[asterisk-users] Sangoma A200
Hi I was wondering if any one has used these cards, I am looking at this as a replacement for the tdm410, I have some issues with installing the tdm410 in a small case because of the power plug being at the end of the board. I am in australia seems like we have a different setup for out fxs voltage, any one in oz using this card ? Thanks Alex signature.asc Description: Digital signature ___ -- 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
Re: [asterisk-users] Skype for Asterisk. Any return of experience ?
On Sat, Jun 27, 2009 at 11:06 AM, Olivier wrote: > Hi, > > Has anyone tried it ? > Is there any available pricelist ? It is possible no one wants to answer this due to the NDA they had to sign? ___ -- 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
Re: [asterisk-users] music on hold file formats
On Tue, Jun 23, 2009 at 5:40 PM, Ron wrote: > > >> I have a portal where a user can upload their own MP3, but when a user > >> is using a g729 codec, the music on hold has a crackly sound. using g711 > >> it's very clear. > > > > That could be for any number of reasons, including a overly lossy mp3 > > to begin with. > > the mp3 moh is very clear when i use g711, only on g729 i'm having the > issue, could that be an issue of lossy MP3 still? > > Ron, Keep in mind that G.729a is a low bit-rate CODEC optimized for speech compression. It does not do well with music, or even DTMF tones. G.711, on the other hand, has greater dynamic range, so music sounds better. It would be nice if Asterisk had an option like CCM that forces MoH to use G.711, while the voice calls still use G.729. Sincerely, Trevor Hammonds ___ -- 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
Re: [asterisk-users] Sangoma A200
Yes, although not for connecting to the PSTN - I've used one for connecting to a legacy NEC PABX. Voltage isn't the issue - the difference is in the impedance. Australia uses complex impedance (220+820Ohm resistors with a 120nF capacitor) whereas the US uses a straight resistor. Alex Samad wrote: > Hi > > I was wondering if any one has used these cards, I am looking at this as > a replacement for the tdm410, I have some issues with installing the > tdm410 in a small case because of the power plug being at the end of the > board. > > I am in australia seems like we have a different setup for out fxs > voltage, any one in oz using this card ? > > Thanks > Alex > > > > > > ___ > -- 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 ___ -- 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
Re: [asterisk-users] Skype for Asterisk. Any return of experience ?
On Sun, 28 Jun 2009, randulo wrote: > On Sat, Jun 27, 2009 at 11:06 AM, Olivier wrote: >> Hi, >> >> Has anyone tried it ? >> Is there any available pricelist ? > > It is possible no one wants to answer this due to the NDA they had to sign? > Though they have written me back twice to say "coming soon" I am still waiting for the software... j ___ -- 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] Looking for real world uses of CallbackAgentLogin()
Hello everyone! I am currently in the middle of writing an article on converting from AgentCallbackLogin() dialplan application (which is deprecated as of 1.4, and removed in the 1.6.x series of Asterisk) to using the AddQueueMember(), and other dialplan based methods of performing the same functionality. However, after looking at the AgentCallbackLogin() application, it seems there is a fair amount of functionality that it could be used in, so I'd like to try and cover the common, and not so common, methods that it is being used in currently. If you could reply directly to me at l...@leifmadsen.com with a snippet of your dialplan and a brief description of what you're doing with AgentCallbackLogin(), it would be appreciated. I will attempt to cover all the most common uses of AgentCallbackLogin() based on this feedback, and cover how to convert from using it to a pure dialplan method in order to ease the transition to 1.6.x for those who are looking to do that. Any dialplan snippets I receive and use will be attributed in the article, which will be freely available on my blog and on http://www.asterisk.org as a developers blog. Thanks to all who contribute! Leif Madsen. http://www.leifmadsen.com http://www.oreilly.com/catalog/asterisk ___ -- 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] Recommendation / doubt about building of dialplan
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi all!
Now that I have a little more time, I was debugging my dialplan and it
was of the following way:
- -
; DGB - 20090615
[macro-dial]
exten => s,1,Dial(${ARG1},15)
exten => s,n,Goto(s-${DIALSTATUS},1)
exten => s-NOANSWER,1,Voicemail(${macro_ext...@voicemail,u)
exten => s-NOANSWER,n,Hangup
exten => s-BUSY,1,Voicemail(${macro_ext...@voicemail,b)
exten => s-BUSY,n,Hangup
exten => s-CHANUNAVAIL,1,Playback(pbx-invalid)
[from-internal]
; Call to SIP extensions
exten => _xxx,1,Macro(dial,SIP/${EXTEN})
exten => _xxx,n,Hangup
; Analog extension
exten => 402,1,Macro(dial,DAHDI/2)
exten => 402,n,Hangup
; Outgoing calls
exten => _9.,1,Dial(DAHDI/1/${EXTEN:1})
exten => _9.,n,Hangup
;exten => 9,1,Dial(DAHDI/1,20,tTr)
; Voicemail
exten => *100,1,Answer
exten => *100,n,Wait(1)
exten => *100,n,VoiceMailMain(${CALLERID(num)}...@voicemail)
exten => *100,n,Hangup
; Echo test
exten => *200,1,Answer
exten => *200,n,Playback(demo-echotest)
exten => *200,n,Echo
exten => *200,n,Playback(demo-echodone)
exten => *200,n,Hangup
; Music on the hold
exten => *300,1,Answer
exten => *300,n,SetMusicOnHold(default)
exten => *300,n,WaitMusicOnHold(2000)
exten => *300,n,Hangup
; Dial-by-name directory
exten => *400,1,Directory(voicemail,from-internal)
;---
[from-pstn]
; incoming calls from FXO port are directed to this context
exten => s,1,Dial(DAHDI/2,15,tTrm)
exten => s,n,Background(if-u-know-ext-dial) ; Dial known extension
exten => s,n,WaitExten()
include => from-internal
- -
Although internally it works as I had thought in such a way that
Asterisk derives to the voicemail indicating the reason by which one
became (nonavailable person or busy extension) and to indicate that the
extension is not valid in case it does not exist or the extension is not
registered when to try to contact (if there is some situation that I'm
ignoring, make to me notice it, please), the problem that I am seeing
with this is that if I include from-iternal context in from-pstn in such
a way that the incoming calls from the PSTN can communicate with both
SIP or DAHDI extensions, I think (with my present knowledge of Asterisk)
that it will be not useful to me so that in case the extension is not
valid a Goto(s,2) of from-pstn are accomplished so that the person can
dial the extension again without having to make a new call.
I suppose that it would be possible to be done defining again the
extensions in context from-pstn, but I suppose that there will be one
more efficient way to obtain the behavior to which I made reference of
one better way, which can be especially useful if we have defined a lot
of extensions.
Thanks in advance for your reply.
Regards,
Daniel
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Re: [asterisk-users] Skype for Asterisk. Any return of experience ?
Jeff LaCoursiere wrote: > On Sun, 28 Jun 2009, randulo wrote: > >> On Sat, Jun 27, 2009 at 11:06 AM, Olivier wrote: >>> Hi, >>> >>> Has anyone tried it ? >>> Is there any available pricelist ? >> It is possible no one wants to answer this due to the NDA they had to sign? >> > > Though they have written me back twice to say "coming soon" I am still > waiting for the software... > So you'd rather have it even when it hasn't been finished? I'm sure that as soon as it is complete and stable there will be pricing and availability announced. ___ -- 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
Re: [asterisk-users] [Asterisk-Users] Prices of g729 codec
Kevin P. Fleming wrote: > - Chris Mason (Lists) wrote: > >> licenses on them, usually $100 each time, and when I install the real >> >> hardware for the client, I can't transfer the licenses. If I scrap >> that > > Our support department is very accomodating when it comes to handling > licensing issues like this; I'm surprised to hear that you "can't transfer > the licenses" as we do that exact thing all the time. > > If you have a specific support ticket number where you requested this and it > was declined, please email it to me off-list. > I have a few lots of G.729 licenses and a digium branded linecard. I can't even email support, presumably since they were bought such a long time ago there is no longer a record that they were sold. (oh and I haven't registered the card, and haven't wanted to since it would involve taking down the machine it's in to find out the S/N.) (I have a 4 channel license for G.729 which I can't use with the most recent versions of codec_g729, presumably because the Product line calls it Digium-G729 rather than G.729 Codec. ___ -- 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] BUG in Asterisk 1.6.1.0 and issue in DAHDI 2.1.0.4
Starting playing with asterisk 1.6.1.0 i found the following problems:
In the cdr_pgsql, the sql statement is wrong:
2009-06-25 12:17:01 COT LOG: statement: INSERT INTO cdr
("accountcode","calldate","src","clid","dst","src","dcontext","dst","clid","dcontext","channel","channel","dstchannel","lastapp","lastapp","lastdata","lastdata","duration","start","billsec","answer","disposition","end","duration","amaflags","billsec","accountcode","disposition","uniqueid","amaflags","userfield","userfield","uniqueid")
VALUES ('4868','2009-06-25 12:14:11','unknown','"bga-gw-1-4"
','4','unknown','from-bga-gw','4','"bga-gw-1-4"
','from-bga-gw','SIP/bga-gw-1-4-09be9960','SIP/bga-gw-1-4-09be9960','DAHDI/5-1','Dial','Dial','DAHDI/g11/4868','DAHDI/g11/4868',170,'2009-06-25
12:14:11',170,'2009-06-25 12:14:11','ANSWERED','2009-06-25
12:17:01',170,3,170,'4868',8,'1245950051.272',3,'4','4','1245950051.272')
2009-06-25 12:17:01 COT ERROR: column "src" specified more than once
and asterisk shows:
[Jun 25 12:18:07] ERROR[15544]: cdr_pgsql.c:309 pgsql_log: Failed to
insert call detail record into database!
[Jun 25 12:18:07] ERROR[15544]: cdr_pgsql.c:310 pgsql_log: Reason:
ERROR: column "src" specified more than once
[Jun 25 12:18:07] ERROR[15544]: cdr_pgsql.c:311 pgsql_log: Connection
may have been lost... attempting to reconnect.
[Jun 25 12:18:07] ERROR[15544]: cdr_pgsql.c:314 pgsql_log: Connection
reestablished.
[Jun 25 12:18:07] ERROR[15544]: cdr_pgsql.c:320 pgsql_log: HARD ERROR!
Attempted reconnection failed. DROPPING CALL RECORD!
[Jun 25 12:18:07] ERROR[15544]: cdr_pgsql.c:321 pgsql_log: Reason:
ERROR: column "src" specified more than once
Also in chan_dahdi i got the following:
[Jun 25 12:19:04] WARNING[15757]: chan_dahdi.c:3090 dahdi_call:
Unrecognized prilocaldialplan NPI modifier: k
[Jun 25 12:19:04] WARNING[15757]: chan_dahdi.c:3090 dahdi_call:
Unrecognized prilocaldialplan NPI modifier: o
[Jun 25 12:19:04] WARNING[15757]: chan_dahdi.c:3090 dahdi_call:
Unrecognized prilocaldialplan NPI modifier: w
Slds.
--
Nestor A. Diaz
Ingeniero de Sistemas
Tel. +57 1-600-5490 x 211
Cel. +57 316-227-3593
Tel. SIP: sip:2...@tiendalinux.com
Email/MSN: nes...@tiendalinux.com
http://www.tiendalinux.com/
Bogota, Colombia
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Re: [asterisk-users] IAX for internet file transfer?
On 27 Jun 2009, at 11:27, Maris wrote: guarantee delivery?, not to mention that IAX2 does not use RTP. Are you suggesting to change the protocol to support such transfers? When it makes sense, yes - see below, otherwise the idea can get into the waste paper backet. ... But why does he want to do it ? Share secret / illegal files LOL ? Transfer files and/or logging data to/from computers anywhere in the intranet of organizations - over the internet. Due to restrictions this computer may not have server functionality. For the purpose, an IAX client can be installed on the remote computer. Of course, such client-client communication can be solved using an intermediate server which two clients that exchange data connects to. The specific features of IAX (NAT transparency) could help, provided that simple TCP channels initiated by the clients can posess problems in establishing connections under certain weird network constellations - it goes beyond my knowledge to judge that. ... to the other side and decode it there Asterisk (or just about any VoIP software) will opt for timely delivery rather than a reliable delivery. Encoding digital data into audio in order to transfer it as digital audio data packets makes no sense for me. Packet problems can be overcome with other methods, as pointed out by other contributors. Rob Maris Hardware developer You should read the protocol spec. http://www.rfc-editor.org/authors/rfc5456.txt It already supports a couple of 'data' transports, including the one that was used to upgrade the IAXy firmware. I don't think you would have to change much (if anything) in the protocol to make it work. Tim. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk smime.p7s Description: S/MIME cryptographic signature ___ -- 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
Re: [asterisk-users] Skype for Asterisk. Any return of experience ?
On Sun, 28 Jun 2009, Thomas Kenyon wrote: > Jeff LaCoursiere wrote: >> On Sun, 28 Jun 2009, randulo wrote: >> >>> On Sat, Jun 27, 2009 at 11:06 AM, Olivier wrote: Hi, Has anyone tried it ? Is there any available pricelist ? >>> It is possible no one wants to answer this due to the NDA they had to sign? >>> >> >> Though they have written me back twice to say "coming soon" I am still >> waiting for the software... >> > So you'd rather have it even when it hasn't been finished? Umm, no, but then when a company says "looking for beta testers - please sign up now!" and then four months later has nothing to let me beta test, I am a bit put off. > > I'm sure that as soon as it is complete and stable there will be pricing > and availability announced. > Indeed. j ___ -- 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] FW: re: Asterisk Outbound with Failover, alarm notification, dial status and hangupcause capturing to CDR from Dialplan
Managed to implement this on asterisk v1.4.24.1,
Also, Hangupcause updating to user field.
However, this only works on the edge of my voice network (demarcation
point)
It does not work on my internal routing boxes as I use IAX to route
between remote sites.
I was thinking of using some sort of SIP variables to transport these
results over the IAX trunk..
Any bright ideas folks???
[outbound]
exten => _XXX.,n(dial),gotoif($[${loadbalance} = 1 ]?balance) ;at dial
decide if we want to load balance
exten => _XXX.,n,goto(50) ;if no balance then outbound call
initiated with failover
exten => _XXX.,n(balance),NoOp("Load Balancing Active")
;NoOp - display Load balancing status
exten => _XXX.,n,Set(counter1=${DB(data/counter)})
;retrieve counter value from astdb
exten => _XXX.,n,NoOp(${counter1})
;display value of counter
exten => _XXX.,n,Set(counter=${MATH(${counter1}+1,int)}) ;
increment counter
exten => _XXX.,n,Set(DB(data/counter)=${counter})
;write incremented value back to asterisk db
exten => _XXX.,n,Set(result=${MATH(${counter}%2)})
;check for Odd/ Even using modulus of ${counter} via MATH function
exten => _XXX.,n,NoOp(${result})
;Display output - 0 for even and 1 for odd
exten => _XXX.,n,GotoIf($[${result} > 0]?50:100);Odd calls route to
outbound-1, even calls to outbound-2
exten => _XXX.,n(balance),NoOp("Load Balancing Active") ;NoOp -
display Load balancing status
exten => _XXX.,n,Set(counter1=${DB(data/counter)})
;retrieve counter value from astdb
exten => _XXX.,n,NoOp(${counter1}) ;display
value of counter
exten => _XXX.,n,Set(counter=${MATH(${counter1}+1,int)});
increment counter
exten => _XXX.,n,Set(DB(data/counter)=${counter}) ;write
incremented value back to asterisk db
exten => _XXX.,n,Set(result=${MATH(${counter}%2)}) ;check
for Odd/ Even using modulus of ${counter} via MATH function
exten => _XXX.,n,NoOp(${result})
;Display output - 0 for even and 1 for odd
exten => _XXX.,n,GotoIf($[${result} > 0]?50:100);Odd calls route
to outbound-1, even calls to outbound-2
exten => _XXX.,50,gotoif($[${dialout} > 0 ]?firstfail)
exten => _XXX.,n,Set(dialout=${MATH(${dialout}+2)})
exten => _XXX.,n,NoOp(${dialout})
exten => _XXX.,n,dial(${route2}/${ext...@${context2})
exten => _XXX.,n,goto(after-dial)
exten => _XXX.,n(firstfail),set(try=${MATH(${dialout}+2)})
exten => _XXX.,n,dial(${route2}/${ext...@${context2})
;attempt to dial out via route 2
exten => _XXX.,n,goto(after-dial);after attempting to dial, go to
after-dial
exten => _XXX.,100,gotoif($[${dialout} > 0 ]?secondfail)
exten => _XXX.,n,Set(dialout=${MATH(${dialout}+1))
exten => _XXX.,n,NoOp(${dialout})
exten => _XXX.,n,dial(${route1}/${ext...@${context1})
;attempt dial out via route 1
exten => _XXX.,n,goto(after-dial)
;after attempting to dial, go to after-dial
exten => _XXX.,n(secondfail),set(try=${MATH(${dialout}+2})
;If call has been dialled by other route and is failing over, set
variable try = 2
exten => _XXX.,n,dial(${route1}/${ext...@${context1})
;attempt to dial out via route 1
exten => _XXX.,n,goto(after-dial)
;after attempting to dial, go to after-dial
exten => _XXX.,n(after-dial),Set(CDR(accountcode)=${DIALSTATUS})
;first step - add ${DIALSTATUS} to CDR in accountcode field
exten => _XXX.,n,goto(${DIALSTATUS})
;go to dial-status received from attempt
exten => _XXX.,n(BUSY),,goto(set_cause)
;if Busy, go to set_cause
exten => _XXX.,n(NOANSWER),goto(set_cause)
;if No answer, go to set_cause
exten => _XXX.,n(CANCEL),goto(set_cause)
;if Cancel, go to set_cause
exten => _XXX.,n(NOANSWER),goto(set_cause)
;if No Answer, go to set_cause
exten => _XXX.,n(CHANUNAVAIL),gotoif($[${try} = 2 ]?emergency-notify)
;If CHANUNAVAIL,check if both routes are down. If yes, send emergency
notification
exten => _XXX.,n,gotoif($[${first-dial} = ${route1} ]?notify-1)
;If this was first attempt from route 1, go to notification for route 1
down
exten => _XXX.,n,goto(notify-2)
;else go to notification for route 2
exten => _XXX.,n(set_cause),hangup()
;at set_cause, firstly hang up channel (if not done already)
exten => _XXX.,n,goto(CDRfield)
;go to CDR field mapping section
exten => _XXX.,n(CDRfield),Set(CDR(userfield)=${HANGUPCAUSE})
;set Hangupcause to user field in CDR
exten => _XXX.,n(notify-1),System(echo "Call redirect detected on
'${route1}'" | mail -s "Calls Fail Over" neeraj.ch...@ocis.com.au)
;send notification route 1 down
exten => _XXX.,n,goto(100) ;attempt dial via route 2
exten => _XXX.,n(notify-2),System(echo "Call redirect detected on
'${route2}'" | mail -s "Calls Fail Over" neeraj.ch...@ocis.com.au)
;send notification route 2 down
exten => _XXX.,n,goto(50) ;attempt dial via route 1
exten => _XXX.,n(emergency-notify),System(echo "Call redirect detected
on BOTH Routes!" | mail -s "Calls Fail Over"
neeraj.ch...@ocis.com.au) ;se
Re: [asterisk-users] Sangoma A200
On Sun, Jun 28, 2009 at 10:10:48PM +1000, Rob Hillis wrote: > Yes, although not for connecting to the PSTN - I've used one for > connecting to a legacy NEC PABX. Thats good > > Voltage isn't the issue - the difference is in the impedance. Australia I get this in my dmesg when I load up the rdm410 modules [1083334.103487] Freed a Wildcard [1083336.171371] ALAW override parameter detected. Device will be operating in ALAW [1083338.040522] Boosting ringer on slot 1 (89V peak) [1083338.040542] Port 1: Installed -- AUTO FXS/DPO [1083340.340472] Boosting ringer on slot 2 (89V peak) [1083340.340492] Port 2: Installed -- AUTO FXS/DPO > uses complex impedance (220+820Ohm resistors with a 120nF capacitor) > whereas the US uses a straight resistor. Did yo buy from the us or local ? Alex > > Alex Samad wrote: > > Hi > > > > I was wondering if any one has used these cards, I am looking at this as > > a replacement for the tdm410, I have some issues with installing the > > tdm410 in a small case because of the power plug being at the end of the > > board. > > > > I am in australia seems like we have a different setup for out fxs > > voltage, any one in oz using this card ? > > > > Thanks > > Alex > > > > > > > > > > > > ___ > > -- 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 > > ___ > -- 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 > -- Some men are born mediocre, some men achieve mediocrity, and some men have mediocrity thrust upon them. -- Joseph Heller, "Catch-22" signature.asc Description: Digital signature ___ -- 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
Re: [asterisk-users] Skype for Asterisk. Any return of experience ?
>>> Though they have written me back twice to say "coming soon" I am still >>> waiting for the software... >>> >> So you'd rather have it even when it hasn't been finished? > > Umm, no, but then when a company says "looking for beta testers - please > sign up now!" and then four months later has nothing to let me beta test, > I am a bit put off. The beta was limited. Digium wants to open it but says Skype themselves are delaying the operation. I have compelling reasons to believe this, even though I can't put them out in public. I was surprised too at the apparent slowness, but I think it will happen in good time. ___ -- 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] Callwithus.com is discontinuing IAX service
Callwithus.com is discontinuing iax service. Can anybody recommend IAX provider - I need somebody with good rates to Philippines. -- Joseph ___ -- 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] about monitored calls storing
Hello all, how can I possibly make the monitoring for all calls through the asterisk, and for those file to be stored with the name of the initiator, in additional to know to whom this call is going, could this functionality be implemented via configurations! in other words, could I configure the asterisk so that the administrator to be able to hear calls coming from who going to whom, as a having a record for each call, I am using trixbox v2.6.2.1 should that functionality be implemented by an external application , such as one written using asterisk-java !!! any help is appreciated? thanks in advance, ___ -- 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
