Re: [Asterisk-Users] Asterisk on MS Virtual Server
Turgut Abacioglu wrote: Hello I downloaded Astwind and get working the network (means can access to Internet through MS Windows). DEbian and Asterisk files are updated from Internet. But When I make install in Zaptel (it was my first make) I got many errors. Acoording to one manual this happens when we do not have modeversion .h kernel header file (according to it, it should reside in /usr/src/linux) which in /usr/src/linux, a make menuconfig will create it. BuT I do not have the linux dir (in /usr/src) and kernel source files thus modversion.h file. In addition I do not know how to download kernel files to linux directory (I tried apt-get but I could not format properly the /etc/spt/source.list file) Could you help. Am I in the correct path? No, you are not. Zaptel is a driver to hardware cards. CoLinux (on which Astwind is based) is a virtual Linux running as a Windows task. Virtual here means - no hardware. In short, you can install or otherwise use any hardware cards, like Zaptel, with Asterisk when running on CoLinux and generally, I'll advise you to not use Astwind for anything other then playing. It's a nice toy, but that is all. Gilad -- Gilad Ben-Yossef [EMAIL PROTECTED] Codefidence. A name you can trust(tm) Web: http://codefidence.com | SIP: [EMAIL PROTECTED] Tel: +972.9.8650475 ext. 201 | Fax: +972.9.8850643 I am Jack's Overwritten Stack Pointer -- Hackers Club, the movie ___ 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: How do I match a D? (Was: RE: [Asterisk-Users] In-band disconn ect problem (legacy PBX) - asterisk doesn't hear the touchtone?)
David Brodbeck wrote: -Original Message- From: David Brodbeck [mailto:[EMAIL PROTECTED] Okay, the problem appears to be that I'm tone deaf. ;) I finally thought to turn on debugging on the channel. The PBX is sending D, not *. The programmer of the previous voice mail system (whose configuration I was cribbing from) seems to have made the same mistake. Is there some trick for matching the letter tones? I added this extension: exten = D,1,Goto(bye,s,1) But it doesn't trigger, even though I see this debugging output when I hang up: [ TYPE: DTMF (1) SUBCLASS: D (68) ] [Zap/1-1] ___ I'm prbably stupid, but wont this do what you want? exten = 1,1,Goto(bye,s,1) -- Gilad Ben-Yossef [EMAIL PROTECTED] Codefidence. A name you can trust(tm) Web: http://codefidence.com | SIP: [EMAIL PROTECTED] Tel: +972.9.8650475 ext. 201 | Fax: +972.9.8850643 I am Jack's Overwritten Stack Pointer -- Hackers Club, the movie ___ 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: How do I match a D? (Was: RE: [Asterisk-Users] In-band disc onn ect problem (legacy PBX) - asterisk doesn't hear the touchtone?)
David Brodbeck wrote: -Original Message- From: Gilad Ben-Yossef [mailto:[EMAIL PROTECTED] I'm prbably stupid, but wont this do what you want? exten = 1,1,Goto(bye,s,1) No, because I wanted to match on D, not 1. I am stupid - I thought you meant the DTMF for the D button (aka 3DEF) :-) Anyway, I figured it out. The extension was working, but Background() ignores the tones A through D by default. I didn't realize this because I wasn't waiting for message playback to finish. Cool, I didn't realize you can match on these DTMF signals as Israeli phones usually don't have them :-) Gilad -- Gilad Ben-Yossef [EMAIL PROTECTED] Codefidence. A name you can trust(tm) Web: http://codefidence.com | SIP: [EMAIL PROTECTED] Tel: +972.9.8650475 ext. 201 | Fax: +972.9.8850643 I am Jack's Overwritten Stack Pointer -- Hackers Club, the movie ___ 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] Happy Wednesday Morning SMS question, slightly OT
Jay Milk wrote: That's a known, yet not feasible work-around over accessing an SMS-center directly. But the question remains how to accept IMCOMING messages with *. It's very simple - you register withj your Telco to receive SMS messages. When an SMS message arrives, your Telco line will ring and send a specific CallerID that indicates that the SMS center is calling. You should answer the line and start the Asterisk SMS application in answer mode. The actuall SMS message is sent via fsk modulation. In case of a voice line, one can only assume that if you have a DID and the codec is uLaw, you might get away with asking the Telco to enable SMS for the DID number and hopefully you might be able to get the fsk modem connection over the VoIP channel. Maybe Gilad ___ 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] answer supervision for POTS FXO interfaces
Samudra E. Haque wrote: hello, using Asterisk, is there any clever way to provide answer supervision based upon the received audio only from the FXO interface (from a public PSTN switch that does not have battery reversal, or CPC). In zapata.conf use either busydetecgt=yes busycount=6 (it will take about 10 seconds to indetify the hangup or busy) If you're lucky, you can try the experimental callprogress=yes Cheers, Gilad ___ 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] kind of urgent
Shoval Tomer wrote: Hi all. Can anyone comment why shouldn't we use FC 3 for an * production system? For the same reason you should not use Fedora Core line for ANY production system, as it designers intend it to be an experimental branch. In particular, FC3 has the NSA's SELinux patches integrated and enabled for the first time in any general purpose distro - one day it will be a great technology, but do you REALLY want to test this on your PBX system? I'm not looking to start a distro war, but we just found out that redhat 9 (and FC 1) don't support SATA drives, and apparently FC 3 does. We are only familiar with red hat and are in a point in time that switching distros is not available. The guy installing the system is already on location. Yes, I know we made a silly mistake. Please help us... I would go with either RedHat Enterprise 3.0 (in case you have the money for the support contract and patience to the stupid local marketing drones that know nothing from Matrix) or with WhiteBox Enterprise Linux (a recompilation of RHEL 3.0 from the same sources minus some trademarked pictures and costly support contract) if you don't. Hope this helps, Gilad ___ 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] Asterisk CPU priorities (nice?)
Matt Schulte wrote: Had a good question for the list, it seems whenever I work in an Asterisk console or on the machine normally I get jitters on any audio going through it. Especially if you did file copies or a 'ps ax' for example. I was wondering if there was a proper way to 'nice' the asterisk proc's? Cisco does this for example to it's EXEC and icmp processes, I tried reniceing the asterisk processes with very bad results, especially when I/O (voicemail, etc) comes into play. I'm not swapping out or anything, ideas? Since VoIP is a real time activity, simple nice really isn't enough. What you should do is mark the Asterisk proccess as a real time task for the Linux kernel to schedule accordingly. You can do this with Asterisk by passing the -p option to the Asterisk command line. A warning is due here: real time priority scheduled tasks are not something to be toyed with. You need to be root to be able to turn on this feature (meaning you have to be running Asterisk as root). A bug in Asterisk, a problem with mpg123 or a red alert on a FXO card can very well leave your system completly non responsive - so use with care. Having said that, I've been running an Asterisk server on a machine which is also used as SOHO firewall and file server for year now and it works great. Hope this helps, Gilad ___ 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] Asterisk CPU priorities (nice?)
Justin Carlson wrote: what is wrong with running asterisk with the -pg flags at startup? Which is exactly what I suggested: Since VoIP is a real time activity, simple nice really isn't enough. What you should do is mark the Asterisk proccess as a real time task for the Linux kernel to schedule accordingly. You can do this with Asterisk by passing the -p option to the Asterisk command line. And the warning still holds: A warning is due here: real time priority scheduled tasks are not something to be toyed with. You need to be root to be able to turn on this feature (meaning you have to be running Asterisk as root). A bug in Asterisk, a problem with mpg123 or a red alert on a FXO card can very well leave your system completly non responsive - so use with care. Having said that, I've been running an Asterisk server on a machine which is also used as SOHO firewall and file server for year now and it works great. Gilad ___ 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] Is asterisk that unstable ????
James wrote: I've seen something with the X101P that lead me to think so: I have two cards and two lines. I also own a small UPS that happend to have a jack for a phone line, to act as a power cleaner and I've put the line that goes to one of these cards there. Surge arrestors used for POTS lines aren't the same as used for digital circuit. I'm not surprised you had trouble. The X101P is a FXO interface card for a POTS line, not digital. Gilad -- Gilad Ben-Yossef [EMAIL PROTECTED] Codefidence. A name you can trust(tm) Web: http://codefidence.com | SIP: [EMAIL PROTECTED] Tel: +972.9.8650475 ext. 201 | Fax: +972.9.8850643 I am Jack's Overwritten Stack Pointer -- Hackers Club, the movie ___ 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] Is asterisk that unstable ????
Damon Estep wrote: Any PC platform is only as stable as the sum of what you run on it, put a single analog interface in a red hat ES on $10,000 worth of hardware and you will have to reboot every 3 days. I'm not seeing these problem with X101P, nor does any of my (not so many) clients. And all that's boils down to is that: a. I'm lucky and b. I've have helped my luck by using only one card per machine, choosing a good MB and making sure the card don't share IRQ with anything. But the point still remains - any software or hardware that needs to be rebooted every 3 days to work is broken and should not be used. Period. If you use such software or hardware you should find out what's wrong and fix it or switch to something else. The reason that the connection between nightly reboots and MS exists is because: a. MS users they can't fix it - it's propritery and b. MS users can't replace it - for a heck load of reasons not interesting to discuss now. Basically what we're saying is that nightly reboot is simply not an option, except in MS shops, that for some reasons are willing to accept such low quality. How and why this comes about is left as excersize to the alert reader... ;-) Gilad ___ 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] Voicemail and Zapatel
Adi Linden wrote: On Thu, 30 Dec 2004, Lyle Giese wrote: Is your X100P set for loop start or Kewl Start? I am betting loop start, try changing to ks instead. Lyle This is what I have in /etc/asterisk/zapata.conf so it should be Kewl Start. It might be that your local telco does not supply disconnect supervision. Try adding: busydetect=yes busycount=6 And if you're lucky Asterisk will guesstimate hangup based on that busy signal you hear when the line is disconnected by the other hand (it takes about 10 seconds with bustcount=6). A word of warning: turning this might cause random disconnect in the middle of calls, so test test test. Gilad ___ 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] Is asterisk that unstable ????
Greg - Cirelle Enterprises wrote: Are you running a stable (v 1.0 - 1.0.3) or cvs Asterisk CVS-v1-0-10/03/04 I've upgraded two months ago to get a feature I wanted (SMS support). It should be round about Asterisk 1.0.2 Gilad ___ 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] Is asterisk that unstable ????
Rich Adamson wrote: The only issue I have with that is there are several people with digium T1 and TDM cards in their systems, and its always the TDM that goes out to lunch; not the T1. No doubt there are less then desirable mobos around (and probably lots of them), but that doesn't explain why stability of the TDM's very different from a T100P (both with Intel 537 chips). You know, this is sort of a crazy guess, but I think the power on the telephone line has a lot to do with the flakiness of these cards. I've seen something with the X101P that lead me to think so: I have two cards and two lines. I also own a small UPS that happend to have a jack for a phone line, to act as a power cleaner and I've put the line that goes to one of these cards there. Now, in two different occasions, during a power out the UPS signaled the server which shutdown gracefully and when the power was back on , the card whose line went through the UPS was in a Red Alert state. Only taking out the line from the UPS and putting it directly into the telco socket a couple of time cleared this alaram (no, even rebooting did not help). After these two inceidents I simply kept the line directly to the socket (like the second card) and we had several power outages since (don't ask) and this did not happen. So my guess is - what makes the FXO/FXS more sensative then the PRI cards is the power on the line. Or not. Did I mention it's crazy guess? :-) Gilad ___ 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] Is asterisk that unstable ????
Greg - Cirelle Enterprises wrote: from voip-info wiki Asterisk automatic daily restart To automatically restart Asterisk you can add something like this to cron # Restart Asterisk PBX once a day to prevent any problems from piling up 10 7 * * * root /usr/sbin/asterisk -rx restart now /dev/null 21 or 10 7 * * * root /usr/sbin/asterisk -r -x restart gracefully /dev/null 21 Does this software have substantial problems that one would have to do this??? I'm runing Asterisk for a year now as the IPBX of our little consulting firm. It stopped working only 4 times: two of these where power failures and the other two turned out to be Telco company problems (dead line). We have 2 PSTN lines (using Digium X101P cards), 5 intrernal VoIP extentions (Grandstream budgettone - one of which is located on another continent, using a Wifi connection to a near by village that hosts an ADSL router... don't ask) and 2 VoIP termination/origination lines. Of course, your mileage may very, but at least here there is no nightly restart script. Hope that helps you in any way. Gilad -- Gilad Ben-Yossef [EMAIL PROTECTED] Codefidence. A name you can trust(tm) Web: http://codefidence.com | SIP: [EMAIL PROTECTED] Tel: +972.9.8650475 ext. 201 | Fax: +972.9.8850643 I am Jack's Overwritten Stack Pointer -- Hackers Club, the movie ___ 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] Asterisk Interface to propriotary system and GPL
Shahed wrote: Hi All, I am wondering if I will be breaking the GPL, if I write for example, a channel driver or make some modifications to the astrisk source code, to interface at RUN TIME, through sockets, with a proprietary system. First, a warning - I am NOT a lawyer. I don't play one on TV, either and you really need a lawyer to answer this question. Having said that... The real testing which matter is whether your socket layer + propritery stuff + Asterisk is considered derived work of Asterisk. This is the the ONLY test that really matters. In most circumstances known to me, if you communicate with the propritery engine via sockets, even if you wrote the layer for this yourself, you are not creating a dervied work anymore then using Asterisk to communicate via SIP to some propritery SIP server/gateway makes a propritery work from the SIP gateway and Asterisk, so you're OK legally. Having said that, and reminding you again that you really need to ask a layer, you should consider two other issues: 1. Moral issue - being legal does no make it moral. 2. Perfomace issues. And why really not make your propritery code wrapper talk some VoIP protocol and be done with it anyway? Gilad ___ 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] Low-latency kernel?
Rich Adamson wrote: While trying to apply the low-latency kernel patches to our RHv9 Linux 2.4.20-31.9, the patches would not apply. In comparing one of the first patch files (lowlatency.h) to that already on the system, it would appear the low latency patches were already applied by RH. The original RHv9 file (lowlatency.h) even had the patch author's name/credit in it. Does anyone know whether RH made an effort to incorporate the patches, and if so, about what kernel version? Redhat kernel's, like most Linux distro vendors, contain around 200 patches over the vanilla tree, including O(1) scheduler, VM system patches and the lowlatency patch. You can find out the exaqct version etc by downloading the kernel SRPM that include the list of patches. Gilad ___ 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] four wildcards in a single pc
Hi Jim, Jim Van Meggelen wrote: Getting dedicated IRQs for the cards is a minor problem compared to what happens when you have four cards hammering away mercilessly at the chipset and CPU of your motherboard; 1000 IRQs per second, per card. Nobody's really sure what's wrong, but it causes problems for pretty nearly everyone. From your description it is very clear what is wrong - the machine is heavily over loaded (or sometime having load spikes) due to interrupt livelock. It spends so much resources dealing with interrupts that it doesn't have enough CPU time to handle any thing else. If anyone is interested in a very more info about this phenomena, simply search google for interrupt livelock and interrupt mitigation. Most of the research pertaining to this problem was done for network cards but it really applies to any source of (too many) interrupts. I've had some expereicne dealing with the problem in network cards. If I can help in any way... Cheers, Gilad -- Gilad Ben-Yossef [EMAIL PROTECTED] Codefidence. A name you can trust(tm) Web: http://codefidence.com | SIP: [EMAIL PROTECTED] Tel: +972.9.8650475 ext. 201 | Fax: +972.9.8850643 I am Jack's Overwritten Stack Pointer -- Hackers Club, the movie ___ 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] Is Gigabit Ethernet necessary?
[EMAIL PROTECTED] wrote: For an office that is using VoIP phones to connect to Asterisk, is gigabit ethernet really necessary for the Asterisk box to connect to the switch? I know that I won't even approach the limits of 100 Mbps, but would gigabit help with latency / collisions when several calls are underway? The fact is, anything going outside the office will be over a data T1, so intuition tells me that 100 Mbps should be fine... The office will have 20 phones, with remote VoIP phones added to the mix later on. The reason to chose a Gigabit Ethernet card has nothing to do with bandwidth - (most of?) these card use some sort of interrupt mitigation technique which takes a hell lot of load off of the processor for dealing with interrupts. VoIP traffic, with it's typical many small packets, is very susceptible to causing interrupt live lock on servers and routers and interrupt mitigation scheme (or even polling, but that's rare) makes a real change in performance. Having said that, there are 100Mb cards that do interrupt mitigation as well (for example AFAIK the Intel e100 cards) and there are drivers that implement interrupt mitigation at the software level (customized drivers for the tulip chip set based cards and the Linux NAPI framework). However, it is simply much easier to just grab a Giga card then research which 100Mb chip and which driver you need to get ;-) Hope this helps, Gilad -- Gilad Ben-Yossef [EMAIL PROTECTED] Codefidence. A name you can trust(tm) Web: http://codefidence.com | SIP: [EMAIL PROTECTED] Tel: +972.9.8650475 ext. 201 | Fax: +972.9.8850643 I am Jack's Overwritten Stack Pointer -- Hackers Club, the movie ___ 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] BLOCKING incoming FAXES on voice line.
Joseph wrote: At time to time somebody is trying their luck and send me most likely a junk fax on my voice line. During normal working hours is not a problem I just pickup the line and hangup the call but after-hours my voice mailbox is intercepting the call and recording those beeps (waisting my CPU cycles). Is there a way to block call / issue hangup command if the incoming call is a fax? Assuming you're getting the calls on some sort of a Zap channel, then in the same context where your extention is defined, add: exten = fax,1,Hangup or even better yet: exten = fax,1,Background(if-this-really-is-a-human-please-press-1) exten = fax,2,Hangup You must also have: faxdetect=incoming In zaptel.conf for this to work. Hope this helps, Gilad -- Gilad Ben-Yossef [EMAIL PROTECTED] Codefidence. A name you can trust(tm) Web: http://codefidence.com | SIP: [EMAIL PROTECTED] Tel: +972.9.8650475 ext. 201 | Fax: +972.9.8850643 I am Jack's Overwritten Stack Pointer -- Hackers Club, the movie ___ 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] Asterisk with SMS
Nguyen Quang Hoa wrote: Hi all, I am trying to setup the SMS of Asterisk. I have a Siemens SMS enable fixed phone which connects to my Asterisk through PSTN. Currently, I can use my fixed phone to edit and send messages to the Asterisk. However, I cannot make my Asterisk to send messages to the fixed phone. The SMS command displays TX and RX records, hang for a while and then stops with non-zero exits. I read somewhere in the technical manual of the phone that the phone should be able to identify the caller id in order to receive messages. My telephone line for the fixed phone has the callerid feature, but I guess I should config the phone as well to identify the SMS calls from the Asterisk, but I don't know how. Have anyone tried Asterisk with SMS? Yes, I did. The phone number you need to have Asterisk dial to send SMS messages is NOT the phone number of the phone you want to receive the SMS message, but that of your local SMS service center. You can receive this number from your local telco or from browsing your PSTN phone menus. Hope this helps, Gilad ___ 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] Asterisk with SMS
B G wrote: My intention is to setup Asterisk to be a message center to receive from and send SMS to fixed phones. Can it be possible? My fixed phone can dial to Asterisk and send SMS to Asterisk, but I cannot setup the other way: make Asterisk dial to fixed phone and send SMS to fixed phone. Ah, I see. In that case your phone must have set in it's menu some caller ID (number) that Asterisk should be made to set the caller ID to when it is trying to send the SMS. That's the only way for the phone to figure out that an SMS is being sent to it when Asterisk rings. Gilad ___ 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] [Fwd: Call Transfer between phones]
Michael Nolan wrote: Receive call, press flash, call other party, wait for answer, press transfer, hangup. Yes, assuming fairly recent firmware (1.0.5.16+). That does an attended transfer. To blind transfer you can just press transfer exten send and hangup. Question: I saw on the Wiki a report that 1.0.5.16 message button does not play well with Asterisk and sends a broken NOTIFY. Is that true? Anyone here have been using 1.0.5.16 (or later?) and care to comment on problems he or she may be having? The reason is that I have a working setup here, except the attended transfer and I'm willing to upgrade the firmware to get that extra feature but not if it breaks something that already works... :-) Thanks! Gilad -- Gilad Ben-Yossef [EMAIL PROTECTED] Codefidence. A name you can trust(tm) Web: http://codefidence.com | SIP: [EMAIL PROTECTED] Tel: +972.9.8650475 ext. 201 | Fax: +972.9.8850643 I am Jack's Overwritten Stack Pointer -- Hackers Club, the movie ___ 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] OS Choice ?
Alex Brecher wrote: Which Distro is the most commonly used distro with Asterisk please ? I don't know which is most commonly used, but I can tell you which is the easiest to install if you're going to install the OS from scratch anyway and plan to use it with Asterisk: Xorcom Rapid is a Debian/Asterisk distribution program that includes an auto-install and special auto-configuration features. It quickly and effortlessly converts any PC to a functioning Asterisk PBX... http://www.voip-info.org/wiki-Xorcom+Rapid -- Gilad Ben-Yossef [EMAIL PROTECTED] Codefidence. A name you can trust(tm) Web: http://codefidence.com | SIP: [EMAIL PROTECTED] Tel: +972.9.8650475 ext. 201 | Fax: +972.9.8850643 I am Jack's Overwritten Stack Pointer -- Hackers Club, the movie ___ 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] how to call s extension from SIP phone?
*Which* SIP phone? Some of them, like the Gradstream Budgettone, have a dial this number when user picks up configuration option. If you have such a phone you can create an alias for the s extention like so: exten = 666,1,Goto(s,1) And instruct the phone to dial '666' when the phone is off the hook. Hope this helps, Gilad -- Gilad Ben-Yossef [EMAIL PROTECTED] Codefidence. A name you can trust(tm) Web: http://codefidence.com | SIP: [EMAIL PROTECTED] Tel: +972.9.8650475 ext. 201 | Fax: +972.9.8850643 I am Jack's Overwritten Stack Pointer -- Hackers Club, the movie ___ 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] Opinions on renice or turning off swap or ramdis k as swap?
Colin Anderson wrote: I have 4 gig in my * box. I'm tuning for performance and I'd like to ask opinions: 1. asterisk -p == renice -20 ?? What asterisk -p does is mark the aterisk process as a POSIX real time priority process. Unless you have other process marked in the same way, the scheduling algorithm will prefer this process to others at all times. which means that if is not blocking, it will be the running process. I've been running like this with Asterisk for a couple of month with no ill effects except that some error conditions cal cause asterisk to go into a loop which will effectively freeze all user space activity on the machine. I keep a shell set to a higher real time priority then asterisk on the machine for these cases. You can use the following tool to get a real time priority shell: http://projects.codefidence.com/realtime.html 2. I've turned off swap with no apparent ill effects. Can anyone commment on long term effects with moderate load (say, 30 SIP phones / 2-3K calls /day) Don't do that. Swap can be useful to allow the kernel to organize memory a little better and avoid fragmentation. 3. Can anyone comment on using ramdisk as swap and whether this is a good idea or bad idea? Very bad idea. Linux memory management is much smarter then DOS, from which you got this idea, I assume. Anyone else have any performance tips? Disable any interrupts not needed on the system. Specifically, use NAPI enabled network device drivers and turn NAPI on. Gilad -- Gilad Ben-Yossef [EMAIL PROTECTED] Codefidence. A name you can trust(tm) Web: http://codefidence.com | SIP: [EMAIL PROTECTED] Tel: +972.9.8650475 ext. 201 | Fax: +972.9.8850643 I am Jack's Overwritten Stack Pointer -- Hackers Club, the movie ___ 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] redhat9 100% CPU
TELUX wrote: Redhat 9 is running 100% cpu usage. I had a couple boxes doing this. upgraded to Fedora and its ok. I would try running asterisk with LD_ASSUME_KERNEL=2.4.1 if it isn't already. Gilad -- Gilad Ben-Yossef [EMAIL PROTECTED] Codefidence. A name you can trust(tm) Web: http://codefidence.com | SIP: [EMAIL PROTECTED] Tel: +972.9.8650475 ext. 201 | Fax: +972.9.8850643 I am Jack's Overwritten Stack Pointer -- Hackers Club, the movie ___ 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] STUN and Asterisk? (Was: SER is a better NAT solution?)
Matthew Boehm wrote: STUN requires 2 NIC interfaces on the machine running the server right? And both interfaces need seperate public IP's right? ' Why ever for? I realize that in order to set up a STUN server you need a public IP, but why two of them and why two different interfaces? Dazed and confused, Gilad -- Gilad Ben-Yossef [EMAIL PROTECTED] Codefidence. A name you can trust(tm) Web: http://codefidence.com | SIP: [EMAIL PROTECTED] Tel: +972.9.8650475 ext. 201 | Fax: +972.9.8850643 I am Jack's Overwritten Stack Pointer -- Hackers Club, the movie ___ 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] Call Status
Shaun Tierney wrote:
When I use the Dial command to connect a call using my Asterisk PBX, it
seems that the PBX says that the call was answered right when the two
channels are bridged together, rather than when the actually callee answers
their phone. I would like to be able to detect the actual call status and
respond to it in the dialplan. ${DIALSTATUS} just seems to tell me whether
or not the channel answered rather than the actual callee, so it equals
ANSWER every time I dial because the bridge is automatic. Is there any
command or variable out there that will allow me to determine the actual
call status?
It's not the Dial command or Asterisk. I'm willing to be you are dialing
an outside line, right?
Well, on analog lines it can sometime happens that from your PBX point
of view the phone line is being answered by a switch or pbx on the other
side which continues to send dialing sounds down the line until the
other side actually answers.
For a simple analog phone it doesn't matter - you simply hear ring
sounds until someone answers. But for PBX equipment it's a problem -
there is no way [1] for Asterisk to know that the line hasn't really
been answered yet.
Gilad
[1] Well, I lied - there is a way, which is the callprogress feature.
it's far from perfect (or in some states, working) though. Look for
callprogress on voip-info.org
--
Gilad Ben-Yossef [EMAIL PROTECTED]
Codefidence. A name you can trust(tm)
Web: http://codefidence.com | SIP: [EMAIL PROTECTED]
Tel: +972.9.8650475 ext. 201 | Fax: +972.9.8850643
I am Jack's Overwritten Stack Pointer
-- Hackers Club, the movie
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Re: [Asterisk-Users] How to generate ringing tone to a calling party.
Joseph wrote: Who to generate ring tone to a calling party when the call is passed to an extension. The asterisk answers correctly, plays welcome message and ring an extension, but the caller does not here the rings. I'm willing to be you didn't set up Music On Hold correctly but are using the m option in the Dial command. Gilad -- Gilad Ben-Yossef [EMAIL PROTECTED] Codefidence. A name you can trust(tm) Web: http://codefidence.com | SIP: [EMAIL PROTECTED] Tel: +972.9.8650475 ext. 201 | Fax: +972.9.8850643 I am Jack's Overwritten Stack Pointer -- Hackers Club, the movie ___ 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] Memory Consumption
Roland Zagler wrote: I use Asterisk 1.0.2 on a RedHat Enterprise Server 3.0 (Kernel 2.4.21) and i experienced that the memory consumption of the asterisk-process started by the init.d-script raises continously. Now, after 3 hours of operation (on our testing-system we have 30 concurrent connections to another asterisk box using IAX2 and GSM codec) there is already 66MB allocated. I think this could be ok, but the memory consumption increases until all memory (including swap) is used, and then there is following entry in /var/log/asterisk/messages: Failed to fork(): Cannot allocate memory This is a just a guess, but an educated one - RHEL3 is using a RedHat back port of NPTL which is, how to say it... completly fscked up :-) If this is really the problem then all you have to do to resolve the issue is to add a line with: export LD_ASSUME_KERNEL=2.4.1 to the init script that runs Asterisk. For an explanation of what this does look here: http://people.redhat.com/drepper/assumekernel.html If this does work, you must do the following: 1. Tell me it worked, I'm a curious bastard :-) 2. Call up RedHat support and tell them to fix their broken Enterprise distro. Good luck, Gilad -- Gilad Ben-Yossef [EMAIL PROTECTED] Codefidence. A name you can trust(tm) Web: http://codefidence.com | SIP: [EMAIL PROTECTED] Tel: +972.9.8650475 ext. 201 | Fax: +972.9.8850643 I am Jack's Overwritten Stack Pointer -- Hackers Club, the movie ___ 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] SysMaster and GPL Violation
Martin List-Petersen wrote: You may or may not be aware that to contribute code to FSF owned and maintained software one needs to assign copyright to the FSF in much the same way one is required to assign copyrights to Digium. True, I don't think the FSF are going to sell licenses to those programs under different terms and Digium does. Could you please go and read Marks (or my) mail again ? I don't know about the FSF (haven't checked) but with Digium you are keeping your copyright, but giving Digium a non-excluse, non-revocable license to your changes. I stand corrected. AFAIK the FSF is a full copyright assigment. BTW, I actually signed and faxed the Digium disclaimer and already contributed code under it thinking it was a copyright assigment. I guess I just trust Mark, which in the end - this is what all this is about. Gilad ___ 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] SysMaster and GPL Violation (lets think before we
Joe Greco wrote: Hey All, Isn't it possible that part of the commercial licenses that is offered is that you (the buyer) are not required to advertise, disclose, or even admit that your products offerings are based on an open source project? What other reason would one have for buying a commercial license for an OS piece of software? It is no doubt a Nazy plot. Can this thread please end now? Gilad ___ 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] SysMaster and GPL Violation
Further, that really does seem to fly in the face of the spirit of the GPL, and this is touched on by the GPL FAQ: http://www.gnu.org/licenses/gpl-faq.html#TOCReleaseUnderGPLAndNF http://www.gnu.org/licenses/gpl-faq.html#TOCDeveloperViolate You may or may not be aware that to contribute code to FSF owned and maintained software one needs to assign copyright to the FSF in much the same way one is required to assign copyrights to Digium. True, I don't think the FSF are going to sell licenses to those programs under different terms and Digium does. FYI, Gilad -- Gilad Ben-Yossef [EMAIL PROTECTED] Codefidence. A name you can trust(tm) Web: http://codefidence.com | SIP: [EMAIL PROTECTED] Tel: +972.9.8650475 ext. 201 | Fax: +972.9.8850643 I am Jack's Overwritten Stack Pointer -- Hackers Club, the movie ___ 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] SysMaster and GPL Violation
Joe Greco wrote: I'm struggling to think of another free software project where contributed code bearing an identical GPL or BSD license would require any such additional disclaimer. How about any softwaer owned by the FSF, MySQL, SleepCat DB, QT. I can continue if you want... :-) Gilad -- Gilad Ben-Yossef [EMAIL PROTECTED] Codefidence. A name you can trust(tm) Web: http://codefidence.com | SIP: [EMAIL PROTECTED] Tel: +972.9.8650475 ext. 201 | Fax: +972.9.8850643 I am Jack's Overwritten Stack Pointer -- Hackers Club, the movie ___ 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] SysMaster and GPL Violation
Joe Greco wrote: The GPL is fundamentally flawed in that it's never been functionally tested and challenged in court, and many IP lawyers believe that there are challenges that it would not survive. The fact that some lawyers may have found further legal loopholes to exploit is not shocking, given the holes in the current implementation. Actually, this is not true. The GPL was tested in a Germen court and survived very well thank you very much. But this is not the most improtant point. The important point is this: The target of a good license (or any legal document for that matter) is not to survive in court. The purpose of a good license is to be so iron clad clear that it never ever gets into court in the first place. And this, my friend, is something the GPL has done *very well*. Gilad -- Gilad Ben-Yossef [EMAIL PROTECTED] Codefidence. A name you can trust(tm) Web: http://codefidence.com | SIP: [EMAIL PROTECTED] Tel: +972.9.8650475 ext. 201 | Fax: +972.9.8850643 I am Jack's Overwritten Stack Pointer -- Hackers Club, the movie ___ 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] SysMaster and GPL Violation
Matt Riddell wrote: 1. The strings prove it is Asterisk 2. They deny it is 3. We have multiple sources who can confirm this So, let's all post the article on GPL violation to Slashdot. All in favour? Danger Will Robinson. I actually heard Welte give a talk about GPL violation in Germany in the last OLS and there are very good legal reasons why a huge public exposure is a very bad idea *prior to getting Digium lawyers ready with a law suit*. IANAL, but my understanding is that if we make this public before Digium lawyers are OK to move forward you are practically stopping them from using certain legal tactics which can only be used if you demonstrate you've reacted at the first possible moment after you learned about the violation. In short, first find out what Mark wants to do with this and let Digium lawyers go through with this or otherwise you're risking making it more difficult for Mark Digium. Only if Mark is not interested in going the legal path should we make this Slashdot material. Gilad ___ 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] Top posting
George Gardiner wrote: I must admit I live in perpetual fear of forgetting to switch of html or rtf formatting (useful for work) and top posting. A: Because otherwise we don't understand what you're replying to. Q: Why top posting is so frowned upon? Cheers, Gilad -- Gilad Ben-Yossef [EMAIL PROTECTED] Codefidence. A name you can trust(tm) Web: http://codefidence.com | SIP: [EMAIL PROTECTED] Tel: +972.9.8650475 ext. 201 | Fax: +972.9.8850643 I am Jack's Overwritten Stack Pointer -- Hackers Club, the movie ___ 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] Amount of time asterisk take to pickup incoming call on ZAP interface
Sophus wrote: Hello, Ok... After trying all this stuff, what still confuses me is - I dial the zap interface -- Asterisk instantly says- --Starting simple switch on 'Zap/1-1' then a couple of seconds later Nov 4 19:11:39 NOTICE[1146895]: chan_zap.c:5356 ss_thread: Got event 2 (Ring/Answered)... then another couple of seconds later Nov 4 19:11:42 NOTICE[1146895]: chan_zap.c:5356 ss_thread: Got event 2 (Ring/Answered)... then another couple of seconds later -- Executing Answer(Zap/1-1, ) in new stack why all the delay in executing the answer? Because teh way caller ID information is passed of analog lines is by sending FSK modulated beep between the first and the second ring. Therefore, what Asterisk does (if you asked to enable caller ID) is to register the first ring, get the the FSK modulated beep, wait for the second beep (thus knowing it is over) and then answering the phone. Cheers, Gilad -- Gilad Ben-Yossef [EMAIL PROTECTED] Codefidence. A name you can trust(tm) Web: http://codefidence.com | SIP: [EMAIL PROTECTED] Tel: +972.9.8650475 ext. 201 | Fax: +972.9.8850643 I am Jack's Overwritten Stack Pointer -- Hackers Club, the movie ___ 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] SMS through Cisco PSTN GW
Nahuel Alejandro Ramos wrote: Hi everyone, I have my asterisk working with a Cisco 2610 PSTN Gateway connected over SIP protocol. Could anybody tell me if I can send and receive SMS through this Gateway with the SMS command in asterisk? Depends on the codec really. Landline SMS is sent via FSK modulation. I'm guessing that if you're using ulaw/alaw codecs for the call you shoudln't have a problem. You might also have to shut off echo cancelation. Cheers, Gilad -- Gilad Ben-Yossef [EMAIL PROTECTED] Codefidence. A name you can trust(tm) Web: http://codefidence.com | SIP: [EMAIL PROTECTED] Tel: +972.9.8650475 ext. 201 | Fax: +972.9.8850643 I am Jack's Overwritten Stack Pointer -- Hackers Club, the movie ___ 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] SIP REGISTER -- Asterisk non-compliant or is it the provider?
Benjamin on Asterisk Mailing Lists wrote: On Sat, 06 Nov 2004 12:00:57 -0500, Karl Brose [EMAIL PROTECTED] wrote: The syntax for the register command is register=username:secret:[EMAIL PROTECTED]:port/extension Trouble is though that this does not have any effect on the username in the digest. Whatever it is intended for, it's not doing anything to untie the From field from the Digest username field. Well, it looks like the digest is being built with authname in build_reply_digest(). It's using sip_pvt.authname which get's initialized to the *username* in create_addr, but is then being copied over by the authuser field from the SIP registery which looks like it should get initalized by sip_register(). But obviously, it doesn't :-) Hope this helps in any way, Gilad ___ 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] SIP REGISTER -- Asterisk non-compliant or is it the provider?
Benjamin on Asterisk Mailing Lists wrote:
It would seem that there is only a single line which has changed in
respect of SIP reigstration ...
*** static int transmit_register(struct sip_
*** 4054,4059
--- 4055,4061
if (!ast_strlen_zero(r-username)) {
strncpy(p-peername, r-username,
sizeof(p-peername)-1);
strncpy(p-authname, r-username,
sizeof(p-authname)-1);
+ strncpy(p-fromuser, r-username,
sizeof(p-fromuser)-1);
}
... and I am trying to make sense of this so as to be confident to
apply the change to the earlier version. Is this likely to be what
fixed this bug or did I mess up with the diff? I'd appreciate if
I don't claim to understand the code at all, but what little I think I
understand from it makes me believe this is not the change you're
looking for.
Cheers,
Gilad
--
Gilad Ben-Yossef [EMAIL PROTECTED]
Codefidence. A name you can trust(tm)
Web: http://codefidence.com | SIP: [EMAIL PROTECTED]
Tel: +972.9.8650475 ext. 201 | Fax: +972.9.8850643
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-- Hackers Club, the movie
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Re: [Asterisk-Users] X100P noise on ADSL line.
WipeOut wrote: Following on from the message below I have discovered that the X100P causes the SNR on my ADSL line to drop even with the Asterisk box **switched off** and the power unplugged... This seems very strange.. Why should a card in a switched off PC cause noise on a line meaning that it drops out and has to reconnect quite often.. Anyone got any other ideas to try and stop it messing up my internet connection cos its causing havoc with my VoIP calls coming in and going out over the ADSL line.. Not really helpfull, but just so you'll know - I had a very similar problem with Digium bought X101P. Every time the card was connected to the microfilter the ADSl would drop dead. I managed to find another bloke with the same problem and had a few techn support emails with Digium but they could not help me. I tried replacing the microfilter, adding another filter, building a reverse filter (don't ask...) - nothing helped. Then we moved offices and it works great in the new office along side the ADSL. I have no idea why. There was just something in the old telco line that made it happen, I guess. Cheers, Gilad ___ 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] VoIP test numbers
All you really need is a list of 1-800 numbers in various countries. Most multi-national corporations have a list buried somewhere on their web site. For example: http://www.microsoft.com/resources/howtotell/ww/windows/what.aspx Gilad ;-) -- Gilad Ben-Yossef [EMAIL PROTECTED] Codefidence. A name you can trust(tm) Web: http://codefidence.com | SIP: [EMAIL PROTECTED] Tel: +972.9.8650475 ext. 201 | Fax: +972.9.8850643 I am Jack's Overwritten Stack Pointer -- Hackers Club, the movie ___ 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] Re: call progress - what are the sticking points?
Joe Greco wrote: More difficult is the problem of knowing when the remote end has gone away. Reversal, loop break, dial tone, and just plain silence are not all that unusual as methods of detection. In some cases, you do actually need to infer that the remote has gone away. I understand that the phone company (sometime) doesn't provide information about remote hangup on POTS lines. What bugs me is the simple question - how does your average 10$ answering machine detects the hang up? I'm guessing the obvious - DSP and some heuristics as to what a hangup sounds like and it sounds to me that it isn't all that hard to do in Asterisk (since it's done in those cheap machines) but I would be very glad to hear some tips from someone that knows a little better then me. Thanks, Gilad -- Gilad Ben-Yossef [EMAIL PROTECTED] Codefidence. A name you can trust(tm) Web: http://codefidence.com | SIP: [EMAIL PROTECTED] Tel: +972.9.8650475 ext. 201 | Fax: +972.9.8850643 I am Jack's Overwritten Stack Pointer -- Hackers Club, the movie ___ 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] Re: call progress - what are the sticking points?
Joe Greco wrote: Answering machines get by on several mechanisms. The ones that come to mind are: 1) Silence detection. 2) Session time limit. Both of these are effective at doing something vaguely right within the requirements of an answering machine. If you've never heard an answering machine that's recorded a minute's worth of dialtone followed by the loud the phone is off the hook tone, then I'm shocked. :-) I never did. I also never owned an answering machine :-) Just because you can engineer around a problem doesn't make the solution right. Agreed and I was not thinking about this as a solution, but rather as a better kludge then my current method of using session time limit which is good enough for voice mail but is not good enough with conference bridge. Cheers, Gilad -- Gilad Ben-Yossef [EMAIL PROTECTED] Codefidence. A name you can trust(tm) Web: http://codefidence.com | SIP: [EMAIL PROTECTED] Tel: +972.9.8650475 ext. 201 | Fax: +972.9.8850643 I am Jack's Overwritten Stack Pointer -- Hackers Club, the movie ___ 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
