[asterisk-users] Telecom Best Practices
OK. I'm getting out the fireproof suit because it's coming and my hackles have been raised by a number of comments on the list of late. Disclaimer: No disrespect intended to the individuals of any *specific* thread. I'm a little frustrated over energy wasted on pedantic top/bottom posting crap rather than understanding the technology and industry best-practices which have been built upon for years. I'm not against change - far from it. I'm against throwing out good work and history done by an entire industry to make telecom one of the most complex and yet stable computing environments (class 4/5 entrants from Nortel, Lucent, etc.) We should learn from and extend best practices where they do not address circumstances which weren't available 20 years ago (or more) but not to ignore proven practices simply because the transport mechanism is now a packet instead of a circuit. I'm not alone. Here's the deal with Asterisk as an Answering Machine - industry best practices. - don't put phones in parallel with the pbx except for the single, emergency phone next to the PBX. - PBX's are directors of calls. For it to direct, it must have control. For it to have control, you can't answer some calls in parallel. - even if it is a home/1-phone-office, the PBX accepts and directs the call to phones *behind* it. The phone rings, if you don't answer it goes to voicemail. If you don't follow this practice you will have: - timing issues with the answering of analogue phones - rings are not always consistent. - people will pick up just in time and will have to compete with voicemail. - you won't get accurate CDR's which means you can do proper billing reconcilliation, chargebacks or help you understand your call paths and volumes to help troubleshoot down the road. (You may not care about bill reconcilliation or chargebacks but remember - this is a PBX (aka business phone system) and that's what business does so that's the business model that is supported by most practices. Just to prove I'm not too old for change and acceptance of new technology... - If you get charged by the connect from your provider, route by DID but don't answer it in an IVR. That way you don't get billed. - Once you are looking to route to a phone behind the PBX, hey - check your jabber status. Is your desktop in IDLE, you're not there - send it to your cell phone. Oh, BTW - change the CLID on the way back out to append H- to the caller so you know it came redirected from the house. This helps you decide context of the caller and decide if you want to answer or *how* you will answer. There is no reason to not have all phones behind the PBX. There is nothing mandating you to dial a 9, or similar to get an outside line. Be creative. Use internal extensions that don't conflict with your local calling area exchanges. Then you write dialplans for the phones that will dial right away and not make you wait to timeout on the 10-digit+ dial. There are *way* too many cool things we can do with Asterisk that worry about top/bottom posting. Let's get back to reading docs - asterisk industry practices. Fireproof suit on and buttoned up. I'm ready. -dbc -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Fwd: Re: Asterisk as a Condo door opener/intercom
Asterisk as a phone system makes perfect sense in a condo. You can get
all the DID's you want and eliminate costs for the owners. You can offer
standard FXO for people who don't care and IP sets for people who want
to upgrade to feature sets.
Your door openner is a piece of cake.
1. Create an option in your dialplan only in the from-access-door
context that reads DTMF from the called station only.
2. Use this to access an external program to turn on a serial port line
for 10 seconds.
3. This line drives a solid state relay (~$30) so you won't blow the
sink current on the PC port that drives a standard door lock.
A commercial door strike is about $100. The program to run the port is
childs play. Here is a test prog I used for turning on a power hungry
last printer. Change the comments and the sleep time and you're done.
/*
* lpon Lineprinter ON
* *** test program only **
*
* (c) David Cook, 1994
*
* Set signlal lines on serial port to turn on 5vdc
* signal. Used for solid-state relay (low current
* draw on RS232C port) to switch high voltage/high
* current load for printer.
*
* Part of an intelligent print spooler to only power
* on/off high draw printer when required.
*
* Usage: lpondevice bits to set
* For example, lpon /dev/cua4 4 to set bit 3 on
* port /dev/cua4.
* 4 = 0100 or bit 3 which is DTR
* 2 = 0010 or bit 2 which is RTS
* 6 = 0110 or both DRT RTS
*/
#includesys/types.h
#includesys/ioctl.h
#includetermios.h
#includefcntl.h
#includeerrno.h
#includestdlib.h
#includeunistd.h
#includestdio.h
#includesignal.h
#include lpswitch.h
/* Main program. */
int main(int argc, char **argv)
{
struct termios port_config;
int fd;
int set_bits = 2;
/* Open monitor device. */
if ((fd = open(SWDEV, O_RDWR | O_NDELAY)) 0) {
fprintf(stderr, lpswtich: %s: %d\n, SWDEV, strerror(errno));
exit(1);}
cfmakeraw(port_config );
port_config.c_iflag=port_config.c_iflag|IXON;
port_config.c_oflag=port_config.c_oflag|CLOCAL|~CRTSCTS;
tcsetattr( fd, TCSANOW,port_config );
ioctl(fd, TIOCMSET,set_bits );
/* wait for printer to warm up */
sleep(45);
/* not say ready and release the printer */
set_bits = 6;
cfmakeraw(port_config );
port_config.c_iflag=port_config.c_iflag|IXON;
port_config.c_oflag=port_config.c_oflag|CLOCAL|~CRTSCTS;
tcsetattr( fd, TCSANOW,port_config );
ioctl(fd, TIOCMSET,set_bits );
close(fd);
}
On 12/04/2011 8:16 AM, asterisk-users-requ...@lists.digium.com wrote:
Message: 3
Date: Mon, 11 Apr 2011 18:21:39 -0500
From: Don Kellyd...@donkelly.biz
Subject: Re: [asterisk-users] Asterisk as a Condo door opener/intercom
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Message-ID:8E20A6A94C9548C8A0E27B502B18F200@DonPC
Content-Type: text/plain; charset=us-ascii
Continuing top posting...
The same argument could be made for any commercial solution. Why use
Asterisk when we could throw $4,000 at our problem for a commercial
solution?
I'd like to have a solution that would have the features you suggest for
$400.
--Don
On Behalf Of C F
Sent: Monday, April 11, 2011 11:43 AM
Search the lists. Some hints:
Viking electronics makes a door box that connects to any analog line
(IIRC e-20).
They also make a DTMF keypad that integrates in series with any analog
line. They might also make a door box with a DTMF keypad on it.
Sandman makes a relay that will get energized when there is a ring on
the line which could be used to unlock the door.
However, why would you use asterisk? Using asterisk for the sole
purpose of MDU entry system is like using windows for asterisk, it
works but why?
Go for the commercial solutions, it comes with a geziilion options for
your setup one of them the ability of chosing an apartment, another
add key fobs, another one is the ability of using a code for the
residence (not guests) to unlock the door. Also the interface with
asterisk you will have to build one from scratch. The commercial
solutions have em built in.
On 4/10/11, Bruce Bbruceb...@gmail.com wrote:
Hi Everyone,
Looking to replace a condo intercom system. Apparently the current one
taps
into the lines and dials phone numbers but needs to be changed as it's
faulty.
I will probably still use the same analogue dialing and back it up with a
VoIP line and use the current cabling that is in place. But as for as the
door opening function goes, I am not sure how to interface and how open
these modules are usually built.
I would appreciate it if someone with experience can throw in some
pointers
as to what I might be
Re: [asterisk-users] asterisk-users Digest, Vol 73, Issue 63
I have 2 FXO channels from which I want to route incoming calls to different contexts in extensions.conf. I edited the context entries in dahdi-channels.conf and created matching entries in extensions.conf. One channel is routed to the new context as I want, but the other channel is stuck going to the default from-pstn context no matter what I do. Can anyone see what I've missed? From dahdi-channels.conf: ;;; line=3 WCTDM/4/2 FXSKS signalling=fxs_ks callerid=asreceived group=0 context=from-pstn-3 channel = 3 callerid= group= context=default ;;; line=4 WCTDM/4/3 FXSKS signalling=fxs_ks callerid=asreceived group=0 context=from-pstn-4 channel = 4 callerid= group= context=default You have multiple context= lines in your file and the order within the file is important. channel = should be the last item. So channel 4 is actually reading the context=default line which is 3 lines under channel=3 in your config file. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] About BRI / ISDN hardware. What to buy?
Hi, http://store.%50honiceq.com has the quadbri card for $400. We can also offer free shipping for this card. The card has 4 BRI ports and is based on the same main chipset (HFC-4S) as Digium/Beronet/Junghanns cards. It does not have EC onboard. best regards On 1/18/07, Cosmin Prund [EMAIL PROTECTED] wrote: I finally found a price tag for the darn thing, at around 500 euros I can handle it. Qustion: Do they behave properly if I've got an other Digium TDM400 card in the system? How about installing two cards in the same server? At the moment I've only got 1 ISDN line plus a few analog lines going into the TDM but in the very near future we might want to get a second ISDN. Alberto Pastore wrote: Jens Vagelpohl ha scritto: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On 18 Jan 2007, at 18:31, Patrick wrote: I think http://www.melware.de carries the Eicon Server ISDN cards which have hardware echo cancellation. They are also the author of the chan_capi driver for Asterisk. I use the Eicon Server BRI cards with Asterisk myself and they work very well. I concur, I have a Eicon DIVA single port BRI card and it works very well. Cosmin, if you want to use it for Fax traffic as well make sure you do *not* get a V-BRI card. Those will not do Fax. jens Tried almost all cards (Junghanns, Sangoma, Beronet, some hfc-based oem cards, Eicon Diva Server). Eicon is expensive but is *REALLY* worth it. The other cards are just a waste of money (even if little money). If you want a reliable PBX (who doesn't want it?), Diva Server cards are the definitive choice. The best card ever. Zero echo problems, superb hardware echo cancellation. Top reliability. Excellent FAX support with Hylafax (only cards with builtin DSPs, that is, NOT the V-series, as pointed out by Jens). Easy driver installation and powerful utilities/configuration tools. I tested BRI-2M, 4BRI-8M, PRI-30M on several installations, even older 1.0 version cards (PCI 5v only) just work great. I use diva server drivers software source rpm from Eicon, chan_capi from www.melware.org (0.7.1) on asterisk 1.2.14 (kernel 2.6.17.3). We've deployed more than 40 PBX (from 1 bri to 8 bri) without a flaw. I'm only a little bit annoyed about not being able to take advantage of the onboard DSPs to perform audio transcoding, because of the lack of a suitable asterisk driver (the cards themselves support hardware gsm/g726 codecs, for instance). Alberto. ___ --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 -- regards Martin * Visit http://pbxhardware.com for alternative T1/E1 Voice/Data cards * ___ --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] Random SIP Phone Problem
I got the same problem with 04/19/05 CVS version. I am using Grandstream phones. I also noticed that when this happens, an already hung-up call was still shown as bridged between a SIP phone and a Zap channel. On 4/18/05, Shaun Tierney [EMAIL PROTECTED] wrote: I am currently running CVS-HEAD-04/15/05-13:15:00 and I have an issue that just recently cropped up. I upgraded to this version of Asterisk last Friday and now twice in the last two hours, all of my Aastra SIP phones lose service suddenly. Network connectivity is still there between the phones and the PBX, and I have restart Asterisk to fix the issue. Would it be worth my time to move to the latest CVS Asterisk release even though it has only been three days since I installed the version in operation? Or would I be better off going with a previous CVS release to fix the problem? I can't use the stable release because I use macro arguments in the dial command. Here are the error messages that seem to show up for the duration of the problem. Apr 18 13:43:50 NOTICE[16997] chan_sip.c: Peer 'brettb' is now UNREACHABLE! Last qualify: 1045 Apr 18 13:44:03 VERBOSE[16997] logger.c: Don't know what to do if second ROSE component is of type 0x6 Apr 18 13:44:07 NOTICE[16997] app_queue.c: Added interface 'SIP/brettb' to queue 'psc' Apr 18 13:44:14 NOTICE[16997] chan_sip.c: Peer 'brettb' is now REACHABLE! (76ms / 2000ms) Regards, Shaun ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New native assisted transfer (atxfer) usage inforequired
Try to change the macro from
exten = s,1,Dial(${ARG2},20)
to
exten = s,1,Dial(${ARG2},20,${ARG3})
--JJL44
On Wed, 26 Jan 2005 20:35:16 -0500, Steven Frazier [EMAIL PROTECTED] wrote:
I got it to work using the standard example of using exten to ring a phone
vs. the newer macro example.
I am still learning how macros work, maybe I can incorporate the atxfer into
the macro at some point.
I was wondering if you could incorporate this is a .call file, I haven't
seen an example of how to do that, but that would be nice if you had a .call
file that was dialing a cell phone and you had the ability to transfer the
person to another extension from your cell phone. Just a thought.
This worked:
exten = 5811,1,Dial(SIP/5811,10,Ttr)
exten = 5810,1,Dial(Zap/1,10,Ttr)
I was trying to make it work with the example and using the macro stdexten
This did not work:
exten = 5810,1,Macro(stdexten,5810,ZAP/1,Ttr)
exten = 5811,1,Macro(stdexten,5811,SIP/5811,Ttr)
[macro-stdexten]
;
; Standard extension macro:
; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here as well
; ${ARG2} - Device(s) to ring
;
exten = s,1,Dial(${ARG2},20)
exten = s,2,Goto(s-${DIALSTATUS},1)
;
exten = s-NOANSWER,1,Voicemail(u${ARG1})
exten = s-NOANSWER,2,Goto(default,s,1)
;
exten = s-BUSY,1,Voicemail(b${ARG1})
exten = s-BUSY,2,Goto(default,s,1)
;
exten = _s-.,1,Goto(s-NOANSWER,1)
;
exten = a,1,VoicemailMain(${ARG1})
;
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New native assisted transfer (atxfer) usage inforequired
On Tue, 25 Jan 2005 19:33:00 -0500, Steven Frazier [EMAIL PROTECTED] wrote: Could I ask you a question? You don't have to flash to use the transfer feature correct or do you? I have tried it both ways and nothing happens. No, I do not have to flash to use the transfer feature. It works pretty much like the old # transfer. The sequence for blind transfer is like this: 1. I press ## keys quickly (two # keys within 0.5 seconds) 2. I hear the sound Transfer, the other party hears hold music 3. I wait for dial tone. After hearing the dial tone, I dial the transferee. I need to dial quickly because the timeout is set to 3 seconds. After the last digit is dialed, I wait for the asterisk to hang up and give me budy tone. At this time I hang up. 4. The other party hears the ring tone for the transferee. If the transferee picks up, they can talk. The sequence for attended transfer is like this: 1. I press ** keys quickly (** is my atxfer key) 2. I hear the sound Transfer, the other party hears hold music 3. I wait for dial tone. After hearing the dial tone, I dial the transferee. I need to dial quickly because the timeout is set to 3 seconds. After the last digit is dialed, I wait for 3 seconds and then I hear the ring tone for the transferee. 4. If the transferee picks up, I talk with the transferee. 5. If I hang up, the transferee is connected with the caller. If the transferee hangs up, I am back with the original caller. I verified that I had beep.gsm and beeperr.gsm in my sounds directory and here is my features.conf file: If I dial *1 (from a Zap or SIP) phone the person I am talking to just hears the tones I press *2 or #1. When I press ## (my blindxfer key) the other party does not hear any DTMF tones. If I press # (single #) the other party hears the # tone after a short delay (I guess asterisk is delaying the tone to make sure there is no more # coming). Does the automon work too? Where does it put the file that it records at? I have not tried this feature yet. I made sure I have Tt in my dial command in my extensions.conf file as well: From my extensions.conf file: exten = 5801,1,Macro(stdexten,5801,SIP/5801,Ttr) exten = 5802,1,Macro(stdexten,5802,SIP/5802,Ttr) exten = 5803,1,Macro(stdexten,5803,SIP/5803,Ttr) exten = 5804,1,Macro(stdexten,5804,SIP/5804,Ttr) exten = 5805,1,Macro(stdexten,5804,SIP/5805,Ttr) exten = 5810,1,Macro(stdexten,5810,ZAP/1,Ttr) I think you can use only one transfer flag, either T or t but not both Tt. Best regards, --JJL44 [general] parkext = 700 ; What ext. to dial to park parkpos = 701-720 ; What extensions to park calls on context = parkedcalls ; Which context parked calls are in parkingtime = 45 ; Number of seconds a call can be parked for ; (default is 45 seconds) transferdigittimeout = 3 ; Number of seconds to wait between digits when transfering a call courtesytone = beep ; Sound file to play to the parked caller ; when someone dials a parked call xfersound = beep; to indicate an attended transfer is complete xferfailsound = beeperr ; to indicate a failed transfer adsipark = yes ; if you want ADSI parking announcements pickupexten = *8; Configure the pickup extension. Default is *8 featuredigittimeout = 500 ; Max time (ms) between digits for ; feature activation. Default is 500 [featuremap] blindxfer = #1 ; Blind transfer disconnect = *0; Disconnect automon = *1 ; One Touch Record atxfer = *2; Attended transfer [EMAIL PROTECTED] asterisk]# Thanks for your help. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New native assisted transfer (atxfer) usage info required
I had it working. My features.conf file is the same as yours except for the [featuremap]. I use ## for blindxfer and ** for atxfer. In my dial plan I use t or T as the Dial() flag. Make sure that you have beep.gsm and beeperr.gsm in the asterisk sound file folder. If these files are missing, transfer will fail. Is there any other special Dial() flag required in the dialplan? If I hang-up the call is not transferred, nor if I presso *0 (disconnect). When I hang up the call is transferred. When the transferee does not want to answer, he/she hangs up and I am back with the original caller. I would also like to know how to go back to the caller, if the other extension is busy or doesn't answer or doesn't want to talk with the caller. My dialplan sends busy calls to voicemail, so that I can simply press # to exit from the transferee's voicemail and get back to the caller. --JJL44 On Tue, 25 Jan 2005 20:37:49 +0100, Marco Menardi [EMAIL PROTECTED] wrote: Hi, I would like to use the new atxfer (native assisted transfer, see mantis item #3241) , but I've partially been able to make it work. I can receive a call and then having the caller hear MOH while talking with another extension (the one I want to transfer to), but then I can't make the caller and the trasferred talk hanging up or pressing any key combination I'm aware of. My features.conf are these ones: [general] language=it parkext = 521 ; What ext. to dial to park parkpos = 522-525 ; What extensions to park calls on context = parkedcalls ; Which context parked calls are in parkingtime = 120 ; Number of seconds a call can be parked for ; (default is 45 seconds) ;transferdigittimeout = 3 ; Number of seconds to wait between digits when transfering a call courtesytone = beep ; Sound file to play to the parked caller ; when someone dials a parked call xfersound = beep ; to indicate an attended transfer is complete xferfailsound = beeperr; to indicate a failed transfer ;adsipark = yes ; if you want ADSI parking announcements pickupexten = *8; Configure the pickup extension. Default is *8 ;featuredigittimeout = 500 ; Max time (ms) between digits for ; feature activation. Default is 500 [featuremap] ; there are different from the default in features.conf.SAMPLE, ; and faster to type when apropriate, OMHO blindxfer = #7; Blind transfer disconnect = *0 ; Disconnect automon = *1 ; One Touch Record atxfer = *7 ; Attended transfer And in the Dial() command I use the tT flag. Is there any other special Dial() flag required in the dialplan? If I hang-up the call is not transferred, nor if I presso *0 (disconnect). I would also like to know how to go back to the caller, if the other extension is busy or doesn't answer or doesn't want to talk with the caller. Could someone provide me the exact settings required, and the keystrokes needed to make it work (successful transfer and aborted transfer, going back to the caller)? A sort of atxfer for dummies :) I'm using the more recent CVS Head. Btw, of course I know that I can have a assisted transfer enabled SIP phone, or use the 3 way calling of my TDM400, but I want to make this feature of asterisk working without any client implementation (that is the goal of atxfer). Thanks a lot Marco Menardi -- --JJL44 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Segmentation Fault after Digitnetwork X100P install
On Thu, 20 Jan 2005 22:41:08 -0600, Steven Critchfield [EMAIL PROTECTED] wrote: Shouldn't you contact your vendor for support and not a different vendors support channel? As far I know, although Digium hosts the asterisk-users list and supports the Asterisk development, Asterisk is still a GPL open source project and asterisk-list is not a Digium support channel. Asking questions about a vendor other than Digium is no difference than asking questions about Digium hardware. --JJL44 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] # Transfers.
Attended transfer, also called supervised transfer, works like this: While on conversation with another party, you dial ** the transfer key sequence. Asterisk says Transfer then gives you a dial tone, while put the other party on hold music. You dial the transferee number and talk with the transferee to introduce the call, then you can hang up and the other party will be connected with the transferee. In case the transferee does not want to answer the call, he/she simply hang up and you will be back to your original conversation. On Thu, 20 Jan 2005 13:59:36 +0100, Robert Spielmann [EMAIL PROTECTED] wrote: What is an attended transfer? :) -- Robert Spielmann --JJL44 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] # Transfers.
I justed edited the Wiki Asterisk config file features.conf for this attended transfer features. Please check Wiki again for details. Best regards, --JJL44 On Thu, 20 Jan 2005 09:04:40 -0800 (PST), Bruce Komito [EMAIL PROTECTED] wrote: Sorry if I missed the beginning of this thread, but I've never heard of the ** transfer key sequence, nor have I found a way to make it work. Would you mind, please explaining this further or pointing me to somewhere where it's documented? (I checked Wiki and Google but no joy.) Thanks Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] # Transfers.
I have no idea if atxfer works with app_queue/chan_agent. Can anyone try it? Best regards, --JJL44 On Thu, 20 Jan 2005 17:38:25 -, Ben Merrills [EMAIL PROTECTED] wrote: Does this work with app_queue/chan_agent? Cheers, Ben -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Asterisk List Sent: 20 January 2005 17:28 To: Bruce Komito Cc: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] # Transfers. I justed edited the Wiki Asterisk config file features.conf for this attended transfer features. Please check Wiki again for details. Best regards, --JJL44 -- --JJL44 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems with loading TE110 module
I got a reply from Digium support. They knew about this problem and are trying to find a fix in hardware. Meanwhile, the only solution right now is to find out what the subsystem ID has changed to and add those new ID's to the wcte11xp.c source code so that they will be recognized by the driver. On Fri, 14 Jan 2005 15:08:25 -0800, Asterisk List [EMAIL PROTECTED] wrote: I encountered the same problem today. 'lspci -nvv' showed that the subsystem ID of the TE110p changed from 79fe to 79fa or 797e. Powering off/on the machine restored the subsystem ID to 79fe and the wcte11xp module could then load. I already emailed digium support for this. On Mon, 20 Dec 2004 11:32:15 +0100, Tamas J [EMAIL PROTECTED] wrote: Hello! I discovered, that I'm unable to load ther kernel module under 2.4.28. Before that I had 2.4.26 and tryed to upgrade the kernel to 2.4.28. After restart (reboot - soft restart) I can't load the module. When I go back to 2.4.26 and try to load the module, I'm getting the same problem. Only turning off/on helps. What can I do? It's very annoying because the box is in hosting and not easy to just restart. The wct1xxp worked fine in the same box (with restarts also). Any idea, hint? Kind regards, Tamas modprobe wcte11xp /lib/modules/2.4.28-magic/misc/wcte11xp.o: init_module: No such device Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg /lib/modules/2.4.28-magic/misc/wcte11xp.o: insmod /lib/modules/2.4.28-magic/misc/wcte11xp.o failed /lib/modules/2.4.28-magic/misc/wcte11xp.o: insmod wcte11xp failed ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] # Transfers.
The current CVS HEAD version already has ## transfer built-in. See
the included configs/features.conf.sample file. You can define your
own transfer key sequence. There is also an attended transfer
feature.
features.conf file:
[featuremap]
blindxfer = ##
atxfer = **
This worked very well for me.
On Wed, 19 Jan 2005 00:32:15 -0500, Ronald Hartmann
[EMAIL PROTECTED] wrote:
So I have read and read and read... google is my friend and the wiki is
by brother...
However, I am still unclear on what the preferred method of using the
pound sign is.
If the Pound sign is set aside for Transfer.. Then when I make an
outbound call to my bank I can not Enter my PIN followed by Pound
Likewise if I turn off the ability to transfer when initiating a call,
my bank pin works great, however I loose that ability to call park the
person I called
So I can pass the call to someone else in the office.
Conf file for park
[parkedcalls]
exten = 70,1,Answer
exten = 70,2,SetMusicOnHold(default)
exten =
70,3,ParkAndAnnounce(PARKED,10,SIP/${DIALED-EXTEN}|ext-local,${DIALED-EX
TEN},1)
exten = 70,4,Hangup
exten = _7X[1-9],1,ParkedCall(${EXTEN})
So I could adopt the doublehash patch. but it does not seam to be
something to make the CVS. therefore I have to patch patch patch
repeatedly.
What is everyone else using.
If pound pound is not something Mark the asterisk God does not wish to
add to CVS, would something like the following work
# would work as normal Conduct a pound transfer
However *#, or *## would send a Pound in DTMF to the called party.
This way it will keep the Pound Transfer in tact.
Anyways I ramble, I am anxious to see how others much brighter than I
have solved this issue.
ron
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] My dialplan just stopped working one day
[inbound] ; This is the list if inbound lines exten = 2181,1,Answer exten = 2181,2,Playback(silence/1) exten = 2181,3,Goto(default,main,1) exten = 2181,3,Hangup Notice there are two 2181,3 entries. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: how to manage Digium TDM04B outgoing calls
On Wed, 19 Jan 2005 15:55:10 -0700, Jason Kawakami [EMAIL PROTECTED] wrote: -Original Message- My question concern outgoing calls. How can I configure my extensions.conf to get a PSTN line on my TDM04B card in the following order : first trying on the channel 4 then if 4 is busy then switch to 3 if 3 is busy then switch to 2 and if 2 is busy then say there's no more line available. I don't want to dial on the first channel as it's my main number and all calls are received by that channel if the line isn't busy. I don't have to manage it for incoming calls (it can't be done anyway) as it's done not on my side but on the side of my provider (with the callforward on busy thing) but for outgoing calls the asterisk server must manage it. -put all of the zap channels in a group and specify that you want the outgoing calls to choose from the back of the group. Look on the wiki for the g and G options to the Dial()application. The above is good suggestion. If you do not wish any calls to be placed from channel 1, you can put channel 1 in a group and 2-4 in another group. The two groups can have the same context so that incoming calls are treated the same way. In your dial plan, you can Dial() outgoing calls using the second group only. --JJL44 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems with loading TE110 module
I encountered the same problem today. 'lspci -nvv' showed that the subsystem ID of the TE110p changed from 79fe to 79fa or 797e. Powering off/on the machine restored the subsystem ID to 79fe and the wcte11xp module could then load. I already emailed digium support for this. On Mon, 20 Dec 2004 11:32:15 +0100, Tamas J [EMAIL PROTECTED] wrote: Hello! I discovered, that I'm unable to load ther kernel module under 2.4.28. Before that I had 2.4.26 and tryed to upgrade the kernel to 2.4.28. After restart (reboot - soft restart) I can't load the module. When I go back to 2.4.26 and try to load the module, I'm getting the same problem. Only turning off/on helps. What can I do? It's very annoying because the box is in hosting and not easy to just restart. The wct1xxp worked fine in the same box (with restarts also). Any idea, hint? Kind regards, Tamas modprobe wcte11xp /lib/modules/2.4.28-magic/misc/wcte11xp.o: init_module: No such device Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg /lib/modules/2.4.28-magic/misc/wcte11xp.o: insmod /lib/modules/2.4.28-magic/misc/wcte11xp.o failed /lib/modules/2.4.28-magic/misc/wcte11xp.o: insmod wcte11xp failed ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Is Asterisk-users down?
David, I found your post on the Digium archives because I too have noticed that the flow of traffic on the list has stopped for the past 24 hours or so. I have replied to many existing threads and started new ones, only to not see my new messages. I take it from your recent post that you too have experienced this. Are you still not getting anything? Nothing. Not a damm thing :( So there must be a block somewere on the outbound side of things if My email made it to the archives. Hmm wonder what is going on? Thanks for confirming that it is not only me that is having a challange. David Thanks. -- Kristian Kielhofner ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM channel shows Offhook when I plug it to the telco
I had a similar problem but not exactly same: when telco lines are plugged into the FXO ports, initially zap show channel 1 says it is Onhook but I cannot make outgoing calls. Once I unplug the telco line and re-plug it, or after there is an incoming call, zap show channel 1 says Offhook but both incoming and outgoing calls work fine. I believe this is related to a incomplete fix to bug #2359 but my limited knowledge does not enable me to track down or fix the problem. - Original Message - From: Felix Pizarro [EMAIL PROTECTED] Date: Fri, 24 Sep 2004 08:53:20 -0700 (PDT) Subject: [Asterisk-Users] TDM channel shows Offhook when I plug it to the telco To: [EMAIL PROTECTED] Hello, everyone I am having problems with a TDM400 that has 3 fxs modules and 1 fxo. When plug a line from the telco to the fxo module it changes state from onhook to offhook, and of course I can not receive any calls. (When I tried to call from the outside to that line it shows as busy). Could someone help me? I am at the customers site ;( Thanks a lot ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Know if a call is answered
Hello: I have an asterisk server answering SIP calls. Whenever a call comes, asterisk answers, plays a gsm file (information) and dials to another SIP phone. Using asterisk Master.csv file I only have one record, and don't know if the second call is answered. I only know this if: - The called phone is busy - The called phone doesn't answer in X seconds (the parameter in Dial) But I can't see a difference between an answerd call and when the caller hangs up whith the other phone is still ringing. The call is always ANSWERED and the last command is Dial with the SIP address as the parameter. How can I do this? Could it be done with an AGI script? Sorry if this is a basic question, I have been searching for a solution for weeks and don't know what to do. Thanks in advance, Rober T. _ Reserva y planifica tu viaje online. http://www.msn.es/Viajes/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Problems with outgoing calls
Thanks Scott, but I already have a script that creates a call-load. That's how I have found the OutgoingSpoolFailed error and the other error with calls being retried without waiting to their retry time. Has anybody found this problem? Robert T. From: Scott Stingel [EMAIL PROTECTED] Date: Fri, 26 Dec 2003 13:44:18 - I have a call generation script (very simple) to generate call load for testing, if that's what you're trying to accomplish. It's good for generating huge call volumes for IVR testing. Let me know if you need it! _ Reserva y planifica tu viaje online. http://www.msn.es/Viajes/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problems with outgoing calls
Hello: I have found the following problems with outgoing calls with asterisk, compiled with an updated CVS on 22 Oct. 1.- Problem with retries: Whenever I set the MaxRetries parameter, to something greater than 0 in a call-fille, Asterisk ignores the RetryTime parameter and retries every file in the outgoing folder when a new call-file is copied into that folder. So, if I make a call placing a call-file in the outgoing folder, nobody answers and my call-file stays in the outgoing folder waiting for 'RetryTime' seconds (say 120 seconds), if another call-file is placed in the outgoing folder the previous call is retried inmediately before the RetryTime finishes. 2.- Problem with high-volume calls: When I put lots of call-files in then outgoing folder, from time to time asterisk show an error in Master.csv. The error is OutgoingSpoolFailed, and shows no info about the call, just failed as the extension. I move one call-file every second, but for several minutes. How can I avoid this two problems? Thanks in advance, Robert T. _ Entra de visita en las decenas de tiendas del nuevo MSN Compras. Compara los precios antes de comprar. http://www.msn.es/compras/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Destination number
Hello: I need to prepare some detailed stats from asterisk, and I'm asked to show data I don't know how to obtain it: It's the 'final' number (don't know what's its name) In the stats I have to show the caller_id (I have it), the called_id (I have it) and the final number that actually accepted the call. In extensions.conf file, I try to pass the call to several numbers in sequence so if one line is busy or doesn't answer I pass it to the next one. I have to know who answered the call, how can I do this? I'm currently looking for a Dial as the last command and getting the data for that command, but doesn't seem a solid solution. Best regards, Robert T. _ Una mejor experiencia en Internet. Prueba gratis dos meses MSN 8. http://join.msn.com/?pgmarket=es-esXAPID=1577DI=1055 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
