Re: [asterisk-users] How accurate is show translation?
Eric ManxPower Wieling wrote: Leo Ann Boon wrote: Hi all, I'm using 'show translation' to help dimension my system, but I confused by the results I get. My 2 test systems (results below): an AthlonXP 2000+ (1.3GHz) and a Pentium D930 (duo-core, 3.0GHz) produced similar results (D930 is slightly faster). Googling shows that others have similar results running on other CPU speeds 2.0GHz. Does show translation recalc 30 show any different results? Eric, Before I posted, I ran tests with various recalc values between 10 and 200. The results are pretty much the same, give and take 1ms on either side. Forgot to add, I'm running 1.2.11 on both systems. Leo ___ --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] How accurate is show translation?
I tried it on a intel 3 ghz p4 box and a athlon 3000 768 mb ram running vista and host for centos 4 ( vmware ) considering the load on athlon running asterisk ( that too under vista plus vmware ) while intel 3 ghz p4 1 GB ram box was sitting idle with centos , there was hardly a 1 ms difference in show translation on both machines . Besides i just compared my p4's results to ur D930 results and there is no difference ( infact my g729 results are better than ) .. But this doesnt mean both are same dual core cpu's will definitely give much higher number of channel transcoding then lower p4's . Put both the box under some cpu load by other programs and then use show translation recalc 30 and you will see performance difference between them ;) g723 gsm ulaw alaw g726 adpcm slin lpc10 g729 speex ilbc g723 - - -- - - - - - - - gsm - - 22 2 2 1 5 82814 ulaw - 3 -1 2 2 1 5 82814 alaw - 3 1- 2 2 1 5 82814 g726 - 3 22 - 2 15 8 2814 adpcm - 3 22 2 - 1 5 8 2814 slin - 2 11 1 1 - 4 7 2713 lpc10 - 4 33 3 3 2 - 9 2915 g729 - 3 2 2 2 2 1 5 - 2814 speex - 4 3 3 3 3 2 6 9 -15 ilbc - 4 3 3 3 3 2 6 9 29 - On 23/12/06, Leo Ann Boon [EMAIL PROTECTED] wrote: Eric ManxPower Wieling wrote: Leo Ann Boon wrote: Hi all, I'm using 'show translation' to help dimension my system, but I confused by the results I get. My 2 test systems (results below): an AthlonXP 2000+ (1.3GHz) and a Pentium D930 (duo-core, 3.0GHz) produced similar results (D930 is slightly faster). Googling shows that others have similar results running on other CPU speeds 2.0GHz. Does show translation recalc 30 show any different results? Eric, Before I posted, I ran tests with various recalc values between 10 and 200. The results are pretty much the same, give and take 1ms on either side. Forgot to add, I'm running 1.2.11 on both systems. Leo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need quality toll free 800 number over IAX?
I have 3 toll free did's with nufone since 1 month .. Until now i dont have a problem with them .. their portal was good enough to do proper configuration and call quality wasnt bad ( even though i havent used them in really huge traffic yet ) . On 23/12/06, John Novack [EMAIL PROTECTED] wrote: Doug Crompton wrote: At the very least the BBB (bbb.org) should be notified. They have a web site SORRY, BUT IMHO the BBB is a joke. I wouldn't waste the time to type in one word and if it is really wire/internet fraud then the FBI (www.fbi.gov/majcases/fraud/internetschemes.htm) has a site you can register a complaint with. The FBI and the FTC, Federal Trade Comission are good choices. They don't require you to fly anywhere or do anything in court, and if enough people have simial experiences, they may very well do something The US government moves slow, but when it moves you REALLY don't want to be in the way of the army of bureaucrats. Don't talk yourself into inaction. If in fact you have any evedence at all, then register it. John Novack ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New astGUIclient VICIDIAL Release: 2.0.2
this really is a great program as far as i have heard even though i am not able to make it work for me _ On 23/12/06, Matt Florell [EMAIL PROTECTED] wrote: Hello, We've released another update to our astGUIclient suite: 2.0.2 http://astguiclient.sf.net/ The client suite runs on most modern web browsers on almost any GUI-capable operating system, and it includes the astGUIclient client-side web app which extends your phone's functionality and the VICIDIAL call center suite. This package is free and GPL. (the suite is not an asterisk configuration tool) This package is geared towards Asterisk installations with SIP,IAX or Zap phones and Zaptel, IAX or SIP trunks. For this release, we have made many changes on the server side, including changing logging from AGI to FastAGI which can cut system load in half on busy VICIDIAL systems. We have also tested the suite on Asterisk versions through 1.2.14(cannot use 1.2.11 or 1.2.12 because of Asterisk bugs) All client web-apps and administration pages are available in English, Spanish, Greek and German, with translations of French, Polish, Italian, Portuguese and Brazillian Portuguese for the client web-apps only. Check out the project blog for more information: http://astguiclient.blogspot.com Let me know what you think. Thanks, MATT--- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] International dialplans for Asterisk?
Am Freitag, den 22.12.2006, 09:53 -0500 schrieb Doug Crompton: Wow what a mess! I can imagine how much easier it would be if the world adopted a country/area/exchange scheme like in the US with known length. It must be complicated in Germany just within the country. At least in the US we know what the length should be so if we don't have that we know the number is in error. Doug Having variable length offers several advantages, it just makes life harder for telcos (and telco-lookalikes like us ;) - Companies with several phones can have DDI, possibly multi-digit, but (as convention) the -0 always is the central phone desk. So if you do not know the number of the person, but just the company (which would be listed in the yellow pages), you call the -0 and ask to be transferred. For example if I need to speak to the University library, I just call 73-0 (local call), and they transfer to extensions 73-1234. I prefer this a lot over the system it seems the US as well as UK use a lot, where you always dial a fixed-length number, the call is connected (from which moment on you are charged for the call--- this was really expensive if international tariffs applied) and a recorded message asks you to dial a four-digit extension or 0 for assistance. I like our system better, as it allows for all those nice digital phone features like Callback-on-Called-Party-ends-call (if the remote side was on a call, so the busy indication works without additional charge), Callback-After-First-Call-Of-Remote-Party (where, if nobody was available, the network will notifiy after the other side got off-hooked and on-hooked again), and whatever there is. - And then, in small populated areas that got their own area code nonetheless, they handed out short numbers. Nowadays with ISDN lines (which give you three up to 10 numbers) being so popular, VoIP geographic numbers taking their share etc, the number room became full in those areas. So they just took free 3-digit codes and made 6-digit blocks out of them. No problem with digital switches on the telco side. Here in Bonn (area 228) policy was that every new landline gets one (analogue line) or three (ISDN) 7-digit numbers, and if you apply for up to 7 more numbers, which was free until 2005, you would get 8-digit numbers. Lots of 6-digit numbers are still in use (handed out until the late 90s), shorter numbers do not exist except for companies. How short a number they have (plus DDI digits) depends on what they pay for it and how many channels their trunk has. They will get 5-digit(+x) here for a reasonable price, 4-digit will be rather expensive, and 2-/3-digit are not available any more - those in use have been assigned in the 50ties when the phone system was built. So the University, the DoD, the D.of Foreign Affairs, the town hall, the police central station... caught them, and bad luck for all the others. Of course, Germany does not have something like area-split and similar hassles - at least not for technical reasons. AFAIK there have been a handful area code changes for political reasons during the last 50 years, and the eastern part of Germany got a completely new phone system after the Iron Curtain fell - quite foresight to keep the 3x area codes unused in the western phone system (except 30 which is Berlin - but that in fact is eastern Germany as well, obviously). I think the most recent area code changes were when the 0-800 numbers came, because some area had the code 8001 or so. European standards and their popularity in germany, ahem... :) BR Anselm ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How accurate is show translation?
On 12/23/06 09:51 Leo Ann Boon said the following: I would love to hear how others are using the results from show translation in system dimensioning. So far, I feel that dimensioning an Asterisk box is still mostly guesstimation :). Currently, I'm using the 30MHz per call rule to dimension. on a Pentium D 2.80Ghz, we've sustained 300 simultaneous IAX2 calls terminating in a dialplan loop that answers the call, waits 2 seconds, plays demo-instruct and loops again. a cursory examination revealed that a large portion of the CPU was used to handle NIC interrupts. occasionally we got a chan_iax2.so error which said, Maximum trunk data space exceeded to... this seems to be controlled by the MAX_TRUNKDATA constant in chan_iax2.c which is set to 40ms of SLIN for 200 calls. it'd be nice to know what this constant is for and what would the implications of increasing it be. [cc'ed to -dev as well] -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --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] How accurate is show translation?
Vicky wrote: I tried it on a intel 3 ghz p4 box and a athlon 3000 768 mb ram running vista and host for centos 4 ( vmware ) considering the load on athlon running asterisk ( that too under vista plus vmware ) while intel 3 ghz p4 1 GB ram box was sitting idle with centos , there was hardly a 1 ms difference in show translation on both machines . Besides i just compared my p4's results to ur D930 results and there is no difference ( infact my g729 results are better than ) .. But this doesnt mean both are same dual core cpu's will definitely give much higher number of channel transcoding then lower p4's . Put both the box under some cpu load by other programs and then use show translation recalc 30 and you will see performance difference between them ;) Vicky, The point of the exercise is that you should run 'show translation' with no load to get the baseline value. Your results confirmed my suspicion that the value is not tied to the number of CPUs - which indicates that the test was run on only 1 CPU. My concern is why the performance plateau. It makes no sense that a 3GHz CPU should take the same amount of time as a 1.3GHz CPU - that is unless there's something else is holding back the transcoder. It's like those graphics benchmarks - at some point, all the CPUs show the same FPS because the refresh rate is the one holding up the CPU. At this point, I don't feel that 'show translation' is a useful indicator of actual transcoding performance. It's OK for relative comparisons but utterly useless if you need the figures for sizing purposes. Leo ___ --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] Determining invalid extensions.
Phil Finkler wrote: Hi all, I’m trying to incorporate using the i extension in my callplan to determine if someone enters an invalid extension. My internal extensions are all 3 digits (100-104). The problem is, the callplan doesn’t see that say, extension 600 is invalid, it just goes back to the beginning of the callplan and repeats. If I enter a single digit, it works perfectly. Anyone have any ideas? Here’s the incoming callplan. It's because 600 will match _XX. Why don't you just use the 's' extension, instead of '_XX.'? Leo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: meetmejoin example
You have to join a conference using the dialplan. If you want the Manager Client to be able to make an existing call join a conference, set up an extension in the dialplan that does what you want, and then use the Manager Redirect command to transfer the channel to that extension. Many thanks, do you know if it is possible to join a conference from different channels with this method? (For example, 2 calls from SIP and one from H323) ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SNOM 200 behind NAT and other xmas woes
I decided to give the whole family IP phones for christmas, all hooked into my asterisk server, so all the nephews can have their own lines. However, one of the phones I got was the SNOM 200. That's worked fine for me on my own network, but I'm having bad luck getting it to work behind NAT talking to Asterisk. It talks to my termination/origination provider, which seems to ruthlessly ignore SDPs and send audio to the address it gets audio from, which works pretty well behind NAT. I've tried all the various NAT settings on the SNOM 200 (with the last firmware rev they made) but reports are that's broken. The SDPs and Contact headers it sends out are always the natted address, even if I tell it to use STUN or static or UPNP etc. If the nat traversal is broken not much I can do on that end. Asterisk, on the other hand, should be handling this with nat=yes on the channel, but it's not. It handles it for the SIP packets, responding to those on the address the requests came from (ignoring the contact header) but it seems to accept the SDP, which contains and address Asterisk can't see. The docs say nat=yes will fix addresses in SDPs. I'm running svn trunk from a few days ago. Is there a way to get Asterisk to send audio to the address the incoming audio comes from, or to take the SDP and replace the IP address in it with the IP address the SIP came from? Otherwise while the phone will be able to make PSTN calls, it is unable to call Asterisk for voicemail and the rest. -- Some other issues: Anybody tried to use vbuzzer with Asterisk with an IVR on the DID? I find when I do this, after the IVR connext to an extension, the reinvite Asterisk sends to vbuzzer is responded to by a very simplified 200 state response which Asterisk seems to get upset at. Asterisk immediately hangs up the channel, but nothing in the debug logs (even level 9) seems to say why. I will use another DID for now. Also, I bought an SPA-2000 on ebay that claimed to be unlocked. It isn't. I will probably get my money back but that's not going to help at christmas. Anybody know a way to unlock these things at the hardware level (shorting pins etc.) The factory reset codes all demand a PW, which means it's locked. Web access disabled too. I have a Cisco ATA 186 but they only do NAT traversal with a static address, no STUN etc. I may end up setting up the SNOM and replacing it if no other solution shows up. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: meetmejoin example
On 12/23/06, nik600 [EMAIL PROTECTED] wrote: You have to join a conference using the dialplan. If you want the Manager Client to be able to make an existing call join a conference, set up an extension in the dialplan that does what you want, and then use the Manager Redirect command to transfer the channel to that extension. And sorry, another question...how can i automatically do a redirect command when i recive a call? for example, my dialplan will be: [default] ; extension called from external user exten = 777,1,Dial(999) exten = 777,2,Dial(SIP/[EMAIL PROTECTED]) exten = 777,3,transfer ... exten = 777,4,Dial(SIP/[EMAIL PROTECTED]) exten = 777,5,transfer ... ; extension to join a conference exten = 999,1,Answer exten = 999,2,Meetme(100) Can i do something like that? ___ --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] SNOM 200 behind NAT and other xmas woes
On Sat, Dec 23, 2006 at 02:02:18AM -0800, Brad Templeton wrote: I've tried all the various NAT settings on the SNOM 200 (with the last firmware rev they made) but reports are that's broken. The SDPs and Contact headers it sends out are always the natted address, even if I tell it to use STUN or static or UPNP etc. If the nat traversal is broken not much I can do on that end. Well, I tried a bit more, including a numeric STUN server address and a reboot and that seemed to have helped. So now I'm down to the DTMF not working (I've tried both inband and rfc2833 and INFO settings but will try again.) And other issue. I have 5 line buttons set to 5 accounts. Incoming calls to these accounts light the right line button, and make the right ringtone. However, pressing the line buttons to make calls still results in all the calls coming from the first line's identity in the From: header. What's a good way to get the source line to identify on the SNOM 200? ___ --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] Help with SUSE 10.2 and Sangoma A104D
Hi I'm not familiar enough with Sangoma. I do hope I can slightly help in isolating the problem. On Fri, Dec 22, 2006 at 01:41:08AM -0200, Josué Conti wrote: Hi all, as good? I try to install asterisk-1.2.14, zaptel-1.2.12,libpri-1.2.4,addons-1.2.5 , sounds-1.2.1 and wanpipe-2.3.4-3 and hwec-utils-beta4-2.3.4 But it is not compiling drivers of the Sangoma, Here lies the problem. This is what you should debug. why udev's for board in /dev/zap(1-31, channel,ctl,pseudo,timer) is not created. But when I install a board TE110P Digium, udev's is created and asterisk functions perfectly. : ) As you see, there is an error building the drivers. Until that is fixed, no point trying to use zaptel. This mine motherboard, has a NIC Gbps and if I leave qualified, the A104D not recognized for motherboar, if I disable the NIC Gbps, through lspci, I identify the A104D. Not recognized == does not show up on lspci? that's a strange problem indeed. I tend to believe that it is unrelated. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma Wanpipe 2.3.4-3 compilation fails under FC2 with Zaptel 1.0.9.2
On Fri, Dec 22, 2006 at 09:58:22AM -0700, Colin Anderson wrote: LD [M] /root/wanpipe/patches/kdrivers/wanec/wanec.o Building modules, stage 2. MODPOST *** Warning: register_wanec_iface [/root/wanpipe/patches/kdrivers/wanec/wanec.ko] undefined! *** Warning: unregister_wanec_iface [/root/wanpipe/patches/kdrivers/wanec/wanec.ko] undefined! CC /root/wanpipe/patches/kdrivers/wanec/wanec.mod.o LD [M] /root/wanpipe/patches/kdrivers/wanec/wanec.ko make[1]: Leaving directory `/lib/modules/2.6.10-1.771_FC2smp/build' make -C /lib/modules/2.6.10-1.771_FC2smp/build SUBDIRS=/root/wanpipe/kdrvtmp CC=gcc KBUILD_VERBOSE=0 modules make[1]: Entering directory `/lib/modules/2.6.10-1.771_FC2smp/build' CC [M] /root/wanpipe/kdrvtmp/sdla_tdmv.o CC [M] /root/wanpipe/kdrvtmp/sdla_remora_tdmv.o /root/wanpipe/kdrvtmp/sdla_remora_tdmv.c: In function `wp_remora_zap_ioctl': /root/wanpipe/kdrvtmp/sdla_remora_tdmv.c:227: error: `ZT_SETPOLARITY' undeclared (first use in this function) Strange. What version of Zaptel do you build against? /root/wanpipe/kdrvtmp/sdla_remora_tdmv.c:227: error: (Each undeclared identifier is reported only once /root/wanpipe/kdrvtmp/sdla_remora_tdmv.c:227: error: for each function it appears in.) make[2]: *** [/root/wanpipe/kdrvtmp/sdla_remora_tdmv.o] Error 1 make[1]: *** [_module_/root/wanpipe/kdrvtmp] Error 2 make[1]: Leaving directory `/lib/modules/2.6.10-1.771_FC2smp/build' make: *** [all] Error 2 -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How accurate is show translation?
On Sat, Dec 23, 2006 at 09:51:24AM +0800, Leo Ann Boon wrote: Hi all, I'm using 'show translation' to help dimension my system, but I confused by the results I get. My 2 test systems (results below): an AthlonXP 2000+ (1.3GHz) and a Pentium D930 (duo-core, 3.0GHz) produced similar results (D930 is slightly faster). Googling shows that others have similar results running on other CPU speeds 2.0GHz. At first glance, it would look like the AthlonXP gives better bang for the buck :). But, I'm sure that are other reasons. I know show translation times how long it takes a convert 1s of full duplex audio. I suspect the test is using a single CPU (since it's in a single thread) and there are some constant overheads that makes a 3.0GHz produce the same numbers as a 1.3GHz. If you had just one call, then adding extra CPUs wouldn't have helped. 'show translations' mainly helps you compare different codecs. It is also handy as a benchmark because it's there. However I would love to hear how others are using the results from show translation in system dimensioning. So far, I feel that dimensioning an Asterisk box is still mostly guesstimation :). Currently, I'm using the 30MHz per call rule to dimension. There are some other factos. For instance, if you test relatively short calls (as someone else in this thread did), then the call set-up and tear-down overheads carry a larger wheight. It is also significant if you have an expensive dialplan (e.g: running an AGI for every call). -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UPDATE - Analog Phones with FSK/Stutter MWI
Doug Crompton wrote: In order for the external MWI to work you must turn on the message indicator and for units that have answering machines the machine must be turned off. Perhaps we could put together a list of analog phones that have this feature. I have been told that both Uniden and ATT have models that work but I have no knowledge of all that do in their entire line. I would suggest you start a page on the Wiki and send the link to the list. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How accurate is show translation?
On Sat, 23 Dec 2006, Leo Ann Boon wrote: At this point, I don't feel that 'show translation' is a useful indicator of actual transcoding performance. It's OK for relative comparisons but utterly useless if you need the figures for sizing purposes. Just to give you another (relative) comparison... This is from a VIA processor running at 533MHz, (64KB Cache) asterisk compiled as i586 as it's missing some of the nicer MMX instructions: g723 gsm ulaw alaw g726 adpcm slin lpc10 g729 speex ilbc g723 - - - - - - - - - - - gsm - -10103212 943 - - 174 ulaw -18 - 124 4 135 - - 166 alaw -18 1 -24 4 135 - - 166 g726 -362020 -221953 - - 184 adpcm -19 3 325 - 236 - - 167 slin -17 1 123 3 -34 - - 165 lpc10 -422626482825 - - - 190 g729 - - - - - - - - - - - speex - - - - - - - - - - - ilbc -49333355353266 - - - This board runs very well with a TDM400P card in it. On another Via processor, this time a C3 running at 1GHz with 128KB cache: g723 gsm ulaw alaw g726 adpcm slin lpc10 g729 speex ilbc g723 - - - - - - - - - - - gsm - - 4 412 4 320 - -72 ulaw - 6 - 110 2 118 - -70 alaw - 6 1 -10 2 118 - -70 g726 -13 9 9 - 9 825 - -77 adpcm - 6 2 210 - 118 - -70 slin - 5 1 1 9 1 -17 - -69 lpc10 -171313211312 - - -81 g729 - - - - - - - - - - - speex - - - - - - - - - - - ilbc -17131321131229 - - - I'm finding these mini ATX boards with VIA processes to be generally very good for small-office applications. Low power, fanless, boot from flash, etc. and seem to just soak up everything thrown at them - until you need to do an excessive amount of transcoding, but thats virtually non-existent in my applications. (Although I'm using G726 on some IAX trunks over office ADSL lines here in the UK. (outgoing b/w typically 256Kbps on older installations, just over 400 on ADSL-MAX installations. In theory we can have up to (just over) 800Kbps on the business-class -MAX installations, but very few people are prepared to pay for it!!! Gordon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] conditional dialplan
Hi
can i set up some conditions in my dialplan?
For example:
exten = 99,1,Answer
exten = 99,2, ... if {RECORD}=yes
then:
monitor...
Dial
else:
Dial.
Or something similar... ?
Many thanks in advance
nik
___
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[asterisk-users] Determining invalid extensions.
I used _XX. Since it was used in the examples I got from voicepulse. Maybe I can modify it so it's standardized by using 's'. Any idea why they'd use something like that for incoming calls? Are you sure 600 would match _XX.? I thought _XX. Was just two digits. Thanks for the help, Phil Phil Finkler wrote: Hi all, I'm trying to incorporate using the i extension in my callplan to determine if someone enters an invalid extension. My internal extensions are all 3 digits (100-104). The problem is, the callplan doesn't see that say, extension 600 is invalid, it just goes back to the beginning of the callplan and repeats. If I enter a single digit, it works perfectly. Anyone have any ideas? Here's the incoming callplan. It's because 600 will match _XX. Why don't you just use the 's' extension, instead of '_XX.'? Leo ___ --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] System Application with java
Are you sure you want to fire up a JVM each and every time you run this
command? that's a resource hog and will anyway cause a delay for system
class loading, etc. Maybe attaching to a resident process would be lighter.
k,
On Fri, 22 Dec 2006 13:48:27 +0100, Andre Gustavo Lomonaco
[EMAIL PROTECTED] wrote:
Hi,
I created a script named example2.sh which goal is read some text from
my HP Service Desk using an application in java and send this text to
the text2wave application for TTS.
example2.sh
java -Xbatch Example10 | text2wave -f 8000 -o
/var/lib/asterisk/sounds/my-sd.wav
When I execute the script in prompt, everything is ok, but when I use
the system() command in my
extensions.conf it isn´t work, just a small file my-sd.wav is created.
Here my extensions.conf configuration
;testing text2wav
exten = 666,1,Answer
exten = 666,2,system(/root/example2.sh /root/log.txt )
exten = 666,3,system(echo ${SYSTEMSTATUS} /root/log.txt)
exten = 666,4,wait(10)
exten = 666,5,Playback(my-sd)
exten = 666,6,Hangup
And here the logging by Asterisk..
Connected to Asterisk 1.2.13 currently running on fedora (pid = 1951)
Verbosity is at least 3
-- Remote UNIX connection
-- Executing Answer(SIP/lomonaco-0945fd18, ) in new stack
-- Executing System(SIP/lomonaco-0945fd18, /root/example2.sh
/root/log.txt ) in new stack
-- Executing System(SIP/lomonaco-0945fd18, echo SUCCESS
/root/log.txt) in new stack
-- Executing Wait(SIP/lomonaco-0945fd18, 10) in new stac
Any help or tip
Thanks in Advanced
Andre Lomonaco
--
Loway Research - Home of QueueMetrics
http://queuemetrics.loway.it
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Re: [asterisk-users] conditional dialplan
nik600 wrote: Hi can i set up some conditions in my dialplan? Yes, look at the following command: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+GotoIf Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: meetmejoin example
In article [EMAIL PROTECTED], nik600 [EMAIL PROTECTED] wrote: You have to join a conference using the dialplan. If you want the Manager Client to be able to make an existing call join a conference, set up an extension in the dialplan that does what you want, and then use the Manager Redirect command to transfer the channel to that extension. Many thanks, do you know if it is possible to join a conference from different channels with this method? (For example, 2 calls from SIP and one from H323) ? Yes, any kind of channel that is executing in the dialplan can join a conference by calling Meetme with the appropriate parameters. Each channel needs to do this separately. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need quality toll free 800 number over IAX?
Vicky [EMAIL PROTECTED] wrote: I have 3 toll free did's with nufone since 1 month .. Until now i dont have a problem with them .. their portal was good enough to do proper configuration and call quality wasnt bad ( even though i havent used them in really huge traffic yet ) . The reason for the warning is to get people to move before they start to have problems. I'm sure everything will work for a while (perhaps for years), but when you have a problem - perhaps an sudden deterioration or complete outage at NuFone itself, you'll find out just how difficult it is to get a helpful, or even just polite, response from their support department. You might find that you have no service through no fault of your own, and have no way to fix it yourself. Usually a threat of fix it or I'll take my business elsewhere tends to work, but if you do follow up on that then NuFone will rob you of any outstanding call credit you have in your account. You'll find lots of examples of this if you search around. I ignored the warnings because the service was working at the time, and I didn't think I'd ever need to contact their support department. Don't make the same mistake. Here are a few references I found with a five-minute Google search. There are lots more where these came from: https://forum.voxilla.com/other-providers/nufone-severe-problem-service-9548.html http://www.voip-info.org/wiki/view/Nufone http://www.shakataganai.com/index.php?/archives/163-NuFone.html http://hansgrueber.blogspot.com/2006/01/nufone-sucks.html There are loads of examples in the various mail list archives. Here are a couple: http://lists.digium.com/pipermail/asterisk-biz/2004-December/001468.html http://lists.digium.com/pipermail/asterisk-biz/2004-December/001457.html As you have only had the DDIs for a month, you could get away with simply going elsewhere and using new numbers (use up your call credit first). If you have vanity numbers, or you want to keep them for some other reason then I wish you good luck getting NuFone to help you port them to your new provider. The bottom line is that you are free to use whichever provider(s) you like, but please be aware of the high likelyhood of having severe service difficulties with NuFone. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Determining invalid extensions.
Leo Ann Boon wrote: Phil Finkler wrote: Hi all, I’m trying to incorporate using the i extension in my callplan to determine if someone enters an invalid extension. My internal extensions are all 3 digits (100-104). The problem is, the callplan doesn’t see that say, extension 600 is invalid, it just goes back to the beginning of the callplan and repeats. If I enter a single digit, it works perfectly. Anyone have any ideas? Here’s the incoming callplan. It's because 600 will match _XX. Why don't you just use the 's' extension, instead of '_XX.'? Because the s extension is only matched when there is NO dialed number. Extension i is designed for use within an IVR. We use 4 digit extensions and use exten = _,1,Whatever to match invalid extensions. ___ --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] UPDATE - Analog Phones with FSK/Stutter MWI
Doug, Thanks for the info. I'm glad it works. One question: Is there some sort of one-button way to dial in to your voicemail? It seems I read something about it, when I was doing similar research? I think it was the Uniden CLX-465, which claims support of Phone Company voicemail. I could not find one locally, however. Happy Holidays Bob... Doug Crompton wrote: After purchasing the Uniden TRU9480 and then the Panansonic 5672, both of which do not have phone company compatible FSK/stutter MWI, I finally got smart and found out just which Panasonic phones have this feature. Only the following 5.8G models in their current line have FXO compatible MWI. I purchased the 5771 unit and one remote. I have confimed it does in fact work with Asterisk and my SPA-3000. When there is a message waiting both the LCD display and a flashing indicator on the phone alert you. This is true for all extensions on the system, up to 8. These work with both FSK and Stutter tone. I did not turn on the tone MWI as the FSK worked fine. KX-TG5776S 5.8 GHz FHSS GigaRange. Expandable Digital Cordless Answering System with a 1.5 Full-Color (65k Color Capable) Backlit LCD on Handset $119.95 KX-TG5771S 5.8 GHz FHSS GigaRange. Expandable Digital Cordless Answering System with Talking Caller ID $99.95 KX-TG5622M 5.8 GHz FHSS GigaRange Dual-Handset Phone System $89.95 KX-TG5761S 5.8 GHz FHSS GigaRange Expandable Digital Cordless Phone with Talking Caller ID $89.95 In order for the external MWI to work you must turn on the message indicator and for units that have answering machines the machine must be turned off. Perhaps we could put together a list of analog phones that have this feature. I have been told that both Uniden and ATT have models that work but I have no knowledge of all that do in their entire line. Each brand has their own features and while the Panasonic is solid - I had a 2.4G system for years and really liked it - the Unidens seems to have more for the money but in this case not MWI. I guess you could tell I really wanted this MWI to work! Doug ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: problems using the 1.4 version of meetme
The nearest I can do is I have a Linksys 3102 and its also set to inband and when I call that extension fromm outside using asterisk 1.4 I can hear the dtmf just fine in my ear -- works about the same as using 1.2. The only app so far which is not working is meetme not detecting the * in asterisk 1.4 using sip and this is what I need help with. Is this something I should file a bug about? on Friday 12/22/2006 Tony Mountifield([EMAIL PROTECTED]) wrote In article [EMAIL PROTECTED], John covici [EMAIL PROTECTED] wrote: The 1.4 and 1.2 are alternately on the same box -- I have to install 1.2 rmmod the zaptel modules delete /usr/lib/asterisk/modules/* and install the zaptel and asterisk. I use the same procedure to go back to the 1.4. I know the ivr works because I have to use a ivr menu and even enter a password to get to the meetme conference and those work fine. The sip provider is using inband as I have requested. Also, I tried calling through the sip provider to my local extension and I hear the tones just fine, so its a mystery to me. Do you have a SIP phone you can register directly with the box and try? If so, try setting it to the three different ways of sending DTMF and see whether any of them work. Just trying to whittle down the possibilities to start with You may find it better to use out-of-band DTMF with SIP. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CLI Errors and warnings
Hi all, I am getting the following popping up in my asterisk CLI. Everything seems to working ok, but I'm curious as to what exactly these messages mean: Dec 23 12:28:44 NOTICE[9858]: chan_sip.c:11123 handle_request_subscribe: Got SUBSCRIBE for extension [EMAIL PROTECTED] from 192.168.1.104, but there is no hint for that extension Thanks for any help. -- Warm Regards, Lee ___ --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] CLI Errors and warnings
Lee Jenkins wrote: Dec 23 12:28:44 NOTICE[9858]: chan_sip.c:11123 handle_request_subscribe: Got SUBSCRIBE for extension [EMAIL PROTECTED] from 192.168.1.104, but there is no hint for that If I'm remembering correctly, it's a message you'd get if you had a Polycom phone (Other as well maybe?) setup with buddy lists. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CLI Errors and warnings
Doug Lytle wrote: Lee Jenkins wrote: Dec 23 12:28:44 NOTICE[9858]: chan_sip.c:11123 handle_request_subscribe: Got SUBSCRIBE for extension [EMAIL PROTECTED] from 192.168.1.104, but there is no hint for that If I'm remembering correctly, it's a message you'd get if you had a Polycom phone (Other as well maybe?) setup with buddy lists. Doug Thanks for responding, Doug. That extension is a Budgetone 100 and after looking in the config utility, I didn't notice anything about buddy lists. -- Warm Regards, Lee ___ --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] Need quality toll free 800 number over IAX?
Folks, with all due respect: this thread is now wy off topic, as it has nothing to do with Asterisk whatsoever. Please take it offline, or to ~biz. thx. B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UPDATE - Analog Phones with FSK/Stutter MWI
Not that I know of. I guess you could speed dial but then my Asterisk voicemail is 80 so how hard is it to pick up the phone and dial that. I never had phone company voicemail on a wired line so I don't know how that works but I suspect you have to dial your own 7 digit or 10 digit number??? Doug On Sat, 23 Dec 2006, Bob Chiodini wrote: Doug, Thanks for the info. I'm glad it works. One question: Is there some sort of one-button way to dial in to your voicemail? It seems I read something about it, when I was doing similar research? I think it was the Uniden CLX-465, which claims support of Phone Company voicemail. I could not find one locally, however. Happy Holidays Bob... Doug Crompton wrote: After purchasing the Uniden TRU9480 and then the Panansonic 5672, both of which do not have phone company compatible FSK/stutter MWI, I finally got smart and found out just which Panasonic phones have this feature. Only the following 5.8G models in their current line have FXO compatible MWI. I purchased the 5771 unit and one remote. I have confimed it does in fact work with Asterisk and my SPA-3000. When there is a message waiting both the LCD display and a flashing indicator on the phone alert you. This is true for all extensions on the system, up to 8. These work with both FSK and Stutter tone. I did not turn on the tone MWI as the FSK worked fine. KX-TG5776S 5.8 GHz FHSS GigaRange. Expandable Digital Cordless Answering System with a 1.5 Full-Color (65k Color Capable) Backlit LCD on Handset $119.95 KX-TG5771S 5.8 GHz FHSS GigaRange. Expandable Digital Cordless Answering System with Talking Caller ID $99.95 KX-TG5622M 5.8 GHz FHSS GigaRange Dual-Handset Phone System $89.95 KX-TG5761S 5.8 GHz FHSS GigaRange Expandable Digital Cordless Phone with Talking Caller ID $89.95 In order for the external MWI to work you must turn on the message indicator and for units that have answering machines the machine must be turned off. Perhaps we could put together a list of analog phones that have this feature. I have been told that both Uniden and ATT have models that work but I have no knowledge of all that do in their entire line. Each brand has their own features and while the Panasonic is solid - I had a 2.4G system for years and really liked it - the Unidens seems to have more for the money but in this case not MWI. I guess you could tell I really wanted this MWI to work! Doug ___ --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 Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UPDATE - Analog Phones with FSK/Stutter MWI
On Sat, Dec 23, 2006 at 04:05:08PM -0500, Doug Crompton wrote: Not that I know of. I guess you could speed dial but then my Asterisk voicemail is 80 so how hard is it to pick up the phone and dial that. I never had phone company voicemail on a wired line so I don't know how that works but I suspect you have to dial your own 7 digit or 10 digit number??? In the UK, you dial 1571. -- Phil Reynolds o mail: [EMAIL PROTECTED] |L_ \ / Web: http://www.tinsleyviaduct.com/phil/ (_)- \/ Waltham 66, Emley Moor 69, Droitwich 79, Windows 95 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] The Good, Bad and Scam VoIP Providers
Brian Capouch I changed the subject I don't think it was right for this message!! // Re: [asterisk-users] Need quality toll free 800 number over IAX? Well I don't agree with you about this thread they are talking about the good and the bad VoIP providers This is information that Asterisk users MUST KNOW. We have to put the SCAMMERS like trxtel.com out of business (That don't pay there users) The BAD VoIP providers must try to get there servers and customer service right or they need to go way. Keep in mind list THE VOIP PROVIDERS and VOIP SUPPLIERS NEED US. SO the providers and suppliers need to get there acts together. The BAD and the SCAMMERS are giving VoIP a bad name and pushing OUR NEW PBX CUSTOMERS TO VONAGE. Is this what the list wants I DON'T THINK SO Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email (VoIP PBX) 1-563-773-6610 EXT: 250 -- For Information on PBX Systems for SOHO http://www.bochterservices.com/?j=PBXt=email -- Need A Toll Free Number? http://www.bochterservices.com/?t=TFdidt=email -- Need Voice Mail? http://www.bochterservices.com/?t=VMSt=email --For new and used security items http://www.bochterservices.com/?j=storet=email --BUY Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email Brian Capouch wrote: Folks, with all due respect: this thread is now wy off topic, as it has nothing to do with Asterisk whatsoever. Please take it offline, or to ~biz. thx. B. ___ --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] The Good, Bad and Scam VoIP Providers
On Sat, Dec 23, 2006 at 05:30:54PM -0500, Al Bochter wrote: Keep in mind list THE VOIP PROVIDERS and VOIP SUPPLIERS NEED US. SO the providers and suppliers need to get there acts together. The BAD and the SCAMMERS are giving VoIP a bad name and pushing OUR NEW PBX CUSTOMERS TO VONAGE. Is this what the list wants I DON'T THINK SO Best regards, Al Bochter Bochter Services And the fact that you offer a competing service naturally has nothing to do with that. So please keep your tone down and stay on-topic. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help with SUSE 10.2 and Sangoma A104D
Hello Tzafrir, all good? It was really a strange and curious problem, but the fact is that I entered in contact with the support technician of the Sangoma and Mr. Alex and Mr. Yuan of the Sangoma, had passed me a new package of the Wanpipe ( wanpipe-2.3.4-2.1) that it compiled normally in SUSE 10.2; I noticed that referring to /dev/zap, it only appears after the packages of wanpipe to have been compiled successfully. Still I am with some problems of synchronism with the A104D, but that now it is more easy of if resolv. Tzafrir thank you will be this attention and would like to desire for you and your family a happy Christmas and prosperous a 2007, full of peace, love and much personal and professional success. :) Best Regards Josue 2006/12/23, Tzafrir Cohen [EMAIL PROTECTED]: Hi I'm not familiar enough with Sangoma. I do hope I can slightly help in isolating the problem. On Fri, Dec 22, 2006 at 01:41:08AM -0200, Josué Conti wrote: Hi all, as good? I try to install asterisk-1.2.14, zaptel-1.2.12,libpri-1.2.4, addons-1.2.5 , sounds-1.2.1 and wanpipe-2.3.4-3 and hwec-utils-beta4-2.3.4 But it is not compiling drivers of the Sangoma, Here lies the problem. This is what you should debug. why udev's for board in /dev/zap(1-31, channel,ctl,pseudo,timer) is not created. But when I install a board TE110P Digium, udev's is created and asterisk functions perfectly. : ) As you see, there is an error building the drivers. Until that is fixed, no point trying to use zaptel. This mine motherboard, has a NIC Gbps and if I leave qualified, the A104D not recognized for motherboar, if I disable the NIC Gbps, through lspci, I identify the A104D. Not recognized == does not show up on lspci? that's a strange problem indeed. I tend to believe that it is unrelated. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma Wanpipe 2.3.4-3 compilation fails under FC2 with Zaptel 1.0.9.2
Hello Colin: Please try ftp://ftp.sangoma.com/linux/custom/Yuan/wanpipe-2.3.4-2.1.gz Best Regards Josue 2006/12/23, Tzafrir Cohen [EMAIL PROTECTED]: On Fri, Dec 22, 2006 at 09:58:22AM -0700, Colin Anderson wrote: LD [M] /root/wanpipe/patches/kdrivers/wanec/wanec.o Building modules, stage 2. MODPOST *** Warning: register_wanec_iface [/root/wanpipe/patches/kdrivers/wanec/wanec.ko] undefined! *** Warning: unregister_wanec_iface [/root/wanpipe/patches/kdrivers/wanec/wanec.ko] undefined! CC /root/wanpipe/patches/kdrivers/wanec/wanec.mod.o LD [M] /root/wanpipe/patches/kdrivers/wanec/wanec.ko make[1]: Leaving directory `/lib/modules/2.6.10-1.771_FC2smp/build' make -C /lib/modules/2.6.10-1.771_FC2smp/buildSUBDIRS=/root/wanpipe/kdrvtmp CC=gcc KBUILD_VERBOSE=0 modules make[1]: Entering directory `/lib/modules/2.6.10-1.771_FC2smp/build' CC [M] /root/wanpipe/kdrvtmp/sdla_tdmv.o CC [M] /root/wanpipe/kdrvtmp/sdla_remora_tdmv.o /root/wanpipe/kdrvtmp/sdla_remora_tdmv.c: In function `wp_remora_zap_ioctl': /root/wanpipe/kdrvtmp/sdla_remora_tdmv.c:227: error: `ZT_SETPOLARITY' undeclared (first use in this function) Strange. What version of Zaptel do you build against? /root/wanpipe/kdrvtmp/sdla_remora_tdmv.c:227: error: (Each undeclared identifier is reported only once /root/wanpipe/kdrvtmp/sdla_remora_tdmv.c:227: error: for each function it appears in.) make[2]: *** [/root/wanpipe/kdrvtmp/sdla_remora_tdmv.o] Error 1 make[1]: *** [_module_/root/wanpipe/kdrvtmp] Error 2 make[1]: Leaving directory `/lib/modules/2.6.10-1.771_FC2smp/build' make: *** [all] Error 2 -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The Good, Bad and Scam VoIP Providers
I seriously doubt trxtel.com scams anyone. I may be wrong but the person behind it has been with this community for a long time and has done nothing post insightful and meaningful things to this list and give back to the community in many other ways as well. It is a unique idea but that is really all I know about it (the service). I fire customers all the time. I would probably fire you if you were my customer based on the way you are ranting. In these cases, the drain is not worth it personally or for the business so bye bye. Bottom line, you get what you pay for. Check out a provider, try their customer service, see if there is a toll free number, call it and see if someone picks up. Try it over and over. Use whois to see how long they have been around, ask questions, and use common sense. It is called due diligence. As for me, I use Asterisk in a very LARGE (although everything is relative) deployment but I use no VoIP providers. I terminate to a T3 (28 T1s), all PSTN ULAW. The only VoIP that we do is INSIDE ONE DATA RACK and is traditional telephony one form or another outside of that rack. /This is information that Asterisk users MUST KNOW. /is simply not true. Expand your horizons, expand your vision. Do not automatically assume that everyone using Asterisk is using a VoIP provider. Post to the biz list where this belongs. Thanks, Steve Al Bochter wrote: Brian Capouch I changed the subject I don't think it was right for this message!! // Re: [asterisk-users] Need quality toll free 800 number over IAX? Well I don't agree with you about this thread they are talking about the good and the bad VoIP providers This is information that Asterisk users MUST KNOW. We have to put the SCAMMERS like trxtel.com out of business (That don't pay there users) The BAD VoIP providers must try to get there servers and customer service right or they need to go way. Keep in mind list THE VOIP PROVIDERS and VOIP SUPPLIERS NEED US. SO the providers and suppliers need to get there acts together. The BAD and the SCAMMERS are giving VoIP a bad name and pushing OUR NEW PBX CUSTOMERS TO VONAGE. Is this what the list wants I DON'T THINK SO Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email (VoIP PBX) 1-563-773-6610 EXT: 250 -- For Information on PBX Systems for SOHO http://www.bochterservices.com/?j=PBXt=email -- Need A Toll Free Number? http://www.bochterservices.com/?t=TFdidt=email -- Need Voice Mail? http://www.bochterservices.com/?t=VMSt=email --For new and used security items http://www.bochterservices.com/?j=storet=email --BUY Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email Brian Capouch wrote: Folks, with all due respect: this thread is now wy off topic, as it has nothing to do with Asterisk whatsoever. Please take it offline, or to ~biz. thx. B. ___ --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] The Good, Bad and Scam VoIP Providers
Tzafrir Cohen, Well if you would have asked I don't aim to sell service to VoIP users. I BUY VOIP TRUNK SERVICE from VoIP Providers. I BUY VOIP DEVICES from suppliers I install Asterisk PBX Servers and point the my customers to VoIP Providers and Suppliers So the fact is I don't offer a competing service. I sell the services to the END USER. Like a supplier said to me once. I will take care of a contractor before I return a call to an End User. The End user is only one sale and alot of time. The happy contractor's are 100's of sales The other way to look at this is the contractor / installer is 100's of end users So what I stated has everything to do with that. If I point a client to a BAD VoIP provider or supplier that make me look bad. And I could lose sales So Trafrir what do you do? I looked at your site it look like you would be a VoIP SUPPLIER? So you are a supplier competing for sales from myself and others on the list? If you would make a note I changed the Subject line I started a new Topic. So I am on-topic. Bad service is a big deal so the tone should be VERY LOUD. Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email (VoIP PBX) 1-563-773-6610 EXT: 250 -- For Information on PBX Systems for SOHO http://www.bochterservices.com/?j=PBXt=email -- Need A Toll Free Number? http://www.bochterservices.com/?t=TFdidt=email -- Need Voice Mail? http://www.bochterservices.com/?t=VMSt=email --For new and used security items http://www.bochterservices.com/?j=storet=email --BUY Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email Tzafrir Cohen wrote: On Sat, Dec 23, 2006 at 05:30:54PM -0500, Al Bochter wrote: Keep in mind list THE VOIP PROVIDERS and VOIP SUPPLIERS NEED US. SO the providers and suppliers need to get there acts together. The BAD and the SCAMMERS are giving VoIP a bad name and pushing OUR NEW PBX CUSTOMERS TO VONAGE. Is this what the list wants I DON'T THINK SO Best regards, Al Bochter Bochter Services And the fact that you offer a competing service naturally has nothing to do with that. So please keep your tone down and stay on-topic. ___ --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] The Good, Bad and Scam VoIP Providers
Steve Totaro, I will contract you off the list about trxtel that is not my base point of this. // Bottom line, you get what you pay for. I agree. // Check out a provider, try their customer service, see if there is a toll free number, call it and see if someone picks up. You forgot word of others that used the server. // Use whois to see how long they have been around, ask questions, and use common sense. It is called due diligence. Whois.org is not going to tell you much about them. Ask questions? HM MM is that not what I am stating here?? And have others tell you how the provider was to them. Sorry you trying to shoot me down on that point. / / This is information that Asterisk users MUST KNOW. /is simply not true. // Expand your horizons, expand your vision. Do not automatically assume that everyone using Asterisk is using a VoIP provider. So are you stating that if the provider (ANY POT or OTHERS) gave you bad service you would stay with them and not tell anyone. // Post to the biz list where this belongs. What am I trying to sell??? This is end user stuff Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email (VoIP PBX) 1-563-773-6610 EXT: 250 -- For Information on PBX Systems for SOHO http://www.bochterservices.com/?j=PBXt=email -- Need A Toll Free Number? http://www.bochterservices.com/?t=TFdidt=email -- Need Voice Mail? http://www.bochterservices.com/?t=VMSt=email --For new and used security items http://www.bochterservices.com/?j=storet=email --BUY Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email Steve Totaro wrote: I seriously doubt trxtel.com scams anyone. I may be wrong but the person behind it has been with this community for a long time and has done nothing post insightful and meaningful things to this list and give back to the community in many other ways as well. It is a unique idea but that is really all I know about it (the service). I fire customers all the time. I would probably fire you if you were my customer based on the way you are ranting. In these cases, the drain is not worth it personally or for the business so bye bye. Bottom line, you get what you pay for. Check out a provider, try their customer service, see if there is a toll free number, call it and see if someone picks up. Try it over and over. Use whois to see how long they have been around, ask questions, and use common sense. It is called due diligence. As for me, I use Asterisk in a very LARGE (although everything is relative) deployment but I use no VoIP providers. I terminate to a T3 (28 T1s), all PSTN ULAW. The only VoIP that we do is INSIDE ONE DATA RACK and is traditional telephony one form or another outside of that rack. /This is information that Asterisk users MUST KNOW. /is simply not true. Expand your horizons, expand your vision. Do not automatically assume that everyone using Asterisk is using a VoIP provider. Post to the biz list where this belongs. Thanks, Steve Al Bochter wrote: Brian Capouch I changed the subject I don't think it was right for this message!! // Re: [asterisk-users] Need quality toll free 800 number over IAX? Well I don't agree with you about this thread they are talking about the good and the bad VoIP providers This is information that Asterisk users MUST KNOW. We have to put the SCAMMERS like trxtel.com out of business (That don't pay there users) The BAD VoIP providers must try to get there servers and customer service right or they need to go way. Keep in mind list THE VOIP PROVIDERS and VOIP SUPPLIERS NEED US. SO the providers and suppliers need to get there acts together. The BAD and the SCAMMERS are giving VoIP a bad name and pushing OUR NEW PBX CUSTOMERS TO VONAGE. Is this what the list wants I DON'T THINK SO Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email (VoIP PBX) 1-563-773-6610 EXT: 250 -- For Information on PBX Systems for SOHO http://www.bochterservices.com/?j=PBXt=email -- Need A Toll Free Number? http://www.bochterservices.com/?t=TFdidt=email -- Need Voice Mail? http://www.bochterservices.com/?t=VMSt=email --For new and used security items http://www.bochterservices.com/?j=storet=email --BUY Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email Brian Capouch wrote: Folks, with all due respect: this thread is now wy off topic, as it has nothing to do with Asterisk whatsoever. Please take it offline, or to ~biz. thx. B. ___ --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 Inbound (clean). Database: 0662-0, 12/22/2006 - 12/23/2006 6:27:07 PM ___ --Bandwidth and Colocation
[asterisk-users] mySQL and to many connections with SQL statement UPDATE
Hi,
If Iam doing UPDATE SQL statements I got an overload for connection.
am doing everytime an Disconnect ${connid}) but this is ignored.
any idea?
best regards
Thomas
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Re: [asterisk-users] Determining invalid extensions.
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Phil Finkler wrote: I used _XX. Since it was used in the examples I got from voicepulse. Maybe I can modify it so it's standardized by using 's'. Any idea why they'd use something like that for incoming calls? Are you sure 600 would match _XX.? I thought _XX. Was just two digits. The fullstop will match one or more digits: http://www.voip-info.org/wiki-Asterisk+Dialplan+Patterns - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://wap.sineapps.com (Daily Asterisk News for your cellphone) http://feeds.feedburner.com/AsteriskNews (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFFjdOvS6d5vy0jeVcRAmFHAJ0XR/VxhpfbM8Kc5ph915JLxCee5gCgjzit 1AL8DrT2EOPpSlVJiHQhpx0= =Nost -END PGP SIGNATURE- ___ --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] Sangoma Wanpipe 2.3.4-3 compilation fails under FC2 with Zaptel 1.0.9.2
On Sat, Dec 23, 2006 at 02:14:14PM +0200, Tzafrir Cohen wrote: On Fri, Dec 22, 2006 at 09:58:22AM -0700, Colin Anderson wrote: LD [M] /root/wanpipe/patches/kdrivers/wanec/wanec.o Building modules, stage 2. MODPOST *** Warning: register_wanec_iface [/root/wanpipe/patches/kdrivers/wanec/wanec.ko] undefined! *** Warning: unregister_wanec_iface [/root/wanpipe/patches/kdrivers/wanec/wanec.ko] undefined! CC /root/wanpipe/patches/kdrivers/wanec/wanec.mod.o LD [M] /root/wanpipe/patches/kdrivers/wanec/wanec.ko make[1]: Leaving directory `/lib/modules/2.6.10-1.771_FC2smp/build' make -C /lib/modules/2.6.10-1.771_FC2smp/build SUBDIRS=/root/wanpipe/kdrvtmp CC=gcc KBUILD_VERBOSE=0 modules make[1]: Entering directory `/lib/modules/2.6.10-1.771_FC2smp/build' CC [M] /root/wanpipe/kdrvtmp/sdla_tdmv.o CC [M] /root/wanpipe/kdrvtmp/sdla_remora_tdmv.o /root/wanpipe/kdrvtmp/sdla_remora_tdmv.c: In function `wp_remora_zap_ioctl': /root/wanpipe/kdrvtmp/sdla_remora_tdmv.c:227: error: `ZT_SETPOLARITY' undeclared (first use in this function) Strange. What version of Zaptel do you build against? My guess was that this was due to building vs. zaptel 1.0 headers. When comparing zaptel.h from 1.0 and 1.2, one of the addition turns out to be: diff -u (wget -O- http://svn.digium.com/svn/zaptel/branches/1.0/zaptel.h) \ (wget -O- http://svn.digium.com/svn/zaptel/tags/1.2.0/zaptel.h) \ | less /* * Set polarity -- implemented by individual driver. 0 = forward, 1 = reverse */ #defineZT_SETPOLARITY _IOW (ZT_CODE, 92, int) -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The Good, Bad and Scam VoIP Providers
Geez Al, let it go. We've heard your rants for what seems like years now (even though it's only been weeks). No one cares anymore. ___ --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] The Good, Bad and Scam VoIP Providers
You guys are missing the point of the message I sent! Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email (VoIP PBX) 1-563-773-6610 EXT: 250 Bill Hackensack wrote: Geez Al, let it go. We've heard your rants for what seems like years now (even though it's only been weeks). No one cares anymore. ___ --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 Inbound (clean). Database: 0662-0, 12/22/2006 - 12/23/2006 9:13:10 PM ___ --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] How accurate is show translation?
Tzafrir Cohen wrote: If you had just one call, then adding extra CPUs wouldn't have helped. 'show translations' mainly helps you compare different codecs. It is also handy as a benchmark because it's there. However I agree with you that with 1 call, more CPU won't help. I'm just surprised that a 3GHz CPU is not much faster than a 1.3GHz CPU. I'm actually trying to find an analytical model to dimension an asterisk box. I need to transcode 120 channels of IAX (speex) into g711 to fed into 4xE1. My current guesstimation is a single Intel D930 should be up to the job. Without hard numbers, it's not very convincing. This is one aspect of asterisk that's annoying - you can't size a system reliably without resorting to lots of empirical testing. IMHO, this usually leads to over-engineering which drives up the cost. Regards and happy holidays. Leo 'In God we trust, others must have numbers.' ___ --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] sip help for newbie
You have to ask what they have open on the box on thier firewall. A good way to learn asterisk is to get a p3 and play at home. - Original Message - From: blackwater dev To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Wednesday, December 20, 2006 11:25 AM Subject: Re: [asterisk-users] sip help for newbie I'm not sure. I'm a linux newb and this is just running on a server I have hosted somewhere. I do have control of the box, just not sure what's open or how to open them. On 12/13/06, Dovid B [EMAIL PROTECTED] wrote: You need port 5060 as well as 1-2 UDP open to the server. Also is the server behind NAT at all ? - Original Message - From: blackwater dev To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Wednesday, December 13, 2006 5:14 AM Subject: Re: [asterisk-users] sip help for newbie Thanks for the info, I've gone through the tutorial and followed it and asterisk is running but I just can't seem to log in. The xten phone just tells me connection timed out. I'm simply running asterisk on a webserver that is also running apache and service content. I simply pinged the box to get the ip to plug into the softphone. Do I need to open a port or something? On 12/12/06, Forrest Beck [EMAIL PROTECTED] wrote: www.asteriskguru.com On 12/12/06, blackwater dev [EMAIL PROTECTED] wrote: Does anyone know of any good step by step tutorials on getting sip set up? I have asterisk installed but can't seem to figure out how to get an account set up and connect from my xTen phone so I can try the demo. The tutorials I read online seem to go into voicepulse stuff and all and I don't have an account there so am a bit lost. Thanks! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk + door opener
Never got the financial backing for it. I looked in to mass producing it in China and I was in way over my head. I am currently working with an investor. I will let you guys know if anything comes up. Dovid - Original Message - From: Jerry [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, December 20, 2006 11:30 PM Subject: Re: [Asterisk-Users] asterisk + door opener Hi Dovid, I am actually now working on massproducing door openers that will work with asterisk. It will have an rj45 port and then a port to plug the door opener in to. Please contact me off list if you are interested. This is an old message, but I was wondering if you are still doing this, and what the specs/cost are. Thanks, J. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] How accurate is show translation?
Just to give you another (relative) comparison... This is from a VIA processor running at 533MHz, (64KB Cache) asterisk compiled as i586 as it's missing some of the nicer MMX instructions: g723 gsm ulaw alaw g726 adpcm slin lpc10 g729 speex ilbc g723 - - - - - - - - - - - gsm - -10103212 943 - - 174 ulaw -18 - 124 4 135 - - 166 alaw -18 1 -24 4 135 - - 166 g726 -362020 -221953 - - 184 adpcm -19 3 325 - 236 - - 167 slin -17 1 123 3 -34 - - 165 lpc10 -422626482825 - - - 190 g729 - - - - - - - - - - - speex - - - - - - - - - - - ilbc -49333355353266 - - - Hey... why does your asterisk allow translation to and from ilbc? This is mine (Dual Celeron 533's): Translation times between formats (in milliseconds) Source Format (Rows) Destination Format(Columns) g723 gsm ulaw alaw g726 adpcm slin lpc10 g729 speex ilbc g723 - - - - - - - - - - - gsm - - 7 713 7 61739 100 - ulaw -17 - 1 8 2 1123495 - alaw -17 1 - 8 2 1123495 - g726 -23 8 8 - 8 71840 101 - adpcm -17 2 2 8 - 1123495 - slin -16 1 1 7 1 -113394 - lpc10 -24 9 915 9 8 -41 102 - g729 -23 8 814 8 718 - 101 - speex -2712121812112244 - - ilbc - - - - - - - - - - - And it won't translation ilbc to/from anything else. I had to buy some g729 licenses (a better option anyway though) because my voip provider only does alaw, g729, and ilbc, and I prefer to use alaw internally... James ___ --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] Happy X-mas
Hello everybody HAPPY and Merry Christmas to all. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Happy X-mas
Ditto, Happy Holidays everyone! On 12/23/06, Carlos Rojas [EMAIL PROTECTED] wrote: Hello everybody HAPPY and Merry Christmas to all. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mitchel Constantin Snap - A desktop user interface for Asterisk www.snapanumber.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] centos4.4 x86_64 and zaptel-1.2.12 compile problems?
Anyone seen this and know how to fix it? (note the Assembler messages at
the end). Thanks in advance:
server# make linux26
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -m64 -DSTANDALONE_ZAPATA
-DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DHOTPLUG_FIRMWARE -c -o
gendigits.o gendigits.c
cc -o gendigits gendigits.o -lm
./gendigits tones.h
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -m64 -DSTANDALONE_ZAPATA
-DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DHOTPLUG_FIRMWAREmakefw.c -o
makefw
./makefw tormenta2.rbt tor2fw tor2fw.h
Loaded 69900 bytes from file
./makefw pciradio.rbt radfw radfw.h
Loaded 42096 bytes from file
ZAPTELVERSION=1.2.12 build_tools/make_version_h version.h.tmp
if cmp -s version.h.tmp version.h ; then echo; else \
mv version.h.tmp version.h ; \
fi
rm -f version.h.tmp
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -m64 -DSTANDALONE_ZAPATA
-DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DHOTPLUG_FIRMWARE -c -o ztcfg.o
ztcfg.c
cc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE -m64
-DBUILDING_TONEZONE -o zonedata.lo zonedata.c
cc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE -m64
-DBUILDING_TONEZONE -o tonezone.lo tonezone.c
ar rcs libtonezone.a zonedata.lo tonezone.lo
cc -o ztcfg ztcfg.o libtonezone.a -lm
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -m64 -DSTANDALONE_ZAPATA
-DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DHOTPLUG_FIRMWARE -c -o
torisatool.o torisatool.c
cc -o torisatool torisatool.o
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -m64 -DSTANDALONE_ZAPATA
-DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DHOTPLUG_FIRMWARE -c -o
ztmonitor.o ztmonitor.c
cc -o ztmonitor ztmonitor.o
cc -o ztspeed.o -c ztspeed.c
cc -o ztspeed ztspeed.o
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -m64 -DSTANDALONE_ZAPATA
-DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DHOTPLUG_FIRMWARE -c -o zttool.o
zttool.c
cc -o zttool zttool.o -lnewt
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -m64 -DSTANDALONE_ZAPATA
-DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DHOTPLUG_FIRMWAREzttest.c -o
zttest
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -m64 -DSTANDALONE_ZAPATA
-DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DHOTPLUG_FIRMWARE -c -o
fxotune.o fxotune.c
cc -o fxotune fxotune.o -lm
/lib/modules/2.6.9-42.ELsmp/build
make -C /lib/modules/2.6.9-42.ELsmp/build SUBDIRS=/usr/src/zaptel-1.2.12
modules
make[1]: Entering directory `/usr/src/kernels/2.6.9-42.EL-smp-x86_64'
CC [M] /usr/src/zaptel-1.2.12/zaptel.o
{standard input}: Assembler messages:
{standard input}:16281: Error: suffix or operands invalid for `mov'
{standard input}:16282: Error: suffix or operands invalid for `mov'
{standard input}:16773: Error: suffix or operands invalid for `mov'
{standard input}:16774: Error: suffix or operands invalid for `mov'
{standard input}:17321: Error: suffix or operands invalid for `mov'
{standard input}:17322: Error: suffix or operands invalid for `mov'
{standard input}:17810: Error: suffix or operands invalid for `mov'
{standard input}:17811: Error: suffix or operands invalid for `mov'
/usr/src/zaptel-1.2.12/zaptel.c:188: warning: 'fcstab' defined but not
used
make[2]: *** [/usr/src/zaptel-1.2.12/zaptel.o] Error 1
make[1]: *** [_module_/usr/src/zaptel-1.2.12] Error 2
make[1]: Leaving directory `/usr/src/kernels/2.6.9-42.EL-smp-x86_64'
make: *** [linux26] Error 2
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RE: [Asterisk-Users] asterisk + door opener
Thers a box we can get here in Australia that has an RJ45 plug, a built in web server that has a config page and URL's to close/open one of 4 included relays. I use phpagi to hit the url and open/close doors that way. If anyone is interested, ill let you know URL's but its being sold from our local Jaycar agent. Regards Kevin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dovid B Sent: Sunday, 24 December 2006 1:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] asterisk + door opener Never got the financial backing for it. I looked in to mass producing it in China and I was in way over my head. I am currently working with an investor. I will let you guys know if anything comes up. Dovid - Original Message - From: Jerry [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, December 20, 2006 11:30 PM Subject: Re: [Asterisk-Users] asterisk + door opener Hi Dovid, I am actually now working on massproducing door openers that will work with asterisk. It will have an rj45 port and then a port to plug the door opener in to. Please contact me off list if you are interested. This is an old message, but I was wondering if you are still doing this, and what the specs/cost are. Thanks, J. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] How accurate is show translation?
When you built Asterisk, it must have refused to build the ilbc codec - I have never seen an Asterisk box that could not transcode ilbc, in over 3 years of working with Asterisk. PaulH On Sun, 2006-12-24 at 14:12 +1100, James Harper wrote: Just to give you another (relative) comparison... This is from a VIA processor running at 533MHz, (64KB Cache) asterisk compiled as i586 as it's missing some of the nicer MMX instructions: g723 gsm ulaw alaw g726 adpcm slin lpc10 g729 speex ilbc g723 - - - - - - - - - - - gsm - -10103212 943 - - 174 ulaw -18 - 124 4 135 - - 166 alaw -18 1 -24 4 135 - - 166 g726 -362020 -221953 - - 184 adpcm -19 3 325 - 236 - - 167 slin -17 1 123 3 -34 - - 165 lpc10 -422626482825 - - - 190 g729 - - - - - - - - - - - speex - - - - - - - - - - - ilbc -49333355353266 - - - Hey... why does your asterisk allow translation to and from ilbc? This is mine (Dual Celeron 533's): Translation times between formats (in milliseconds) Source Format (Rows) Destination Format(Columns) g723 gsm ulaw alaw g726 adpcm slin lpc10 g729 speex ilbc g723 - - - - - - - - - - - gsm - - 7 713 7 61739 100 - ulaw -17 - 1 8 2 1123495 - alaw -17 1 - 8 2 1123495 - g726 -23 8 8 - 8 71840 101 - adpcm -17 2 2 8 - 1123495 - slin -16 1 1 7 1 -113394 - lpc10 -24 9 915 9 8 -41 102 - g729 -23 8 814 8 718 - 101 - speex -2712121812112244 - - ilbc - - - - - - - - - - - And it won't translation ilbc to/from anything else. I had to buy some g729 licenses (a better option anyway though) because my voip provider only does alaw, g729, and ilbc, and I prefer to use alaw internally... James ___ --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
