Re: [Asterisk-Users] Syncronize Monitored Calls
ok im sorta confused.
when I save a * email'ed voicemail and I check the properties on th file It says
BitRate 13kbps
Channnels 1 mono
Audio Sample Rate 8 kHz
Audio Format GSM 6.10
when I look at the sox'd files from you script I see
BitRate 128kbps
Audio Sample Size 16bit
Channnels 1 mono
Audio Sample Rate 8 kHz
Audio Format PCM
I dont really think that the monitor files are getting GSM'd correctly.
Ill RTFM on sox and see what I can find
Dave
[EMAIL PROTECTED] 8/25/2003 3:41:07 PM
My mux script does the gsm compression using sox
On Mon, 25 Aug 2003, Dave Packham wrote:
and we could GSM compress them to be email friendly I think sox does gsm compress
Dave again
[EMAIL PROTECTED] 8/25/2003 2:30:25 PM
ok now lets modify that mix script to pick up on who started the monitored call and
look them up in the voicemail.conf and email it to em
Dave
[EMAIL PROTECTED] 8/25/2003 2:14:16 PM
Note that h will not be called if you park the call and pick it backup.
bkw
On Mon, 25 Aug 2003, David Harris wrote:
I thought I would post this in case it might be of any use to anyone.
Not anything special but it does work. Keep in mind you need sox and
wmix.
Here is some relevant exerpts of my extensions.conf using John Todds
macro.
[globals]
CALLFILENAME=foo
FOO=foo
CALLERIDNUM=foo
[default]
exten = 287,1,Macro(dial,SIP/agent20002|20)
exten = 287,2,Voicemail(u287)
exten = h,1,Macro(hangup)
[macro-dial]
exten = s,1,AGI(set-timestamp.agi)
exten =
s,2,SetVar(CALLFILENAME=${timestamp}-${CALLERIDNUM}-${MACRO_EXTEN})
exten = s,3,Monitor(wav,${CALLFILENAME})
exten = s,4,Dial(${ARG1},${ARG2},${ARG3})
[macro-hangup]
exten = s,1,SetVar(MONITORDIR=/var/spool/asterisk/monitor)
exten = s,2,GotoIf($[${CALLFILENAME} = ${FOO}]?6:3)
exten = s,3,System(/usr/local/bin/mix_monitor_files.pl ${MONITORDIR}
${CALLFILENAME}-in.wav ${CALLFILENAME}-out.wav ${CALLFILENAME}.wav)
exten = s,6,NoOp
Here is mix_monitor_files.pl. sox and wmix must be in the path of your
perl script
#!/usr/bin/perl
$monitordir = shift;
$infile = shift;
$outfile = shift;
$finishfile = shift;
chdir($monitordir);
$infile_output = `sox $infile -e stat 21`;
$outfile_output = `sox $outfile -e stat 21`;
$infile_output =~ /Samples read:\s+(\d+)/;
$infile_samples = $1;
$outfile_output =~ /Samples read:\s+(\d+)/;
$outfile_samples = $1;
if($outfile_samples $infile_samples)
{
$diff_samples = $outfile_samples - $infile_samples;
system(sox $outfile temp${outfile} trim ${diff_samples}s);
system(wmix $infile temp${outfile} $finishfile);
system(rm -f $infile temp${outfile} $outfile);
}
elsif($infile_samples $outfile_samples)
{
$diff_samples = $infile_samples - $outfile_samples;
system(sox $infile temp${infile} trim ${diff_samples}s);
system(wmix temp${infile} $outfile $finishfile);
system(rm -f temp${infile} $outfile $infile);
}
else
{
system(wmix $infile $outfile $finishfile);
system(rm -f $infile $outfile);
}
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Re: [Asterisk-Users] 0 out of voicemail to different secretaries
Almost... right now it lets you press * to cancel and enter a different mailbox, but that just lets you leave a message rather than ringing the extension. I guess exit vm and go back to the automated attendant is a typical type of feature. Maybe it would be cool if there were a way to quit voicemail and dump the call to an extension similar to the o operator extension, say a for automated attendant? Or maybe voicemail should take care of acting as the automated attendant and ask for the extension number and then send the call there? Hard to say. On Mon 25 Aug 2003 15:30, Brian West wrote: ah I see now.. I didn't notice that but it does atleast give you someway to exit and go to another extension doesn't it? On Mon, 25 Aug 2003, Brad Bergman wrote: I certainly contemplated that very thing... but somehow it escaped implementation. Even as things are now, the PBX administrator can set something like this up by putting engineering, accounting, etc in different contexts, and setting different o extensions for them. I was thinking of a couple of relevant features, one giving the individual mailbox user the ability to store a different extension (i.e., the target attendant) in the database that would override the o extension in the dialplan when a caller presses '0'. The other is to allow another number(s) to be stored in the DB so that a caller could press, say, 4, 5, or 6, and be transferred to whatever number the mailbox owner has stored there. Of course, all of this subject to whatever restrictions are imposed on the use of these features. I will look into this. Cheers, Brad On Mon 25 Aug 2003 14:25, Brian West wrote: http://bugs.digium.com/bug_view_page.php?bug_id=156 patients grass hopper! bkw On Mon, 25 Aug 2003, Don Pobanz wrote: Is it possible to configure * so that if a caller reaches voicemail for someone in Engineering, but doesn't want to leave a message they can press zero (0) and reach the Engineering Secretary or if they are calling someone in Accounting and reach voicemail, pressing '0' would reach the Accounting secretary, not the Engineering secretary? Don Pobanz ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dialed Number Identification in analog hunt group
Does anyone out there know if it is possible to discover the dialed number when a line in an analog hunt group rings? I can't get a straight answer from our IT folks. We have a 5ess switch delivering 4 analog lines which are in a simple hunt group servicing our lab. I would like to have a different call attendant based on which number is dialed so that I can route the calls to the appropriate group. I know that Asterisk can easily do this once I have the information to pass into the dial plan. The problem is getting the information. While I know that this is possible with T1, it is, unfortunately, a bit overkill for 4 lines. Anyone have any suggestions? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dialed Number Identification in analoghunt group
On Tue, 26 Aug 2003 17:48:55 -0500 Don Pobanz [EMAIL PROTECTED] wrote: *This message was transferred with a trial version of CommuniGate(tm) Pro* On Tuesday, August 26, 2003 4:55 PM, Stephen R. Besch [SMTP:[EMAIL PROTECTED] wrote: Does anyone out there know if it is possible to discover the dialed number when a line in an analog hunt group rings? I can't get a straight answer from our IT folks. We have a 5ess switch delivering 4 analog lines which are in a simple hunt group servicing our lab. I would like to have a different call attendant based on which number is dialed so that I can route the calls to the appropriate group. I know that Asterisk can easily do this once I have the information to pass into the dial plan. The problem is getting the information. While I know that this is possible with T1, it is, unfortunately, a bit overkill for 4 lines. Anyone have any suggestions? If they are pots lines in a hunt group, you won't be able to. If they are analog DID trunks then the dialed number would be passed. My guess is you have pots lines and there is no way to find out the dialed number. Don Pobanz Again, not near my asterisk box so I can't check this out, but is it possible to have the different ports drop into * in a different context for each line? That way you could just set up an 's' extension in that context for the different attendants. Regards, Jamie Carl Jazz Inc. Email: [EMAIL PROTECTED] Web:www.jazz-inc.net Phone: +61-414-365-466 Jabber: [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dialed Number Identification in analog huntgroup
Again, not near my asterisk box so I can't check this out, but is it possible to have the different ports drop into * in a different context for each line? That way you could just set up an 's' extension in that context for the different attendants. Yup. Set up different contexts in zapata.conf and extensions.conf for each line (I'm making a rash assumption you're using a zaptel FXO device). ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] H.323 channel problems
What part of IN OTHER WORDS: Run Open H.323 v1.11.7, nothing newer, nothing older if u want this to work. don't you understand? Jeremy McNamara Jan Rychter wrote: I have hit a problem where chan_h323 sometimes doesn't hang up properly and stays stuck in the Up state, with asterisk consuming 100% of CPU: *CLI show channels Channel (ContextExtensionPri ) State Appl. Data H323/ip$127.0.0.1:30008/21552 (local 123 1 ) Up (None)(None) 1 active channel(s) *CLI show ch channel channels *CLI show channel H323/ip$127.0.0.1:30008/21552 -- General -- Name: H323/ip$127.0.0.1:30008/21552 Type: H323 UniqueID: 1061946140.22 Caller ID: Jan DNID Digits: (N/A) State: Up (6) Rings: 0 NativeFormat: 8 WriteFormat: 1024 ReadFormat: 1024 1st File Descriptor: 26 Frames in: 47575 Frames out: 94850 Time to Hangup: 0 -- PBX -- Context: local Extension: 123 Priority: 1 Call Group: 0 Pickup Group: 0 Application: (N/A) Data: (None) Stack: -1 Blocking in: ast_waitfor_nandfds *CLI That's after hanging up (in gnomemeeting) on a H.323 call that is then bridged to IAX2. Now, before I go running to the bugtracker, I'd like to ask some general questions. The H.323 channel readme says: NOTICE: Whatever you do, DO NOT USE distrubution specific installs of Open H.323 and PWLib. In fact you should check to make sure your distro didn't install them for you without your knowledge. Check everything out of CVS. If you dont know how to deal with cvs, learn. Also, if you are not using the listed versions of Open H.323 or PWlib you are on your own, sorry. And: Some chan_h323 users have reported success and others have reported dramatic failures when using newer versions of Open H.323. We haven't personally tested this and will not be able to assist you if you have 'issues'. Sorry. IN OTHER WORDS: Run Open H.323 v1.11.7 nothing newer nothing older if u want this to work. How does this relate to my bug? I'm using openh323-1.12 and pwlib-1.5.0 that I compiled myself. Do they have problems? Does this mean I am on my own? Perhaps it's worth trying to report the bugs to distribution maintainers if indeed the distribution-specific installs of openh323 are this buggy? The requirement of using this particular version of openh323 is a problem for those of us who also use other H.323 software (such as gnomemeeting) which specifically requires newer libraries. Briefly, do I have a chance of reporting this bug with my versions of libraries, or is chan_h323 completely unsupported if I use anything other than 1.11.7? many thanks, --J. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Chan_h323 support for phone numbers via gateway?
Does chan_h323 support phone number calling via a gateway? ie., something like calling 5000 forwarded to: exten = 5000,1,Dial(h323/[EMAIL PROTECTED]) if so - what format should the exten be in? Thanks. Regards, Steven Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Chan_h323 does not seem to send the destimation number to gateway
Continuing my problems with h323. I think I am getting closer. SJPhone works direct to the gateway - calls and answers fine on the pstn. So the gateway is working. Inbound calls from PSTN = Gateway = Asterisk = Phone work great! Outbound from Asterisk = Gateway = PSTN still remains a problem. The debug stuff on the gateway receives the call signal from asterisk - but does not receive the number to call - its errors with callID is -1 (nothing to call) Any ideas for the correct format to use within extenensions.conf for outbound phone number via chan_h323 and a gateway? h323 works fine if it is just an IP address that it is calling, ie, a softphone. Thanks for your help Regards, Steven Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] conference authorization
Hello all ! How can I make conference authorization based on pin number ? I have: exten = 1,1,Meetme,1234|ps| where is a pin number and this doesn't works Where do I have to add information about pin number ?? Greetings Andrzej Radke ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] conference authorization
Well, going by apps/app_meetme.c, for some of it, we see
inpin = strchr(inflags, '|');
if (inpin) {
*inpin = '\0';
inpin++;
/* XXX Need to do something with pin XXX */
ast_log(LOG_WARNING, MEETME WITH PIN=(%s)\n, inpin);
}
and a bit further down, we see:
/* XXX Should prompt user for pin if pin is required XXX */
/* Run the conference */
res = conf_run(chan, cnf, confflags);
Therefore, we conclude asterisk does not do conference authentication, yet.
On Wed, Aug 27, 2003 at 09:10:46AM +0200, radan wrote:
Hello all !
How can I make conference authorization
based on pin number ?
I have:
exten = 1,1,Meetme,1234|ps|
where is a pin number
and this doesn't works
Where do I have to add information about pin number ??
Greetings
Andrzej Radke
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[Asterisk-Users] SNOM 200 bugs
Title: Message Does anyone have the same issues and is there any work arounds. I have a SNOM 200 which seems to work fine for so long but after an undetermined time when I make a call I hear no audio. If I reboot the SNOM all is fine again. Also when I reboot the SNOM it only ever picks up the NTP timeand registers correctly after the second reboot. Thanks for any info. Rgds, Stuart
Re: [Asterisk-Users] SNOM 200 bugs
Stuart Hirst wrote: Does anyone have the same issues and is there any work arounds. I have a SNOM 200 which seems to work fine for so long but after an undetermined time when I make a call I hear no audio. If I reboot the SNOM all is fine again. The same here. Version sip-1.16w. You have to go down to 1.16b if you want to get a temporary solution. When the problem occurs, for some reason a RTP media stream is never disconnected ( SNOM to * ). The good news is that SNOM is aware of this and has promised a beta fix shortly. -- Pertti Also when I reboot the SNOM it only ever picks up the NTP time and registers correctly after the second reboot. Thanks for any info. Rgds, Stuart ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SNOM 200 bugs
Does anyone have the same issues and is there any work arounds. I have a SNOM 200 which seems to work fine for so long but after an undetermined time when I make a call I hear no audio. If I reboot the SNOM all is fine again. Also when I reboot the SNOM it only ever picks up the NTP time and registers correctly after the second reboot. Thanks for any info. Rgds, Stuart The audio bug is resolved in ver 1.16x but it looks like they have done a whole lot more so we just wait for them to finish testing.. -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] H323 caller ID
Hi, I use asterisk-oh323 and a gatekeeper (gnugk) and netmeeting In asterisk i can have Caller ID when I do "show channel " I have (N/A). Does anyone know how I can have this caller ID ? Notice that in the gatekeeper I can see the user login of the netmeeting caller. Regards Rattana
Re: [Asterisk-Users] conference authorization
Perhaps you should check out the AGI module. Write a perl script to compare
DTMF(pin) with any data storage(text file, Database). See this doc
http://home.cogeco.ca/~camstuff/.
The other solution is of course modify the source code to check for pin.
You can also use the Autheticate module.
I found that the first option is easier to implement and provide more
control.
Foong
- Original Message -
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, August 27, 2003 3:19 PM
Subject: Re: [Asterisk-Users] conference authorization
Well, going by apps/app_meetme.c, for some of it, we see
inpin = strchr(inflags, '|');
if (inpin) {
*inpin = '\0';
inpin++;
/* XXX Need to do something with pin XXX
*/
ast_log(LOG_WARNING, MEETME WITH
PIN=(%s)\n, inpin);
}
and a bit further down, we see:
/* XXX Should prompt user for pin if pin is
required XXX */
/* Run the conference */
res = conf_run(chan, cnf, confflags);
Therefore, we conclude asterisk does not do conference authentication,
yet.
On Wed, Aug 27, 2003 at 09:10:46AM +0200, radan wrote:
Hello all !
How can I make conference authorization
based on pin number ?
I have:
exten = 1,1,Meetme,1234|ps|
where is a pin number
and this doesn't works
Where do I have to add information about pin number ??
Greetings
Andrzej Radke
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[Asterisk-Users] Registering via IAX2 succeeds, but bridging to the registered peerfails
Setup as follows: [private*] - Natting Router - [public*] [private*] cannot register via IAX2 correctly while [public*] is running. Status remains UNKNOWN even after minutes, calls from [public*] to [private*] are not possible. Console output of [public*]: | *CLI iax2 show peers | Name/UsernameHost Mask Port Status | iaxtest/iaxtest (Unspecified) (D) 255.255.255.255 0 UNKNOWN | -- Registered 'iaxtest' (AUTHENTICATED) at 217.187.160.28:4569 | *CLI iax2 show peers | Name/UsernameHost Mask Port Status | iaxtest/iaxtest 217.187.160.28 (D) 255.255.255.255 4569 UNKNOWN [private*]: | *CLI iax2 show registry | Host UsernamePerceived Refresh State | 62.75.137.248:4569iaxtest 217.187.160.28:456960 Registered BUT if I restart [public*], everything looks perfectly okay and calls work well: | Asterisk Ready. | *CLI iax2 show peers | Name/UsernameHost Mask Port Status | iaxtest/iaxtest 217.187.160.28 (D) 255.255.255.255 4569 OK (72 ms) Am I doing anything wrong or do I hit a bug? Config of [private*]: | # grep -v \^\; /etc/asterisk/iax.conf | [general] | bindaddr=192.168.11.1 | bandwidth=low | disallow=g723.1 ; Hm... Proprietary, don't use it... | disallow=lpc10 ; Icky sound quality... Mr. Roboto. | allow=gsm ; Always allow GSM, it's cool :) | register = iaxtest:[EMAIL PROTECTED] | tos=lowdelay | [guest] | type=user | context=voip Config of [public*]: | [general] | bindaddr=62.75.137.248 | bandwidth=low | disallow=g723.1 ; Hm... Proprietary, don't use it... | disallow=lpc10 ; Icky sound quality... Mr. Roboto. | allow=gsm ; Always allow GSM, it's cool :) | tos=lowdelay | [guest] | type=user | context=public | [iaxtest] | type=peer | username=iaxtest | context=voip | auth=plaintext | secret=iax | host=dynamic | qualify=yes ; Make sure this peer is alive Both * are compiled from latest CVS sources. Tia Manuel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] conference authorization
Use extension logic.
Is there an echo in here?
Jeremy McNamara
[EMAIL PROTECTED] wrote:
Well, going by apps/app_meetme.c, for some of it, we see
inpin = strchr(inflags, '|');
if (inpin) {
*inpin = '\0';
inpin++;
/* XXX Need to do something with pin XXX */
ast_log(LOG_WARNING, MEETME WITH PIN=(%s)\n, inpin);
}
and a bit further down, we see:
/* XXX Should prompt user for pin if pin is required XXX */
/* Run the conference */
res = conf_run(chan, cnf, confflags);
Therefore, we conclude asterisk does not do conference authentication, yet.
On Wed, Aug 27, 2003 at 09:10:46AM +0200, radan wrote:
Hello all !
How can I make conference authorization
based on pin number ?
I have:
exten = 1,1,Meetme,1234|ps|
where is a pin number
and this doesn't works
Where do I have to add information about pin number ??
Greetings
Andrzej Radke
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Re: [Asterisk-Users] conference authorization
Well, considering I replied as soon as I got it, it could always be a delay in smtp or whatever... Anyways, the documention advertises a feature which isn't present in the module it indicates it is in. This would normally be classifed as a bug, or do you feel it doesn't need to be raised? - Andrew Griffiths On Wed, Aug 27, 2003 at 05:38:51AM -0400, Jeremy McNamara wrote: Use extension logic. Is there an echo in here? Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Chan_h323/g729 - X100P connecting to non-Digium Partner
I have on Chan_h323 with G729 and X100P trying to connect to a Planet VOIP400 gateway box(http://www.planet.com.tw) I uncommented g729 in the Makefile and I'm setting g729 in h323.conf I'm receving in my side: 1:20.906 H225 Caller:810f070 h323ep.cxx(1537) H323 Clearing connection ip$localhost/4112 reason=EndedByRemoteUser and the other side(Planet) says: 15- RADH 2 HSMU RAD: cmHookSend(masterSlaveDeterminationAck) 11- HSMU 0 Remote capabilities list: 0- HSMU 0 [1] g729AnnexA: Audio Receive 0- HSMU 0 Try matching local element: 0- HSMU 0 [1] g7231: Audio Receive and Transmit 0- HSMU 0 Try matching local element: 0- HSMU 0 [2] g729: Audio Receive and Transmit 0- HSMU 0 Try matching local element: 0- HSMU 0 [3] g711Ulaw64k: Audio Receive and Transmit 0- HSMU 0 Try matching local element: 0- HSMU 0 [4] t38fax: Data Receive and Transmit 0- HSMU 0 Try matching local element: 0- HSMU 0 [5] g729: Audio Receive and Transmit 0- HSMU 3 HSMU 2: Capabilities: NO MATCH FOUND! 0- HSSM 2 HSMU 2: SM H245 WAIT COMPLETE - error release 1- HSMU 2 HSMU 2: abort timer in state H245 WAIT COMPLETE 0- HSSM 2 HSMU 2: SM H245 WAIT COMPLETE == RELEASING 10- RADH 2 HSMU RAD: cmHookSend(endSessionCommand) 3- RAD 2 HSMU 2: cmEvCallControlStateChanged(cmControlStateTransportDisconnected, - ) Anybody has any idea? Thanks, Isamar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] iax.conf order important?
Is the order important in IAX.CONF? I DID have this: [iaxtel] type=user host=12.37.165.130 context=iax-caller callerid=Guest IAX User accountcode=iaxuser ;auth=rsa ;inkeys=iaxtel [other] host=dynamic type=friend context=default auth=md5,rsa secret=secret inkeys=testinkey outkeys=testoutkey With this configuration, I was NOT able to receive calls via iaxtel? My messages log kept saying (everytime I called myself on my 1700 number): Aug 27 23:01:36 NOTICE[10251]: File chan_iax2.c, Line 4268 (socket_read): Host 12.37.165.130 failed to authenticate as other When I reversed the iax.conf file, so that other was above iaxtel, I could then call myself and saw an accept UNAUTHENTICATED call ... message. It seems that the host= line was ignored and assumed that the incoming call from iaxtel should have been authenticated using the other stanza. Does that mean that all incoming authenticated calls will be defined as per the iaxtel stanza? Shouldnt I be able to create stanzas for each incoming call from other defined iax servers? Also, it seems that the auth=rsa and the key wouldnt work (I didnt get any incoming connections from iaxtel)? Is that right? Do I have a dud key - or does iaxtel not try and authenticate with it? ...deon --- Have you looked at the A/NZ Tivoli User Group website? http://www.tuganz.org Deon George, IBM Tivoli Software Engineer, IBM Australia Office: +61 3 9626 6058, Fax: +61 3 9626 6622, Mobile: +61 412 366 816, IVPN +70 66058 mailto:[EMAIL PROTECTED], http://www.ibm.com/tivoli ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] call center - operators not using phone keys
On Mon, 2003-08-25 at 18:05, Ernest W. Lessenger wrote: At 11:09 AM 8/25/2003 -0500, you wrote: Perhaps some day there will be a client side product/widget/whatever for Asterisk, but right now it doesn't exist, to my knowledge that is. I believe the Asterisk Manager will do everything Miguel wants. I will be giving this a try in the next month or two, so if Miguel is willing to wait until then I'll let him know. Sure, I'm interested. The setup I'm looking for is.. something like... Operators should login on their linux desktops, in X11 mode. (as an example) A tcl/tk application is lauched, giving the operator his work environment. This application launches a non-gui VoIP client, and will connect the agent to *, setting the agent as available for inbound calls and/or ready for outbound calls. If the agent is idle and inbound calls are available, the application should respond with a welcome screen, load caller profile information, if available, and start the correct script for the called queue. If the operator is idle, no inbound calls waiting and there are available outbound calls, the script should initiate the outbound call and position the operator in that outbound script. During any call, call transfer, put on hold, call pickup, or call recording should all be controlled by the tcl/tk application. No phone keypad should be available. After any calls, the operator should be able to use the same tcl/tk application to put himself not-ready, to perform any duties, rest, or continue to any next calls. Everything should happen without the need for multiple logins or having to press any keys or dial any numbers, besides the user login and maybe selecting the queues the operator wishes to connect to. I'm considering a bunch of small pc's, with 1 sound card each or a USB headset. (searched the web, and really don't know which has a higher quality) I can look into other equipment if anyone has some other recommendation. I used tcl/tk as an example for an application, but a web interface, even ncurses, perl+tk, ticatk or any rapid development graphical interface should/could do. Regards, Miguel Dias --Ernest -Original Message- From: Miguel Bettencourt Dias (Netopia) [mailto:[EMAIL PROTECTED] Sent: Monday, August 25, 2003 09:53 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] call center - operators not using phone keys On Mon, 2003-08-25 at 15:09, Mark Spencer wrote: yes, you'll need outbound spool (see sample.call) Sure, This will allow me to start a call. but what about call transfers, agent login, accepting a call, and others ? Regards, mbd Mark On Mon, 25 Aug 2003, Miguel Bettencourt Dias (Netopia) wrote: Hi, I'm considering setting up a small call centre, but I don't want operators to need to use their phone keypads. Supposedly, all required call functions (dialling, answering, transfer, on hold, hang up, etc), should be done via their scripts (be is a web interface, curses or whatnot) and not using either a regular phone, nor a gnome-phone type interface. Also, all this will be happen in a 100% free MS zone. I think this can be done, but my experience is limited. Would this be possible with asterisk ? (more than a confirmation that it is possible, I was looking for a I did it this way kinda of answer) :-) Regards, Miguel Dias ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users signature.asc Description: This is a digitally signed message part
[Asterisk-Users] include context
hi, how can I add or remove this line "include =context"by the command CLI ? regards Rattana
Re: [Asterisk-Users] Dialed Number Identification in analog huntgroup
Don Pobanz wrote: On Tuesday, August 26, 2003 4:55 PM, Stephen R. Besch [SMTP:[EMAIL PROTECTED] wrote: Does anyone out there know if it is possible to discover the dialed number when a line in an analog hunt group rings? I can't get a straight answer from our IT folks. We have a 5ess switch delivering 4 analog lines which are in a simple hunt group servicing our lab. I would like to have a different call attendant based on which number is dialed so that I can route the calls to the appropriate group. I know that Asterisk can easily do this once I have the information to pass into the dial plan. The problem is getting the information. While I know that this is possible with T1, it is, unfortunately, a bit overkill for 4 lines. Anyone have any suggestions? If they are pots lines in a hunt group, you won't be able to. If they are analog DID trunks then the dialed number would be passed. My guess is you have pots lines and there is no way to find out the dialed number. Don Pobanz Don, Thanks for the info so far. At least now I know to ask the IT people if they can provide me with DID from the 5ess. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Chan_h323/g729 - X100P connecting to non-DigiumPartner
Hi The endpoint seems to be running Radvision h323 stack, and I know chan_h323 works with Radvision, there could be a couple of reasons!! 1) You dont have G729A in the capabilities of remote endpoint 2) The packetization interval is way off The best way would be to run ethereal or dump323 and see what is being negotiated. Also try to use fastConnect on both sides and force same packetization, (you can use my patch posted a couple of days ago to force packetization interval in G729 in chan_h323) Isamar Said I have on Chan_h323 with G729 and X100P trying to connect to a Planet VOIP400 gateway box(http://www.planet.com.tw) I uncommented g729 in the Makefile and I'm setting g729 in h323.conf I'm receving in my side: 1:20.906 H225 Caller:810f070 h323ep.cxx(1537) H323 Clearing connection ip$localhost/4112 reason=EndedByRemoteUser and the other side(Planet) says: 15- RADH 2 HSMU RAD: cmHookSend(masterSlaveDeterminationAck) 11- HSMU 0 Remote capabilities list: 0- HSMU 0 [1] g729AnnexA: Audio Receive 0- HSMU 0 Try matching local element: 0- HSMU 0 [1] g7231: Audio Receive and Transmit 0- HSMU 0 Try matching local element: 0- HSMU 0 [2] g729: Audio Receive and Transmit 0- HSMU 0 Try matching local element: 0- HSMU 0 [3] g711Ulaw64k: Audio Receive and Transmit 0- HSMU 0 Try matching local element: 0- HSMU 0 [4] t38fax: Data Receive and Transmit 0- HSMU 0 Try matching local element: 0- HSMU 0 [5] g729: Audio Receive and Transmit 0- HSMU 3 HSMU 2: Capabilities: NO MATCH FOUND! 0- HSSM 2 HSMU 2: SM H245 WAIT COMPLETE - error release 1- HSMU 2 HSMU 2: abort timer in state H245 WAIT COMPLETE 0- HSSM 2 HSMU 2: SM H245 WAIT COMPLETE == RELEASING 10- RADH 2 HSMU RAD: cmHookSend(endSessionCommand) 3- RAD 2 HSMU 2: cmEvCallControlStateChanged(cmControlStateTransportDisconnected, - ) Anybody has any idea? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom SoundPoint 500 with Asterisk
Hello All, Does anyone use a Plolycom SIP-based phone with Asterisk? Does anyone use a Polycom SoundPoint 500 with Asterisk using SIP? If so, please share your experiences, both good and bad. Thanks, Tim ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom SoundPoint 500 with Asterisk
On Wed, 2003-08-27 at 09:59, Timothy Soos wrote: Hello All, Does anyone use a Plolycom SIP-based phone with Asterisk? Does anyone use a Polycom SoundPoint 500 with Asterisk using SIP? If so, please share your experiences, both good and bad. I tried to contact Polycom regarding their VoIP products and they were less than helpful. They will not offer any documentation about the phones. That being said... Feel free to give them a shot, but realize that supply and support can ba hard to come by for those phones. --Karl Thanks, Tim ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Karl Putland [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ADSI Programs
I just received an unlocked ADSI phone and I am playing with the ADSI script. I was wondering how I can include Voicemail functions (Check new messages, Delete message) into the soft buttons. I checked in app_voicemail.c and it looks like these functions have already been programmed. Is there a voicemail.adsi script somewhere? If not, then how do I get the functions I want onto my phone? Thank you for your time. __ Do you Yahoo!? Yahoo! SiteBuilder - Free, easy-to-use web site design software http://sitebuilder.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
AW: [Asterisk-Users] ADSI Programs
Hi, one question: What you mean with unlocked ? -Ursprungliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von jerk face Gesendet: Mittwoch, 27. August 2003 18:31 An: [EMAIL PROTECTED] Betreff: [Asterisk-Users] ADSI Programs I just received an unlocked ADSI phone and I am playing with the ADSI script. I was wondering how I can include Voicemail functions (Check new messages, Delete message) into the soft buttons. I checked in app_voicemail.c and it looks like these functions have already been programmed. Is there a voicemail.adsi script somewhere? If not, then how do I get the functions I want onto my phone? Thank you for your time. __ Do you Yahoo!? Yahoo! SiteBuilder - Free, easy-to-use web site design software http://sitebuilder.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: AW: [Asterisk-Users] ADSI Programs
Some ADSI phones come locked to a certain service provider. You cannot load your own adsi scripts into these phones - you need one that isn't tied to a specific company or pbx. -d At 06:35 PM 8/27/2003 +0200, you wrote: Hi, one question: What you mean with unlocked ? -Ursprungliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von jerk face Gesendet: Mittwoch, 27. August 2003 18:31 An: [EMAIL PROTECTED] Betreff: [Asterisk-Users] ADSI Programs I just received an unlocked ADSI phone and I am playing with the ADSI script. I was wondering how I can include Voicemail functions (Check new messages, Delete message) into the soft buttons. I checked in app_voicemail.c and it looks like these functions have already been programmed. Is there a voicemail.adsi script somewhere? If not, then how do I get the functions I want onto my phone? Thank you for your time. __ Do you Yahoo!? Yahoo! SiteBuilder - Free, easy-to-use web site design software http://sitebuilder.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] include context
check 'help' include contexta in contextb regards Martin On Wed, 27 Aug 2003, Rattana BIV wrote: hi, how can I add or remove this line include = context by the command CLI ? regards Rattana ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sample configs / load module failure
Hi List, I am trying to locate some detailed documentation and sample configs. I downloaded and compiled Asterisk, and I haven't been able to find much detailed docs on the config files. The distribution I compiled and installed doesn't have any config files, and the handbook is good but doesn't cover all of the configs. Here's my specific problem, when launching Asterisk for the first time, it fails to launch with the following information: [res_parking.so]WARNING[1024]: File loader.c, Line 212 (ast_load_resource): /usr/lib/asterisk/modules/res_parking.so: undefined symbol: ast_moh_start WARNING[1024]: File loader.c, Line 368 (load_modules): Loading module res_parking.so failed! If this is merely a matter of not using the parking module, that's fine, but I can't find the docs on how to NOT use a specific module. Thanks, Ted ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: AW: [Asterisk-Users] ADSI Programs
I know all about most ADSI phones being locked. The first line of my email was I just received an unlocked ADSI phone and I am playing with the ADSI script. I have a Cybiolink P-I, and it is completely unlocked. --- denon [EMAIL PROTECTED] wrote: Some ADSI phones come locked to a certain service provider. You cannot load your own adsi scripts into these phones - you need one that isn't tied to a specific company or pbx. -d At 06:35 PM 8/27/2003 +0200, you wrote: Hi, one question: What you mean with unlocked ? -Ursprungliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von jerk face Gesendet: Mittwoch, 27. August 2003 18:31 An: [EMAIL PROTECTED] Betreff: [Asterisk-Users] ADSI Programs I just received an unlocked ADSI phone and I am playing with the ADSI script. I was wondering how I can include Voicemail functions (Check new messages, Delete message) into the soft buttons. I checked in app_voicemail.c and it looks like these functions have already been programmed. Is there a voicemail.adsi script somewhere? If not, then how do I get the functions I want onto my phone? Thank you for your time. __ Do you Yahoo!? Yahoo! SiteBuilder - Free, easy-to-use web site design software http://sitebuilder.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users __ Do you Yahoo!? Yahoo! SiteBuilder - Free, easy-to-use web site design software http://sitebuilder.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Question About BRI Cards
Gustavo Villaran wrote: Hi, im new in the list and i want to buy a BRI card that works with Asterisk PBX software for testing purpose, but i dont know which one works with that software. If someone knowns something that can help me, please write to me. Thanks Gustavo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users Nearly everything that is supported by I4l should work, though I´d personally recommend getting a passive AVM-Card and run it using the CAPI Channel Driver. Holger ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PCI X100P card interrupt problems
My X100P card seems to have interrupt clashes with my Sound card, any ideas to prevent this ? Thanks and Regards Ajit ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Default Flash Time
Anyone know offhand what the default flash time is? Where to find and adjust if necessary? Going to test out some analog sets with * and wanted to know. Sincerely, Andy Hester Consero (817)375-1244 (817)937-7977 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] PCI X100P card interrupt problems
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Ajit M Kallingal Sent: Wednesday, August 27, 2003 1:29 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] PCI X100P card interrupt problems My X100P card seems to have interrupt clashes with my Sound card, any ideas to prevent this ? Thanks and Regards Ajit ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users Ajit, Disable any unnecessary devices(ie serial or parallel port or usb) in your bios and put your X100P on its own interrupt. Sincerely, Andy Hester Consero ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Default Flash Time
zaptel.h:#defineZT_DEFAULT_FLASHTIME750 /* 750 ms default flash time */ Martin On Wed, 27 Aug 2003, Andy Hester wrote: Anyone know offhand what the default flash time is? Where to find and adjust if necessary? Going to test out some analog sets with * and wanted to know. Sincerely, Andy Hester Consero (817)375-1244 (817)937-7977 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Configuration Adtran TA 750
hello, Is this configuratoin possible: --FXO --FXO ADTRAN TA 750 - T1Card --- ASTERISK -FXOT1 line -FXO Telephone lines from Telcomp. regards, Bart
Re: [Asterisk-Users] Configuration Adtran TA 750
Yes it is possible. Please describe what you want in the future. As you can see below your mail looks like crap and wasted all your time drawing this mess out. You really should look at the source to your last message and see how nasty it was. On Wed, 2003-08-27 at 14:38, Bartosz Jozwiak wrote: hello, Is this configuratoin possible: --FXO --FXO ADTRAN TA 750 - T1Card --- ASTERISK -FXO T1 line -FXO Telephone lines from Telcomp. regards, Bart -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Configuration Adtran TA 750
Yes. Jared Smith On Wed, 2003-08-27 at 13:38, Bartosz Jozwiak wrote: hello, Is this configuratoin possible: --FXO --FXO ADTRAN TA 750 - T1Card --- ASTERISK -FXO T1 line -FXO Telephone lines from Telcomp. regards, Bart ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
