$agi"callerid"."> \n");
fputs($cf,"MaxRetries: 100\n");
fputs($cf,"RetryTime: 30\n");
fclose($cf);
fclose($in);
fclose($err);
?>

Extensions.conf

• in extensions.conf add something like this

exten => s-BUSY,1,ChanIsAvail(SIP/${MACRO_EXTEN}|s)
exten => s-BUSY,2,GotoIf($"${AVAILSTATUS}"<="1"?s-NOANSWER,1)
exten => s-BUSY,3,Read(digit|callback|1)
exten => s-BUSY,4,Gotoif($ "${digit}" = "5"?callback,1:5)
exten => s-BUSY,5,Voicemail(${MACRO_EXTEN},b)
exten => s-BUSY,6,Hangup

exten => callback,1,AGI(callback|${MACRO_EXTEN})
exten => callback,2,Hangup

;in cbXXXXXX first three digits are FROM exten and last ones are TO exten
;this exten is used by CALL FILE, logic for this is to make sure extension we are calling is IDLE or NOT_INUSE state.

exten => _cbXXXXXX,1,Set(FROM=${EXTEN:2:3})
exten => _cbXXXXXX,2,Set(TO=${EXTEN:5:3})
exten => _cbXXXXXX,3,ChanIsAvail(SIP/${TO}|s)
exten => _cbXXXXXX,4,GotoIf($"${AVAILSTATUS}" <= "1"?5:7)
exten => _cbXXXXXX,5,Set(CALLERID(all)="CB ${TO} <${TO}>")
exten => _cbXXXXXX,6,Dial(SIP/${FROM}|10)
exten => _cbXXXXXX,7,Hangup

• Copy Callback script in agi-bin directory.
• Copy callback.gsm in /var/lib/asterisk/sounds/ -

Asterisk phone xten eyeBeam - How to configure video calls with Xten eyeBeam through Asterisk

History
Xten eyeBeam is one of a few SIP soft-video-phones on the market. Starting with release 1.1 3004t (16741), video calls work well with Asterisk HEAD 02/25/2005.
Update: Asterisk CVS HEAD 03/05/2005 now supports pass-thru video calls with h.263p codec between eyeBeams (potentially other video clients supporting H.263+).
General configuration See a good section here for configuring Xten X-lite, which is similar.


Video Configuration

In your sip.conf:
[general]
videosupport=yes ; enable Asterisk video support
[101] type=friend
username=101
secret=hidden
host=dynamic
context=internal
callerid=Video Phone <101>
disallow=all ; better for custom-tunning codec selection
allow=ulaw
allow=alaw
allow=gsm
allow=h263 ; H.263 is our video codec
allow=h263p ; H.263p is the enhanced video codec
dtmfmode=rfc2833 ; inband is not supported in compressed codecs like gsm, so we better set it to rfc2833
canreinvite=no ; canreinvite must be set to 'no'


TrixBox

There is a slight difference in configuring video conferencing working for TrixBox and it is as follows:
in your sip.conf:
[general]
videosupport=yes ; enable Asterisk video support

in your sip_additional.conf, (same as above):
[101] type=friend
username=101
secret=hidden
host=dynamic
context=internal
callerid=Video Phone <101>
disallow=all ; better for custom-tunning codec selection
allow=ulaw
allow=alaw
allow=gsm
allow=h263 ; H.263 is our video codec
allow=h263p ; H.263p is the enhanced video codec
dtmfmode=rfc2833 ; inband is not supported in compressed codecs like gsm, so we better set it to rfc2833
canreinvite=no ; canreinvite must be set to 'no'

By doing this you will find that video conferencing works for both extensions whereas if you implemented the above configuration you would find one or both extensions are unable to send or receive video. By inserting the above script in the sip_additional.conf file you will see that the extensions you enter will appear in FreePBX/AMP along with your other extensions.


Other prep

1. Download the latest release eyeBeam 1.11 3004w. For latest news and changelog see the thread on Xten website.


Making the video-call

1. Start eyeBeam
2. Configure SIP subscription information / calibrate audio/video input.
3. Make sure phone registered ok. Also, you should see your own video in the video 'drawer.'
4. Dial another eyeBeam user.
5. Click "Start video" button (the one in upper-left corner next to the "wrench").
6. Bingo!

PBX Screen Popper

http://www.codesonics.com

En este link se encuentra la version 1.5.0 disponible para bajar (trial de 30 dias).
Cada licencia cuesta U$S 15 (para mas de 50 licencias)

* Opens a web page to a specified URL when you receive a phone call to your Asterisk extension.
* Use all or part of the CallerID as a formfield in the URL
* Specify conditions under which the page is opened
* Uses the Asterisk Manager API
* Opens page on answer, ring or when asked
* Open page for any CallerID or if matched to a specific value, starts with, contains or ends with a specific value
* Allows you to strip characters from the start or end of CallerID

AsteriskMonitor - Current: version 3 beta

http://www.asteriskmonitor.org/

AsteriskMonitor is a FREE powerful software completely written in Java and distributed under the GPL license. AsteriskMonitor pretends to be a bridge between Asterisk and any Java-based application trying to get statistics and data from the asterisk server.

AsteriskJavaEye is a FREE web-based java application, released under the GPL license too, that connects to AsteriskMonitor and shows you what's happening NOW in your Asterisk server: who is connected to which queues, for how long today, how many calls has he/she answered, is it ringing now, etc.

AsteriskJavaStat is a FREE web-based java module for AsteriskJavaEye, released under the GPL as well, that allows you to view formatted statistics from your PBX. AsteriskJavaStat won't read the CDR; it uses the AsteriskMonitor's statistic database, which is easier to read and mantain than the CDR.

Features:

• Automatic recognition of any kind of generic event (login/logout of agents, joining/leaving queues, new calls, voicemails, recordings, dials, transfers, etc)

• Real-time updated data-model of Asterisk's state on the server

• Server accesible trought RMI, with an open API that allows you just to deal with Terminals, Agents, Calls, Queues, etc, as Java objects.

• Expandable: You can easily develope new modules for AsteriskMonitor, to recognize new kinds of events for your own profit. You just need to write it in Java, compile your single .java file and leave it in the right directory. AsteriskMonitor will load it dinamycally and start using it in seconds. No compile or restart needed.

• Inverse mode: you can also modify Asterisk's state, adding agents to queues, reloading config files, restarting Asterisk, etc.

• Works over the Asterisk's Manager Api interface. It has got an integrated connection-repeater system, which will open as many new Manager Api ports as you want, allowing you to connect multiple programs to Asterisk trought AsteriskMonitor, Asterisk won't even notice about it.

• Includes a web application (AsteriskJavaEye) you can deploy on any Tomcat engine, which will connect to AsteriskMonitor trought RMI and allow you to see any statistical data you need about your agents, and will let you add/remove them from queues, connect/disconnect them, etc.

• All the system is 100% Java-Based and independent on the machine. You can have Asterisk working in a PC and run AsteriskMonitor + AsteriskJavaEye in any other one.

• Reliable CDR: AsteriskMonitor can register its own CDR, with detailed, exact and well-documented information about call times, steps, holds, transfers, etc. It can record agent connection times, total time spent in a queue, etc. AsteriskMonitor's statistical functions are able to connect to any database which has got a jdbc connector: MySQL, Postgre, Oracle, MS SQL, etc. No compilation or restart required; just put the driver in the right directory (if it hasn't got it already), and go to see your new statistics working.

• Statistics browser module: AsteriskJavaStat is an independent module that can be loaded into the AsteriskJavaEye, and will show you nice and real statistics about your contact center.

Version information:

AsteriskMonitor is currently in a beta version. It has been proved to be stable and reliable, and it's currently being used in a production system with 20 agents answering about 500 calls per day.

About the author:

AsteriskMonitor has been enterely designed and developed by Roger de la Fuente (rdelafuente a_t ticop.com).
Currently, the AsteriskMonitor, AsteriskJavaStat and AsteriskJavaEye name are owned by TICOP (www.ticop.com), an Asterisk-Developing Company from SPAIN, and the source-codes are released under the GPL license. The program can be used for free if you just keep a link to our website in the main visual screen of any software using our's.

Downloading AsteriskMonitor

by Roger de la Fuente on Monday 11 of February, 2008 [15:13:35]

The sourceforge project is:http://sourceforge.net/projects/asteriskmonitor

In the SVN you will find the source code.

Activa brings the Asterisk IP PBX to the call center

Activa for Asterisk...
Activa brings the Asterisk IP PBX to the call center. Its components aim to enable call center implementations with functionality such as computer telephony, screenpop&click2dial, agent control, automatic dialing...
The Activa package includes an Asterisk TAPI Service Provider (TSP) we called Asterisk ActivaTSP. This TSP enables integration of TAPI third party applications and Asterisk. So for instance you could initiate outbound calls through Asterisk using your Microsoft Outlook, Microsoft Dialer, ACT!, TapiCall and many more. Call screen-popup is also supported. The TSP itself has been developed using the basic ASTProvider framework.
The package also includes a basic framework that enables integration of Asterisk with your C++ applications
Current version is 1.4.2 (Version readme)
We invite you to download the package, configure ActivaTSP and try it.

ActivaTSP FAQ

What is Activa TSP ?
Generally speaking a TAPI Service Provider (TSP) is a 'driver' that enables Microsoft TAPI to communicate to a telephony device such as a switch or modem.
So Activa Asterisk TSP enables TAPI and TAPI-compatible applications to use the Asterisk IP-PBX: Microsoft Outlook, ACT!, TapiCall, MS Dialer,...

Helius Fax

Javier mira estos emails cortos..sobre Sangoma, Siemens, faxesHay muchas variables a ver, ademas de los problemas reales con los aparatos de fax (falta de papel, roturas, etc) y las limitantes actuales entre trixbox y la Siemens. Por eso creo, que seria muy importante tener un ambiente de pruebas..!!Y no pasar las penurias en el cliente como hoy...ademas de la falta de posibilidades por el tipo de conexion. ------------------------------Siememes HiPath 3550------------------------------------------------------No son los mismos valores que para Uruguay ojo!!Muy parecido a lo egregamos hoy al DryTek..!Estava tendo alguns problemas com uma Siememes HiPath 3550. Coloquei no indications.conf [br]description = Brazilringcadence = 1000,4000dial = 425busy = 425/250,0/250ring = 425/1000,0/4000congestion = 425/250,0/250,425/750,0/250callwaiting = 425/50,0/1000 ; Dialrecall not used in Brazil standard (using UK standard) dialrecall = 350+440; Record tone is not used in Brazil, use busy tonerecord = 425/250,0/250; Info not used in Brazil standard (using UK standard) info = 950/330,1400/330,1800/330stutter = 350+440 zaptel.conf defaultzone=brloadzone=br zapata.conflanguage=br sip.conf [general] country=br E resolvi o problema. ---------------------------Fax through Sangoma------------------------------------------------------- in our company we are trying to do this: Fax <--> Traditional PBX <--> Asterisk <--> PSTN In practice, we have put an Asterisk equipped with a Sangoma A102 (2 PRI ports) between our PBX ( Siemens HiCom) and the PSTN in order to have a VoIP network along the traditional telephony network. The problem is with the fax. We just want to send and receive faxes from/to our fax machine connected to the Siemens (without needing any interaction with our VoIP network, the faxes are sent to/received from PSTN). Unfortunately we are experiencing a lot of problems: the faxes not always work and when they work, it's likely to have incomplete pages. I know that faxing with VoIP is very troublesome, but maybe someone else is using a similar configuration and he found a good configuration or maybe has some hints to improve the results. We are using Asterisk 1.2.13.-------------------------Sangoma will detect the tones and disable echo cancel---------------------Incoming faxes, the Sangoma will detect the tones and disable echo cancel. To send outbound, you will have to add another trunk group, of one or more channels and disable echo cancellation and use that to dial out. Example (/etc/asterisk/zapata.conf)blah blahechocancel=yesblah blah group = 1channel =>1-20 blah blahecho cancel=nogroup = 2channel=22-23 So you would use for faxing specifically Zap/g2/ and it will use channel 22 or 23 but with echo turned off. Use Zap/g1 and it will use the first group of channels with echo cancel on (or whatever other parameters come before the group command)----------------------------------- ULAW ou ALAW ------------------------------------------------------Renato, FAX sobre VoIP é um negócio chato, nunca funciona 100%.Pense na possibilidade de colocar uma Placa FXS direto no seu Asterisk.DIGIVOICE -> ASTERISK -> FXS -> Aparelho de FAX.Assim pode ligar o FAX direto no Asterisk e não precisa converter sua chamada para VoIP e jogar na rede para o ATA, pois a latência,perda de pacotes e outros motivos influenciam diretamente no envio de FAX.Considerações,FAX e Multifuncionais modernas enviam FAX na velocidade de 33.6Kbps, enviando através de ATA / VoIP não vai funcionar.veja se consegue abaixar a velocidade no seu aparelho.Ative o recurso de correção de erro do seu fax (Se tiver)Lembre-se de sempre usar ULAW ou ALAW para envio de fax, se for outro codec com maior compressão não funciona.Certifique-se que está fixando o codec no sip.conf.Experimente desligar o cancelamento de echo do seu ATA, normalmente ele faz isso automaticamente ao detectar o sinal de fax, masé necessário forçar-----------------------------------T.38 support in Asterisk------------------------------------------The T.38 support in Asterisk 1.4 is only pass through. Asterisk is notan endpoint or gateway of any sort. It only works passthrough inSIP to SIP calls without chan_local or chan_agent involved. Both endpoints of the call will need T.38 support and have T.38enabled in sip.conf.Asterisk 1.4 as I see it is still a bit unstable and not ready forproduction use. Be careful out there.-------------------------------------------------------------------------------------------------------------------Podria seguir pegando Emails de Asterisk, trixbox y Siemens...pero no es el caso...Solo queria enviarte los que considere importantes..!! --------------------------------------------------------------------------------------------------------------------- "Nós somos o que fazemos no dia-a-dia. A excelência, portanto não é um ato, mas sim um hábito" - Aristóteles Grupo de Usuarios y Desarrolladores Debian del Uruguay Debian Uruguay (Guddú)http://debianuruguay.org/Debian (デビアン) は、あなたのコンピュータのためのフリーなオペレーティングシステム (OS) です しかし、OS の基本ツールのほとんどが GNU プロジェクトに由来しているため、GNU/Linux と名付けています。---

Helius tonos antel

Me imagino que esto debe ser cosa de cada dia, por ser una empresa de telefonia...
Por las dudas, se los envios.
Tonos Uruguay
http://www.3amsystems.com/wireline/tone-search.htm?start=0&kTone=1004
Country Tone Freq/Hz Cadence/s
Uruguay Busy tone 425 0.5 on 0.5 off
Uruguay Congestion tone 425 0.25 on 0.25 off
Uruguay Dial tone - I 425 continuous
Uruguay Dial tone - II 450 0.7 on 0.8 off 0.2 on 0.3 off
Uruguay Negative indication tone 425 0.5 on 0.5 off
Uruguay Positive indication tone 425 2x(0.1 on 0.2 off) 0.1 on 1.3 off
Uruguay Ringing tone 425 1.0 on 4.0 off
Uruguay Special information tone 950/1400/1800 3x0.333 on 1.0 off
Uruguay Waiting tone 425 0.2 on 0.2 off 0.2 on 4.4 off
Uruguay Warning tone - operator intervening 425 0.15 on 0.25 off 0.15 on 1.45 off
Ademas tiene ejemplos de configuracion:
Switch to a different display format: [Sipura/Linksys] [Zaptel/Asterisk]

Ejemplo Country Tone:
Sipura/Linksys
Uruguay Busy tone 425@xx;yy(0.5/0.5/1)
Uruguay Congestion tone 425@xx;yy(0.25/0.25/1)
Uruguay Dial tone - I 425@xx;yy(*/0/1)
Uruguay Dial tone - II 450@xx;yy(0.7/0.8/1,0.2/0.3/1)
Uruguay Negative indication tone 425@xx;yy(0.5/0.5/1)
Uruguay Positive indication tone 425@xx;yy(0.1/0.2/1,0.1/0.2/1,0.1/1.3/1)
Uruguay Ringing tone 425@xx;yy(1/4/1)
Uruguay Special information tone 950@xx,1400@xx,1800@xx;yy(0.333/0/1,0.333/0/2,0.333/1/3)
Uruguay Waiting tone 425@xx;yy(0.2/0.2/1,0.2/4.4/1)
Uruguay Warning tone - operator intervening 425@xx;yy(0.15/0.25/1,0.15/1.45/1)
Country Tone
Zaptel/Asterisk string
Uruguay Busy tone 425/500,0/500
Uruguay Congestion tone 425/250,0/250
Uruguay Dial tone - I 425
Uruguay Dial tone - II 450/700,0/800,450/200,0/300
Uruguay Negative indication tone 425/500,0/500
Uruguay Positive indication tone 425/100,0/200,425/100,0/200,425/100,0/1300
Uruguay Ringing tone 425/1000,0/4000
Uruguay Special information tone 950/333,1400/333,1800/333,0/1000
Uruguay Waiting tone 425/200,0/200,425/200,0/4400
Uruguay Warning tone - operator intervening 425/150,0/250,425/150,0/1450

Helius Fax

por lo de "Pobres condiciones linea"Did you change the rxgain or the rx setting? I changed the rxgain from 0.0 to 20.0. I might make some more test calls, reducing the rxgain to see at what setting the incoming fax detection breaks. Might invoke ztmonitor as well.Are you using the rxfax module or have you changed to nvfaxdetect? This time around I was using the rxfax module only. One variation that I noticed with AAH 1.5 was: (no son Trixbox, pero vienen bien)Without NVFaxDetect-Internal SIP fax -> 666 = OK. Internal SIP fax-> 7777 = Failure (Does not recognise fax & dials my default test ext). Incoming PSTN fax -> AAH 1.5 = OK.With NVFaxDetect-Internal SIP fax -> 666 = OK.Internal SIP fax-> 7777 = OK.Incoming PSTN fax -> AAH 1.5 = OK.Now I understand what you mean by fax detection as distinct from fax handshaking/negotiation.How I sent the test FAXHere is my ztmonitor during sending. (I split the rx and tx onto separate lines to get around the wp formatting) and then during the reception of the Telstra fax report.SENDING A FAXpbx:~# /usr/src/zaptel-1.2.2/ztmonitor 1 -vvVisual Audio Levels.-------------------- Use zapata.conf file to adjust the gains if needed.( # = Audio Level * = Max Audio Hit )<----------------(RX)----------­ ------><----------------(TX)----------­ ------>######################* ########################*########################*########################*####################### *######################## *######################## *######################## *######################## * ######################## *##########################*RECEIVING A FAXpbx:~# /usr/src/zaptel-1.2.2/ztmonitor 1 -vvVisual Audio Levels.--------------------Use zapata.conf file to adjust the gains if needed. ( # = Audio Level * = Max Audio Hit )<----------------(RX)----------­ ------>#########*#########*#########*#########*##########*######### *######### *##########*<----------------(TX)----------­ ------> Telstra FaxStream reported:Transmission level: -17.1dBm EXCELLENTLine noise level: -54.3dBm EXCELLENTTelstra recommends fax transmission levels be of -15.0 to -17.0 dBmother reported scores were all EXCELLENT with 100.00% error free (yay!!)My ztmonitor txgain is about 2/3 to 3/4 up the scale and is a smidge OVER the Telstra recommendation so I could drop it a notch or two.looks like the notion of getting the ztmonitor vu meter about half way is indeed what we need (I like to test these things for myself!)Like I thought before.... I could up my rxgain a bit and reduce the txgain but I found a bit of audible hiss crept into the voice calls with rxgain over 13.I did change my setting to:rxgain=11.0txgain=-3.0-- "Nós somos o que fazemos no dia-a-dia. A excelência, portanto não é um ato, mas sim um hábito" - Aristóteles Grupo de Usuarios y Desarrolladores Debian del Uruguay Debian Uruguay (Guddú)http://debianuruguay.org/Debian (デビアン) は、あなたのコンピュータのためのフリーなオペレーティングシステム (OS) です しかし、OS の基本ツールのほとんどが GNU プロジェクトに由来しているため、GNU/Linux と名付けています。---

Helius Fax

First off, do not use any version over 0.0.3. I am using 0.0.3 on centos 4.5, asterisk 1.2.24 and freepbx 2.3 and it is working very well. One very important thing to keep in mind is that faxing over voip will only work reliably with ulaw or alaw and your internet connection MUST be able to sustain a constant data stream with low jitter. If your interested I have a shell script to install asterisk 1.2.24 and
freepbx-2.3 with rxfax and txfax on centos 4 and working on centos 5.

Gustavo JAVA

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IBM anunció un servicio gratuito que permitirá a los usuarios de
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Polycom, proveedor líder mundial en soluciones de comunicación
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Una de las charlas del reciente developerWorks Forum que se
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Manipulando mensajes del tipo XMLNS


Este texto de Jorge Fernández Lara provee información específica
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IBM Rational Software Architect (RSA) nos permite crear conexiones
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Helius Click to call

Asterisk Click To Dial & Click To Call
http://www.callfire.com/dialer/cm/info/click_to_dial_asterisk.html
screenshots
http://www.callfire.com/dialer/cm/screenshots.html;jsessionid=5AFA1FD3367C97FA99A202CB21DE5317

Gustavo SWIK

http://swik.net/Asterisk

Helius Softphone

Para plataformas Windows, Linux, Mac.
http://iaxclient.sourceforge.net/iaxcomm/
iaxComm is an Open Source softphone for the Asterisk PBX.
iaxComm is cross-platform program developed on aWindows XP system.
iaxComm is targeted for the Win32, Linux and Mac OS X systems. I try to avoid anything that will cause a problem under Mac OS, but have no way of building or testing on a Mac.
Features:
* iLBC support
* speex support
* Music on Hold.
* Blind Transfer.
* Custom Ringtones per CallerID
* Speakerphone mode.
* Register with multiple servers (ie enterprise server and iaxtel).
* Multiple call appearances.
* User selectable audio devices.
* User defined ringtones.
* Autoanswer intercom calls (with password protection).
* Registration with an asterisk server prior to dialing is no longer required.
* Simplified Directory dialogs.
* Incoming callers automatically added to phone book.
Los hay solo es cuestion de buscar con ganas..!!

Helius softphone

Freedom to Call, Freedom to Code

OpenWengo is a community of enthusiasts and developers, creating free software products related to communication over IP. The flagship product of the OpenWengo project is a softphone which allows you to make free PC to PC video and voice calls, and to integrate all your IM contacts in one place.
OpenWengo was started and is supported by the french VoIP provider Wengo . Through our partnership with Wengo, we also offer very cheap PC to telephone and SMS rates. OpenWengo is an active community - come join us
http://www.openwengo.org/
OpenWengo is a communication tool based on the SIP protocol, developed by the OpenWengo community and released under the GNU General Public License. It is more than just VoIP software: it is a full-fledged communications tool that lets you enjoy free calls, video conferencing, SMS, chat over many existing protocols (MSN, AIM/ICQ, Yahoo!, Jabber)... Everything you need to stay in touch with your friends and family!
Firefox Extension
*
Windows
o 0.68 XPI - 1.5 Mb
*
Mac OS X
o 0.68 XPI - 2.3 Mb
*
GNU/Linux
o 0.68alpha XPI - 2.4 Mb


WengoPhone screenshot
http://www.openwengo.org/index.php/openwengo/public/homePage/openwengo/public/projectsNgScreenShots

Helius SIP

on este tipo de documentacion es la que hay que trabajar desarrollando...ya que existen muchas cosas en los protocolos..que no estan siendo usadas o subaprovechadas...
RFC 3261 - SIP: Session Initiation Protocol http://www.faqs.org/rfcs/rfc3261.html

Helius Digium appliance

Te interesa una solución comercial de Asterisk paquetizada como el Digium Appliance? Finalizado el 14/4/2007 19:27 Sí, como integrador 66 % 66 % (14) Sí, como cliente final 14 % 14 % (3) No 19 % 19 % (4) Depende (explicar en comentarios) 0 % (0) Total de Votos: 21 Total de electores: 21
------------------------------------------------------------------------------------------------
Digium : Digium anuncia el lanzamiento de su PBX empotrada 'Asterisk Appliance'
Enviado por HeLL on 14/9/2006 7:31:38 (262 Lecturas)
Digium Appliance es una PBX con un Asterisk empotrado especial para este sistema que corre uCLinux, una distribución de Linux para sistemas empotrados y que incluye una versión de Asterisk Business Edition (ABE) especial para esta plataforma.
Incorpora una exclusiva GUI para programarla desarrollada por Digium y que facilita la forma de programar Asterisk y adaptarla a las soluciones basadas en este software.
Esta PBX está diseñada para pequeñas y medianas empresas (PYMES) e incorpora una conexión a la red telefónica mediante líneas analógicas (FXS y FXO).
Incluye un completo manual de desarrollo y de uso para que cualquier persona pueda configurarla y adaptarla a sus necesidades así como soporte para tarjetas Compact Flash donde almacenar datos externos (mensajes de buzón de voz, sonidos, etc)
También incluye puertos LAN y WAN para conectar terminales IP y conseguir una separación completa con la red de datos existente; documentación y manuales y soporte completo de Digium para la configuración y/o desarrollo de nuevas aplicaciones para este Asterisk.
Más información, en la web de Digium
-----------------------------------------------------------------------------------------------
Digium : Digium anuncia el lanzamiento de su nueva tarjeta para líneas analógicas Enviado por armstrong on 27/10/2005 13:46:37 (295 Lecturas )
La empresa norteamerica Digium, principal impulsora del proyecto Asterisk ha anunciado la comercialización de un nuevo modelo de tarjeta denominada genéricamente TDM2400P.
Esta tarjeta surge para cubrir la gran demanda existente de numerosas líneas FXS/FXO en un entorno empresarial existente. La tarjeta sigue la filosofía de la familia TDM400P, salvo que los módulos FXS/FXO van agrupados de 4 en 4, hasta un total de 6 módulos (24 puertos) por tarjeta.
El conector de salida es un amphenol de 25 pares (o rj21).
Desde el 1 de Noviembre se estarán aceptando pedidos, y el 28 de noviembre es la fecha prevista para su lanzamiento en todo el mundo.
Tendremos que probarla!
Para más información: http://www.digium.com/index.php?menu=product_detail&category=hardware&product=TDM2400P

Gustavo Click to call

Click-to-Call con tu propio softphone
Muchas veces, cuando vemos un número de teléfono en una web o en nuestra agenda, nos gustaría poner hacer clic con el ratón y que el sistema llame utilizando nuestro propio softphone, pero a menudo los números de teléfono vienen con el protocolo “callto:” y no viene definida ninguna aplicación para utilizar este protocolo.
En Linux es bastante sencillo asociar este protocolo a una aplicación, en el gestor de protocolos (de KDE o de GNOME) se configura con una facilidad digna de admirar, pero en Windows generalmente a de ser una aplicación la que asocie este protocolo para que sea utilizado.
Inicialmente el protocolo “callto:” venía asociado a la aplicación NetMeeting, y más tarde Skype hizo lo propio, de manera que muchas personas que ponen su número de teléfono ya lo enlazan con el prefijo callto.
Pero si queremos poder hacer llamadas desde nuestro softphone a través de nuestro Asterisk, tendremos que marcarlo a mano.
Por fín he encontrado los registros de Windows necesarios para asociar este protocolo a un softphone cualquiera enviándole como parámetro el número de teléfono indicado en el enlace y os lo pongo aquí para que podais hacer llamadas con vuestro softphone a cualquier número con el enlace 900900900 (por ejemplo).
Estoy seguro que esto ya lo conocía bastantes personas, aunque por mucho que lo he buscado jamás lo había encontrado.
Podeis editar el archivo y modificarlo por la ruta de vuestro softphone añadiendo al final el %1 que será sustituido por el número de teléfono al que se va a llamar.
No lo he podido probar porque no tengo ningún Windows a mano, pero os dejo la web donde podeis descargaros el tutorial y el archivo para el registro de windows.
http://www.a-enterprise.ch/content/view/170/111/