Re: [asterisk-users] WiFi SIP phones
Hi, I see the offer from Singtel here: http://www.singtel.com/mio . All was based on Nokia handsets. And other offer from Neuf Telecom http://www.cbronline.com/article_news.asp?guid=79B12F42-9EB5-47ED-9CEF-78BD3D8F7D1A They use handset based on Qtopia. So it seem that those dual mode phone has come out from the lab. Personally I use an E61. Working. Does anyone has experience with GreenPhone (Qtopia) from Trolltech? On 5/23/07, SIP [EMAIL PROTECTED] wrote: In all honesty, things have NOT moved very far since you last saw them. Battery life has, overall, gotten somewhat better. Range is still abominable in most of them, and they're not, as a general rule, all that easy to deal with. We've mucked about with the Linksys WIP3XX series, the UTStarCom F1000G, and F3000, and the Nokia WiFi-enabled GSM phones (E60, E61). Of those we've played with, the Nokias are by far the most reliable to connect and remain connected (but perhaps the most annoying to set up). For in-office stuff, you're still not liable to beat ATAs + DECT phones. Outside the office, it really doesn't hurt to try and consolidate your Mobile and SIP service in one of the Nokias that support it (granted, if you're in the US, you'll have to buy them elsewhere, as the US Mobile providers have done their best to avoid using any sort of WiFi-capable GSM phones in their networks (Nokia has, for instance, the E62 for the US Cingular/ATT market which is, in every way, like the E61 except that it doesn't come with WiFi/SIP capability)). Some people swear by the UTStarCom phones, but we found their support to be incredibly substandard, their phone programming to be lacking in an understanding of the market, and their hardware to be flashy (in the case of the F3000), but troublesome with constant loss of connection and a difficulty to reestablish. The Linksys phones are, alas, just as annoying. It's kind of a shame, really. One would expect that to be a decent market -- IP phones that, you know, actually work well. But apparently, not so much. N. Chris Bagnall wrote: Greetings list, What are people's experiences with WiFi SIP phones? When I last looked into them about 18 months ago, they were incredibly expensive, had very limited range and poor battery life. In the end, it worked out much more cost effective to simply use ATAs + DECT cordless phones where there was a requirement for portable devices. I assume things must have moved on somewhat since then. What models are currently out there people would recommend I look at? Thanks in advance. Regards, Chris ___ --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 -- With best regards, Nguyen ___ --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] WiFi SIP phones
Am Donnerstag, den 24.05.2007, 08:23 +0300 schrieb Cosmin Prund: -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Remco Post Sent: Wednesday, May 23, 2007 10:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] WiFi SIP phones Tony Plack wrote: Are the DECT phones two channel or do they share a channel like most other portable phones? DECT is a digital standard, quite distantly comparable to GSM. There are multiple channels (I believe the standard allows for 12 channels, but the last time I actually worked on DECT is ages ago). A siemens S450IP can have up two 6 handsets with 2 'external' (SIP or POTS) phonecalls concurently. You cannot decline a phonecall, but you can ignore it. I'm curious: is the impossibility of declining an call a DECT limitation or is it that agest ago DECT phones were backed by POTS line and declining a call would make no sense since the POTS doesn't support it? For ISDN DECT phones (like the Siemens Gigaset S100isdn), there is very well the option to reject a call. Mine displays reject on the left button and stop ringing on the right button. Pressing reject will immediately give busy to the opposite side. So: No, it is not a DECT limitation, but handsets that are made for analog base stations probably do not have that button, and it might be a Siemens propriatory addition. BTW I use that DECT phone behind a Fritz!Box, which serves as ISDN ATA on my *, and pressing reject yields -- Executing Dial(...) in new stack -- Called sip501 -- SIP/sip501-08459061 is ringing [...] -- Got SIP response 486 Busy Here back from 80.136.199.41 -- SIP/sip501-08459061 is busy BR Anselm ___ --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] SCCP + hint
Thanks for your reply I use asterisk 1.4.4. Thanks in advance. Cheers, Alex. Michiel van Baak a écrit : On 12:19, Wed 23 May 07, Alexandre VERNIOL wrote: Hi all, Does someone know if it's possible to use hint function with skinny ? Can anyone send me an example ? Thanks in advance, Alex. What version of asterisk are you using? hints on chan_skinny work in -trunk ___ --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] There is no tone on an outgoing call
Hello, everyone. I'm having a strange problem with my asterisk. After dialing and before the other side picks up the phone I should hear the tones (I'm not sure what are they called: p---pii) and in almost all cases that is true. However there is a range of numbers where I'm having this problem. There is no tones, just silence, until someone picks up the phone. This does not occur when I call to those numbers with a mobile or regular PSTN phone. Has anyone experienced anything similar? Where I should be looking to correct this? Thanks in advance P.S. I'm using asterisk 1.2.18. The Dial command is the same for all calls: _X.,n,Dial(SIP/[EMAIL PROTECTED],45) ___ --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] redirect on AT-530 IP Phone
Good morning everybody! I have two AT-530 IP phones, when a call entry from outside (zap channel9 it goes to 101 extension. When I take the call and start to speak with the other person, how can I redirect this call to 102 extension? Thank to all. Bye!!! ___ --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] Caller ID matching
got it. thanx
On 5/24/07, Matthew Yingling [EMAIL PROTECTED] wrote:
We use this macro, which works quite well:
[macro-checkuservoicemail]
; ${ARG1} - Device extension(s) to check for mail
; Usage
; in main context do exten = 1000,1,Macro(checkuservoicemail,101)
exten = s,1,NoOp(Entering CheckUserVoiceMail for ${MACRO_EXTEN})
exten = s,n,GotoIf($[${MACRO_EXTEN} = ${ARG1}]?:NoMatchVM)
exten = s,n,Playback(beep) ; Hack for UIP200 clipping bug
exten = s,n,VoicemailMain([EMAIL PROTECTED] [EMAIL PROTECTED]) ;
Check vmail
exten = s,n,Hangup ; Hangup after checking vmail
exten = s,n(NoMatchVM),NoOp(End checkuservoicemail)
-Original Message-
*From:* [EMAIL PROTECTED] [mailto:
[EMAIL PROTECTED] Behalf Of *Mike Hammett
*Sent:* Tuesday, May 22, 2007 9:37 AM
*To:* 'Asterisk Users Mailing List - Non-Commercial Discussion'
*Subject:* RE: [asterisk-users] Caller ID matching
Yeah, I was trying to have it match the caller ID with what they're
dialing so that I don't have a separate entry for every customer.
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
*From:* [EMAIL PROTECTED] [mailto:
[EMAIL PROTECTED] *On Behalf Of *Rizwan Hisham
*Sent:* Tuesday, May 22, 2007 5:14 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Caller ID matching
I did it anyway. i used another way around to do it:
suppose 88777 is your number
exten= 88777,1,Dial(SIP/you)
exten= 88777/88777,1,VoiceMailMain()
but in this case you will have to make a separate vm extension for every
user.
On 5/22/07, *Rizwan Hisham* [EMAIL PROTECTED] wrote:
well i have tried to solve your problem, making your extensions in my
dialplan and reloading dialplan gives me segmentation fault. im afraid i
cant help u :)
exten = 555*,1,NoOp(${CALLERID(num)})
exten = 555*,2,Hangup
On 5/20/07, *Mike Hammett* [EMAIL PROTECTED] wrote:
What's going on here? 555* seems to indicate that the number is being
passed as the callerID because NoOp says the phone number.
I'm trying to emulate cell phone voicemail where you call your own number
to check your voicemail.
-- Accepting AUTHENTICATED call from 65.182.165.XXX:
requested format = gsm,
requested prefs = (),
actual format = ulaw,
host prefs = (ulaw),
priority = mine
-- Executing NoOp(IAX2/815748-16, 815748) in new stack
-- Executing Hangup(IAX2/815748-16, ) in new stack
== Spawn extension (outbound-scripted, 555*, 2) exited non-zero on
'IAX2/815748-16'
May 20 11:10:34 ERROR[3286]: cdr_addon_mysql.c:144 mysql_log: cdr_mysql:
cannot connect to database server localhost.
-- Hungup 'IAX2/815748-16'
May 20 11:11:00 NOTICE[3275]: chan_iax2.c:7323 socket_read: Rejected
connect attempt from 65.182.165.XXX, request '[EMAIL PROTECTED]'
does not exist
exten = ${CALLERID(NUM)}/${CALLERID(NUM)},1,Answer
exten = ${CALLERID(NUM)}/${CALLERID(NUM)},2,NoOp(It's here)
exten =
${CALLERID(NUM)}/${CALLERID(NUM)},3,VoicemailMain(${CALLERID(NUM)})
exten = ${CALLERID(NUM)}/${CALLERID(NUM)},4,Hangup()
exten = 555*,1,NoOp(${CALLERID(num)})
exten = 555*,2,Hangup
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
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--
Rizwan Hisham
Software Engineer
AXVOICE Inc.
--
Rizwan Hisham
Software Engineer
AXVOICE Inc.
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--
Rizwan Hisham
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AXVOICE Inc.
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[asterisk-users] asterisk-backports.org giveaway
hi all I created asterisk-backports.org a year back for backporting stuff from unstable or new versions of asterisk to stable or older versions. now, since we're leaving asterisk for callweaver, and since the domain is about to expire, I have no more needs for this project. However, if someone wants to take over, be my guest :) roy --- Roy Sigurd Karlsbakk [EMAIL PROTECTED] Tlf: 98013356 --- Why is it drug addicts and computer afficionados are both called users? -- Clifford Stol ___ --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] There is no tone on an outgoing call
On Thursday 24 May 2007 09:44, dima wrote: in almost all cases that is true However there is a range of numbers where I'm having this problem. There is no tones, just silence, until someone picks up the phone. This does not occur when I call to those numbers with a mobile or regular PSTN phone. If it's working for most numbers and these few then I would suggest it's an issue at the other end and not yours. P.S. I'm using asterisk 1.2.18. The Dial command is the same for all calls: _X.,n,Dial(SIP/[EMAIL PROTECTED],45) Try using: _X.,n,Dial(SIP/[EMAIL PROTECTED],45,r) r- Indicate ringing to the calling party. Pass no audio to the calling party until the called channel has answered. It's possibly what your mobile and PSTN supplier do themselves... - Barry -- Kind regards, Barry O'Donovan http://www.barryodonovan.com/ ___ --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] modprobe
Hello every boy again I have some problems with modprobe. When I type modprobe zaphfc, this error happens FATAL: Module zaphfc not found. And when I tyoe ztcfg -vv this error happens: Notice: Configuration file is /etc/zaptel.conf line 0: Unable to open master device '/dev/zap/ctl' 1 error(s) detected Someone can help me??? Thanks to all. ___ --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] There is no tone on an outgoing call
Am Donnerstag, den 24.05.2007, 10:44 +0200 schrieb dima: Hello, everyone. I'm having a strange problem with my asterisk. After dialing and before the other side picks up the phone I should hear the tones (I'm not sure what are they called: p---pii) and in almost all cases that is true. However there is a range of numbers where I'm having this problem. There is no tones, just silence, until someone picks up the phone. This does not occur when I call to those numbers with a mobile or regular PSTN phone. Has anyone experienced anything similar? Where I should be looking to correct this? Thanks in advance P.S. I'm using asterisk 1.2.18. The Dial command is the same for all calls: _X.,n,Dial(SIP/[EMAIL PROTECTED],45) Same thing here for some calls (randomly, seems not to depend on the number called, but I do not keep statistics of it). I have been to lazy to investigate (it seems only my personal SIP provider does this, but the other SIP lines my * is connected to) -- I think there is a Dial() option that makes sure that a ringing sound is provided even if the upstream provider does not send it. See the voip-info.org page about Dial(), it should be covered there. BR Anselm ___ --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] modprobe
We would need more details to help -- version of asterisk and zaptel and what you did to try to install them -- hardware you have, etc and why you did that modprobe statement. on Thursday 05/24/2007 Josu Lazkano([EMAIL PROTECTED]) wrote Hello every boy again I have some problems with modprobe. When I type modprobe zaphfc, this error happens FATAL: Module zaphfc not found. And when I tyoe ztcfg -vv this error happens: Notice: Configuration file is /etc/zaptel.conf line 0: Unable to open master device '/dev/zap/ctl' 1 error(s) detected Someone can help me??? Thanks to all. Hello every boy againbrbrI have some problems with modprobe. When I type quot;modprobe zaphfcquot;, this error happens quot;FATAL: Module zaphfc not found.quot;brbrAnd when I tyoe quot;ztcfg -vvquot; this error happens: brbrNotice: Configuration file is /etc/zaptel.confbrline 0: Unable to open master device #39;/dev/zap/ctl#39;brbr1 error(s) detectedbrbrSomeone can help me???brbrThanks to all.br ___ --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 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [EMAIL PROTECTED] ___ --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] modprobe
On Thu, May 24, 2007 at 11:17:57AM +0200, Josu Lazkano wrote: Hello every boy again I have some problems with modprobe. When I type modprobe zaphfc, this error happens FATAL: Module zaphfc not found. zaphfc is part of bristuff. have you installed brisuff (or any other bristuffed zaptel package, such as the one from Debian)? And when I tyoe ztcfg -vv this error happens: Notice: Configuration file is /etc/zaptel.conf line 0: Unable to open master device '/dev/zap/ctl' This is normally an indication that the module zaptel is not loaded. Which makes sense, as the driver you modprobed for did not exist nd hence could not pull zaptel with it. What zaptel hardware do you have? How have you installed Zaptel? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] modprobe
Hi Josu, I had the same problem with wctdm.I just loaded zaptel before wctdm and it was all ok. Hope it can help you. :) Giorgio Incantalupo Josu Lazkano wrote: Hello every boy again I have some problems with modprobe. When I type modprobe zaphfc, this error happens FATAL: Module zaphfc not found. And when I tyoe ztcfg -vv this error happens: Notice: Configuration file is /etc/zaptel.conf line 0: Unable to open master device '/dev/zap/ctl' 1 error(s) detected Someone can help me??? Thanks to all. ___ --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 -- _ Giorgio Incantalupo, mailto:[EMAIL PROTECTED] FGA srl - http://www.fgasoftware.com - [EMAIL PROTECTED] - The Agile PBX http://www.voiceatwork.eu Tel: 02997663.14, Fax: 0291390172 ___ --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] modprobe
Hello john, thanks for response. I am trying to install a Billion ISDN on Asterisk I have Debian Etch and I installed theese packages: apt-get install linux-headers-`uname -r` apt-get install make apt-get install ncurses-base ncurses-bin ncurses-term apt-get install libncurses5 libncurses5-dev apt-get install bison apt-get install openssl apt-get install libssl0.9.8 apt-get install libssl-dev apt-get install libeditline0 libeditline-dev libedit-dev libedit2 apt-get install gcc apt-get install zlib1g-dev To install Asterisk with Bristuff I do that: in usr/src: wget http://212.91.251.199/~junghanns.net/downloads/bristuff-0.3.0-current.tar.gz http://212.91.251.199/%7Ejunghanns.net/downloads/bristuff-0.3.0-current.tar.gz tar zxvf bristuff-0.3.0-current.tar.gz cd bristuff-0.3.0-PRE-1r ./install.sh That could help? Thanksss 2007/5/24, John covici [EMAIL PROTECTED]: We would need more details to help -- version of asterisk and zaptel and what you did to try to install them -- hardware you have, etc and why you did that modprobe statement. on Thursday 05/24/2007 Josu Lazkano([EMAIL PROTECTED]) wrote Hello every boy again I have some problems with modprobe. When I type modprobe zaphfc, this error happens FATAL: Module zaphfc not found. And when I tyoe ztcfg -vv this error happens: Notice: Configuration file is /etc/zaptel.conf line 0: Unable to open master device '/dev/zap/ctl' 1 error(s) detected Someone can help me??? Thanks to all. Hello every boy againbrbrI have some problems with modprobe. When I type quot;modprobe zaphfcquot;, this error happens quot;FATAL: Module zaphfc not found.quot;brbrAnd when I tyoe quot;ztcfg -vvquot; this error happens: brbrNotice: Configuration file is /etc/zaptel.confbrline 0: Unable to open master device #39;/dev/zap/ctl#39;brbr1 error(s) detectedbrbrSomeone can help me???brbrThanks to all.br ___ --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 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [EMAIL PROTECTED] ___ --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 ___ --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] modprobe
Hello Tzafrir, thanks for response. I am trying to install a Billion ISDN on Asterisk I have Debian Etch and I installed theese packages: apt-get install linux-headers-`uname -r` apt-get install make apt-get install ncurses-base ncurses-bin ncurses-term apt-get install libncurses5 libncurses5-dev apt-get install bison apt-get install openssl apt-get install libssl0.9.8 apt-get install libssl-dev apt-get install libeditline0 libeditline-dev libedit-dev libedit2 apt-get install gcc apt-get install zlib1g-dev To install Asterisk with Bristuff I do that: in usr/src: wget http://212.91.251.199/~junghanns.net/downloads/bristuff-0.3.0-current.tar.gz http://212.91.251.199/%7Ejunghanns.net/downloads/bristuff-0.3.0-current.tar.gz tar zxvf bristuff-0.3.0-current.tar.gz cd bristuff-0.3.0-PRE-1r ./install.sh That could help? Thanksss 2007/5/24, Tzafrir Cohen [EMAIL PROTECTED]: On Thu, May 24, 2007 at 11:17:57AM +0200, Josu Lazkano wrote: Hello every boy again I have some problems with modprobe. When I type modprobe zaphfc, this error happens FATAL: Module zaphfc not found. zaphfc is part of bristuff. have you installed brisuff (or any other bristuffed zaptel package, such as the one from Debian)? And when I tyoe ztcfg -vv this error happens: Notice: Configuration file is /etc/zaptel.conf line 0: Unable to open master device '/dev/zap/ctl' This is normally an indication that the module zaptel is not loaded. Which makes sense, as the driver you modprobed for did not exist nd hence could not pull zaptel with it. What zaptel hardware do you have? How have you installed Zaptel? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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 ___ --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] modprobe
On Thu, May 24, 2007 at 12:16:26PM +0200, Giorgio Incantalupo wrote: Hi Josu, I had the same problem with wctdm.I just loaded zaptel before wctdm and it was all ok. Hope it can help you. :) Actually, you don't need to modprobe zaptel explicitly. You just need to not run ztcfg at modprobe time. On some systems (notably RHEL4/centos4, and probably some matching fedoras) generation of the udev nodes was horribly slow, and hence you need to wait a few seconds after you insmod zaptel (which happens implicitly when you modproe any other zaptel module) till you can run ztcfg. But this is not really his problem. In his case zaptel did not get loaded at all. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] modprobe
On Thu, May 24, 2007 at 12:23:07PM +0200, Josu Lazkano wrote: Hello john, thanks for response. I am trying to install a Billion ISDN on Asterisk I have Debian Etch and I installed theese packages: apt-get install linux-headers-`uname -r` apt-get install make apt-get install ncurses-base ncurses-bin ncurses-term [ snip] Yeah. That sounds familiar... You seem to have ignored my response to your previous post. So a short summary: A. it would be much faster for you to set up a system with standard packages from Debian Etch B. The package you're trying to use actually pulls lder zaptel/asterisk than the ones in Etch. See that previous post for links to newer bristuff. C. You had a problem installing zaptel . Read the README there regarding setting up the /usr/src/linux link that nobody really needs except the bristuff build script. Ignoring previous replies and starting totally new threads is *not* a way to gin popularity here. I also hope you realise that it is a good method to get a bunch of irrelevant replies. Please follow-up on the original thread. Reply to your or my message there so mail programs will consider it part of the same thread. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] PRI problem, pri_fixup_principle: Call specified, but not found?
Hi, in a PRI setup, the receiving side is changing the B channel at proceeding. It seems this sometimes breaks some logic (pri_fixup_principle) and then the hangup kind of breaks, release is not answered and a restart cycle is triggered (by remote side). Anyone can help me debug this ? I've seen many posts with simmilar issues but no answer/solution. This is happening on Asterisk 1.2.16 + libpri 1.2.4 on a sangoma A104D. On a general side, where can I find a document (other than sources :) to start digging where the Zap-pri mapping is done and how, or at leaset what pri_fixup_principle is supposed to do... TIA, -- Carlos G Mendioroz [EMAIL PROTECTED] ___ --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] modprobe
Thanks Giorgio!!! I made modprobe zaptel and then ztcfg -vv anI have this: Zaptel Configuration == SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Clear channel (Default) (Slaves: 01) Channel 02: Clear channel (Default) (Slaves: 02) Channel 03: D-channel (Default) (Slaves: 03) 3 channels configured. ZT_SPANCONFIG failed on span 1: No such device or address (6) I think is better but not enough, thanks for that. Anyone uses the Billion ISDN PCI? Thanks every body!!! 2007/5/24, Giorgio Incantalupo [EMAIL PROTECTED]: Hi Josu, I had the same problem with wctdm.I just loaded zaptel before wctdm and it was all ok. Hope it can help you. :) Giorgio Incantalupo Josu Lazkano wrote: Hello every boy again I have some problems with modprobe. When I type modprobe zaphfc, this error happens FATAL: Module zaphfc not found. And when I tyoe ztcfg -vv this error happens: Notice: Configuration file is /etc/zaptel.conf line 0: Unable to open master device '/dev/zap/ctl' 1 error(s) detected Someone can help me??? Thanks to all. ___ --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 -- _ Giorgio Incantalupo, mailto:[EMAIL PROTECTED] FGA srl - http://www.fgasoftware.com - [EMAIL PROTECTED] - The Agile PBX http://www.voiceatwork.eu Tel: 02997663.14, Fax: 0291390172 ___ --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 ___ --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] WiFi SIP phones
I've gotten SIP calls to work via hotspots on a Dell Axim running SJ-Phone. I've also had reasonable success with a Nokia E60. I've had ZERO luck from a hotspot on the UTStarCom phones, nor on the Linksys wifi phones. I'm not quite sure yet why something which is ONLY a wifi phone has more issues connecting to wifi hotspots than, say, multi-function devices. However, looking at the shortcuts both developers have taken in their firmware, I'm of the opinion it's just sloppy code. Michael Graves wrote: I travel a lot for work. I frequently find hotels that have wifi, free or otherwise available. But I've yet to find it anywhere near sufficient to support voip applications. At least not good enough to compel me to not use my cell phone. If you have control of the host LAN then you can ensure it meets the needs of a wifi SIP phone, otherwise why bother. Has anyone ever seen anyone making a voip call on a wif handset ata public hotspot? While that would score many geek points I doubt it would work in many places. About 18 mo ago I bought the Hitachi Cable WIP5000 handset. It was seriously flawed so I resold it after a few months and settled on the Aastra desk phone. I do wish the cordless handsets were a little more like a Panasonic cordless phone...more buttons...easier to program, etc. Michael On Wed, 23 May 2007 21:59:03 -0400, Justin Moore wrote: On 5/23/07, Michael Graves [EMAIL PROTECTED] wrote: I must say that I've VERY happy with my Aastra 4801 CT phones. I think that they're DECT. Each can have up to six cordless handsets. Technically its a 9 line phone, but if you use G.729 you can only sustain two calls at once. I can have a call on the portable and easily take another on the base. I am also an extremely happy user of an Aastra 480i CT. Awesome phone. However, I was under the impression that the OP was looking for a WiFi phone that could be carried from place to place, but I may be wrong... -- Justin Moore aka wantmoore --- _www.wantmoore.com_ ___ --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_ ___ --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 ___ --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] PRI problem, pri_fixup_principle: Call specified, but not found?
in a PRI setup, the receiving side is changing the B channel at proceeding. Please post the layer 3 trace for this call scenario so that members of this forum can help you debug your problem. John Treble Ottawa, Canada -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Carlos G Mendioroz Sent: May 24, 2007 7:53 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] PRI problem, pri_fixup_principle: Call specified,but not found? Hi, in a PRI setup, the receiving side is changing the B channel at proceeding. It seems this sometimes breaks some logic (pri_fixup_principle) and then the hangup kind of breaks, release is not answered and a restart cycle is triggered (by remote side). Anyone can help me debug this ? I've seen many posts with simmilar issues but no answer/solution. This is happening on Asterisk 1.2.16 + libpri 1.2.4 on a sangoma A104D. On a general side, where can I find a document (other than sources :) to start digging where the Zap-pri mapping is done and how, or at leaset what pri_fixup_principle is supposed to do... TIA, -- Carlos G Mendioroz [EMAIL PROTECTED] ___ --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 ___ --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] CDR on channel 'IAX2/u92613106-3' already started
On Wed, 2007-05-23 at 20:51 -0600, Mike Diehl wrote: Hi all, I'm having a problem with an asterisk server being unable to call certain cellphones and answering machines. Anytime the person answers the phone call, everything works well. But when the call goes to voicemail or an answering machine, I get the error message below: --- snip --- Mike-- I assume you're using 1.4 or trunk; normally this shouldn't be a big deal. Exactly which version of 1.4 are you using? Have you tried the latest SVN version? I made some fixes concerning check_start. murf -- Steve Murphy [EMAIL PROTECTED] Digium ___ --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] Asterisk Time Card
Hello All, I have been looking for this solution for quite sometimes Asterisk Time Card System. I found some discussion from Digium forum but not quite helpful. Can anyone redirect me to the correct path? Thanks, Nitesh ___ --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] PSP Voip
This isn't a serious post but Interesting announcement yesterday about PSP's being hooked up with voip services to not only their 'home' pc's but also public wifi hotspots. http://gigaom.com/2007/05/23/sony-psp-voip-usa/ Anyone have any further information? Are they allowing inbound call services as well or outbound only? Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). http://click.mexuar.com/webuser/click/7/userurl/Cognation http://click.mexuar.com/webuser/nojs/7/userurl/Cognation image001.gif___ --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] Re: Kernel Panic in wct4xxp during unload on Zaptel-1.4.4
On 5/22/07, Axel Thimm [EMAIL PROTECTED] wrote: Have you tried using the 1.4.x atrpms packages? I did try the 1.4 packages from atrpms overnight yesterday, with similar results. I was able to address the kernel panic when unloading by commenting out ztcfg -s in the stop() function of the init script (based on suggestions on this list). The system appeared stable (went through several clean startup/shutdown cycles), but then proceeded to kernel panic four times in three hours when calls were being processed, forcing me to downgrade to 1.2 again. Unfortunately, I was remote and unable to capture the full kernel panic details, so I'm kind of stuck at square one until I can upgrade again during a maint window and attempt to force a panic during call processing. -- j. ___ --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] Parking Lot CallerID
Is there anyway of storing an incoming calls CallerID on a parked call
and having it restored when someone picks up the parked call?
I've tried storing the CID as a global variable and restoring it in my
dialplan, and while NoOp shows it working, the phone ignores it and uses
the parking lot extension for callerid instead. I believe this is
because the phone is calling out instead of a call coming in, is there
anyway around this?
This is a basic idea of what I've done to try to capture the CID in
testing:
exten = 200,1,SetGlobalVar(P1NAME=${CALLERID(NAME)})
exten = 200,n,SetGlobalVar(P1NUM=${CALLERID(NUM)})
exten = 200,n,Park()
exten = _20x,1,Wait(1)
exten = _20x,n,NoOp(${DIAL_OPTIONS})
exten = _20x,n,Set(CALLERID(NAME)=${P1NAME})
exten = _20x,n,Set(CALLERID(NUM)=${P1NUM})
exten = _20x,n,NoOp(${CALLERID(NAME)} ${CALLERID(NUM)})
exten = _20x,n,ParkedCall(${EXTEN})
Beings the call is originating on the phone I'm not sure there's a way
to push the CID back to it, any thoughts?
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[asterisk-users] bridging calls between two numbers with extensions
I am using the Originate Manager API call to bridge a call between two numbers. The first number is called by originate and the action is set to Dial the other number. If the second number has an extension that needs to be sent using DTMF I can use the D option when I call Dial like this: Local/[EMAIL PROTECTED]||D(ww1ww2ww3ww4) This works fine. My question is how do I send DTMF to the first number called? As far as I know there is no way to send DMTF using Originate like you can with Dial. Thanks, Matthew Boedicker ___ --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] Parking Lot CallerID
On Thu, 24 May 2007, Ken Williams wrote: Beings the call is originating on the phone I'm not sure there's a way to push the CID back to it, any thoughts? It is possible that the phone accepts some form of SIP NOTIFY message for revising its caller ID display. Some caller ID / CNAM is implemented this way because the lookup takes place subsequent to the establishment of end-to-end signaling parameters, mirroring the ISDN setting for the same. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 ___ --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] Asterisk Time Card
On 5/24/07, Nitesh Divecha [EMAIL PROTECTED] wrote: Hello All, I have been looking for this solution for quite sometimes Asterisk Time Card System. I found some discussion from Digium forum but not quite helpful. Hi what is the mean of time card system ? is this kind of attendent system ? kindly give some more details ram ___ --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] Asterisk Clusters
On 5/23/07, Nitesh Divecha [EMAIL PROTECTED] wrote: Hello All, I need to implement a clustered PBX System where parent * is connected to one of the outbound carrier and other child * will register to parent *. Reason for this implementation is because some of the child * are behind NAT. Parent * is on Public IP Address and its connected to outbound carrier. Child * will only send out long distances calls to Parent * to terminate, rest are internal calls. Now which is the best way to implement this type of scenario... DUNDi? or custom context? Hi why dont you looking this kind of solution DS3TDMOVERIPSER-ASterisk ram Thanks, Nitesh ___ --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 ___ --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] PRI problem, pri_fixup_principle: Call specified, but not found?
Here... Please advise if any special flags/options are needed. [ 02 01 01 27 ] Supervisory frame: SAPI: 00 C/R: 1 EA: 0 TEI: 000EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 019 P/F: 1 0 bytes of data -- ACKing all packets from 18 to (but not including) 19 -- Since there was nothing left, stopping T200 counter -- Stopping T203 counter since we got an ACK -- Nothing left, starting T203 counter -- Unsolicited RR with P/F bit, responding Sending Receiver Ready (51) [ 02 01 01 67 ] Supervisory frame: SAPI: 00 C/R: 1 EA: 0 TEI: 000EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 051 P/F: 1 0 bytes of data -- Restarting T203 counter -- Restarting T203 counter T203 counter expired, sending RR and scheduling T203 again Sending Receiver Ready (51) [ 00 01 01 67 ] Supervisory frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 051 P/F: 1 0 bytes of data -- Restarting T203 counter [ 00 01 01 27 ] Supervisory frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 019 P/F: 1 0 bytes of data -- ACKing all packets from 18 to (but not including) 19 -- Since there was nothing left, stopping T200 counter -- Stopping T203 counter since we got an ACK -- Nothing left, starting T203 counter -- Got RR response to our frame -- Restarting T203 counter [ 02 01 01 b1 ] Supervisory frame: SAPI: 00 C/R: 1 EA: 0 TEI: 000EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 088 P/F: 1 0 bytes of data -- ACKing all packets from 87 to (but not including) 88 -- Since there was nothing left, stopping T200 counter -- Stopping T203 counter since we got an ACK -- Nothing left, starting T203 counter -- Unsolicited RR with P/F bit, responding Sending Receiver Ready (6) [ 02 01 01 0d ] Supervisory frame: SAPI: 00 C/R: 1 EA: 0 TEI: 000EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 006 P/F: 1 0 bytes of data -- Restarting T203 counter -- Restarting T203 counter T203 counter expired, sending RR and scheduling T203 again Sending Receiver Ready (6) [ 00 01 01 0d ] Supervisory frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 006 P/F: 1 0 bytes of data -- Restarting T203 counter [ 00 01 01 b1 ] Supervisory frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 088 P/F: 1 0 bytes of data -- ACKing all packets from 87 to (but not including) 88 -- Since there was nothing left, stopping T200 counter -- Stopping T203 counter since we got an ACK -- Nothing left, starting T203 counter -- Got RR response to our frame -- Restarting T203 counter -- Accepting AUTHENTICATED call from 10.8.0.6: requested format = alaw, requested prefs = (), actual format = alaw, host prefs = (), priority = mine -- Executing Dial(IAX2/10.8.0.6:4569-3, Zap/g2/113) in new stack -- Requested transfer capability: 0x00 - SPEECH [ 00 01 26 66 08 02 0e 89 05 04 03 80 90 a3 18 04 e9 82 83 81 28 08 43 61 72 6c 6f 73 20 4d 6c 06 41 81 31 31 30 30 70 04 c1 31 31 33 a1 ] Informational frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 N(S): 019 0: 0 N(R): 051 P: 0 41 bytes of data -- Restarting T203 counter Stopping T_203 timer Starting T_200 timer Protocol Discriminator: Q.931 (8) len=41 Call Ref: len= 2 (reference 3721/0xE89) (Originator) Message type: SETUP (5) [04 03 80 90 a3] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: A-Law (35) [18 04 e9 82 83 81] Channel ID (len= 6) [ Ext: 1 IntID: Explicit, PRI Spare: 0, Exclusive Dchan:0 ChanSel: Reserved Ext: 1 DS1 Identifier: 2 Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] [28 08 43 61 72 6c 6f 73 20 4d] Display (len= 8) [ Carlos M ] [6c 06 41 81 31 31 30 30] Calling Number (len= 8) [ Ext: 0 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number passed network screening (1) '1100' ] [70 04 c1 31 31 33] Called Number (len= 6) [ Ext: 1 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '113' ] [a1] Sending Complete (len= 1) -- Called g2/113 [ 00 01 01 28 ] Supervisory frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 020 P/F: 0 0 bytes of data -- ACKing all packets from 18 to (but not including) 20 -- ACKing packet 19, new txqueue is -1 (-1 means empty) -- Since there was nothing left, stopping T200
[asterisk-users] Nokia release
Hi all, sorry to ask you something not related to asterisk, but i really want to know whether the Nokia N95 cell phone is released in the USA or not? if somebody from USA knows, plz reply. -- Rizwan Hisham Software Engineer AXVOICE Inc. ___ --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] WiFi SIP phones
So many hotspots require some form of web based login, possibly even with acceptance of a browser cookie. That takes a lot fo wifi SIP devices out of the game. The best sucess I had while travelling with the WIP5000 in my bag was also carrying an Apple Airport Express. This allowed me to use wired network access in hotels on a wlan with known properties. However, while it was an interesting project, it was never as practical as my cell phone. Michael On Thu, 24 May 2007 08:00:54 -0400, SIP wrote: I've gotten SIP calls to work via hotspots on a Dell Axim running SJ-Phone. I've also had reasonable success with a Nokia E60. I've had ZERO luck from a hotspot on the UTStarCom phones, nor on the Linksys wifi phones. I'm not quite sure yet why something which is ONLY a wifi phone has more issues connecting to wifi hotspots than, say, multi-function devices. However, looking at the shortcuts both developers have taken in their firmware, I'm of the opinion it's just sloppy code. Michael Graves wrote: I travel a lot for work. I frequently find hotels that have wifi, free or otherwise available. But I've yet to find it anywhere near sufficient to support voip applications. At least not good enough to compel me to not use my cell phone. If you have control of the host LAN then you can ensure it meets the needs of a wifi SIP phone, otherwise why bother. Has anyone ever seen anyone making a voip call on a wif handset ata public hotspot? While that would score many geek points I doubt it would work in many places. About 18 mo ago I bought the Hitachi Cable WIP5000 handset. It was seriously flawed so I resold it after a few months and settled on the Aastra desk phone. I do wish the cordless handsets were a little more like a Panasonic cordless phone...more buttons...easier to program, etc. Michael On Wed, 23 May 2007 21:59:03 -0400, Justin Moore wrote: On 5/23/07, Michael Graves [EMAIL PROTECTED] wrote: I must say that I've VERY happy with my Aastra 4801 CT phones. I think that they're DECT. Each can have up to six cordless handsets. Technically its a 9 line phone, but if you use G.729 you can only sustain two calls at once. I can have a call on the portable and easily take another on the base. I am also an extremely happy user of an Aastra 480i CT. Awesome phone. However, I was under the impression that the OP was looking for a WiFi phone that could be carried from place to place, but I may be wrong... -- Justin Moore aka wantmoore --- _www.wantmoore.com_ ___ --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_ ___ --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 ___ --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 ___ --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] WiFi SIP phones
Has anyone installed Linux on your ABP phones, and got all functionality (including GSM and WiFi)? Will these phones work in the US (which radio frequency modes)? On Thu, 2007-05-24 at 00:49 -0700, [EMAIL PROTECTED] wrote: Date: Thu, 24 May 2007 00:10:23 -0500 From: Shanon Swafford [EMAIL PROTECTED] Subject: RE: [asterisk-users] WiFi SIP phones To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii I work for ABP Technology and lurk on this list so I hope I'm not breaking any taboos... ABP is now carrying a dual GSM/Wifi phone. We tested 2 models, 1 had Windows-CE on it. Some reason we only have the Non-CE version public right now. http://www.abptech.com/products/Pirelli/DPL10.html blocked::http://www.abptech.com/products/Pirelli/DPL10.html blocked::http://var.abptech.com/s.nl/it.A/id.2041/.f?sc=2category=31 VARs/Resellers/ITSPs/Consultants: http://www.abptech.com/support/qa/index.php?target=become_reseller blocked::http://www.abptech.com/support/qa/index.php?target=become_reseller End Users go here and we'll help you find a place to buy one: http://www.abptech.com/aboutus/find_reseller.php Shanon ABP Technology -- (C) Matthew Rubenstein ___ --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] Asterisk Time Card
Thanks for your reply, The basic system would work as follows: - Method 1 === An employee would call in to the system and a welcome message is prompted. After that a employee is asked to enter the employee ID and PIN number and once verified Employee ID, Caller ID, and time of day is stored into MySQL DB. By end of the day employee will call in again to logout from the system and same information is stored into the DB. Method 2 === This time employee is verified with Caller ID, so the employee ID and PIN number is skipped and time of day is logged into the DB. Is it possible? Thanks, Nitesh ram wrote: On 5/24/07, *Nitesh Divecha* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hello All, I have been looking for this solution for quite sometimes Asterisk Time Card System. I found some discussion from Digium forum but not quite helpful. Hi what is the mean of time card system ? is this kind of attendent system ? kindly give some more details ram ___ --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 ___ --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] Asterisk Time Card
On Thu, 24 May 2007, Nitesh Divecha wrote: I have been looking for this solution for quite sometimes Asterisk Time Card System. I found some discussion from Digium forum but not quite helpful. Are you by chance referring to chipsets that provide hardware timing / Real-Time Clock functionality used by Asterisk? -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 ___ --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] SLA with SIP-only environment
Hello, All the examples of SLA talk about Zap channels from one side and SIP on the other side, while my system is a pure SIP one. I would like to have two phones having extensions 1 2 defined on them, and when someone calls extension 1 it rings on both, each one can see its status, and when one station puts line 1 on hold the other one can pick it. Is it possible at all? If so, can someone give the relevant fragments for sla.conf, sip.conf and extensions.conf? Thanks! __Yehavi: ___ --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] Asterisk Time Card
This is all definitely possible by using Asterisk database interfaces, but I cannot find an existing implementation of something of this nature. It is an unusual and clever application of Asterisk. :-) -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 ___ --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] PSP Voip
What was the content of the message you sent? And what is the deal with these messages the list delivers scrubbed of their content? Maybe the listbot can't handle multipart/alternative MIME messages. On Thu, 2007-05-24 at 05:52 -0700, [EMAIL PROTECTED] wrote: Date: Thu, 24 May 2007 08:50:58 -0400 From: Dean Collins [EMAIL PROTECTED] Subject: [asterisk-users] PSP Voip To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Skipped content of type multipart/alternative-- next part -- A non-text attachment was scrubbed... Name: not available Type: image/gif Size: 2775 bytes Desc: image001.gif Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20070524/7908e840/attachment.gif -- (C) Matthew Rubenstein ___ --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] Asterisk Time Card
On 5/24/07, Nitesh Divecha [EMAIL PROTECTED] wrote: Thanks for your reply, The basic system would work as follows: - Method 1 === An employee would call in to the system and a welcome message is prompted. After that a employee is asked to enter the employee ID and PIN number and once verified Employee ID, Caller ID, and time of day is stored into MySQL DB. By end of the day employee will call in again to logout from the system and same information is stored into the DB. Method 2 === This time employee is verified with Caller ID, so the employee ID and PIN number is skipped and time of day is logged into the DB. Is it possible? Thanks, Nitesh Anything is possible. But I haven't seen one off-the-shelf. It really won't be a big deal to write, though. We created a timeclock application and toyed with allowing people to clock in via phone, and I even wrote the extension logic, but we opted to not enable it because we don't trust our employees that much. This was years ago, when we were running pre-1.0 code. We've switched servers a few times, so the logic is long gone, but it only took an afternoon to write and debug. ___ --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] meetme sounds
I am playing around with dynamic meetme conferences, and wanted to have one person constantly in the conference, with calls popping in and out. Is there an option / any way of playing enter / leave sounds to the person who created the conference only, and not the people leaving / joining ? TIA Julian. ___ --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] Additional commands for MeetMeAdmin
Would anybody mind if the the following command options where added to MeetMeAdmin? 0 - 9, * and # I'm considering hacking the code to add these commands to play the DTMFs to the specified user as tones and hope that the SIP or IAX channels then work with these correctly. -HJC ___ --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] Asterisk Time Card
Quoting Alex Balashov [EMAIL PROTECTED]: On Thu, 24 May 2007, Nitesh Divecha wrote: I have been looking for this solution for quite sometimes Asterisk Time Card System. I found some discussion from Digium forum but not quite helpful. Are you by chance referring to chipsets that provide hardware timing / Real-Time Clock functionality used by Asterisk? I think he means the prepaid phone card discussions -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 ___ --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 Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com This message was sent using IMP, the Internet Messaging Program. ___ --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] Asterisk Time Card
On 5/24/07, Alex Balashov [EMAIL PROTECTED] wrote: On Thu, 24 May 2007, Nitesh Divecha wrote: I have been looking for this solution for quite sometimes Asterisk Time Card System. I found some discussion from Digium forum but not quite helpful. Are you by chance referring to chipsets that provide hardware timing / Real-Time Clock functionality used by Asterisk? Unless I'm very much mistaken, he's referring to a Time and Attendance system. The idea is to capture times that a person clocks in and when the person clocks out, to simplify running payroll. ___ --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] Asterisk Time Card
Nitesh Divecha wrote: Thanks for your reply, The basic system would work as follows: - Method 1 === An employee would call in to the system and a welcome message is prompted. After that a employee is asked to enter the employee ID and PIN number and once verified Employee ID, Caller ID, and time of day is stored into MySQL DB. By end of the day employee will call in again to logout from the system and same information is stored into the DB. Method 2 === This time employee is verified with Caller ID, so the employee ID and PIN number is skipped and time of day is logged into the DB. Is it possible? Thanks, Nitesh Nitesh: This would be pretty easy using AGI. We haven't done time and attendance, but have implemented some reasonably complex IVR payment systems integrating with MySQL. Many others have done similar and even more extensive applications in this manner. Google Asterisk AGI and this should get you started. Mike Clark ___ --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] Integrated T1
Can an asterisk box equipped with a Digium T1 card handle Integrated T1 circuits? I have a T1 with 768k data and the remaining channels voice, can the asterisk box do the Data routing + Voice processing? It's only going to support 4-5 users(the voice channels won't all be active obviously). This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] SCCP
Any one knows where to install chan_sccp for asterisk 1.4 ???. Please guide me from where can I download the asterisk 1.4 sccp channel driver and how to install it because I tried to get chan_sccp-mayday05.tar.gz When I trying to install it ,error happened like this. Please help me how to solve this issue. [EMAIL PROTECTED] t]# cd chan_sccp [EMAIL PROTECTED] chan_sccp]# make clean rm -rf chan_sccp.so .tmp [EMAIL PROTECTED] chan_sccp]# make install \Now compiling chan_sccp.c 742 lines chan_sccp.c: In function `sccp_devicestate': chan_sccp.c:133: error: `AST_DEVICE_UNKNOWN' undeclared (first use in this function) chan_sccp.c:133: error: (Each undeclared identifier is reported only once chan_sccp.c:133: error: for each function it appears in.) chan_sccp.c: In function `reload_config': chan_sccp.c:397: warning: implicit declaration of function `ast_load' chan_sccp.c:397: warning: assignment makes pointer from integer without a cast chan_sccp.c:555: error: incompatible type for argument 1 of `ast_inet_ntoa' chan_sccp.c:555: error: too many arguments to function `ast_inet_ntoa' chan_sccp.c:562: error: incompatible type for argument 1 of `ast_inet_ntoa' chan_sccp.c:562: error: too many arguments to function `ast_inet_ntoa' chan_sccp.c:566: error: incompatible type for argument 1 of `ast_inet_ntoa' chan_sccp.c:566: error: too many arguments to function `ast_inet_ntoa' chan_sccp.c:574: error: incompatible type for argument 1 of `ast_inet_ntoa' chan_sccp.c:574: error: too many arguments to function `ast_inet_ntoa' chan_sccp.c: In function `setcalledparty_exec': chan_sccp.c:601: error: structure has no member named `type' chan_sccp.c: At top level: chan_sccp.c:666: warning: function declaration isn't a prototype chan_sccp.c:700: warning: no previous prototype for 'reload' chan_sccp.c:706: warning: function declaration isn't a prototype chan_sccp.c:730: warning: function declaration isn't a prototype chan_sccp.c:738: warning: function declaration isn't a prototype chan_sccp.c:740: warning: function declaration isn't a prototype make: *** [.tmp/chan_sccp.o] Error 1 * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --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] Fwd: TC400B load problem
-- Forwarded message -- From: Arun Kumar [EMAIL PROTECTED] Date: May 13, 2007 5:40 PM Subject: TC400B load problem To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Hi Im trying to install my TC400B trans coder card when I do: modprobe wctc4xxp tail -f /var/log/messages says: May 13 14:56:36 pbx2 kernel: Registered codec translator 'DTE Encoder' with 92 transcoders (srcs=000c, dsts=0101) May 13 14:56:36 pbx2 kernel: Registered codec translator 'DTE Decoder' with 92 transcoders (srcs=0101, dsts=000c) May 13 14:56:36 pbx2 kernel: Zaptel DTE (g.729a / g.723.1 5.3kbps) Transcoder support LOADED (firm ver = 56) May 13 14:56:49 pbx2 kernel: wctc4xxp: probe of :03:01.0 failed with error -5 please help thanks arun ___ --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] WiFi SIP phones
I took the SIM card out of my Cingular Razor, installed it in the demo we have, and it went right to work on GSM here in Dallas. Here are some particulars I pulled from the web page below: -Form Factor: Candy Bar Type (106Lx46Wx18H mm) -LCD Display: 1.5, 128*128 pixel, 65K colors -Protocols/Bands: GSM 900/1800/1900MHz -Wi-Fi: 802.11b/g. 2.4 GHz -Network Selection Choices: Wi-Fi Only / GSM Only / Wi-Fi Preferred / GSM Preferred -WLAN Security: WEP, WPA-PSK TKIP, WPA2-PSK AES -VoIP Codecs: G.711, G.726, G.729 -Camera: 300k pixel VGA http://www.abptech.com/products/Pirelli/DPL10.html Not sure about Linux, but our sales dept could find that out. Regards, Shanon 972-831-1600 -Original Message- From: Matthew Rubenstein [mailto:[EMAIL PROTECTED] Sent: Thursday, May 24, 2007 9:06 AM To: Shanon Swafford Cc: Asterisk-Users Subject: RE: [asterisk-users] WiFi SIP phones Has anyone installed Linux on your ABP phones, and got all functionality (including GSM and WiFi)? Will these phones work in the US (which radio frequency modes)? On Thu, 2007-05-24 at 00:49 -0700, [EMAIL PROTECTED] wrote: Date: Thu, 24 May 2007 00:10:23 -0500 From: Shanon Swafford [EMAIL PROTECTED] Subject: RE: [asterisk-users] WiFi SIP phones To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii I work for ABP Technology and lurk on this list so I hope I'm not breaking any taboos... ABP is now carrying a dual GSM/Wifi phone. We tested 2 models, 1 had Windows-CE on it. Some reason we only have the Non-CE version public right now. http://www.abptech.com/products/Pirelli/DPL10.html blocked::http://www.abptech.com/products/Pirelli/DPL10.html blocked::http://var.abptech.com/s.nl/it.A/id.2041/.f?sc=2category=31 VARs/Resellers/ITSPs/Consultants: http://www.abptech.com/support/qa/index.php?target=become_reseller blocked::http://www.abptech.com/support/qa/index.php?target=become_reseller End Users go here and we'll help you find a place to buy one: http://www.abptech.com/aboutus/find_reseller.php Shanon ABP Technology -- (C) Matthew Rubenstein ___ --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] Integrated T1
On 5/24/07, Jeremy Mann [EMAIL PROTECTED] wrote: Can an asterisk box equipped with a Digium T1 card handle Integrated T1 circuits? I have a T1 with 768k data and the remaining channels voice, can the asterisk box do the Data routing + Voice processing? Yes, zaptel will create a device node for you. Take a look at the set-hdlc tool in zaptel and the less common channel types in the default zaptel config file (rawhdlc is one, there are also others). ___ --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] Asterisk Time Card
On 5/24/07, David Gomillion [EMAIL PROTECTED] wrote: On 5/24/07, Nitesh Divecha [EMAIL PROTECTED] wrote: Thanks for your reply, The basic system would work as follows: - Method 1 === An employee would call in to the system and a welcome message is prompted. After that a employee is asked to enter the employee ID and PIN number and once verified Employee ID, Caller ID, and time of day is stored into MySQL DB. By end of the day employee will call in again to logout from the system and same information is stored into the DB. Method 2 === This time employee is verified with Caller ID, so the employee ID and PIN number is skipped and time of day is logged into the DB. Is it possible? Thanks, Nitesh Anything is possible. But I haven't seen one off-the-shelf. It really won't be a big deal to write, though. We created a timeclock application and toyed with allowing people to clock in via phone, and I even wrote the extension logic, but we opted to not enable it because we don't trust our employees that much. This was years ago, when we were running pre-1.0 code. We've switched servers a few times, so the logic is long gone, but it only took an afternoon to write and debug. with the AGI you can do all ram ___ --Bandwidth and Colocation provided by Easynews.com http://easynews.com/-- asterisk-users mailing list To 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
Re: [asterisk-users] Asterisk Time Card
This can be accomplished by writing an IVR to prompt and then using AGI or dialplan commands the query strings can be executed. I have a setup like this for a inegrating a in house time keeping system with asterisk. On 5/24/07, Nitesh Divecha [EMAIL PROTECTED] wrote: Thanks for your reply, The basic system would work as follows: - Method 1 === An employee would call in to the system and a welcome message is prompted. After that a employee is asked to enter the employee ID and PIN number and once verified Employee ID, Caller ID, and time of day is stored into MySQL DB. By end of the day employee will call in again to logout from the system and same information is stored into the DB. Method 2 === This time employee is verified with Caller ID, so the employee ID and PIN number is skipped and time of day is logged into the DB. Is it possible? Thanks, Nitesh ram wrote: On 5/24/07, *Nitesh Divecha* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hello All, I have been looking for this solution for quite sometimes Asterisk Time Card System. I found some discussion from Digium forum but not quite helpful. Hi what is the mean of time card system ? is this kind of attendent system ? kindly give some more details ram ___ --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 ___ --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 -- Bruce Reeves Nortex Networks ___ --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] Asterisk Time Card
On 5/24/07, Alex Balashov [EMAIL PROTECTED] wrote: This is all definitely possible by using Asterisk database interfaces, but I cannot find an existing implementation of something of this nature. It is an unusual and clever application of Asterisk. :-) Don't know how unusual. When I do contract work, most of the jobs I do have a phone number to log in and out thru. By the way, when I wrote the module, I cheated and used a System call (although I would use the TrySystem if I were to do it again) and called a very simple PHP script. Oh, and I authenticated within the dialplan so that I could easily play useful error messages without checking the returned value of the PHP script. Not the best system, but it worked in my testing. ___ --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] meetme sounds
You can specify different options to start meetme with (announcements, etc.) in the dialplan by having a separate extension for the person who wants to here the sounds. I've never tried this, but I think it should work. -kn0x On 5/24/07, Julian Lyndon-Smith [EMAIL PROTECTED] wrote: I am playing around with dynamic meetme conferences, and wanted to have one person constantly in the conference, with calls popping in and out. Is there an option / any way of playing enter / leave sounds to the person who created the conference only, and not the people leaving / joining ? TIA Julian. ___ --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 ___ --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] Asterisk Time Card
Alex, No, I don't refer to hardware timing... It is just a Unix time stamp as used by CDR's. Thanks, Nitesh Alex Balashov wrote: On Thu, 24 May 2007, Nitesh Divecha wrote: I have been looking for this solution for quite sometimes Asterisk Time Card System. I found some discussion from Digium forum but not quite helpful. Are you by chance referring to chipsets that provide hardware timing / Real-Time Clock functionality used by Asterisk? -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 ___ --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 ___ --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] meetme sounds
On Thu, 24 May 2007, Julian Lyndon-Smith wrote: I am playing around with dynamic meetme conferences, and wanted to have one person constantly in the conference, with calls popping in and out. Is there an option / any way of playing enter / leave sounds to the person who created the conference only, and not the people leaving / joining ? Not out of the box. I did this for a customer a couple of years ago. As I remember, the key was to add code to conf_run() to take the user out of the conference, play the custom sound file, and put them back into the conference. These in/out steps are needed to keep that user in sync with the conference. Otherwise, their audio will be offset by the length of the sound file. Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --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] Asterisk Time Card
Thanks David, Any code you can share... I just need a kick start... Nitesh David Gomillion wrote: On 5/24/07, *Nitesh Divecha* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Thanks for your reply, The basic system would work as follows: - Method 1 === An employee would call in to the system and a welcome message is prompted. After that a employee is asked to enter the employee ID and PIN number and once verified Employee ID, Caller ID, and time of day is stored into MySQL DB. By end of the day employee will call in again to logout from the system and same information is stored into the DB. Method 2 === This time employee is verified with Caller ID, so the employee ID and PIN number is skipped and time of day is logged into the DB. Is it possible? Thanks, Nitesh Anything is possible. But I haven't seen one off-the-shelf. It really won't be a big deal to write, though. We created a timeclock application and toyed with allowing people to clock in via phone, and I even wrote the extension logic, but we opted to not enable it because we don't trust our employees that much. This was years ago, when we were running pre-1.0 code. We've switched servers a few times, so the logic is long gone, but it only took an afternoon to write and debug. ___ --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 ___ --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] - SOLVED - stream file not working but get data and exec background work
Patrick Fortin wrote: Hi While testing I found a solution to my problem. I don't understand it maybe someone here can explain it. In my script, if I call a Playback just before my stream file then everything works ok. Without the playback then the digits are not captured I will playback a silence to patch my scripts. Playback will issue an Answer before playing the audio. Without the line being answered, you can't receive audio (or inband digits). ___ --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] Asterisk Time Card
David, You are correct... thats the whole scenario to simplify running payroll... I am planning to do three level of verifications which will make sure the employee is in right location, so he is not spoofing anything... 1) Verify by Employee ID and PIN. 2) Verify by Location ID. This will be printed at location site. 3) Verify by token ID, generated by http://www.mypw.com/ 4) Login the time or Logout the employee. If the Caller is calling from the registered Caller ID, then step 1 will be ignored. Kinda like Caller ID authentication. Thanks, Nitesh David Gomillion wrote: On 5/24/07, *Alex Balashov* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: On Thu, 24 May 2007, Nitesh Divecha wrote: I have been looking for this solution for quite sometimes Asterisk Time Card System. I found some discussion from Digium forum but not quite helpful. Are you by chance referring to chipsets that provide hardware timing / Real-Time Clock functionality used by Asterisk? Unless I'm very much mistaken, he's referring to a Time and Attendance system. The idea is to capture times that a person clocks in and when the person clocks out, to simplify running payroll. ___ --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 ___ --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] Asterisk Time Card
Thanks Mike, Will look into Asterisk AGI... Cheers, Nitesh Mike Clark wrote: Nitesh Divecha wrote: Thanks for your reply, The basic system would work as follows: - Method 1 === An employee would call in to the system and a welcome message is prompted. After that a employee is asked to enter the employee ID and PIN number and once verified Employee ID, Caller ID, and time of day is stored into MySQL DB. By end of the day employee will call in again to logout from the system and same information is stored into the DB. Method 2 === This time employee is verified with Caller ID, so the employee ID and PIN number is skipped and time of day is logged into the DB. Is it possible? Thanks, Nitesh Nitesh: This would be pretty easy using AGI. We haven't done time and attendance, but have implemented some reasonably complex IVR payment systems integrating with MySQL. Many others have done similar and even more extensive applications in this manner. Google Asterisk AGI and this should get you started. Mike Clark ___ --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 ___ --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] SCCP
On Thu, 2007-05-24 at 17:46 -0700, Khaled Chehab wrote: Any one knows where to install chan_sccp for asterisk 1.4 ???. Have you tried the chan_skinny driver that comes with 1.4? Alternatively I saw a chan_sccp version for 1.4.3 or 1.4.4 here: http://ting.ip-phone-forum.de/downloads.php?do=fileid=342 Regards, Patrick ___ --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] Additional commands for MeetMeAdmin
meetme DTMF doesn't work very well in SIP or IAX even if set to inband. I had to resort to playing DTMF tones as audio files to get it working for SIP and IAX, not a pretty thing. MATT--- On 5/24/07, Henry Cobb [EMAIL PROTECTED] wrote: Would anybody mind if the the following command options where added to MeetMeAdmin? 0 - 9, * and # I'm considering hacking the code to add these commands to play the DTMFs to the specified user as tones and hope that the SIP or IAX channels then work with these correctly. -HJC ___ --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 ___ --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] Integrated T1
On Thu, 24 May 2007, Jeremy Mann wrote: Can an asterisk box equipped with a Digium T1 card handle Integrated T1 circuits? I have a T1 with 768k data and the remaining channels voice, can the asterisk box do the Data routing + Voice processing? The Zaptel/Asterisk infrastructure can definitely break particular timeslots out of the T1 for voice, but it is not my impression that any existing WAN drivers for Linux support Digium cards or cohabitation with Zapata and can give you a serial data interface on other channels. This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : +1-678-954-0670 ___ --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] Re: meetme sounds
In article [EMAIL PROTECTED], Julian Lyndon-Smith [EMAIL PROTECTED] wrote: I am playing around with dynamic meetme conferences, and wanted to have one person constantly in the conference, with calls popping in and out. Is there an option / any way of playing enter / leave sounds to the person who created the conference only, and not the people leaving / joining ? Not without custom modifications to the code of app_meetme itself. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --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] Integrated T1
Here's a link that will get you most of the way there: http://www.voip-info.org/wiki/view/Asterisk+Data+Configuration If you have any issues with setup, I recommend you contact Digium's support to help you since I'm sure they've had the most experience with it. On 5/24/07, Alex Balashov [EMAIL PROTECTED] wrote: On Thu, 24 May 2007, Jeremy Mann wrote: Can an asterisk box equipped with a Digium T1 card handle Integrated T1 circuits? I have a T1 with 768k data and the remaining channels voice, can the asterisk box do the Data routing + Voice processing? The Zaptel/Asterisk infrastructure can definitely break particular timeslots out of the T1 for voice, but it is not my impression that any existing WAN drivers for Linux support Digium cards or cohabitation with Zapata and can give you a serial data interface on other channels. This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : +1-678-954-0670 ___ --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 ___ --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] Re: Additional commands for MeetMeAdmin
In article [EMAIL PROTECTED], Henry Cobb [EMAIL PROTECTED] wrote: Would anybody mind if the the following command options where added to MeetMeAdmin? I'm sure no-one minds what you do to your own copies of Asterisk! :-) 0 - 9, * and # Rather than create that many commands, why not have a d or D command (for digit, or dtmf), and then the 0-9, *, #, A-D as a parameter to that command? I'm considering hacking the code to add these commands to play the DTMFs to the specified user as tones and hope that the SIP or IAX channels then work with these correctly. Hmmm, what problem are you trying to solve? In Asterisk 1.4 and Trunk, there is an option to MeetMe for passing DTMF frames through the conference. Your command should not generate tones, but should use conf_queue_dtmf() to queue the dtmf into the conference. That way, the DTMF is delivered to each participant channel as an asterisk DTMF frame, and the channel can then handle it in a way appropriate to the technology (e.g. Zap channels would re-generate tones, VoIP channels would deliver DTMF control packets, etc). Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --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] Integrated T1
On Thu, 24 May 2007, William Moore wrote: On 5/24/07, Jeremy Mann [EMAIL PROTECTED] wrote: Can an asterisk box equipped with a Digium T1 card handle Integrated T1 circuits? I have a T1 with 768k data and the remaining channels voice, can the asterisk box do the Data routing + Voice processing? Yes, zaptel will create a device node for you. Take a look at the set-hdlc tool in zaptel and the less common channel types in the default zaptel config file (rawhdlc is one, there are also others). Interesting! Is this a relatively recent development? I stand corrected. Is there a way to make PPP-encapsulated T1 work as well? -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : +1-678-954-0670 ___ --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] Bottom line on fax reception
So what is the bottom line? Does it work or not. I've heard stories it works, it doesn't work, it kinda sorta works when it's not raining out side. Everything under the rainbow. What's the bottom line with recent updates on 1.2.x? Is it production ready for fax? By production ready I mean that it just works all the time and doesn't need any babysitting. Do I have to worry about dropped lines, sometimes not detecting incoming fax toneyada yada. I know I don't have to use fax on Asterisk but I really want to for various reasons. Mostly incoming but outgoing is a nice to have. Should I use an addon package and if so which one? Any help would be appreciated. ___ --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] vmoutcall
Hello guys, I have been looking for a way to call a cell phone after someone has left a voice mail and allow the user to enter the mailbox password to listen to it and found a very old entry of vmoutcall. I tried unsuccessfully to get it to work with 1.4 and is beyond me. Has anyone gotten this to work or has a way to accomplishing this? Most analog pbx's have this feature and I am amazed Asterisk does not natively. Any thoughts on getting similar functionality into the main code too? Regards, Paul Aviles ___ --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] SCCP
On 17:46, Thu 24 May 07, Khaled Chehab wrote: Any one knows where to install chan_sccp for asterisk 1.4 ???. Please guide me from where can I download the asterisk 1.4 sccp channel driver and how to install it because I tried to get chan_sccp-mayday05.tar.gz When I trying to install it ,error happened like this. Please help me how to solve this issue. try chan_skinny that is in asterisk by default -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called 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
Re: [asterisk-users] meetme sounds
On Thursday 24 May 2007 11:30 am, Steve Edwards wrote: As I remember, the key was to add code to conf_run() to take the user out of the conference, play the custom sound file, and put them back into the conference. These in/out steps are needed to keep that user in sync with the conference. Otherwise, their audio will be offset by the length of the sound file. Eep; I wonder if it would have been easier to mix the sound in to just their copy of the conference... if people were entering or leaving when others were talking it would create holes in the creator's audio. -A. ___ --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] Asterisk Time Card
Thanks David, Is it possible if you could share your code... All I need is just an idea and develop my own. Cheers, Nitesh David Gomillion wrote: On 5/24/07, *Alex Balashov* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: This is all definitely possible by using Asterisk database interfaces, but I cannot find an existing implementation of something of this nature. It is an unusual and clever application of Asterisk. :-) Don't know how unusual. When I do contract work, most of the jobs I do have a phone number to log in and out thru. By the way, when I wrote the module, I cheated and used a System call (although I would use the TrySystem if I were to do it again) and called a very simple PHP script. Oh, and I authenticated within the dialplan so that I could easily play useful error messages without checking the returned value of the PHP script. Not the best system, but it worked in my testing. ___ --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 ___ --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] Asterisk Time Card
Thanks Bruce, If possible could you share your code...? I just need an idea how to integrate and store info in DB. Cheers, Nitesh Bruce Reeves wrote: This can be accomplished by writing an IVR to prompt and then using AGI or dialplan commands the query strings can be executed. I have a setup like this for a inegrating a in house time keeping system with asterisk. On 5/24/07, *Nitesh Divecha* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Thanks for your reply, The basic system would work as follows: - Method 1 === An employee would call in to the system and a welcome message is prompted. After that a employee is asked to enter the employee ID and PIN number and once verified Employee ID, Caller ID, and time of day is stored into MySQL DB. By end of the day employee will call in again to logout from the system and same information is stored into the DB. Method 2 === This time employee is verified with Caller ID, so the employee ID and PIN number is skipped and time of day is logged into the DB. Is it possible? Thanks, Nitesh ram wrote: On 5/24/07, *Nitesh Divecha* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hello All, I have been looking for this solution for quite sometimes Asterisk Time Card System. I found some discussion from Digium forum but not quite helpful. Hi what is the mean of time card system ? is this kind of attendent system ? kindly give some more details ram ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves Nortex Networks ___ --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 ___ --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] Asterisk Time Card
Hi, I think is it easily doable with an AGI script, but i am not sure if there is any builtin function to do it. So you might want to look into that (AGI script) Kido Nitesh Divecha a écrit : Thanks for your reply, The basic system would work as follows: - Method 1 === An employee would call in to the system and a welcome message is prompted. After that a employee is asked to enter the employee ID and PIN number and once verified Employee ID, Caller ID, and time of day is stored into MySQL DB. By end of the day employee will call in again to logout from the system and same information is stored into the DB. Method 2 === This time employee is verified with Caller ID, so the employee ID and PIN number is skipped and time of day is logged into the DB. Is it possible? Thanks, Nitesh ram wrote: On 5/24/07, *Nitesh Divecha* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hello All, I have been looking for this solution for quite sometimes Asterisk Time Card System. I found some discussion from Digium forum but not quite helpful. Hi what is the mean of time card system ? is this kind of attendent system ? kindly give some more details ram ___ --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 ___ --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 ___ --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] Basic connection between Mitel 3300 ICP and Asterisk (trixbox) - from a clueless newbie....
Hi all, Our company has deployed a Mitel 3300 system (only about 2.5 years ago) and we are experimenting with setting up Asterisk in our head office (for business continuity, ie we have a bird flu epidemic and no-one can come in, therefore use SIP softphones at home to co-ordinate activity) and at a remote site in the Isle of Man (connected via 2Mbps SDSL) Ideally we'd like anyone on either Asterisk servers (IOM and London) to be able to dial anyone internally on the Mitel 3300 and vice-versa. We have got *one* SIP license so far for the Mitel for testing purposes. I am a bit crap on telephony, but as I have gathered so far we should be able to connect the two systems via either QSIG (with an appropriate card on the Asterisk server), DPNSS (which I'm not sure if any Asterisk compatible hardware supports) or SIP (I'm happy setting up clients, but have no clue with inter-PBX stuff). I don't really care about any special features as long as the Mitel numbers can call SIP users in London or IOM and the other way round. I am planning to get at least 1 BRI pulled into the IOM office for PSTN access, btw. Any help you can offer would be gratefully received. Cheers Alex ___ --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] Asterisk Time Card
Something like this would be fairly easy to write, but it will cost you. :) Nitesh Divecha wrote: Thanks Bruce, If possible could you share your code...? I just need an idea how to integrate and store info in DB. Cheers, Nitesh Bruce Reeves wrote: This can be accomplished by writing an IVR to prompt and then using AGI or dialplan commands the query strings can be executed. I have a setup like this for a inegrating a in house time keeping system with asterisk. On 5/24/07, *Nitesh Divecha* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Thanks for your reply, The basic system would work as follows: - Method 1 === An employee would call in to the system and a welcome message is prompted. After that a employee is asked to enter the employee ID and PIN number and once verified Employee ID, Caller ID, and time of day is stored into MySQL DB. By end of the day employee will call in again to logout from the system and same information is stored into the DB. Method 2 === This time employee is verified with Caller ID, so the employee ID and PIN number is skipped and time of day is logged into the DB. Is it possible? Thanks, Nitesh ram wrote: On 5/24/07, *Nitesh Divecha* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hello All, I have been looking for this solution for quite sometimes Asterisk Time Card System. I found some discussion from Digium forum but not quite helpful. Hi what is the mean of time card system ? is this kind of attendent system ? kindly give some more details ram ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves Nortex Networks ___ --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 ___ --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 ___ --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] basic 3+ way conference call on plain old phones
hi guys, is it possible to do a basic 3-or-more-way conference call when the phones dont support it? I am fully aware of this concept on expensive phones like this one: Grandstream GXP 2000 -Conference call 3-way http://www.youtube.com/watch?v=hlZ6JqE1MT4 The problem is that the basic plain old commercial PBX supports 3-way calling in ugly old phones like this one: http://www.neo-shop.com/tiendas/0009/varios/telefono%20TEIDE-1.jpg connected to an ata like this one: http://www.egk.com.ar/imagenes/hardware/sipura2.jpg The idea is to be caller (A): dial calle (B), once (B) answers press on HOOK or something else to send them to MOH, then dial callee (C), talk to him a little too, then press the same HOOK or something else and the 3, (A)(B) and (C) in a conference call. Unlike the grandstream, this would definitelly have to be done by *, isnt this part of the basic functionality like voicemail that is already done and a couple lines in the config files it will work on all phones done by *? if not, then, how do you recommend me to it? the closest I have seen to shat I am looking for is http://www.voip-info.org/wiki/view/Asterisk+Dynamic+conferences+macro is there a better alternative? any thoughts? thanks a lot! Got a little couch potato? Check out fun summer activities for kids. http://search.yahoo.com/search?fr=oni_on_mailp=summer+activities+for+kidscs=bz ___ --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] Asterisk Time Card
Thanks alot Shanon... That helped me to kick start my work...
Cheers,
Nitesh
Shanon Swafford wrote:
I was messing with something similar one day for a trucking company to track
progress of their drivers.
It is HIGHLY beta, but should get you started:
## extensions.conf ###
exten = s,1,NoOp(FXO Line is Ringing : ${CALLERID(all)})
exten = s,n,NoOp(${CALLERID(all)})
exten = s,n,NoOp(${CALLERID(num)})
exten = s,n,NoOp(${CALLERID(name)})
exten = s,n,GotoIf($[${CALLERID(num)}=9728311600]?agitest|s|1)
exten = s,n,GotoIf($[${CALLERID(num)}=200]?agitest|s|1)
[agitest]
exten = s,1,AGI(test.php)
exten = s,n,Answer
exten = s,n,Background(shanon-welcome) ; Thanks for calling press
1 for sales, 2 for support, ...
exten = s,n,WaitExten
###test.php###
?php
set_time_limit(6);
require('/var/lib/asterisk/agi-bin/phpagi/phpagi.php');
$agi = new AGI();
$agi-answer();
$cidnum = $agi-request['agi_callerid'];
$cidname = $agi-request['agi_calleridname'];
$agi-text2wav(Hello $cidname);
$agi-text2wav('We are testing so please call our cell phones. ');
$test = 0;
while ( $test 1 ) {
$agi-text2wav(Enter your Order Number);
$load_num = $agi-get_data('beep', 3000, 6);
$tmp = strsplit($load_num);
$mydata = ;
foreach ($tmp as $value) {
$mydata .= $value . ;
}
$agi-text2wav(You entered $mydata. Enter 1 if this is correct);
$test = $agi-get_data('beep', 3000, 1);
$agi-conlog(Customer Entered: $test);
}
/* Add code here to insert $test into a database */
$agi-text2wav('Goodbye');
// $agi-hangup();
function strsplit($str, $l=1) {
do {$ret[]=substr($str,0,$l); $str=substr($str,$l); }
while($str != );
return $ret;
}
?
Regards,
Shanon
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nitesh Divecha
Sent: Thursday, May 24, 2007 9:08 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk Time Card
Thanks for your reply,
The basic system would work as follows: -
Method 1
===
An employee would call in to the system and a welcome message is
prompted. After that a employee is asked to enter the employee ID and
PIN number and once verified Employee ID, Caller ID, and time of day is
stored into MySQL DB. By end of the day employee will call in again to
logout from the system and same information is stored into the DB.
Method 2
===
This time employee is verified with Caller ID, so the employee ID and
PIN number is skipped and time of day is logged into the DB.
Is it possible?
Thanks,
Nitesh
ram wrote:
On 5/24/07, *Nitesh Divecha* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Hello All,
I have been looking for this solution for quite sometimes
Asterisk Time
Card System. I found some discussion from Digium forum but not quite
helpful.
Hi
what is the mean of time card system ?
is this kind of attendent system ?
kindly give some more details
ram
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Re: [asterisk-users] Integrated T1
Quoting Alex Balashov [EMAIL PROTECTED]: On Thu, 24 May 2007, William Moore wrote: On 5/24/07, Jeremy Mann [EMAIL PROTECTED] wrote: Can an asterisk box equipped with a Digium T1 card handle Integrated T1 circuits? I have a T1 with 768k data and the remaining channels voice, can the asterisk box do the Data routing + Voice processing? Yes, zaptel will create a device node for you. Take a look at the set-hdlc tool in zaptel and the less common channel types in the default zaptel config file (rawhdlc is one, there are also others). Interesting! Is this a relatively recent development? its one of the oldest things related to asterisk in general. I stand corrected. Is there a way to make PPP-encapsulated T1 work as well? -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : +1-678-954-0670 ___ --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 Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com This message was sent using IMP, the Internet Messaging Program. ___ --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] Integrated T1
On Thu, 24 May 2007, Jeremy Mann wrote: Can an asterisk box equipped with a Digium T1 card handle Integrated T1 circuits? I have a T1 with 768k data and the remaining channels voice, can the asterisk box do the Data routing + Voice processing? I'm not certain, but I believe the Sangoma WANrouter/WANPipe cards are capable of this. Call Sangoma and ask them if it is possible. Tom ___ --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] Bottom line on fax reception
shadowym wrote: So what is the bottom line? Does it work or not. I've heard stories it works, it doesn't work, it kinda sorta works when it's not raining out side. Everything under the rainbow. What's the bottom line with recent updates on 1.2.x? Is it production ready for fax? By production ready I mean that it just works all the time and doesn't need any babysitting. Do I have to worry about dropped lines, sometimes not detecting incoming fax toneyada yada. One simple question - VOIP or PSTN? ___ --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] Bottom line on fax reception
shadowym wrote: So what is the bottom line? Does it work or not. I've heard stories it As it has been said many many times before, Fax detection is an art and most of the time is not reliable. Faxing on the other hand, using iaxmodem along with HylaFAX+ works very well. Search the archives. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --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] Bottom line on fax reception
On Thu, 24 May 2007, shadowym wrote: So what is the bottom line? Does it work or not. I've heard stories it works, it doesn't work, it kinda sorta works when it's not raining out side. Everything under the rainbow. If you're talking about running it as analog pass-thru over G.711u over SIP, especially over the Internet-at-large, that's the answer; it mostly works, unless it doesn't. Analog fax is highly susceptible to even very minute distortions introduced by jitter, 1-2% packet loss, and/or out-of-sequence RTP payloads. That's not something software can really fix, except to the extent that T.38 may do so. If you're talking about picking faxes up over TDM/POTS, it should work fine. I've set it up a number of times and never had any problems. -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : +1-678-954-0670 ___ --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] vmoutcall
Paul Aviles wrote: Hello guys, I have been looking for a way to call a cell phone after someone has left a This can easily be done with database lookups and .call files to accomplishing this? Most analog pbx's have this feature and I am amazed Asterisk does not natively. It can be done natively; within the dial plan. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --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] Conference room as Music on Hold
Here is what I am trying to do. I have a SIP soft phone running on a PC that is streaming a local radio station. I assigned mono out in XP Equalizer as the mic so now I have the softphone streaming audio. I then create a conference room and dial that room from the softphone. Now anyone who joins the conference room hears the streaming audio. How can I configure Asterisk so that when musiconhold is invoked it automatically joins that conference room? The reason I want to do this is because the radio station uses windows media and I haven't been able to get it to work directly on Asterisk. Even after following the mplayer instructions on the Asterisk wiki. ___ --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] Re: Integrated T1
On Thu, 24 May 2007, Jeremy Mann wrote: Can an asterisk box equipped with a Digium T1 card handle Integrated T1 circuits? I have a T1 with 768k data and the remaining channels voice, can the asterisk box do the Data routing + Voice processing? The Zaptel/Asterisk infrastructure can definitely break particular timeslots out of the T1 for voice, but it is not my impression that any existing WAN drivers for Linux support Digium cards or cohabitation with Zapata and can give you a serial data interface on other channels. There are obvious risk factors with the scenario of your Asterisk box being your CSU/DSU/Firewall Router but for a small office this can actually be a good thing. Sangoma cards with their Wanpipe drivers can do this for you. dbc. -- David Cook ___ --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] transfer call sip to zap
how to transfer a call from sip channel to zap channel thanks -- // DiegoF // ___ --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] vmoutcall
Perhaps someone can share how? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle Sent: Thursday, May 24, 2007 3:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] vmoutcall Paul Aviles wrote: Hello guys, I have been looking for a way to call a cell phone after someone has left a This can easily be done with database lookups and .call files to accomplishing this? Most analog pbx's have this feature and I am amazed Asterisk does not natively. It can be done natively; within the dial plan. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --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 ___ --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] Login log out support
is there a way to support login and logout functionality in a phone? We are using Cisco 7940 and 7960 phones and have 2 shift. We want to be able to use the same phone using like 2 different extensions. The phone will then remember your settings if possible, if anyone has left you a voice mail etc. Is this possible? Regards, Paul ___ --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] Integrated T1
Both Digium and Sangoma support this configuration but something tells me that Sangoma will be easier to setup. That something is trying it several years ago when there were little bits and pieces on how to do it spread all over. I never did get it to work. Digium listed it as a feature but when called for support, they said it was not a supported feature. You may have better luck these days. Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tom Rymes Sent: Thursday, May 24, 2007 3:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Integrated T1 On Thu, 24 May 2007, Jeremy Mann wrote: Can an asterisk box equipped with a Digium T1 card handle Integrated T1 circuits? I have a T1 with 768k data and the remaining channels voice, can the asterisk box do the Data routing + Voice processing? I'm not certain, but I believe the Sangoma WANrouter/WANPipe cards are capable of this. Call Sangoma and ask them if it is possible. Tom ___ --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] Cisco CP-7970G
Hi all, I just bought the 7970G phone. It's a beautiful phone. In trying to make it work with Asterisk, I've read many posts on the net. However, all of them make reference to having to install the SIP firmware on the phone. Where can I get it? Thanks ___ --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] Conference room as Music on Hold
On Thu, 24 May 2007, shadowym wrote: How can I configure Asterisk so that when musiconhold is invoked it automatically joins that conference room? Can't say for sure, but the best intuition would be to try to look at how your phone's hold button implements its function on the SIP layer and see if you can build something into the dial plan and/or features.conf that would have the effect of placing the user into a MeetMe room. The reason I want to do this is because the radio station uses windows media and I haven't been able to get it to work directly on Asterisk. Even after following the mplayer instructions on the Asterisk wiki. Why doesn't it work? Live transcoding via mplayer should work fine. That said, it can be a rather CPU-intensive process and is rather pointless. By the time the audio is butchered by that process + some load, not to mention the underlying nature of a 64kbps 3.1 KHz speech bearer capability, it won't sound nearly as good as conventional radio-on-hold most likely. Another thing you could try is hook mpg123 into the picture, instead of using native MP3 decoding. Set up the WMA to be streamed as MP3 on some Shoutcast server or something, then feed mpg123 a file that actually contains a reference to an MP3 stream URI. See how that works for you. -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : +1-678-954-0670 ___ --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] Conference room as Music on Hold
See: http://www.voip-info.org/wiki/index.php?page=Asterisk+config+musiconhold.conf Look for: Using a sound card as the source. You can then plug in a radio or what ever. shadowym wrote: Here is what I am trying to do. I have a SIP soft phone running on a PC that is streaming a local radio station. I assigned mono out in XP Equalizer as the mic so now I have the softphone streaming audio. I then create a conference room and dial that room from the softphone. Now anyone who joins the conference room hears the streaming audio. How can I configure Asterisk so that when musiconhold is invoked it automatically joins that conference room? The reason I want to do this is because the radio station uses windows media and I haven't been able to get it to work directly on Asterisk. Even after following the mplayer instructions on the Asterisk wiki. ___ --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 ___ --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] Cisco CP-7970G
[EMAIL PROTECTED] wrote: Hi all, I just bought the 7970G phone. It's a beautiful phone. In trying to make it work with Asterisk, I've read many posts on the net. However, all of them make reference to having to install the SIP firmware on the phone. Where can I get it? Thanks ___ --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 Download it from cisco www.cisco.com ___ --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] Cisco CP-7970G
On Thu, May 24, 2007 6:27 pm, Anthony Francis [EMAIL PROTECTED] said: [EMAIL PROTECTED] wrote: Hi all, I just bought the 7970G phone. It's a beautiful phone. In trying to make it work with Asterisk, I've read many posts on the net. However, all of them make reference to having to install the SIP firmware on the phone. Where can I get it? Thanks ___ --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 Download it from cisco www.cisco.com Just tried that. It seems that you need a Cisco Service Agreement before you can download it. Is that correct? Is that crazy? ___ --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] Integrated T1
On 5/24/07, Alex Balashov [EMAIL PROTECTED] wrote: On Thu, 24 May 2007, William Moore wrote: On 5/24/07, Jeremy Mann [EMAIL PROTECTED] wrote: Can an asterisk box equipped with a Digium T1 card handle Integrated T1 circuits? I have a T1 with 768k data and the remaining channels voice, can the asterisk box do the Data routing + Voice processing? Yes, zaptel will create a device node for you. Take a look at the set-hdlc tool in zaptel and the less common channel types in the default zaptel config file (rawhdlc is one, there are also others). Interesting! Is this a relatively recent development? I stand corrected. Is there a way to make PPP-encapsulated T1 work as well? Yes. I've got a client with a TE212P with a PRI channelized on one port and a PPP encapsulated IP T1 on the other. It works fine. BJ -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --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] Call Center Application
Hi list; I am looking for an application that can be used with call center, in this application we can integrate the telephony part of the call center (like CTI Client ad so on), any one can advise for a good application to be used with Asterisk Call Center? - Note: The application to be customized easy, to be able to use it with Banking, Telecom, Oil, .. etc. Regards Bilal Get the Yahoo! toolbar and be alerted to new email wherever you're surfing. http://new.toolbar.yahoo.com/toolbar/features/mail/index.php ___ --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] Call Center Application
bilal ghayyad wrote: Hi list; I am looking for an application that can be used with call center, in this application we can integrate the telephony part of the call center (like CTI Client ad so on), any one can advise for a good application to be used with Asterisk Call Center? - Note: The application to be customized easy, to be able to use it with Banking, Telecom, Oil, .. etc. Regards Bilal Try PBXware call centre edition. Full call centre stats, real time monitoring, unlimited agents etc. http://www.bicomsystems.com/products/C/P/319/154_2573/ Regards, Senad www.bicomsystems.com ___ --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] vmoutcall]
-- Perhaps someone can share how?
First you need to give them the option of turning the feature on and
off. I do it with the following:
[callback-activate]
; ***
; Callback activate/deactivate. If this function
; is enabled and there is a call file in the form
; of ${EXTEN}.call, then Asterisk will call the
; phone number contained within the .call file 150
; seconds after a voicemail has been left.
; ***
exten = 80*,1,Set(CALLBACK=${DB(vmcallback/${CALLERIDNUM})})
exten = 80*,2,GotoIf($[${CALLBACK} = YES]?80*,3:80*,101)
exten = 80*,3,Set(DB(vmcallback/${CALLERIDNUM})=NO)
exten = 80*,4,Playback(local/stutter)
exten = 80*,5,Playback(de-activated)
exten = 80*,6,Hangup()
exten = 80*,101,Set(DB(vmcallback/${CALLERIDNUM})=YES)
exten = 80*,102,Playback(local/stutter)
exten = 80*,103,Playback(activated)
exten = 80*,104,Hangup()
Then you need to do a database look up every place in your dial plan
where voice mail may be left, I do it as such:
[macro-sip.extensions]
exten = s,1,Set(CALLBACK=${DB(vmcallback/${ARG1})})
exten = s,n,SetMusicOnHold(cd)
exten = s,n,Dial(SIP/${ARG1},28,tWw)
exten = s,n,NoOP(Dial Status: ${DIALSTATUS})
exten = s,n,NoOP(Hangup Cause: ${HANGUPCAUSE})
exten = s,n,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,GotoIf($[${CALLBACK} =
YES]?s-NOANSWER,2:s-NOANSWER,3)
exten = s-NOANSWER,2,System(/usr/local/bin/vm-callout.sh ${ARG1})
exten = s-NOANSWER,3,Voicemail([EMAIL PROTECTED])
If CALLBACK=YES, then run the script that copies the call file into the
outgoing directory. It uses touch to set the date on the file 150
seconds into the future. This prevents the system calling the user
while voice mail is still being left.
The call file links into the dial plan that loops the message 4 times
waiting for acknowledgment by pressing 1 to collect voice mail.
[voice-mail-callback]
;
; Set timeouts
;
exten = s,1,Set(TIMEOUT(response)=6)
exten = s,2,Set(TIMEOUT(digit)=3)
exten = s,3,Wait(1)
exten = s,4,Set(COUNT=0)
; ***
; Play, your attention is required, press 1 to
; collect voice mail
; ***
exten = s,5,Background(attention-required)
exten = s,6,Background(press-1)
exten = s,7,Background(to-collect-voicemail)
; *
; If 1 is pressed, then play transfer and
; then jump to voice-mail context.
; *
exten = 1,1,Playback(pbx-transfer)
exten = 1,2,Goto(voice-mail,s,1)
;
; Setup a variable to count the number of
; times the message has been played, when
; $COUNT reaches 3, play you've taken
; to long to dial and hangup.
;
exten = t,1,Set(COUNT=$[${COUNT} + 1])
exten = t,2,NoOP(${COUNT})
exten = t,3,GotoIf($[ ${COUNT} 3 ]?103)
exten = t,4,Goto(voice-mail-callback,s,5)
exten = t,103,Playback(local/tolong-todial)
exten = t,104,Playback(goodbye)
exten = t,105,Hangup()
exten = i,1,Playback(local/sorry-invalid-choice)
exten = i,2,Set(COUNT=$[${COUNT} + 1])
exten = i,3,NoOP(${COUNT})
exten = i,4,Goto(voice-mail-callback,s,5)
exten = h,1,NoOP(Hungup)
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary Safety,
deserve neither Liberty nor Safety.
___
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Re: [asterisk-users] Nokia release
Nokia N95 available via ATT/Cingular for $795 with a 2 year contract. It was advertised in the New Jersey Star Ledger this morning. Mark On Thu, 2007-05-24 at 18:42 +0500, Rizwan Hisham wrote: Hi all, sorry to ask you something not related to asterisk, but i really want to know whether the Nokia N95 cell phone is released in the USA or not? if somebody from USA knows, plz reply. -- Rizwan Hisham Software Engineer AXVOICE Inc. ___ --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 ___ --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
