[asterisk-users] Fax with a AEx410P and Beronet BN4S0 = Sending Problem
Hi all, I have problem with one of my configuration : FAX - AEX410P (One FXS port) --- BN4S0 PSTN Case 1 : Receiving Fax is Ok ( PSTN --- BN4SO -- AEX410P -- FAX ) Case 2 : Sending Fax is nok ( FAX --- AEX410P -- BN4SO -- PSTN ) I think we have some synchronisation problem because , de beginning of the fax is correct but I have some blank line on document and Asterisk do not release the line. This is the result of dahdi show channel 1 : Channel: 1LI File Descriptor: 13 Span: 1 Extension: Dialing: no Context: from-internal Caller ID: 70 Calling TON: 0 Caller ID name: device Mailbox: 7...@device Destroy: 0 InAlarm: 0 Signalling Type: FXO Kewlstart Radio: 0 Owner: None Real: None Callwait: None Threeway: None Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: no Busy Detection: no TDD: no Relax DTMF: no Dialing/CallwaitCAS: 0/0 Default law: ulaw Fax Handled: no Pulse phone: no DND: no Echo Cancellation: 128 taps (unless TDM bridged) currently OFF Actual Confinfo: Num/0, Mode/0x Actual Confmute: No Hookstate (FXS only): Onhook And this is configration of midn port : [PORT 2] - name: swisscom - allowed_bearers: all - far_alerting: no - rxgain: 0 - txgain: 0- te_choose_channel: no - pmp_l1_check: no - reject_cause: 21 - block_on_alarm: no - hdlc: no - context: ext-did-0002- language: en - musicclass: default - callerid: - method: standard - dialplan: 0 - localdialplan: 0 - cpndialplan: 0 - nationalprefix: 0- internationalprefix: 00 - presentation: -1 - screen: -1 - always_immediate: no - nodialtone: no - immediate: no- senddtmf: no - astdtmf: no - hold_allowed: no - early_bconnect: yes - incoming_early_audio: no - echocancel: 128 - need_more_infos: no - noautorespond_on_setup: no - nttimeout: no - bridging: yes- jitterbuffer: 4000 - jitterbuffer_upper_threshold: 0 - callgroup: - pickupgroup: - max_incoming: -1 - max_outgoing: -1 - l1watcher_timeout: 0 - overlapdial: 0 - msns: * - faxdetect: no- faxdetect_context: - faxdetect_timeout: 5 - ptp: no Somebody already have this problem ? Thanks for your precious help, Vincent ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Web Browser Pop-up
Heelo, I currently search a program that can make a web browser Pop-up on an incoming call on a specific URL like : http://directorie.ch?CALLNUMBER:00451849799 I have found ADM, but it's a bit more complex for my purpose an it's not very stable. Do you know a simple software for that ? An other part of my project is to eneable click-to-call from a web page, do you know a kind of project that implement callto protocol, at this time I use Noojee click but It only work with Firefox. Thanks for your help, Vincent ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] kirk Wireless + Asterisk 1.2.7 Mysql Realtime problem
Hello, We have 32 DECT clients connected to a Kirk Wireless 600/v3, the Kirk server is connected to an Asterisk 1.2.17 with realtime configuration (MySQL). Our problem is that our Asterisk Server uses the latest inserted user to places calls each time a call is made. Exemple: we have 3 phones with number: 618, 670, 610. The number 610 is the latest inserted phone in the Asterisk server. If the user 618 calls the number 670 the user of the 670 phone will see the number 610 on the phone display. someone have a solution ? Thanks for your help, Vincent Renaville ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CSTA support for asterisk
Hello,I have a project, to make a Call-center tool kit for Asterisk and other PBX, so we start to look to implement CSTA in Asterisk to have a one-to-many PABX interface.If somebody is interest to work with us (It's a GPL product in very early development). or if you have documention on csta.VincentOn 7/31/06, Olivier [EMAIL PROTECTED] wrote: 2006/7/29, Steve Underwood [EMAIL PROTECTED]: A number of people have talking about providing open CSTA support, butI've never seen anything happen. I might be a key to allowing a lot ofexisting applications to make use of open source telephony solutions like Asterisk.SteveHi,My knowledge of CSTA is :- the document describing CSTA is one thousand pages thick,- CTSA is mostly used to build large call center ACDs for which previous proprietary interfaces could not protect customers investment. Though Asterisk Gateway or Management Interface are not approved standards, both rely on widely knowntools (stdin, stdout, ...) so that, for investment protection, Asterisk seems on par to CSTA-compliant PBX. So my question is, do you have anything in mind that could be done with CSTA and not done with AGI or AMI ?Regards ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] (no subject)
Hi,
I have the same problem with the queue configuration
When I receive 2 calls only 1 phone ring even if more agent's phone are free.
The second call will go to an other agent only if the first call is pickup.
Somebody have a solution ?This is my config file :Queue.conf[general]
;
; Global settings for call queues
;
; Persistent Members
; Store each dynamic agent in each queue in the astdb so that
; when asterisk is restarted, each agent will be automatically
; readded into their recorded queues. Default is 'yes'.
;
persistentmembers = yes
;
; Note that a timeout to fail out of a queue may be passed as part of
; an application call from extensions.conf:
; Queue(queuename|[options]|[optionalurl]|[announceoverride]|[timeout])
; example: Queue(dave|t|||45)
autofill = yes
[ticketix]
;
; A sample call queue
;
; Musiconhold sets which music applies for this particular
; call queue (configure classes in musiconhold.conf)
;
autofill=yes
musiconhold = default
;
; An announcement may be specified which is played for the member as
; soon as they answer a call, typically to indicate to them which queue
; this call should be answered as, so that agents or members who are
; listening to more than one queue can differentiated how they should
; engage the customer
;
;announce = queue-ticketix
;
; A strategy may be specified. Valid strategies include:
;
; ringall - ring all available channels until one answers (default)
; roundrobin - take turns ringing each available interface
; leastrecent - ring interface which was least recently called by this queue
; fewestcalls - ring the one with fewest completed calls from this queue
; random - ring random interface
; rrmemory - round robin with memory, remember where we left off last ring pass
;
strategy = roundrobin
;
; Second settings for service level (default 0)
; Used for service level statistics (calls answered within service level time
; frame)
servicelevel = 60
;
; A context may be specified, in which if the user types a SINGLE
; digit extension while they are in the queue, they will be taken out
; of the queue and sent to that extension in this context.
;
;context = qoutcon
;
; How long do we let the phone ring before we consider this a timeout...
;
timeout = 15
;
; How long do we wait before trying all the members again?
;
retry = 5
;
; Weight of queue - when compared to other queues, higher weights get
; first shot at available channels when the same channel is included in
; more than one queue.
;
;weight=0
;
; After a successful call, how long to wait before sending a potentially
; free member another call (default is 0, or no delay)
;
wrapuptime=15
;
; Maximum number of people waiting in the queue (0 for unlimited)
;
maxlen = 0
;
;
; How often to announce queue position and/or estimated holdtime to caller (0=off)
;
announce-frequency = 90
;
;
; How often to make any periodic announcement (see periodic-announce)
;
periodic-announce-frequency=60
;
; Should we include estimated hold time in position announcements?
; Either yes, no, or only once.
; Hold time will be announced as the estimated time,
; or less than 2 minutes when appropriate.
;
announce-holdtime = yes
;
; What's the rounding time for the seconds?
; If this is non-zero, then we announce the seconds as well as the minutes
; rounded to this value.
;
announce-round-seconds = 10
;
; Use these sound files in making position/holdtime announcements. The
; defaults are as listed below -- change only if you need to.
;
queue-youarenext = queue-youarenext ; (You are now first in line.)
queue-thereare = queue-thereare; (There are)
queue-callswaiting = queue-callswaiting ; (calls waiting.)
queue-holdtime = queue-holdtime; (The current est. holdtime is)
queue-minutes = queue-minutes ; (minutes.)
queue-seconds = queue-seconds ; (seconds.)
queue-thankyou = queue-thankyou; (Thank you for your patience.)
queue-lessthan = queue-less-than; (less than)
queue-reporthold = queue-reporthold ; (Hold time)
periodic-announce = queue-periodic-announce ; (All reps busy / wait for next)
;
; Calls may be recorded using Asterisk's monitor resource
; This can be enabled from within the Queue application, starting recording
; when the call is actually picked up; thus, only successful calls are
; recorded, and you are not recording while people are listening to MOH.
; To enable monitoring, simply specify monitor-format; it will be disabled
; otherwise.
;
; You can specify the monitor filename with by calling
; Set(MONITOR_FILENAME=foo)
; Otherwise it will use MONITOR_FILENAME=${UNIQUEID}
;
monitor-format = wav49
;
; If you wish to have the two files joined together when the call ends, set this
; to yes.
;
monitor-join = yes
;
; This setting controls whether callers can join a queue with no members. There
; are three
[Asterisk-Users] Dcap Test
Hello, I will pass the Dcap certification in June at Astricon Paris. I search some information about Dcap test - Book the we need to read - Previous Dcap test (for training purpose) - List of knowledge that we need to have If you have information, can send me an email Thanks a lot, Vincent ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dcap Test
Hello, I will pass the Dcap certification in June at Astricon Paris. I search some information about Dcap test - Book the we need to read - Previous Dcap test (for training purpose) - List of knowledge that we need to have If you have information, can send me an email Thanks a lot, Vincent ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
