RE: [Asterisk-Users] Hardware IAX phone (please read and reply!)
Steven Critchfield wrote: On Fri, 2003-09-05 at 03:25, Marcel Prisi wrote: This is a completely open-source and open-hardware hardware phone based on Linux on an ARM embedded platform ... they already had lots of experience ... but might need some different software ... bzzzt. wrong. There is a lot known about the hardware but it is not open. The software is only open after it was reloaded with debian. Also while the site you list was cheap, if you dig round, the manufacturing cost was over $300 each and target retail was over %600. Granted that was over 3 years ago, it wouldn't have dropped in price too significantly. The site you list was liquidating the last known inventory of those units. The other problem with the touchscreens and VoIP is that the telephone audio circuitry was not accessible by software running on the phone. Here is a block diagram: http://www.blurbco.com/~gork/tuxscreen/shanblock.gif A modification (ShanIP2) was designed to make the handset/speakerphone audio to/from the dsp accessible via the UCB1200 audio chip, and I had designed a PCB for the circuit here: http://www.blurbco.com/~gork/tuxscreen/shanip2-gork8.gif So have a look there : http://www.lart.tudelft.nl/ You will find there the hardware that evolved from what was in the Tuxscreen. It's license is open. It runs a 220Mhz StrongARM with more than 200 MIPS and has options for ethernet and sound i/o, all is linux-compatible ... The LART was actually around before the tuxscreen, and although it is similar, you'll find that most SA1XXX based designs are. It is still a good little board and fun to work with, as is the Tuxscreen if you can still pick one up used from someone. Anyway, since this is starting now getting pretty offtopic, I should probably leave it at this... John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Ordering digital trunks?
OK, this is probably a dumb question for a lot of you, but I have no experience with digital lines outside of a tiny bit of ISDN, so I'll just bite the bullet and ask some newbie questions. I am attempting to plan an asterisk installation with about 20 SIP phones and the following incoming lines: 1) At least 6 (as many as 10) lines for voice to the SIP phones 2) 2 incoming/outgoing fax 3) 4-10 lines for an IVR application I feel like a T1 with 24 channels should suffice, but what exactly do I order and what to I have to have in my asterisk unit to interface? Does the line they terminate just plug into a T100P or do I need some extra hardware? What services do I need to be sure I order on the T1? Is there a way using the T100P to dynamically allocate unused voice channels for data? Finally, how do I get the best fax performance for two analog fax machines out of this setup? Will an ATA-186 serve, or will I get much better performance using a TDM400P with zaptel bridging? Thanks, John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Ordering digital trunks?
I feel like a T1 with 24 channels should suffice, but what exactly do I order and what to I have to have in my asterisk unit to interface? Does the line they terminate just plug into a T100P or do I need some extra hardware? What services do I need to be sure I order on the T1? Is there a way using the T100P to dynamically allocate unused voice channels for data? There is config options for this, but you will want to have a PRI to do this. A PRI only gives you 23 channels to work with. This will also help make sure your FAX lines are clean as can be. You will want at least a block of DID numbers. OK, that basically covers the main question. Thanks very much for the response. With your question about dynamic allocation of empty channels for data, you would have to find someone who will let you dial in and bond up channels. Then you need to calculate the estimated number of lines used a any time, add in a bit for the idle lines so new calls come in, and your left with the number of possible 64k channels for data. To take your low estimates, 6+4+1 = 11 + 3 for spares and you have 14. 23-14 = 9*64kbit. So your low usage, high speed bandwidth is ~576kbit. If you don't like this speed, you really need to think about maybe a DSL line to augment or be your bandwidth. This kind of speed is fine for my purpose. This connection would be only for emergency use when our primary (6 megabit wireless) link fails, and our ISP would allow us to use it for backup purposes only at no additional cost. Finally, how do I get the best fax performance for two analog fax machines out of this setup? Will an ATA-186 serve, or will I get much better performance using a TDM400P with zaptel bridging? The TDM400P would be a good choice, and price isn't too much different. Not to mention that it will help making sure your T1 is up as it is easier to configure. Well, I was asking because I already have some ATA-186's that I am only dinking around with that could actually be used. I guess I should test them to see how (non-T.38) faxing works via asterisk and if the performance is not good, I'll move to the TDM400P. Thanks again, John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] AGI.pm?
I've seen references to this module in the mailing list archives, but it isn't in the 0.4.0 tarball, nor is it in CVS. I can roll my own and was planning to do so anyhow, but that doesn't seem to make a lot of sense if it already exists. Am I not looking somewhere I should be looking? Most of the Google hits just point to the mailing list. The asterisk-perl tools written by James Golovich are at: http://asterisk.gnuinter.net/ The following modules are available: Asterisk::Outgoing - manipulate spool files for pbx_spool Asterisk::Manager - talk to the manager interface Asterisk::AGI - helpful routines to simplify writing AGI's For those of you using it, but not checking the site regularly, version 0.07 was released recently on 09-Jul to fix bugs, add a new AGI command, and add MD5 authentication to the Asterisk::Manager. Also note that I'm just posting this for the benefit of the list readers. I don't have anything to do with the software itself other than that I am a happy user. Asterisk::Manager rocks! John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] the 'pound' and '#' are the same? (OT Rambling)
In the US, and probably some other English speaking countries, # is the pound key on telephones. In the UK it's called the hash key. The technical name for the punctuation mark is octothorpe. A lot of punctuation has strange technical names that you don't hear every day (or maybe you do if you are a typeset designer): ^ - chevron (Look at a Chevron gas station's logo next time you see one. It's two stacked chevrons.) / - virgule, solidus ` - grave = - quadrathorpe # - octothorpe * - asterisk (our favorite) - ampersand I haven't ever found any really interesting technical terms for @, %, or ?. I'd be interested in hearing some.. For you programmer types out there, take a look at this for some punctuation pronouncing fun: http://www.latenighthacking.com/projects/2002/spokenPunctuation/ Sorry for the ramble, John -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of johncn Sent: Thursday, July 24, 2003 1:16 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] the 'pound' and '#' are the same? Hi, I am translating the voice files of voicemail now. I don't know if the POUND and # are the same key in the telephone's keypad. If they are same, how could we understand the following message: %vm-msginstruct.gsm%To hear the next message press 6, to repeat this message press 5, to hear the previous message press 4, to delete or undelete this message press seven, to quite voicemail press pound. During message playback, you may press * to rewind, and # to fast forward. Regards. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] OT: list format vs newsgroup format
Who else on here prefers the newsgroup/threaded approach? If you haven't already, check out news.gmane.org for mailing lists turned into newsgroups readable by news readers... If you want threads, get a MUA that is capable of threading. Most are. The In-Reply-To header makes mail threading on lists trivial (and you can easily spot the people that hit reply and change the subject without actually starting a new thread...) Maybe you just have not yet found where to turn it on. only problem being that this list requires list membership before postingShrug Which is why the mailing list will probably continue to be used... John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Example: Writing a click-to-call application using pbx_spool
I have written a small perl CGI script that demonstrates how one might use the asterisk spooler 'pbx_spool' to make a 'click-to-dial' type application. The script is intended to be a demonstration example only and since it has little security, should not be deployed. I was just experimenting with the spooler and wrote this to try some things, and I though it'd be a good example to share.. The file 'placecall-example.cgi.gz' can be downloaded here: http://www.blurbco.com/~gork/asterisk/ Hope it helps someone, John Short readme embedded in the script: This application is written to demonstrate how you can do PC-to-phone integration with the asterisk outgoing queue mechanism, pbx_spool. WARNING! Take note that this applicaiton provides no security by default. Anyone using it can set up a call between any two extensions in the list you specify, and thus it is not suitable for any kind of real deployment! It is intended to serve only as an example. some omitted instructions Usage: Load placecall.cgi in your web browser. Select the extension from which you want the call to originate. Then, select the extension you wish to call. Your phone will ring and the caller id will show as the value you configured below. When you pick up, you'll hear Please hold while I try that extension and then your call will be connected to the party you wish to call. You should be able to apply the same basic technique seen here to write your own click-to-call style application. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Poll - Would you pay $30-$50 for high quality speech synthesis?
Also almost forgot. They sell the demo voices on their site for 29.99. Linux and windows versions. Since I believe what they use is based off festival, perhaps the voices could be made to plug into the existing festival plugin for asterisk? I have been working with app_festival for about a week or so now trying to figure out what is going on with it... The existing app_festival has a serious bug in it that makes it unsuitable for production use. I posted about it before, but the jist is that the more channels that are trying to use app_festival at the same time, the more problems there are -- the channel will abort, and this will result in asterisk abandoning the call. I have tried tracing it with all sorts of things including stepping back through some truly monumental gdb logs. The problem is not on the festival side of things, as the preforking festival keeps up fine with asterisk, and the problem occurs even when using the festival cache (broken in the current code - my patch fixes it) I cannot find the bug after a week of poking at the code. As I intended to use app_festival as a temporary replacement for recorded voice prompts in an AGI application, it was no big deal at first as it works 100% if if you always wait for the speech to finish and only use one channel (fine for testing), but after having the flexibility to do speech synthesis, I can see that it would be a tremendously good application for even the IVR that I am working on... Anyway, I hope this speech project gets off the ground. Staying all OSS is very nice, but after spending a lot of time mucking with it, I'd be easily willing to spend $50 for a great sounding, working solution. I'm still going to be poking at app_festival, though, so if anyone has suggestions or understands some of the internals and wants to work on this with me, please mail me off-list. Having a 100% working and production-ready solution for festival would be good for a number of reasons and I'd like to see this happen too! ~John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Multiple Phones for 1 Extension
I'd like to have a SIP phone at home and at the office and have them both ring when my extension is dialed. Right now I used the same config for the phones (Cisco 7960's). So they both register with the same login pw. This doesn't seem to work quiet right, where only the last phone to register seems to get the calls. What is the proper way to set this up? Have the phones register with different names (make a separate entry for each phone in sip.conf) then specify them both in the dial string separated by '' .. you can specify as many as you want and all will ring. The one to answer gets the call, naturally. exten = 3000,1,Dial(SIP/phoneoneSIP/phonetwo) John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Problems getting 7960's to play nice with Asterisk
Other things: The phone constantly says Ethernet Disconnected. Even though it tftp's configs and registers with the proxy. Something is wrong with either the phone hardware itself, the network port on the hub or switch it is attached to, the ethernet cable it is connected with, or the 10/100 autonegotiation between the phone and the switch. Any other problems you are having are quite likely due to these network problems, and so you should fix this problem before you attempt to debug any of the others. The Ethernet Disconnected message on the phones is the equivalent of having no link light on your NIC. Standard network connection debugging applies. Some tips: This problem is likely due to a bad cable or 10/100 auto negotiation issues. Try known, working cables on a known working switch/hub port to see if the problem occurs. Try plugging it into another switch (preferably of different brand) if you have one. The 7960 cannot be forced to either 10 or 100mb operation, so if you do determine that it's not the cable or phone hardware, you'll have to either find a way to force 10 or 100mb operation on your switch or you'll have to get a new one... Once you get your phone hooked up to the network without problems, then you can start looking at the software issues.. John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk on Cygwin?
Hey all, quick question: does asterisk work okay in a Cygwin environment? I want to install it on my cygwin setup for local testing/demoing and save me the hassle of using a pure linux machine I had suggested to the fellow asking about running Asterisk in VMWare (It won't work BTW) that cygwin or mingw compilation might be possible and , in fact, feasible to use for a small IP-only setup. As long as it doesn't take a huge huge performance hit from running out of Cygwin, then I'll have a go there for a start First, you won't have zaptel timings from hardware on windows, so things like musiconhold and meetme will be out of the question. I suppose you could write something similar to zaprtc for windows if you really felt like squeezing your brain out your ears. Cygwin itself won't be the problem speed wise, as it is essentially providing native calls directly to the application, but I would suggest a couple things: 1) Find a way to set the thread priority for the asterisk processes high. I believe that you will probably have to write some code specific to Cygwin to do this. Normal priority execution on Win32 will probably not be good enough even for the transcoders and whatnot to get packets in and out in a timely fashon. 2) Please share any experiences you have with this project. I never intend to run asterisk on Win32 personally, but have been interested in seeing if it's possible for fun/hack value. John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] module : cdr_sybase.so
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Roy Sigurd Karlsbakk Sent: Monday, July 14, 2003 3:16 AM To: [EMAIL PROTECTED]; cvasiliu Subject: Re: [Asterisk-Users] module : cdr_sybase.so nice this can probably be used with mssql as well :) our developers only uses that Implementing this with FreeTDS would be a better choice for the standard distribution since it has no dependencies on non-free software libraries like Sybase Open Client (sic) libs. I have had no problems doing anything I needed to with Sybase and SQL Server using FreeTDS, so for CDR logging (just inserts) it should be more than sufficient. Have a look at www.freetds.org John On Friday 11 July 2003 21:56, cvasiliu wrote: If anyone is interested ... just in case! :-)... I have tried to write , based on the cdr_mysql.so module, an Sybase module. To compile you can use something like that: export SYBPLATFORM=linux export SYBASE=/opt/sybase cc -I$SYBASE/include -c -o cdr_sybase.o cdr_sybase.c cc -shared -Xlinker -x -o cdr_sybase.so cdr_sybase.o -lsybdb -lm -L$SYBASE/lib (anyone could write the corect Makefile inside the cdr dir.?) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk and VMWare
Dan, Your problems are all the result of your computer and your software. It's not going to work for you in your setup. Repeat: It's not going to work for you in your setup. Repeat again for increased clarity: It's not going to work for you in your setup. I really don't understand why you keep asking the question because you keep getting the same answer from every single person. For the $299 that VMWare costs, you can build a barebones machine with a small HDD that is sufficient to run asterisk. Even if you'd rather run it all on the same machine, IT IS THE ONLY WAY YOU WILL GET ASTERISK TO RUN PROPERLY. VMware Workstation is NOT DESIGNED to do this kind of job. As I said in a post before, VMWare GSX Server which is designed to do this sort of thing (but still may be insufficient for asterisk) is priced at $2500. If you bought a support contract from VMWare, they'd tell you the same thing. Software running inside of VMWare with a Win32 host is not going to give you good performance when it needs to be interactive, and Asterisk needs to be interactive a lot of the time. No matter how many performance tweaks you make to the Win32 box, you're still going to have problems with asterisk. With the amount of RAM you have, Windows WILL swap the VM's main memory to disk after a while. This will cause you insurmountable performance problems with asterisk or any service-type application running in the VM. You can look at a SIP-Proxy only solution like SEP that doesn't do transcoding or IVR and maybe get things working IF you can figure out how to force windows to never swap VMWare to disk (ie buy another 640MB of ram and force VMWare to run in the highest priority even in the background) Here are your options. Both one of these will give you a 100% working solution to your problem: 1) Return VMWare if you have already purchased it for this purpose and use the $299 to build a standalone computer suitable for the task. If you don't want to build one, you can buy one already built: http://www.compgeeks.com/details.asp?invtid=MC1740-1 2) Purchase a VoIP or IVR application that runs and is supported under Windows that suits your purpose. If you need all the functionality that Asterisk provides, are stuck on Windows, and already have some cisco equipment, I hear that they have a product called CallManager that might do what you need :) No amount of belief on your part is going to make your computer and VMWare do this. John -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Dan Sent: Monday, July 14, 2003 3:23 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk and VMWare Hi, 1. run VMWARE in Full screen windows. Tried this... same problem 2. is your Linux kernel SMP? (see VM knowledge base) I have the RH9 downloaded from Redhat site. 3. what about your Linux guest CPU usage? Swap usage? Windows might report 5% but its what the linux guest sees that counts. VMWARE is a very good emulation but it is still an emulation. Doing near real time codec conversion on a AMD 1GH machine with 386MB might be too much. I'll check this, but still I don't think that the CPU power or memory is the problem, more the interrupts and timing... 4. Did you do bridge networking on the guest OS? NAT will invoke additional performance penalty, and have a big effect on your SIP call. Bridging, using another IP address from the same subnet. 5. What about the other cards in your system? Do they need a lot of interrupts from the PC? Check your perfmon for interrupts per second. CPU usage is only one piece of the pie. I think yes, a lot of interrupts are shared between cards. I have: - 1x Firewire, 2xUSB2.0, 1xUSB1.1, PCI Soft modem, USB Modem, 4xSerial Ports, 1xgraphic card + TV Tunner (ATI All-in-Wonder 128) and a HA Box (serial based). I have succeeeded using USB under VMWare (a flash memory stick) , but still not able to use ztdummy or zaptelrtc (it uses USB for timing, not?) Thanks, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Using 2 PhoneJacks with Asterisk for Data calls.
Thanks for the heads-up. Do you know of any alternatives? There is the zaptel hardware from digium.. the TDM400P for FXS ports, X100P for FXO, or the combination of T400P + channel banks. I recently posted a similair query to comp.linux.hardware relating to setting up a virtual telephone exchange and the result given there my own previous search results appeared to be a lot more expensive than that of the quicknet products (more in the $800+ range). Zaptel has the capability to do 'clear-channel' and/or 'data-quality' calls. See 'show application Dial' on the asterisk and have a look at the 'd' and 'c' options that can be passed to Dial. Someone else may be more familiar with how it works on the X100P/TDM400P hardware though as I haven't ever tested it. It was probably designed to work with channel banks, which is going to go over your budget if you are trying to find something cheaper than $800. My main concern is compatibility according to the asterisk docs, the quicknet products are and drivers are also supplied as part of the kernel. AFAIK, Quicknet is the only folks who use the Linux telephony API but even the Quicknet products often have problems with the driver that is distributed in the standard kernel... OpenH323 is about the only thing that is out there that can actually interface with the stuff 100% and Quicknet has a big hand in that project too. John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Line Override Device
You can build a UPS for that, but the better option here is to attach a phone to the phone side of the X100P that is always connected to the POTS line so that even when the computer goes down you can send and receive calls. If you don't want it to ring *unless* the power is out, you could wire it through a normally-closed relay hooked to something simple like the parallel port (there are schematics everywhere for this). When the computer is off, the relay closes, and the phone rings with the line. Heck, if you have an analog set on FXS you want to ring when power goes, you could get a SPDT relay and wire one line into open and one line into closed and switch between them. If you don't care much about incoming calls during the outage, just plugging a phone into the other end of X100p and turning off the ringer will do the trick. The specs are available on the net to show you how to wire POE (Power over ethernet). In fact I did my own so I can use the 7960 before we found a suitable wall wart. Basicaly all I did was punch down a keystone with the ethernet data lines, then punched down the power lines so that one side had power and the other didn't so I didn't chance blowing up my switch that was made before they thought of doing POE. I used the power supply from a CAC AB1 that had the ringer module broke on it. It produces 1amp of 48volts and was more than adequate for the 7960. If I had a lot of phones to power, I have a 6amp 48volt PSU from a Premisys channel bank that I picked up at a hamfest for $10. If you do this and plug anything other than the 7960 into it like a NIC you can easily damage it! (google for 'etherkiller' for more) Real power over ethernet injectors provide power only to devices that 'ask' for it, but for small setups they are very much more expensive than the price of a UPS that could power the 7960 for hours (a $30 ups running only the 7960 should go for at least a couple hours) - Compare this to paying $100+ per port for PoE injectors! Putting 'raw' 48V on the Ethernet in an office environment where someone else might accidentally plug something into the wall jack incorrectly would be a disaster! Of course there are some cost savings associated with not having to maintain and upkeep 48 UPS's for 48 phones that make PoE worth it, but I'd say that for less than 12 users it becomes harder to justify. BTW, for the UPS, we have some powerware UPSs that have plugs for external batteries. In our former server room we have one with 5 car battery sized batteries attached to it. We feel we had about a 10 hour run time. Even a lot of the UPS's that don't have plugs for this stuff can still be run with external batteries if they have a good charge controller that can deal with it. I have seen a lot of weird setups that use car and tractor batteries glommed into APC ups's that run lots of important systems! Go for it if you can justify the risk.. John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Configuring BudgeTone and ringer over TFTP
I noticed that the BudgeTone (I have the 102) with the latest firmware tries to download a file called cfg.txt (presumably the configuration) and a file called ring.bin (presumably a ringer) from the tftp server. The ring-in sound on the budgetone is the same as the ring-out sound and that is going to be confusing for users. I contacted GrandStream and was informed that both of these formats are available for licensing which sounded a bit odd - does anyone have other info? I really like this phone as an entry-level IP phone. It has great sound and works perfectly with asterisk. For the price it will be hard to beat. Too bad it still has these few usability issues... I'm looking forward to seeing the 102D. John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] app_festival not cleaning up properly?
I am still poking at app_festival a little bit and have found a problem
I don't really understand, and therefore, I don't really know what to
try to do about fixing it. I have done a lot of things in the code, but
the best I can do is rewrite a bunch of stuff and still have the
problem. Perhaps someone more familiar with the * internals can lend
some in helping me to understand what might be going on so I can fix it
:)
The problem is that after the Festival application executes, it
sometimes leaves the channel in such a state that the next application
called will always fail (return -1) for reasons I have not been able to
determine. The biggest head scratcher is that this seems to be more
likely to occur the more extensions are making calls to app_festival at
the same time. This bug is unrelated to the app_festival.c patch I
submitted yesterday, though testing with FestivalBG seemed to cause the
problem to happen more often - but that is because since the stream is
interruptable, you can get a lot more calls to Festival going.
Here is a minimal AGI that you can use to duplicate the bug (I am using
asterisk-perl 0.06)
#!/usr/bin/perl
use Asterisk::AGI;
my $AGI = new Asterisk::AGI;
my %input = $AGI-ReadParse();
$AGI-answer();
while ($tries 100) {
# Use FesivalBG instead of Festival to make the problem easier
to see
# if you have applied my Festival patch.
my $retval = chr($AGI-exec('Festival', 'i'));
print STDERR $$: Festival returned: ($retval)\n;
my $input = chr($AGI-wait_for_digit('5000'));
print STDERR $$: Button pressed: ($input)\n;
$tries++;
}
$AGI-hangup();
Set up an extension to call this AGI and then call it from one or more
phones. (the more channels you get on it the easier it is to duplicate
the problem, though it will very rarely occur with only one active
channel in the system). Begin pressing a DTMF digit as rapidly as
possible on all phones, and eventually you will see that the agi call to
wait_for_digit will fail, return -1, and your channel will hang up. This
may happen on one or more of the phones at the same time.
I originally thought that perhaps connections to the festival server had
to be serialized, but this seems not to be needed as festival forks a
new child to handle each connection. Plus, the problem also happens when
reading festival speech from the cache that app_festival can create, and
that action most definitely does not need to be serialized.
Thanks anyone, for any insight you can offer... I am beginning to
understand the * internals a lot better after today, but still have a
long way to go obviously!
John Laur
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RE: [Asterisk-Users] * Video changes
Does anyone know if someone makes a hard video phone for SIP. Dave I was curious about this too as the video support has been going in. Below are links to what I found. As with some of this stuff, I can't really find who the manufacturer is Leadtek is possibly the manufacturer of this device, but they themselves only specify H323 operation.. See for yourself: http://www.8x8.com/products/home_office/ip_videophone/index.asp.html (SIP) Looks like the same phone but no mention of SIP: http://www.leadtek.com/videophone/bvp_8770_1.html (H323) Two other phones that I found: http://www.umec-web.net/Videophones/videoproduct.htm http://www.innomedia.com/videophone/videophone.htm (H323) John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] defaultip= in sip.conf doesnt work?
I have several (various brand) sip devices with static IP's. I understand that asterisk will not accept a registration from these devices if the host= parameter is not set to 'dynamic' in sip.conf. I want calls to these extensions to be routable even before the device registers. I understand that is what defaultip= is supposed to do, but it doesn't work. I get a busy tone when dialing the extension until the phone reregisters. Here is what the entry looks like for a Cisco7960: [cisco] type=friend username=cisco secret=supersecret host=dynamic defaultip=192.168.0.55 canreinvite=no ; Cisco poops on reinvite sometimes qualify=200 ; Qualify peer is no more than 200ms away context=local callerid=Cisco Phone 2010 mailbox=2010 Of course, if I set the host= parameter to the phone's IP and set the phone not to register, everything works also, but the icons on the phone indicate that it is not registered. Is there any way around this? John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Feasability of preserving SIP (other?) registrations over restart?
Would it be feasible to look into implementing a way to preserve SIP (and possibly other protocols') registrations during a scheduled or otherwise requested restart of asterisk? I am thinking somewhat along the lines that the SIP registration has a certain duration during which it is valid that is in some ways akin to a dhcp lease. In most cases, I would think that a momentary outage by restarting the asterisk server would not affect registrations by SIP devices. Asterisk could either maintain constantly or write on demand (ie immediately before a requested restart) a cache file of SIP registrations and their expiry times that is read in on startup. Remote (dynamic ip) phones would not have to wait to re-register in order to immediately receive calls. Perhaps the issues surrounding sip registration internally to asterisk are more complicated, and this cannot be solved by a simple cache of sorts, but the idea is out here anyway.. John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Distinctive Ring Macro Example
Cool trick!
You could simplify this:
[macro-std-exten]
; Caller*ID is 4 digits (internal call)
exten = s/_,1,Dial(${ARG1}r2,${ARG2})
; Caller*ID is not 4 digits (external call)
exten = s,1,Dial(${ARG1},${ARG2})
; Both of the above lines go here next
exten = s,2,Voicemail(u${MACRO_EXTEN})
exten = s,3,Hangup
exten = s,102,Voicemail(b${MACRO_EXTEN})
exten = s,103,Hangup
For those of you running Cisco 7960's and using the ALERT_INFO stuff,
You can use this version of the same thing. (I am now using this in my
config thanks to Eric's example):
[macro-std-exten]
; Caller*ID is 4 digits (internal call)
exten = s/_,1,SetVar(ALERT_INFO=1)
; Caller*ID is not 4 digits (external call)
exten = s,1,NoOp
; Both of the above lines go here next
exten = s,2,Dial(${ARG1},${ARG2})
exten = s,3,Voicemail(u${MACRO_EXTEN})
exten = s,4,Hangup
exten = s,103,Voicemail(b${MACRO_EXTEN})
exten = s,104,Hangup
John
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Eric Wieling
Sent: Tuesday, June 24, 2003 4:38 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Distinctive Ring Macro Example
I use the following macro for my extensions. It only works with Zap
channels and assumes that any Caller*ID number that is 4 digits is an
internal call and all other calls are external calls.
Use like this: exten = 1234,1,Macro(std-exten,Zap/4,20)
[macro-std-exten]
;
; Caller*ID is 4 digits (internal call)
;
exten = s/_,1,Dial(${ARG1}r2,${ARG2})
exten = s/_,2,Voicemail(u${MACRO_EXTEN})
exten = s/_,3,Hangup
exten = s/_,102,Voicemail(b${MACRO_EXTEN})
exten = s/_,103,Hangup
;
; Caller*ID is not 4 digits (external call)
;
exten = s,1,Dial(${ARG1},${ARG2})
exten = s,2,Voicemail(u${MACRO_EXTEN})
exten = s,3,Hangup
exten = s,102,Voicemail(b${MACRO_EXTEN})
exten = s,103,Hangup
--Eric
--
BTEL Consulting
850-484-4535 x2111 (Office)
504-595-3916 x2111 (Experimental)
877-552-0838 (Backup Phone)
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RE: [Asterisk-Users] X100P creating a short-circuit on line
Did you cvs update zaptel and recompiled ? Yes. I followed the instructions on the Digium download page, namely: I looked at the log file and there was no commit on this recently. It seems that if this change has been made, it's just not in CVS yet :) Looking forward to trying it though... John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] X100P Dialing either Too Soon or Too Fast?
Quite frequently, outgoing calls from the X100P cards here will not dial properly. Instead of hearing the ringing after the Zap interface picks up, I'll hear silence for a while then the 'If you'd like to make a call please hang up and try again' recording as if zaptel picked up the line, punched only a couple digits and then left it. I assume that either zaptel is sending the dtmf before the telco is ready to get it or is sending the digits too fast. I have tried setting toggling overlapdial in zaptel.conf, and I have tried fiddling with the timing parameters a little, but I don't know what each number really does (and think that bad values here may be bad anyway) Does anyone have any ideas to accomplish: 1) Delay zaptel sending dtmf by a few milliseconds 2) Have zaptel send dtmf digits slower Thanks, John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
