Re: [Asterisk-Users] Asterisk and VMWare
This is a solutioin for my question? I don't want another box in my house running 24/7... Why to buy another disk as the system MUST run Win XP for some other important reasons?? Dan . - Original Message - From: Brancaleoni Matteo [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, July 14, 2003 12:05 AM Subject: Re: [Asterisk-Users] Asterisk and VMWare buy a 30$ hd, install linux on it and use it natively. Matteo. Il sab, 2003-07-12 alle 21:49, Dan ha scritto: Hi, I have installed Asterisk in a VM under VMWare Workstation 4.x (on WinXP). The computer is an Athlon @1GHz with 384MB RAM, 128MB allocated for RH9. Linux is fully installed (but without any X stuff). I have the latest Astrisk distribution (DL today) I have no Digium card installed on this machine. When I call Echotest, Asterisk play the message a l ittle bit choppy, but the echotest is perfect (no interruptions). The processor is used max 5% (peak) by the VMWare engine during the message playing. I have used a Cisco 7960 (G711) to call the Echotest. It seems that the GSM to G711 conversion inside VMWare virtual machine is the cause of this. It can be done something to improve this behaviour? Thanks, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Matteo Brancaleoni Espia System Administrator - IT services Website : http://www.espia.it Email : [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DTMF control for TDM device?
I'm not sure I'm sniffing up the right tree here. Using a TDM200 and X100P talking to a POTS circuit. Recently (unfortunately, can't say just *how* recently) I noticed when I called using my credit card that the DTMF tones I'm sending are not recognized by the processor at the other end. I used this exact same hardware, on the same lines, to make the same calls for a couple of months. I called another phone so I could listen, and indeed the tones are clipped very short and sound distorted. I have looked around in the sample configs and in the source and I don't see any config parameters for the Zap devices that might affect this. Have I missed something? Does anyone know what might have changed? Thx. B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and VMWare
On Monday 14 July 2003 01:22, Dan wrote: This is a solutioin for my question? I don't want another box in my house running 24/7... Why to buy another disk as the system MUST run Win XP for some other important reasons?? Then run Linux natively, with Asterisk, and run Win XP within VMWare on top of Linux. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AUSTEL Certified
It's available already, but the certification will not be complete until the end of summer. We're already recommending it as an E400P substitute but not yet as a T400P substitute since the T400P *does* have certification. Are prices also available? I couldn't find anything about it on digium.com or with google. (And the local resellers here in Germany don't have it too) Thanks, Rainer -- http://graphics.cs.uni-sb.de/VoIP/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and VMWare
Hi, I cannot do it WinXP must have access to my home automation system (proprietary cards) Dan - Original Message - From: Tilghman Lesher [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, July 14, 2003 9:44 AM Subject: Re: [Asterisk-Users] Asterisk and VMWare On Monday 14 July 2003 01:22, Dan wrote: This is a solutioin for my question? I don't want another box in my house running 24/7... Why to buy another disk as the system MUST run Win XP for some other important reasons?? Then run Linux natively, with Asterisk, and run Win XP within VMWare on top of Linux. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Setting up A TDM400P
Thanks for the reply, I have made those changes and still get the following error: WARNING[16384]: File chan_zap.c, Line 576 (zt_open): Unable to specify channel 1: No such device or address ERROR[16384]: File chan_zap.c, Line 4746 (mkintf): Unable to open channel 1: No such device or address here = 0, tmp-channel = 0, channel = 1 ERROR[16384]: File chan_zap.c, Line 6404 (load_module): Unable to register channel '1' WARNING[16384]: File loader.c, Line 299 (ast_load_resource): chan_zap.so: load_module failed, returning -1 WARNING[16384]: File loader.c, Line 394 (load_modules): Loading module chan_zap.so failed! I find it quite odd, expecially since the card is detected and showing when I do a dmesg. Jay On Sun, 13 Jul 2003 23:54:13 -0500, John Bigelow [EMAIL PROTECTED] wrote: The channel has to come after the signalling and other configuration lines. It should look something like this: signalling=fxo_ks context=internal channel = 1-3 Don't forget to configure zaptel.conf either. Add this line to it. fxoks=1-3 -John - Original Message - From: Jay Tyndall [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, July 13, 2003 11:38 PM Subject: [Asterisk-Users] Setting up A TDM400P Hi, I am having some trouble getting a TDM400P working, and would be very appriciative of some ideas. I have installed a TDM400P, and downloaded the appropriate files from CVS, compiled, installed and modprobe'd the devices. From dmesg: Module 0: Initialized Module 1: Initialized Module 2: Initialized Module 3: Not installed Found a Wildcard FXS: Wildcard S400P Prototype (4 modules) So, it has found the card OK, but I am little confused as to what I need to put in zapata.conf to make these 3 ports work with the analog phones I have plugged into these ports. I have tried looking at the docs on the digium site, but cannot seem to get it worked out. I tried putting the following in zapata.conf channel = 1 signalling = fxo_ks And this in extensions.conf exten = 200,1,Dial(Zap/1) When I start asterisk, I get the following: [chan_zap.so] = (Zapata Telephony) WARNING[16384]: File chan_zap.c, Line 6654 (load_module): Ignoring switchtype WARNING[16384]: File chan_zap.c, Line 6654 (load_module): Ignoring rxwink WARNING[16384]: File chan_zap.c, Line 576 (zt_open): Unable to specify channel 1: No such device or address ERROR[16384]: File chan_zap.c, Line 4746 (mkintf): Unable to open channel 1: No such device or address here = 0, tmp-channel = 0, channel = 1 ERROR[16384]: File chan_zap.c, Line 6404 (load_module): Unable to register channel '1' WARNING[16384]: File loader.c, Line 299 (ast_load_resource): chan_zap.so: load_module failed, returning -1 WARNING[16384]: File loader.c, Line 394 (load_modules): Loading module chan_zap.so failed! Thanks for your help Jay ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Using M2, Opera's revolutionary e-mail client: http://www.opera.com/m2/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Drop the call in 10min
[EMAIL PROTECTED] wrote: There is a way to drop every running call every 10 min of conversation? Isamar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users Check the whentohangup value from cdr struct . Set the value to 600 . ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and VMWare
Your machine is way underpower and underRAM, I am amazed you can even run VMWARE and a Linux guest on it. Have you try: 1. run VMWARE in Full screen windows. 2. is your Linux kernel SMP? (see VM knowledge base) 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. 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. 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. On Sun, 2003-07-13 at 23:22, Dan wrote: This is a solutioin for my question? I don't want another box in my house running 24/7... Why to buy another disk as the system MUST run Win XP for some other important reasons?? Dan . - Original Message - From: Brancaleoni Matteo [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, July 14, 2003 12:05 AM Subject: Re: [Asterisk-Users] Asterisk and VMWare buy a 30$ hd, install linux on it and use it natively. Matteo. Il sab, 2003-07-12 alle 21:49, Dan ha scritto: Hi, I have installed Asterisk in a VM under VMWare Workstation 4.x (on WinXP). The computer is an Athlon @1GHz with 384MB RAM, 128MB allocated for RH9. Linux is fully installed (but without any X stuff). I have the latest Astrisk distribution (DL today) I have no Digium card installed on this machine. When I call Echotest, Asterisk play the message a l ittle bit choppy, but the echotest is perfect (no interruptions). The processor is used max 5% (peak) by the VMWare engine during the message playing. I have used a Cisco 7960 (G711) to call the Echotest. It seems that the GSM to G711 conversion inside VMWare virtual machine is the cause of this. It can be done something to improve this behaviour? Thanks, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Matteo Brancaleoni Espia System Administrator - IT services Website : http://www.espia.it Email : [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Sunny Woo [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and VMWare
On Mon, 2003-07-14 at 09:15, Dan wrote: Hi, I cannot do it WinXP must have access to my home automation system (proprietary cards) Dan Frankly it amazes me you are serious about running a PBX inside vmware on top of an appalling OS and then asking if some gsm - g711 conversion can be improved cause it don't work too good within vmware. I don't know of any vendor that supports their application if it's running on an unsupported OS. So,the answers you get all say the same: either install Linux on the WinXP box and run your home automation system inside vmware. Or get one of those mini-atx boxes for $300, install Linux and Asterisk on it and you are set. They are silent, low on energy consumption and the 1GHz Via C3 should be fine for home Asterisk use. On a final note, if you do a search on google you will find many references to X10, home automation and Linux so there maybe a way out of those evil proprietary Mickeysoft products. Patrick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
AW: [Asterisk-Users] mod tor2 takes 20-30% from CPU (20-30% System)
Please can anybody help me with this, have anybody experiences with the tor2 driver? -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von Thomas Haeger Gesendet: Freitag, 11. Juli 2003 13:23 An: Asterisk User Betreff: [Asterisk-Users] mod tor2 takes 20-30% from CPU (20-30% System) Hi all, i have a E400P in my P III 1,4 GHz machine. When i start the tor2 driver (modprobe tor2) then i can see (with top) that the System takes 20 - 30 % CPU usage. Is this normal ? Thanks for help, Thomas. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Drop the call in 10min
[EMAIL PROTECTED] wrote: There is a way to drop every running call every 10 min of conversation? Isamar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ast_channel_setwhentohangup(chan,600); ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and VMWare
Get a cheapo system with a VIA C3 or something. Windoze + vmware destroys all the nice timing you might need to get good sound. With a VIA based system, you won't be able to support too many concurrent calls, but it's silent and cost effective (low-power etc) On Saturday 12 July 2003 21:49, Dan wrote: Hi, I have installed Asterisk in a VM under VMWare Workstation 4.x (on WinXP). The computer is an Athlon @1GHz with 384MB RAM, 128MB allocated for RH9. Linux is fully installed (but without any X stuff). I have the latest Astrisk distribution (DL today) I have no Digium card installed on this machine. When I call Echotest, Asterisk play the message a l ittle bit choppy, but the echotest is perfect (no interruptions). The processor is used max 5% (peak) by the VMWare engine during the message playing. I have used a Cisco 7960 (G711) to call the Echotest. It seems that the GSM to G711 conversion inside VMWare virtual machine is the cause of this. It can be done something to improve this behaviour? Thanks, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Roy Sigurd Karlsbakk, Datavaktmester ProntoTV AS - http://www.pronto.tv/ Tel: +47 9801 3356 Computers are like air conditioners. They stop working when you open Windows. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
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 - Original Message - From: Sunny Woo [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, July 14, 2003 10:54 AM Subject: Re: [Asterisk-Users] Asterisk and VMWare Your machine is way underpower and underRAM, I am amazed you can even run VMWARE and a Linux guest on it. Have you try: 1. run VMWARE in Full screen windows. 2. is your Linux kernel SMP? (see VM knowledge base) 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. 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. 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. On Sun, 2003-07-13 at 23:22, Dan wrote: This is a solutioin for my question? I don't want another box in my house running 24/7... Why to buy another disk as the system MUST run Win XP for some other important reasons?? Dan . - Original Message - From: Brancaleoni Matteo [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, July 14, 2003 12:05 AM Subject: Re: [Asterisk-Users] Asterisk and VMWare buy a 30$ hd, install linux on it and use it natively. Matteo. Il sab, 2003-07-12 alle 21:49, Dan ha scritto: Hi, I have installed Asterisk in a VM under VMWare Workstation 4.x (on WinXP). The computer is an Athlon @1GHz with 384MB RAM, 128MB allocated for RH9. Linux is fully installed (but without any X stuff). I have the latest Astrisk distribution (DL today) I have no Digium card installed on this machine. When I call Echotest, Asterisk play the message a l ittle bit choppy, but the echotest is perfect (no interruptions). The processor is used max 5% (peak) by the VMWare engine during the message playing. I have used a Cisco 7960 (G711) to call the Echotest. It seems that the GSM to G711 conversion inside VMWare virtual machine is the cause of this. It can be done something to improve this behaviour? Thanks, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Matteo Brancaleoni Espia System Administrator - IT services Website : http://www.espia.it Email : [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Sunny Woo [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] * with external sip proxy
Hi all, i'm tring ro use sip with an external sip proxy as vocal or ser. My scenario is Vocal or SER Asterisk with cnah_oh323 - Gatekeeper I would like that sip termial register themself to Vocal or ser and the h.323 terminal to gatekeeper. When i place a call from h323 side to sip side all work When a try to place a call form sip to h323 nothing happen Does someone try this??? Any suggestion will be appreciate Tnx Marco ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] .gsm voice format
Hello- What is the .gsm format? Ie: what's the encoding method and sample rate please? Thanks Scott Scott M. Stingel Emerging Voice Technology Inc. Palo Alto, California and London, England Email: [EMAIL PROTECTED] URL:www.evtmedia.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] mgcp problems
Pavel Zheltouhov wrote: When I connected over two mgcp channels and sending numerical indication to cisco ata it seems hangup one channel (receving ) and generate 'fast busy' tone. I hack chan_mgcp and my threewaycalling works ok! But why indications are sent after I press hookflash on answering end? Is it possible to do this hack in chan_sip? Threewaycalling is the only thing i am missing with ATA/SIP/Asterisk! -or- does ATA/MGCP/Asterisk complete working (CallerID-transfer, MSG-Waiting-Indicator...)? Can you post a chan_mgcp.diff or your modified chan_mgcp? I really like to use threewaycalling with my ATAs. Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] .gsm voice format
Gsm is wav in 8/mono srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Scott Stingel Enviado el: lunes, 14 de julio de 2003 12:33 Para: [EMAIL PROTECTED] Asunto: [Asterisk-Users] .gsm voice format Hello- What is the .gsm format? Ie: what's the encoding method and sample rate please? Thanks Scott Scott M. Stingel Emerging Voice Technology Inc. Palo Alto, California and London, England Email: [EMAIL PROTECTED] URL:www.evtmedia.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] .gsm voice format
Thanks - do you know the bit rate? I'm trying to play these prompts with other voice application software, and so far have been unable to. I've tried: Windows Media player, Vox Studio, Envox prompt editor - no luck with any of these. Cheers Scott Scott M. Stingel Emerging Voice Technology Inc. Email: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] URL:www.evtmedia.com http://www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sergio Serrano Revuelto Sent: Monday, July 14, 2003 12:25 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] .gsm voice format Gsm is wav in 8/mono srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Scott Stingel Enviado el: lunes, 14 de julio de 2003 12:33 Para: [EMAIL PROTECTED] Asunto: [Asterisk-Users] .gsm voice format Hello- What is the .gsm format? Ie: what's the encoding method and sample rate please? Thanks Scott Scott M. Stingel Emerging Voice Technology Inc. Palo Alto, California and London, England Email: [EMAIL PROTECTED] URL:www.evtmedia.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Line Override Device
This power failure thing does not have to be complicated. A few solutions come to mind: 1) A 3,5,12 (whatever is needed) power supply (wall wart)used with a relay (DPDT). When the wall wart has power, the computer takes the call. When power fails, the POTS line falls in to place. Now, this does not delay while the computer is booting up. 2) A basic stamp computer - about $25-30. It has 8 programmable i/o pins that will drive relays. One pin monitors either a wall wart or 5v from one of the plugs on your computer's power supply. When pin 1 goes low (no power) relay kicks in to bypass computer and connect POTS line direct. When power returns program jumps to a sleep or delay statement for xMINS until computer boots. And then releases relay for normal operation. www.parallaxinc.com and resellers. James Taylor [EMAIL PROTECTED] 903-793-1953 -- Original Message -- From: Steven Critchfield [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] Date: 13 Jul 2003 17:35:55 -0500 On Sun, 2003-07-13 at 15:55, John Laur wrote: 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. It is easier to wire to a 12 volt(yellow) wire off of the PSU, plus this lets you drive larger relays. 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. etherkillers are 110 volts AC to data pins, POE is 48 volts DC on non data pins. This should not blow devices that are not expecting PoE. Think about it, how would a device ask for power if it doesn't have power to make the request? -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- James Taylor [EMAIL PROTECTED] 903-793-1953 -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] .gsm voice format
Quick time will pay them okay -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Scott Stingel Sent: Monday, July 14, 2003 9:46 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] .gsm voice format Thanks - do you know the bit rate? I'm trying to play these prompts with other voice application software, and so far have been unable to. I've tried: Windows Media player, Vox Studio, Envox prompt editor - no luck with any of these. Cheers Scott Scott M. Stingel Emerging Voice Technology Inc. Email: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] URL:www.evtmedia.com http://www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sergio Serrano Revuelto Sent: Monday, July 14, 2003 12:25 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] .gsm voice format Gsm is wav in 8/mono srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Scott Stingel Enviado el: lunes, 14 de julio de 2003 12:33 Para: [EMAIL PROTECTED] Asunto: [Asterisk-Users] .gsm voice format Hello- What is the .gsm format? Ie: what's the encoding method and sample rate please? Thanks Scott Scott M. Stingel Emerging Voice Technology Inc. Palo Alto, California and London, England Email: [EMAIL PROTECTED] URL:www.evtmedia.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users smime.p7s Description: S/MIME cryptographic signature
Re: AW: [Asterisk-Users] mod tor2 takes 20-30% from CPU (20-30%System)
No it isn't normal. I have a machine with a T400P in it and I don't even see that load continuously on my machine even with calls being routed. On Mon, 2003-07-14 at 03:08, Thomas Haeger wrote: Please can anybody help me with this, have anybody experiences with the tor2 driver? -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von Thomas Haeger Gesendet: Freitag, 11. Juli 2003 13:23 An: Asterisk User Betreff: [Asterisk-Users] mod tor2 takes 20-30% from CPU (20-30% System) Hi all, i have a E400P in my P III 1,4 GHz machine. When i start the tor2 driver (modprobe tor2) then i can see (with top) that the System takes 20 - 30 % CPU usage. Is this normal ? Thanks for help, Thomas. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] .gsm voice format
Use sox to put it is MSGSM and a RIFF header on it. sox file.gsm -g file.wav You need to do this because gsm takes 160 samples and compresses it to 32.5 bytes. On unix systems, they let the half byte go to waste. On windows they slide a second frame down a half byte and combine it with a first frame to put 2 frames into 65 bytes. On Mon, 2003-07-14 at 06:45, Scott Stingel wrote: Thanks - do you know the bit rate? I'm trying to play these prompts with other voice application software, and so far have been unable to. I've tried: Windows Media player, Vox Studio, Envox prompt editor - no luck with any of these. Cheers Scott Scott M. Stingel Emerging Voice Technology Inc. Email: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] URL:www.evtmedia.com http://www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sergio Serrano Revuelto Sent: Monday, July 14, 2003 12:25 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] .gsm voice format Gsm is wav in 8/mono srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Scott Stingel Enviado el: lunes, 14 de julio de 2003 12:33 Para: [EMAIL PROTECTED] Asunto: [Asterisk-Users] .gsm voice format Hello- What is the .gsm format? Ie: what's the encoding method and sample rate please? Thanks Scott Scott M. Stingel Emerging Voice Technology Inc. Palo Alto, California and London, England Email: [EMAIL PROTECTED] URL:www.evtmedia.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and VMWare
On Sat, 2003-07-12 at 14:49, Dan wrote: Hi, I have installed Asterisk in a VM under VMWare Workstation 4.x (on WinXP). The computer is an Athlon @1GHz with 384MB RAM, 128MB allocated for RH9. Linux is fully installed (but without any X stuff). If you where to look through the archive you would find that I have a similar machine running linux on bare hardware and when my xscreensaver would come on it would cause call quality to drop below usable. Of course I used a T100P so I also had an interupt timer that asterisk needed. So please understand that your _WILL_NOT_ get the performance you need from within windows. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] unsubscribe
-- Yagdzhyyev Vladislav Dnepropetrovsk, Ukraine ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk and modem
hi, i have to do a demo with asterisk, unfortunately i don't have yet an x100p card, so i need to use a 56k voice modem on my motherboard... could someone tell me how i can configure asterisk to use this modem to call? thanks a lot for the help!!! Angelo ___ 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] AUSTEL Certified
Same price as the E400P. Mark On Mon, 14 Jul 2003, Rainer Jochem wrote: It's available already, but the certification will not be complete until the end of summer. We're already recommending it as an E400P substitute but not yet as a T400P substitute since the T400P *does* have certification. Are prices also available? I couldn't find anything about it on digium.com or with google. (And the local resellers here in Germany don't have it too) Thanks, Rainer -- http://graphics.cs.uni-sb.de/VoIP/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk and modem
Hi, On Mon, 2003-07-14 at 15:58, Angelo Sampietro wrote: i have to do a demo with asterisk, unfortunately i don't have yet an x100p card, so i need to use a 56k voice modem on my motherboard... could someone tell me how i can configure asterisk to use this modem to call? Forget about it. If you'd ever get it to work, you would demo something that is below acceptable standards. Rather demo voip-asterisk-voip without any PSTN functionality. Or, if you have an ISDN BRI, get an ISDN card and use that (chan_modem_i4l or chan_capi) depending on the ISDN card. Or, just delay the demo until after the X100P has arrived. wkr, -- Envida http://www.envida.net/ Armand A. Verstappen Graadt van Roggenweg 328 [EMAIL PROTECTED] 3531 AH Utrecht tel: +31 (0)30 298 2255Postbus 19127 fax: +31 (0)30 298 21113501 DC Utrecht signature.asc Description: This is a digitally signed message part
Re: AW: [Asterisk-Users] mod tor2 takes 20-30% from CPU (20-30% System)
It is normal. What you see depends on which version of various things are on your system. The tor2 driver spends a lot of time in the interrupt service routine (about 60% of the time on the 700MHz Athlon I use). Whether the interrupt service times shows up as system usage, or falls down a hole without being reported at all, as I said, depends on which versions of things you have on your machine. Regards, Steve Steven Critchfield wrote: No it isn't normal. I have a machine with a T400P in it and I don't even see that load continuously on my machine even with calls being routed. On Mon, 2003-07-14 at 03:08, Thomas Haeger wrote: Please can anybody help me with this, have anybody experiences with the tor2 driver? -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von Thomas Haeger Gesendet: Freitag, 11. Juli 2003 13:23 An: Asterisk User Betreff: [Asterisk-Users] mod tor2 takes 20-30% from CPU (20-30% System) Hi all, i have a E400P in my P III 1,4 GHz machine. When i start the tor2 driver (modprobe tor2) then i can see (with top) that the System takes 20 - 30 % CPU usage. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
AW: AW: [Asterisk-Users] mod tor2 takes 20-30% from CPU (20-30% System)
Steve, thanks for your explanation. This is the cause for the fact that if i change the pci slot, the problem is blown away, i think. Maybe the IRQ sharing is the cause ... Thanks a lot and best regards, Thomas. -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von Steve Underwood Gesendet: Montag, 14. Juli 2003 16:27 An: [EMAIL PROTECTED] Betreff: Re: AW: [Asterisk-Users] mod tor2 takes 20-30% from CPU (20-30% System) It is normal. What you see depends on which version of various things are on your system. The tor2 driver spends a lot of time in the interrupt service routine (about 60% of the time on the 700MHz Athlon I use). Whether the interrupt service times shows up as system usage, or falls down a hole without being reported at all, as I said, depends on which versions of things you have on your machine. Regards, Steve Steven Critchfield wrote: No it isn't normal. I have a machine with a T400P in it and I don't even see that load continuously on my machine even with calls being routed. On Mon, 2003-07-14 at 03:08, Thomas Haeger wrote: Please can anybody help me with this, have anybody experiences with the tor2 driver? -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von Thomas Haeger Gesendet: Freitag, 11. Juli 2003 13:23 An: Asterisk User Betreff: [Asterisk-Users] mod tor2 takes 20-30% from CPU (20-30% System) Hi all, i have a E400P in my P III 1,4 GHz machine. When i start the tor2 driver (modprobe tor2) then i can see (with top) that the System takes 20 - 30 % CPU usage. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] .gsm voice format
Ok, thanks. I should have asked this in the first place: what I'm really getting at is that I need to record (or convert) prompts in many languages. I have a number of Windows based tools to do this. I want to end up with prompts that asterisk can play with the least CPU effort, ie. without transcoding. Would this be the gsm format? If so, it sounds like I should record in Wav format, 8-bit samples? How to convert to gsm? Thanks again! Scott Scott M. Stingel Emerging Voice Technology Inc. Email: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] URL:www.evtmedia.com http://www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Critchfield Sent: Monday, July 14, 2003 2:16 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] .gsm voice format Use sox to put it is MSGSM and a RIFF header on it. sox file.gsm -g file.wav You need to do this because gsm takes 160 samples and compresses it to 32.5 bytes. On unix systems, they let the half byte go to waste. On windows they slide a second frame down a half byte and combine it with a first frame to put 2 frames into 65 bytes. On Mon, 2003-07-14 at 06:45, Scott Stingel wrote: Thanks - do you know the bit rate? I'm trying to play these prompts with other voice application software, and so far have been unable to. I've tried: Windows Media player, Vox Studio, Envox prompt editor - no luck with any of these. Cheers Scott Scott M. Stingel Emerging Voice Technology Inc. Email: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] URL:www.evtmedia.com http://www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sergio Serrano Revuelto Sent: Monday, July 14, 2003 12:25 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] .gsm voice format Gsm is wav in 8/mono srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Scott Stingel Enviado el: lunes, 14 de julio de 2003 12:33 Para: [EMAIL PROTECTED] Asunto: [Asterisk-Users] .gsm voice format Hello- What is the .gsm format? Ie: what's the encoding method and sample rate please? Thanks Scott Scott M. Stingel Emerging Voice Technology Inc. Palo Alto, California and London, England Email: [EMAIL PROTECTED] URL:www.evtmedia.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ 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] asterisk and modem
Hi Armand, second common project ? cheers Michael On Monday 14 July 2003 15:27, Armand A. Verstappen wrote: Hi, On Mon, 2003-07-14 at 15:58, Angelo Sampietro wrote: i have to do a demo with asterisk, unfortunately i don't have yet an x100p card, so i need to use a 56k voice modem on my motherboard... could someone tell me how i can configure asterisk to use this modem to call? Forget about it. If you'd ever get it to work, you would demo something that is below acceptable standards. Rather demo voip-asterisk-voip without any PSTN functionality. Or, if you have an ISDN BRI, get an ISDN card and use that (chan_modem_i4l or chan_capi) depending on the ISDN card. Or, just delay the demo until after the X100P has arrived. wkr, -- Michael Bielicki Managing Director TAAN Consultants Ltd http://www.global-gateway.net/ -- This correspondence is for the named person's use only. It may contain confidential or legally privileged information or both. No confidentiality or privilege is waived or lost by any mistransmission. If you receive this correspondence in error, please immediately delete it from your system and notify the sender. You must not disclose, copy or rely on any part of this correspondence if you are not the intended recipient. Any opinions expressed in this message are those of the individual sender. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] EZ-Install
Has anyone thought about an ISO file that could be used to make a CD for a bootable install for a basic Linux/Asterisk system? Just re-boot and config. -- James Taylor [EMAIL PROTECTED] 903-793-1953 -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] .gsm voice format
On Mon, 2003-07-14 at 09:47, Scott Stingel wrote: Ok, thanks. I should have asked this in the first place: what I'm really getting at is that I need to record (or convert) prompts in many languages. I have a number of Windows based tools to do this. I want to end up with prompts that asterisk can play with the least CPU effort, ie. without transcoding. Would this be the gsm format? If so, it sounds like I should record in Wav format, 8-bit samples? How to convert to gsm? First learn sox, it will save you a lot of time dealing with format conversions. Next record at 8bit 8k so that what you hear on the speakers is the same as what will go out to the phone. Then realize that if you record via PCM and let sox convert to whatever you like, you will be well off. The question about transcoding, if you are going to PSTN via a hardware interface, then you might want to store your audio in alaw or ulaw format, if it is VoIP then whatever codec you might use there, or again alaw/ulaw so it is a one hop conversion. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] EZ-Install
On Mon, 2003-07-14 at 10:34, jltaylor wrote: Has anyone thought about an ISO file that could be used to make a CD for a bootable install for a basic Linux/Asterisk system? Just re-boot and config. Might be interesting to build based off of a knoppix cd, but then what do you store the configs to? -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] EZ-Install
Not CD based. Just CD install. When you reboot Linux with asterisk is installed. You could add any other tools you think are necessary. User then just does config. -- Original Message -- From: Steven Critchfield [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] Date: 14 Jul 2003 10:18:24 -0500 On Mon, 2003-07-14 at 10:34, jltaylor wrote: Has anyone thought about an ISO file that could be used to make a CD for a bootable install for a basic Linux/Asterisk system? Just re-boot and config. Might be interesting to build based off of a knoppix cd, but then what do you store the configs to? -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- James Taylor [EMAIL PROTECTED] 903-793-1953 -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] h323 Ringing sound
Hi When i make a call using oh323 is posible to make the ringing sound thanks
X100P mod or USB relay box, RE: [Asterisk-Users] Line Override Device
The best solution would be an enhancement to the X100P card. If the 2nd RJ jack was a pass through for the line except when the card had power and was initialized. Some kind of watchdog functionality would also be nice so that if, for example, Asterisk dies then pass through functionality would take effect after n seconds. This would probably mean adding a relay to the board which would raise to cast a little. But, as the original poster indicated this is critical for a serious system. An alternative would be an extra relay box, maybe powered by USB. One mode could be to switch based on presence of power, another mode could require periodic watchdog pings via the USB. I always wanted to build something using a USB flavored PIC... I can see this for small offices (like ours). We have 4 incoming lines in a hunt group. If Asterisk is not running I want one of those lines to ring the receptionist (maybe using a simple dedicated phone since they'd otherwise have an IP phone) and the others looped for busy. I can see a box with USB and 12 RJ jacks (4 x (1 in, 2 outs)) to make that work. Would anyone buy a product like that? -reed At 07:12 AM 7/14/2003 -0500, jltaylor wrote: This power failure thing does not have to be complicated. A few solutions come to mind: 1) A 3,5,12 (whatever is needed) power supply (wall wart)used with a relay (DPDT). When the wall wart has power, the computer takes the call. When power fails, the POTS line falls in to place. Now, this does not delay while the computer is booting up. 2) A basic stamp computer - about $25-30. It has 8 programmable i/o pins that will drive relays. One pin monitors either a wall wart or 5v from one of the plugs on your computer's power supply. When pin 1 goes low (no power) relay kicks in to bypass computer and connect POTS line direct. When power returns program jumps to a sleep or delay statement for xMINS until computer boots. And then releases relay for normal operation. www.parallaxinc.com and resellers. James Taylor [EMAIL PROTECTED] 903-793-1953 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MGCP-H323 interoperability
Hello everybody, Anyone knows where I can find information for configure the Asterisk as MGCP-H323 transcoder? May be an example or something. Thank you very much Best regards ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] EZ-Install
Maybe it's just me... But I fail to see the reasoning behind branching to a whole new distribution just to support an easy, out of the box Asterisk install. Perhaps just the creation of an RPM package with a basic configuration would be the ticket? The one potential exception to this would be if you wrote a distribution with advanced hardware detection and preconfiguration such that during the install process, Digium hardware is detected and you can go ahead and configure spans and channels, etc. In that case, the distribution might have some unique value. Short of that, I cannot imagine a new distribution just to package together a pre-configured Asterisk configuration. Even if you wrote an installation process like that, couldn't it be just as well implemented with a clever RPM-based installation and some nice plain old userspace configuration tools? Matt Hardeman PaperSoft -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of jltaylor Sent: Monday, July 14, 2003 11:00 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] EZ-Install Not CD based. Just CD install. When you reboot Linux with asterisk is installed. You could add any other tools you think are necessary. User then just does config. -- Original Message -- From: Steven Critchfield [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] Date: 14 Jul 2003 10:18:24 -0500 On Mon, 2003-07-14 at 10:34, jltaylor wrote: Has anyone thought about an ISO file that could be used to make a CD for a bootable install for a basic Linux/Asterisk system? Just re-boot and config. Might be interesting to build based off of a knoppix cd, but then what do you store the configs to? -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- James Taylor [EMAIL PROTECTED] 903-793-1953 -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] mgcp problems
Thomas Dingermann wrote:
When I connected over two mgcp channels and sending numerical
indication to cisco ata it seems hangup one channel (receving )
and generate 'fast busy' tone.
I hack chan_mgcp and my threewaycalling works ok!
But why indications are sent after I press hookflash on answering end?
Is it possible to do this hack in chan_sip?
I think it's too dificult for me )
Threewaycalling is the only thing i am missing with ATA/SIP/Asterisk!
-or-
does ATA/MGCP/Asterisk complete working (CallerID-transfer,
No, as i know.
MSG-Waiting-Indicator...)?
Maybe.
Can you post a chan_mgcp.diff or your modified chan_mgcp? I really like
to use threewaycalling with my ATAs.
it's simple :
1539a1540,1545
if (strpbrk(tone,0123456789*#)) {
if (mgcpdebug) {
ast_verbose(VERBOSE_PREFIX_3 MGCP Asked to indicate
filtered tone,cisco workaround enabled \n);
}
return 0;
}
works for me (tm)
You need cvs version, 0.4 does not work with flashhook messages at all.
--
Pavel Zheltouhov, Comlink ISP, Voronezh, Russia
phone/fax +7(0732) 727172, http://www.comlink.ru
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[Asterisk-Users] Getting started
Hi, I am a total newbie to asterisk and can't find any useful documentation for asterisk...how are people supposed to get started? I'd like to know, how I create User Accounts, so that a SIP UA can login into asterisk with a password, for example. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] .gsm voice format
Hi, The question about transcoding, if you are going to PSTN via a hardware interface, then you might want to store your audio in alaw or ulaw format, It is possible to use Voicemail prompts in alaw or ulaw format? Thanks, Dan - Original Message - From: Steven Critchfield [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, July 14, 2003 6:10 PM Subject: RE: [Asterisk-Users] .gsm voice format On Mon, 2003-07-14 at 09:47, Scott Stingel wrote: Ok, thanks. I should have asked this in the first place: what I'm really getting at is that I need to record (or convert) prompts in many languages. I have a number of Windows based tools to do this. I want to end up with prompts that asterisk can play with the least CPU effort, ie. without transcoding. Would this be the gsm format? If so, it sounds like I should record in Wav format, 8-bit samples? How to convert to gsm? First learn sox, it will save you a lot of time dealing with format conversions. Next record at 8bit 8k so that what you hear on the speakers is the same as what will go out to the phone. Then realize that if you record via PCM and let sox convert to whatever you like, you will be well off. The question about transcoding, if you are going to PSTN via a hardware interface, then you might want to store your audio in alaw or ulaw format, if it is VoIP then whatever codec you might use there, or again alaw/ulaw so it is a one hop conversion. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: X100P mod or USB relay box, RE: [Asterisk-Users] Line OverrideDevice
On Mon, 2003-07-14 at 10:42, Reed Wade wrote: The best solution would be an enhancement to the X100P card. If the 2nd RJ jack was a pass through for the line except when the card had power and was initialized. Some kind of watchdog functionality would also be nice so that if, for example, Asterisk dies then pass through functionality would take effect after n seconds. This would probably mean adding a relay to the board which would raise to cast a little. But, as the original poster indicated this is critical for a serious system. One wouldn't use a X100P in a serious system. Maybe a appliance expected in the home, but then again in such a system you would probably wire up a adapter that could bridge all the lines together and connect them to a single outside extension on power failure. This way on failure, you revert to a single line analog setup. Possibly with a transformer on the loop to help out with ren limits. An alternative would be an extra relay box, maybe powered by USB. One mode could be to switch based on presence of power, another mode could require periodic watchdog pings via the USB. I always wanted to build something using a USB flavored PIC... Only if you aren't pulling power from the USB bus. There isn't much there. I can see this for small offices (like ours). We have 4 incoming lines in a hunt group. If Asterisk is not running I want one of those lines to ring the receptionist (maybe using a simple dedicated phone since they'd otherwise have an IP phone) and the others looped for busy. I can see a box with USB and 12 RJ jacks (4 x (1 in, 2 outs)) to make that work. Would anyone buy a product like that? -reed At 07:12 AM 7/14/2003 -0500, jltaylor wrote: This power failure thing does not have to be complicated. A few solutions come to mind: 1) A 3,5,12 (whatever is needed) power supply (wall wart)used with a relay (DPDT). When the wall wart has power, the computer takes the call. When power fails, the POTS line falls in to place. Now, this does not delay while the computer is booting up. 2) A basic stamp computer - about $25-30. It has 8 programmable i/o pins that will drive relays. One pin monitors either a wall wart or 5v from one of the plugs on your computer's power supply. When pin 1 goes low (no power) relay kicks in to bypass computer and connect POTS line direct. When power returns program jumps to a sleep or delay statement for xMINS until computer boots. And then releases relay for normal operation. www.parallaxinc.com and resellers. James Taylor [EMAIL PROTECTED] 903-793-1953 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] .gsm voice format
On Mon, 2003-07-14 at 11:25, Dan wrote: Hi, The question about transcoding, if you are going to PSTN via a hardware interface, then you might want to store your audio in alaw or ulaw format, It is possible to use Voicemail prompts in alaw or ulaw format? Yes, just run through the prompts directory and let sox convert them. You may wish to pull the gsm copies to make sure the wqv copies are used. - Original Message - From: Steven Critchfield [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, July 14, 2003 6:10 PM Subject: RE: [Asterisk-Users] .gsm voice format On Mon, 2003-07-14 at 09:47, Scott Stingel wrote: Ok, thanks. I should have asked this in the first place: what I'm really getting at is that I need to record (or convert) prompts in many languages. I have a number of Windows based tools to do this. I want to end up with prompts that asterisk can play with the least CPU effort, ie. without transcoding. Would this be the gsm format? If so, it sounds like I should record in Wav format, 8-bit samples? How to convert to gsm? First learn sox, it will save you a lot of time dealing with format conversions. Next record at 8bit 8k so that what you hear on the speakers is the same as what will go out to the phone. Then realize that if you record via PCM and let sox convert to whatever you like, you will be well off. The question about transcoding, if you are going to PSTN via a hardware interface, then you might want to store your audio in alaw or ulaw format, if it is VoIP then whatever codec you might use there, or again alaw/ulaw so it is a one hop conversion. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] EZ-Install
Sounds like you needed to start a new thread. One of these days I will either need to look up a good resource for mail list rules, or write it for all these newer users. On Mon, 2003-07-14 at 11:11, Todd Lieberman wrote: Hi All, I need some help w/supervised transfer and conference w/a 7940 phone. When I do a blind transfer the calls go through great, but when I do supervised transfer the 7940 tells me Transfer Denied. When I do a conference call I hit the conf key and then dial the next extension. The new call connects and I hit conf again but the calls do not get bridged. Any Suggestions? I'm using the config files from http://www.loligo.com/asterisk/Cisco/79xx/2003-06-20.from-ftpeng.cisco.c om/ Thanks, TL -- Todd Lieberman 800-675-3192 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Open H.323 and cdr problem
Hi all, I have probe cdr feature again and I realize when I make a call from H.323 endpoint, I don't see any log in cdr table. My asterisk box is the next: AVM FRITZ-- | |--EP |- ASTERISK -OH323GATEKEEPER- |--EP X100P-- | |--EP If * receives call from chan_capi or chan_zap I can see the log in cdr table, but if call is made from a H.323 endpoint I can't see any log in cdr table both /var/log/asterisk/cdr_csv/Master.csv. Any idea? Thanks in advance, srsergio ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Hardware Vendors
Hi All! Can anyone direct me to any websites / manufacturers out there who are making small, put-it-in-the-closet-and-forget-it type systems for building routers, home gateway servers, that sort of thing? My fantasy machine for this purpose would be along the lines of a mini-itx system with external power supply, dual Ethernet interfaces on board, and one PCI slot available. If it had one real serial port on it, that would be great too. Am I dreaming, or does it exist for a reasonable price? I would be willing to go the 500 MHz 1 GHz range. Something without a fan would be really nice. Im basically looking for a system that someone out there is stamping out in quantities and isnt too outrageous in price. Does it exist, and if so who sells it? It seems to me a system like the above described would be perfect for building out a home gateway / home asterisk server Matt Hardeman PaperSoft
Re: [Asterisk-Users] Getting started
Read the documentation, read the sip.conf file. And if it still doesn't make sense try one more time through the documentation and config file. At that point you should at least know enough to ask pointed questions at specific problems in your configs. On Mon, 2003-07-14 at 11:19, Johannes Herlitz wrote: Hi, I am a total newbie to asterisk and can't find any useful documentation for asterisk...how are people supposed to get started? I'd like to know, how I create User Accounts, so that a SIP UA can login into asterisk with a password, for example. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Odd output from X100P
I'm attempting to configure a machine with a DevKit Lite (X100P S100U). After I modprobe wcfxo, the machine goes into some kind of loop after about 10 seconds, where it spits out what appears to be 32-bit addresses, ad infinitum. At this point, the machine becomes completely unresponsive to keyboard input (i.e. Ctrl-C, Ctrl-Alt-Del, even CapsLock and NumLock states [lights] cannot be changed). /etc/zaptel.conf contains: fxsks=1 fxoks=2 loadzone = us defaultzone=us The addresses are to the screen and are not logged to syslog. They are of the form [c053dc09] [c8009ec0], etc. The addresses appear to all be in the ranges C0xx and C8xx. The hardware is a PII-266, with 48MB RAM. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] EZ-Install
On Mon, 2003-07-14 at 18:36, Steven Critchfield wrote: Sounds like you needed to start a new thread. One of these days I will either need to look up a good resource for mail list rules, or write it for all these newer users. http://www.freeradius.org/list/users.html comes a long way... wkr, -- Envida http://www.envida.net/ Armand A. Verstappen Graadt van Roggenweg 328 [EMAIL PROTECTED] 3531 AH Utrecht tel: +31 (0)30 298 2255Postbus 19127 fax: +31 (0)30 298 21113501 DC Utrecht signature.asc Description: This is a digitally signed message part
Re: X100P mod or USB relay box, RE: [Asterisk-Users] Line Override Device
At 11:34 AM 7/14/2003 -0500, Steven Critchfield wrote: One wouldn't use a X100P in a serious system. How so? I assume you're talking about scale and not reliability. We get a relatively small number of calls but any one of them could be worth a large stack of cash for our business. A stinky phone system can make us look bad. The main reason I'm looking at Asterisk is to improve the reliability and control over our phone system. All the other great things it provides really are secondary for the folks who pay my salary. Only if you aren't pulling power from the USB bus. There isn't much there. There may be just enough depending on how many relays are needed, but it would be too close. I agree, better off not trying to get power from there. I do like the idea of some kind of watchdog functionality. Simply having power isn't sufficient to trust that a call is getting routed. -reed ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Open H.323 and cdr problem
In oh323.conf, section [general]: amaFlags=billing Michael. Sergio Serrano Revuelto wrote: Hi all, I have probe cdr feature again and I realize when I make a call from H.323 endpoint, I don't see any log in cdr table. My asterisk box is the next: AVM FRITZ-- | |--EP |- ASTERISK -OH323GATEKEEPER- |--EP X100P-- | |--EP If * receives call from chan_capi or chan_zap I can see the log in cdr table, but if call is made from a H.323 endpoint I can't see any log in cdr table both /var/log/asterisk/cdr_csv/Master.csv. Any idea? Thanks in advance, srsergio ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Open H.323 and cdr problem
I can't speak for oh323, but my channel driver, chan_h323, most certainly works with CDRs. Jeremy McNamara Sergio Serrano Revuelto wrote: Hi all, I have probe cdr feature again and I realize when I make a call from H.323 endpoint, I don't see any log in cdr table. My asterisk box is the next: AVM FRITZ-- | |--EP |- ASTERISK -OH323GATEKEEPER- |--EP X100P-- | |--EP If * receives call from chan_capi or chan_zap I can see the log in cdr table, but if call is made from a H.323 endpoint I can't see any log in cdr table both /var/log/asterisk/cdr_csv/Master.csv. Any idea? Thanks in advance, srsergio ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MSN Messenger 4.7 vs 5.0
Hi. This is perhaps a little bit off-topic here, but I couldn't find informations elsewhere, so perhaps someone can help me: I've been using Messenger 4.7 for a while on my W2k Laptop to place SIP-Calls via asterisk with messengers communications service feature. Works fine. Then I thought that it would be a good idea to do an upgrade to v5.0... (yes, never touch a running system ;) But now - there's only the possibility to create and use Passport-accounts. Did I just don't find it or isn't there any chance to do SIP calls with Messenger 5.0 any more? TIA Rainer -- http://graphics.cs.uni-sb.de/VoIP/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] audio pause/delay problems
I'm curious. Isn't anyone else noticing these problems? Or are people simply not using asterisk for VoIP connectivity over wide-area networks this way? Or does it go away with g729 or other proprietary codecs? --J. Jan == Jan Rychter [EMAIL PROTECTED] writes: John == John Todd [EMAIL PROTECTED] writes: John For what it's worth, I have noticed the same problem, but I think John the problem is in IAX2, since my long-haul portions of the John diagram were over IAX2, while my SIP clients are almost always John sitting on the same LAN as the Asterisk server. Jan I have noticed these problems both in this kind of setup and in a Jan SIP call to a remote Asterisk server. John What codec were you testing with over IAX2? Jan GSM. Having investigated this a bit more, it turns out that using alaw instead of gsm on the IAX2 link makes the problem go away. It seems the jitter settings start working then. Any hints? I'd prefer not to be stuck with 80kbps per call... --J. [I have sent a message about SIP problems via gmane, but it seems the list is gatewayed one-way only...] The message was: I've been trying to use Asterisk as a SIP-PSTN gateway. It runs fine when the SIP client is on the local network and there is not packet loss. But now I've tried running a remote client (halfway around the globe) -- this works great until some packets get lost. After that it seems that either my client (linphone) or Asterisk doesn't want to resynchronize -- what gets played back is all voice packets as they have been received. This creates an increasing lag in the conversation and the only way I've found to fix it is to disconnect and reconnect again. Is anyone else seeing this? Is it linphone's fault, or is it expected behavior? Now, I have tried running another * on my side of the link. The setup then becomes: linphone - * - internet (IAX2) - * - PSTN (or echo). I'm testing with the echo application (GSM used everywhere) and I'm getting the same thing: everything seems to work, but sooner or later there is an audio pause and the delay grows. It never gets back to normal. I've had it grow to as much as 10s. What makes it even more surprising is the network performance. I've had ping running in the background, same TOS settings, 10 packets per second. It shows that my RTT is (min/avg/max/mdev) 220/229/287/8.85 with 0% loss! That's a pretty good network. So where do the pauses and delays come from? --J. pgp0.pgp Description: PGP signature
RE: [Asterisk-Users] Open H.323 and cdr problem
Thanks, it works srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Michael Manousos Enviado el: lunes, 14 de julio de 2003 19:18 Para: [EMAIL PROTECTED] Asunto: Re: [Asterisk-Users] Open H.323 and cdr problem In oh323.conf, section [general]: amaFlags=billing Michael. Sergio Serrano Revuelto wrote: Hi all, I have probe cdr feature again and I realize when I make a call from H.323 endpoint, I don't see any log in cdr table. My asterisk box is the next: AVM FRITZ-- | |--EP |- ASTERISK -OH323GATEKEEPER- |--EP X100P-- | |--EP If * receives call from chan_capi or chan_zap I can see the log in cdr table, but if call is made from a H.323 endpoint I can't see any log in cdr table both /var/log/asterisk/cdr_csv/Master.csv. Any idea? Thanks in advance, srsergio ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Odd output from X100P
Hi Tilghman- I recently had a lot of problems getting the DevLit kit working out of the box, even using the configurations supplied on the floppy that came with the kit. I didn't have the problem you are experiencing though, which sounds like some kind of hardware conflict to me. I would suggest contacting [EMAIL PROTECTED], and they are pretty good about diagnosing the problem (if you give them a login to your machine via ssh etc) I did notice that you are running on a slower machine. Digium finally got my configuration running by inserting some delays into my startup file /etc/rc.d/rc.local, which now looks like this: rmmod usb-uhci modprobe usb-uhci modprobe wcfxo modprobe wcusb sleep 1 ztcfg -vv sleep 1 (and then asterisk if you want to start it automatically) Good luck, Scott Scott M. Stingel Emerging Voice Technology Inc. Email: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] URL:www.evtmedia.com http://www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tilghman Lesher Sent: Monday, July 14, 2003 6:03 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Odd output from X100P I'm attempting to configure a machine with a DevKit Lite (X100P S100U). After I modprobe wcfxo, the machine goes into some kind of loop after about 10 seconds, where it spits out what appears to be 32-bit addresses, ad infinitum. At this point, the machine becomes completely unresponsive to keyboard input (i.e. Ctrl-C, Ctrl-Alt-Del, even CapsLock and NumLock states [lights] cannot be changed). /etc/zaptel.conf contains: fxsks=1 fxoks=2 loadzone = us defaultzone=us The addresses are to the screen and are not logged to syslog. They are of the form [c053dc09] [c8009ec0], etc. The addresses appear to all be in the ranges C0xx and C8xx. The hardware is a PII-266, with 48MB RAM. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] EZ-Install
That sounds interesting... -- Original Message -- From: Matthew Hardeman [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] Date: Mon, 14 Jul 2003 11:11:35 -0500 Maybe it's just me... But I fail to see the reasoning behind branching to a whole new distribution just to support an easy, out of the box Asterisk install. Perhaps just the creation of an RPM package with a basic configuration would be the ticket? The one potential exception to this would be if you wrote a distribution with advanced hardware detection and preconfiguration such that during the install process, Digium hardware is detected and you can go ahead and configure spans and channels, etc. In that case, the distribution might have some unique value. Short of that, I cannot imagine a new distribution just to package together a pre-configured Asterisk configuration. Even if you wrote an installation process like that, couldn't it be just as well implemented with a clever RPM-based installation and some nice plain old userspace configuration tools? Matt Hardeman PaperSoft -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of jltaylor Sent: Monday, July 14, 2003 11:00 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] EZ-Install Not CD based. Just CD install. When you reboot Linux with asterisk is installed. You could add any other tools you think are necessary. User then just does config. -- Original Message -- From: Steven Critchfield [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] Date: 14 Jul 2003 10:18:24 -0500 On Mon, 2003-07-14 at 10:34, jltaylor wrote: Has anyone thought about an ISO file that could be used to make a CD for a bootable install for a basic Linux/Asterisk system? Just re-boot and config. Might be interesting to build based off of a knoppix cd, but then what do you store the configs to? -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- James Taylor [EMAIL PROTECTED] 903-793-1953 -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- James Taylor [EMAIL PROTECTED] 903-793-1953 -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] audio pause/delay problems
I use IAX2 over a 2000mile loop from my home to the office using GSM and have no problems as long as the lag is low. Most of the time you can't tell the difference between VoIP and PSTN on the phones at home. On Mon, 2003-07-14 at 12:30, Jan Rychter wrote: I'm curious. Isn't anyone else noticing these problems? Or are people simply not using asterisk for VoIP connectivity over wide-area networks this way? Or does it go away with g729 or other proprietary codecs? --J. Jan == Jan Rychter [EMAIL PROTECTED] writes: John == John Todd [EMAIL PROTECTED] writes: John For what it's worth, I have noticed the same problem, but I think John the problem is in IAX2, since my long-haul portions of the John diagram were over IAX2, while my SIP clients are almost always John sitting on the same LAN as the Asterisk server. Jan I have noticed these problems both in this kind of setup and in a Jan SIP call to a remote Asterisk server. John What codec were you testing with over IAX2? Jan GSM. Having investigated this a bit more, it turns out that using alaw instead of gsm on the IAX2 link makes the problem go away. It seems the jitter settings start working then. Any hints? I'd prefer not to be stuck with 80kbps per call... --J. [I have sent a message about SIP problems via gmane, but it seems the list is gatewayed one-way only...] The message was: I've been trying to use Asterisk as a SIP-PSTN gateway. It runs fine when the SIP client is on the local network and there is not packet loss. But now I've tried running a remote client (halfway around the globe) -- this works great until some packets get lost. After that it seems that either my client (linphone) or Asterisk doesn't want to resynchronize -- what gets played back is all voice packets as they have been received. This creates an increasing lag in the conversation and the only way I've found to fix it is to disconnect and reconnect again. Is anyone else seeing this? Is it linphone's fault, or is it expected behavior? Now, I have tried running another * on my side of the link. The setup then becomes: linphone - * - internet (IAX2) - * - PSTN (or echo). I'm testing with the echo application (GSM used everywhere) and I'm getting the same thing: everything seems to work, but sooner or later there is an audio pause and the delay grows. It never gets back to normal. I've had it grow to as much as 10s. What makes it even more surprising is the network performance. I've had ping running in the background, same TOS settings, 10 packets per second. It shows that my RTT is (min/avg/max/mdev) 220/229/287/8.85 with 0% loss! That's a pretty good network. So where do the pauses and delays come from? --J. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 7960 Transfer Conference
Hi All, I need some help w/supervised transfer and conference w/a 7940 phone. When I do a blind transfer the calls go through great, but when I do supervised transfer the 7940 tells me Transfer Denied. When I do a conference call I hit the conf key and then dial the next extension. The new call connects and I hit conf again but the calls do not get bridged. Any Suggestions? I'm using the config files from http://www.loligo.com/asterisk/Cisco/79xx/2003-06-20.from-ftpeng.cisco.com/ Thanks, TL -- Todd Lieberman 800-675-3192
Re: [Asterisk-Users] Odd output from X100P
On Monday 14 July 2003 12:39 pm, Scott Stingel wrote: On Monday, July 14, 2003, Tilghman Lesher wrote: I'm attempting to configure a machine with a DevKit Lite (X100P S100U). After I modprobe wcfxo, the machine goes into some kind of loop after about 10 seconds, where it spits out what appears to be 32-bit addresses, ad infinitum. At this point, the machine becomes completely unresponsive to keyboard input (i.e. Ctrl-C, Ctrl-Alt-Del, even CapsLock and NumLock states [lights] cannot be changed). /etc/zaptel.conf contains: fxsks=1 fxoks=2 loadzone = us defaultzone=us The addresses are to the screen and are not logged to syslog. They are of the form [c053dc09] [c8009ec0], etc. The addresses appear to all be in the ranges C0xx and C8xx. The hardware is a PII-266, with 48MB RAM. I recently had a lot of problems getting the DevLit kit working out of the box, even using the configurations supplied on the floppy that came with the kit. I didn't have the problem you are experiencing though, which sounds like some kind of hardware conflict to me. I would suggest contacting [EMAIL PROTECTED], and they are pretty good about diagnosing the problem (if you give them a login to your machine via ssh etc) Unfortunately, the machine is with a friend and not accessible via ssh. I did notice that you are running on a slower machine. Digium finally got my configuration running by inserting some delays into my startup file /etc/rc.d/rc.local, which now looks like this: rmmod usb-uhci modprobe usb-uhci modprobe wcfxo modprobe wcusb sleep 1 ztcfg -vv sleep 1 Noted. However, given that at the time of this problem, I have not yet probed wcusb, so the problem seems to be with the X100P driver, not the USB device. And the messages don't hit immediately after probing wcfxo, but there's a lag of approximately 10 seconds (where the keyboard continues to be functional) before the messages start. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hardware Vendors
On Mon, 2003-07-14 at 11:37, Matthew Hardeman wrote: Hi All! Can anyone direct me to any websites / manufacturers out there who are making small, put-it-in-the-closet-and-forget-it type systems for building routers, home gateway servers, that sort of thing? My fantasy machine for this purpose would be along the lines of a mini-itx system with external power supply, dual Ethernet interfaces on board, and one PCI slot available. If it had one real serial port on it, that would be great too. Am I dreaming, or does it exist for a reasonable price? I would be willing to go the 500 MHz 1 GHz range. Something without a fan would be really nice. Im basically looking for a system that someone out there is stamping out in quantities and isnt too outrageous in price. Does it exist, and if so who sells it? It seems to me a system like the above described would be perfect for building out a home gateway / home asterisk server Just because a company makes a lot doesn't mean the price drops. The type of device you are asking for is built by cisco, but the cost isn't near what you asked for. Asterisk can fill that role for the most part, but it expects you to do some work to get it there. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: X100P mod or USB relay box, RE: [Asterisk-Users] Line OverrideDevice
On Mon, 2003-07-14 at 12:10, Reed Wade wrote: At 11:34 AM 7/14/2003 -0500, Steven Critchfield wrote: One wouldn't use a X100P in a serious system. How so? I assume you're talking about scale and not reliability. We get a relatively small number of calls but any one of them could be worth a large stack of cash for our business. A stinky phone system can make us look bad. The main reason I'm looking at Asterisk is to improve the reliability and control over our phone system. All the other great things it provides really are secondary for the folks who pay my salary. I agree that reliability is THE most important item on a phone system, and if you read the list you will see that most the problems are analog related. So my point is that analog signaling is too problematic for a phone system most of the time. Only if you aren't pulling power from the USB bus. There isn't much there. There may be just enough depending on how many relays are needed, but it would be too close. I agree, better off not trying to get power from there. I do like the idea of some kind of watchdog functionality. Simply having power isn't sufficient to trust that a call is getting routed. This makes me think that you could take this a step further too and incorporate an external power supply and a relay that could interupt mains power so that you could power cycle the PC if the watchdog had power to operate and the PC wasn't responding or generating pings. Then a properly configured machine would start the services up on it's own and move on. This power cycle type of device would have saved me a few minutes of downtime the other day when I froze the kernel on our main phone system. As it was, I just called our colo facility and told them what machine to power cycle. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MSN Messenger 4.7 vs 5.0
On Monday 14 July 2003 12:24 pm, Rainer Jochem wrote: Then I thought that it would be a good idea to do an upgrade to v5.0... (yes, never touch a running system ;) But now - there's only the possibility to create and use Passport-accounts. Did I just don't find it or isn't there any chance to do SIP calls with Messenger 5.0 any more? That's correct. MSN Messenger 5.0 removed all interoperability from the client. Considering it's from Microsoft, are you really surprised? -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] G729 licensing
Missing something? No... So far as I'm aware there is no freely available G729 codec available that will run under Linux... Kind of funny that there *is* one for Windows, isn't it? As an aside, though, what kind of equipment are you using, and what circumstances are you communicating in? ALAW ULAW make great codecs for use on a LAN. :) I've also had great luck using GSM with the Snom200 running the very latest firmware. (1.18s) It's not yet posted on their website, but they will give you a link to it should you write their support team... Matt Hardeman PaperSoft -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jan Rychter Sent: Monday, July 14, 2003 12:31 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] G729 licensing This is a MIME-formatted message. If you see this text it means that your E-mail software does not support MIME-formatted messages. --=_megabox.papersoft.com-1525-1058205320-0001-2 Content-Transfer-Encoding: quoted-printable Hi, I'm looking for a good codec to use on a personal VoIP setup. It is strictly for my personal use, I'll never resell it, make money or it, or whatever. It seems a free personal-use G729 codec is available as a WIN32 library. I find it puzzling that at the same time one has to pay license fees to use it under Linux, even non-commercially. I was wondering -- am I missing something? =2D-J. --=_megabox.papersoft.com-1525-1058205320-0001-2 Content-Type: application/pgp-signature Content-Transfer-Encoding: 7bit -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.2 (GNU/Linux) iD8DBQA/EujILth4/7/QhDoRAty1AJ9z5xm6Zkj/oiYkm1buSfjceuC2UQCgh91y FqD3wdGNWgDXjrZLPD5nkcY= =SUZa -END PGP SIGNATURE- --=_megabox.papersoft.com-1525-1058205320-0001-2-- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and VMWare
Hi John, Thanks for your effort to make me buy Call Manager..:-) Maybe a 2K$ server with a couple of 2+ GHZ Xeons and 4GB of RAM will be good enough to run just the Web interface of the Call Manager... If running a maximum of two simultaneous audio calls through Asterisk installed over VMWare is a far too big job for my computer, then you're right. In between I have found an old Compaq Armada notebook who does the job very well, but unfortunately without any possibility to add any Digium hardware to it. Thanks to all of you who have tried to answer me to my question and I consider this issue closed. Dan - Original Message - From: John Laur [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, July 14, 2003 5:49 PM Subject: 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
Re: X100P mod or USB relay box, RE: [Asterisk-Users] Line Override Device
At 12:57 PM 7/14/2003 -0500, you wrote: This makes me think that you could take this a step further too and incorporate an external power supply and a relay that could interupt mains power so that you could power cycle the PC if the watchdog had power to operate and the PC wasn't responding or generating pings. i like that -reed ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Hardware Vendors
I live in California, and saw one of those cube PC's in Frye's for a few hundred dollars ($400??). Really tiny, everything contained inside. Had one PCI slot. I thought it would be nice for demos since it was so easily shipped. I think the processor was 800 MHz or so. Would have preferred 2 PCI slots... I'm in the UK now - when I'm back next week I'll try and find the manufacturer. Regards Scott Scott M. Stingel Emerging Voice Technology Inc. Palo Alto, California and London, England Email: [EMAIL PROTECTED] URL:www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Hardeman Sent: Monday, July 14, 2003 5:38 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Hardware Vendors Hi All! Can anyone direct me to any websites / manufacturers out there who are making small, put-it-in-the-closet-and-forget-it type systems for building routers, home gateway servers, that sort of thing? My fantasy machine for this purpose would be along the lines of a mini-itx system with external power supply, dual Ethernet interfaces on board, and one PCI slot available. If it had one real serial port on it, that would be great too. Am I dreaming, or does it exist for a reasonable price? I would be willing to go the 500 MHz - 1 GHz range. Something without a fan would be really nice. I'm basically looking for a system that someone out there is stamping out in quantities and isn't too outrageous in price. Does it exist, and if so who sells it? It seems to me a system like the above described would be perfect for building out a home gateway / home asterisk server. Matt Hardeman PaperSoft ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MSN Messenger 4.7 vs 5.0
You can have both 4.7 and 5.0 installed at the same time. Just download the 4.7 version from: http://www.microsoft.com/exchange/downloads/2000/IMClient47.asp Or, better yet, try X-Lite from www.xten.com. It's an excellent Windows SIP client, and supports low bandwidth codecs (iLBC, Speex, and GSM) with Asterisk. -wade -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Tilghman Lesher Sent: Monday, July 14, 2003 1:56 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] MSN Messenger 4.7 vs 5.0 On Monday 14 July 2003 12:24 pm, Rainer Jochem wrote: Then I thought that it would be a good idea to do an upgrade to v5.0... (yes, never touch a running system ;) But now - there's only the possibility to create and use Passport-accounts. Did I just don't find it or isn't there any chance to do SIP calls with Messenger 5.0 any more? That's correct. MSN Messenger 5.0 removed all interoperability from the client. Considering it's from Microsoft, are you really surprised? -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MSN Messenger 4.7 vs 5.0
That's correct. MSN Messenger 5.0 removed all interoperability from the client. Considering it's from Microsoft, are you really surprised? Not really. But my first thoughts were that I just missed some checkbox or very hidden option behind three advanced option-buttons :) Thanks, Rainer -- http://graphics.cs.uni-sb.de/VoIP/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 7960 Transfer Call drop problem
Title: Message
Hi,
I'm having problemswith transfer from an analog line via a X100p and Cisco
7960's running SIP.
With an
attended transfer the a call comes in, I transfer it to another 7960, they
answer I announce the call, press transfer again, the two parties talk for 1-2
seconds then the analog line drops, though the Cisco phone is not aware of this,
i.e. nothing on the screen changes. The console output for this is below.
Interestingly enough I seem to have the same problem with an incoming SIP call,
transferring it to another SIP ext, console output from that below as
well.
With a blind transfer a call comes in, I
transfer it to another extension, the analog caller hears the hold music, the
7960 that was transferred the call acts as if it is online with the call but
isn't. If the extension that was transferred the call puts the line on hold and
picks it up then the lines are connected fine.
Analog to SIP
transfer--
-- Zap/1-1 answered
SIP/206-369e -- Started music on hold, class 'default', on
Zap/1-1 -- Executing Macro("SIP/206-bcd1",
"stdexten|SIP/202|202") in new stack -- Executing
Dial("SIP/206-bcd1", "SIP/202|15") in new stack -- Called
202 -- SIP/202-7264 is ringing --
SIP/202-7264 answered SIP/206-bcd1 -- Attempting native
bridge of SIP/206-bcd1 and SIP/202-7264 -- Started music
on hold, class 'default', on SIP/202-7264 -- Stopped music
on hold on SIP/202-7264 -- Stopped music on hold on
Zap/1-1 == Spawn extension (intern-ext, 91415XXX, 1) exited non-zero on
'SIP/206-369e' == Spawn extension (macro-stdexten, s, 1) exited
non-zero on 'Zap/1-1' in macro 'stdexten' == Spawn extension
(intern-ext, 202, 1) exited non-zero on 'Zap/1-1' --
Hungup 'Zap/1-1'
SIP to SIP
transfer--
--
Executing Macro("SIP/206-effd", "stdexten|SIP/255|255") in new
stack -- Executing Dial("SIP/206-effd", "SIP/255|15") in
new stack -- Called 255 --
SIP/255-8cd8 is ringing -- SIP/255-8cd8 answered
SIP/206-effd -- Attempting native bridge of SIP/206-effd
and SIP/255-8cd8 -- Started music on hold, class
'default', on SIP/255-8cd8 -- Executing
Macro("SIP/206-8437", "stdexten|SIP/202|202") in new stack
-- Executing Dial("SIP/206-8437", "SIP/202|15") in new
stack -- Called 202 -- SIP/202-5c6b
is ringing -- SIP/202-5c6b answered
SIP/206-8437 -- Attempting native bridge of SIP/206-8437
and SIP/202-5c6b -- Started music on hold, class
'default', on SIP/202-5c6b -- Stopped music on hold on
SIP/202-5c6b -- Stopped music on hold on
SIP/255-8cd8 -- Attempting native bridge of SIP/206-effd
and SIP/255-8cd8 -- Attempting native bridge of
SIP/255-8cd8 and SIP/202-5c6b -- Got SIP response 481
"Call Leg/Transaction Does Not Exist" back from 67.xxx.xxx.xxx ==
Spawn extension (macro-stdexten, s, 1) exited non-zero on 'SIP/206-effd' in
macro 'stdexten' == Spawn extension (intern-ext, s, 1) exited non-zero
on 'SIP/206-effd' == Spawn extension (macro-stdexten, s, 1) exited
non-zero on 'SIP/255-8cd8' in macro 'stdexten'
Ideas?
Thanks,
Justin
[Asterisk-Users] Remote Agents
Hi, First a little background: the company that I work at have currently been using a Windows-based PBX solution called Televantage. We are primarily a linux-based development shop but originally went with Televantage on the recomendation of someone who no longer works with us. Suffice to say, we have not been happy with that solution and I have been investigating Asterisk as a replacement to that product. I have recently ordered and recieved the Developers Kit (TDM), and have got a working system up an running with one extension. So far I have really enjoyed working with Asterisk, however I have a few questions about some features. One of the key features we need is the Remote Agent, I am not sure how this works and was wondering if someone could give me some information on that. We would like to have calls routed through Asterisk to remote agents at home and then have a screen-pop on their PCs that would give details of the incoming call. We have an ISDN PRI connection through which the calls will be routed. My other question is concerning the scability of Asterisk - what sort of stats are there on how Asterisk can scale? Also, is there some distributed architecture features? I would like to be able to scale Asterisk over multiple servers and databases, does anyone have any information on whether this can currently be done with Asterisk and if so, how it would be accomplished? If anyone could give some information about this it would be greatly appreciated. Many thanks, Derek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] audio pause/delay problems
This happens to me as I mention below, but only rarely. What is your CVS version? JT I'm curious. Isn't anyone else noticing these problems? Or are people simply not using asterisk for VoIP connectivity over wide-area networks this way? Or does it go away with g729 or other proprietary codecs? --J. Jan == Jan Rychter [EMAIL PROTECTED] writes: John == John Todd [EMAIL PROTECTED] writes: John For what it's worth, I have noticed the same problem, but I think John the problem is in IAX2, since my long-haul portions of the John diagram were over IAX2, while my SIP clients are almost always John sitting on the same LAN as the Asterisk server. Jan I have noticed these problems both in this kind of setup and in a Jan SIP call to a remote Asterisk server. John What codec were you testing with over IAX2? Jan GSM. Having investigated this a bit more, it turns out that using alaw instead of gsm on the IAX2 link makes the problem go away. It seems the jitter settings start working then. Any hints? I'd prefer not to be stuck with 80kbps per call... --J. [I have sent a message about SIP problems via gmane, but it seems the list is gatewayed one-way only...] The message was: I've been trying to use Asterisk as a SIP-PSTN gateway. It runs fine when the SIP client is on the local network and there is not packet loss. But now I've tried running a remote client (halfway around the globe) -- this works great until some packets get lost. After that it seems that either my client (linphone) or Asterisk doesn't want to resynchronize -- what gets played back is all voice packets as they have been received. This creates an increasing lag in the conversation and the only way I've found to fix it is to disconnect and reconnect again. Is anyone else seeing this? Is it linphone's fault, or is it expected behavior? Now, I have tried running another * on my side of the link. The setup then becomes: linphone - * - internet (IAX2) - * - PSTN (or echo). I'm testing with the echo application (GSM used everywhere) and I'm getting the same thing: everything seems to work, but sooner or later there is an audio pause and the delay grows. It never gets back to normal. I've had it grow to as much as 10s. What makes it even more surprising is the network performance. I've had ping running in the background, same TOS settings, 10 packets per second. It shows that my RTT is (min/avg/max/mdev) 220/229/287/8.85 with 0% loss! That's a pretty good network. So where do the pauses and delays come from? --J. Content-Type: application/pgp-signature Attachment converted: PrivateSpace:Untitled 302 (/) (04203330) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk and VMWare
Agreed. Do not try and run Asterisk within VMWare. I use VMWare day in and day out but VMWare (even GSX) is not the place to be running Asterisk. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Dan Sent: Monday, July 14, 2003 1:15 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk and VMWare Hi John, Thanks for your effort to make me buy Call Manager..:-) Maybe a 2K$ server with a couple of 2+ GHZ Xeons and 4GB of RAM will be good enough to run just the Web interface of the Call Manager... If running a maximum of two simultaneous audio calls through Asterisk installed over VMWare is a far too big job for my computer, then you're right. In between I have found an old Compaq Armada notebook who does the job very well, but unfortunately without any possibility to add any Digium hardware to it. Thanks to all of you who have tried to answer me to my question and I consider this issue closed. Dan - Original Message - From: John Laur [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, July 14, 2003 5:49 PM Subject: 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
Re: [Asterisk-Users] MSN Messenger 4.7 vs 5.0
On Mon, 14 Jul 2003 12:55:51 -0500, Tilghman Lesher [EMAIL PROTECTED] wrote: On Monday 14 July 2003 12:24 pm, Rainer Jochem wrote: Then I thought that it would be a good idea to do an upgrade to v5.0... (yes, never touch a running system ;) But now - there's only the possibility to create and use Passport-accounts. Did I just don't find it or isn't there any chance to do SIP calls with Messenger 5.0 any more? That's correct. MSN Messenger 5.0 removed all interoperability from the client. Considering it's from Microsoft, are you really surprised? -Tilghman Part of the problem may be that you installed MSN Messenger in place of Windows Messenger that came with win2k. MSN Messenger is the OLDER protocol. I'm running Windows Messenger on WinXP 4.7 and it seems to be the latest (could be mistaken about it being the latest, but I just ran a windows update on it the other day...) Hope this helps... Stefan Johnson ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP immediate hangups with latest CVS
The resolution to this problem was provided by Martin @Digium, who said that there must be at least one or in the codec permissions line. In other words, I have allow=all at the top of the sip.conf file, but I should have something like: disallow=all allow=alaw Asterisk expects an ordering of some type to be defined with or statements, which is a bit confusing, but I suppose it makes logical sense. An easier way would have been to leave out the allow= line entirely. JT No change. I am unable to use SIP at all, apparently, in this latest revision. JT I had this a while back, and set canreinvite=no, and it fixed it. -d At 08:42 PM 7/11/2003 -0700, you wrote: I've been banging my head on this for several hours, and I have no idea what's going on. Maybe there is a very simple result, and I've been looking too hard at this this evening. This is a brand new system, and I'm wondering if there have been SIP bugs introduced in the latest CVS that are preventing from working what should be a stupendously simple test. - Cisco 7960 (non-NATed) - RH 8.0 - Asterisk CVS update as of ~8:00 PM EDT - full make clean; make install on [asterisk,zaptel,libpri] - 2ghz box with E1 card (that's pretty much not part of the equation) I have boiled the configuration down to an extremely (_extremely_) simple setup, and it does not work. SIP calls from the 7960 are hanging up almost immediately, with no audio getting through. It seems that the hangup happens just after the moment that the 7960 sends the ACK message (judging from the debug below, at least.) I have verified that demo-congrats is there, as my original problem stemmed from strange behavior with Zap dialing, and I kept simplifying, so this is the culmination of winnowing down the options to the most basic config. The same phone works flawlessly with other lines that are configured on it to other * servers. Here is my entire relevant configuration. It's as simple as you can get, really. I dial 14109850123 (as a test number - it matches the _1X. list) and I get an almost instant hangup. --- ;sip.conf [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = default ; Default for incoming calls dtmfmode=rfc2833 allow=all [3015321510] type=friend username=3015321510 secret=fluffernutter host=dynamic context=from-sip allow=all --- ;extensions.conf [general] static=yes writeprotect=yes [from-sip] exten = _1X.,1,SetCallerID(3015321510) exten = _1X.,2,Answer exten = _1X.,3,Playback(demo-congrats) exten = h,1,Hangup exten = t,1,Hangup exten = i,1,Hangup --- Other strange notes: - quite often, when launching with -gcd I get a segfault. I have the cores, if anyone is interested. - I have almost identical systems (same hardware, same MB, etc.) churning away with no problems with slightly older revs of code *CLI Sip read: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 128.151.224.33:5060 From: 3015321510 sip:[EMAIL PROTECTED];tag=0002b9eb0ef4012c1a228361-11beb479 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] Date: Sat, 12 Jul 2003 03:24:34 GMT CSeq: 101 INVITE User-Agent: CSCO/4 Contact: sip:[EMAIL PROTECTED]:5060 Expires: 180 Content-Type: application/sdp Content-Length: 247 Accept: application/sdp Remote-Party-ID: 3015321510 sip:[EMAIL PROTECTED];party=calling;id-type=subscriber;privacy=off;screen=no v=0 o=Cisco-SIPUA 15975 21108 IN IP4 128.151.224.33 s=SIP Call c=IN IP4 128.151.224.33 t=0 0 m=audio 19364 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 14 headers, 11 lines Using latest request as basis request Sending to 128.151.224.33 : 5060 (non-NAT) Found audio format 0 Found audio format 8 Found audio format 18 Found audio format 101 Found description format PCMU Found description format PCMA Found description format G729 Found description format telephone-event Capabilities: us - 2147483647, them - 268/0, combined - 268 Non-codec capabilities: us - 1, them - 1, combined - 1 Reliably Transmitting (no NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 128.151.224.33:5060 From: 3015321510 sip:[EMAIL PROTECTED];tag=0002b9eb0ef4012c1a228361-11beb479 To: sip:[EMAIL PROTECTED];tag=as74174b76 Call-ID: [EMAIL PROTECTED] CSeq: 101 INVITE User-Agent: Asterisk PBX Contact: Proxy-Authenticate: Digest realm=asterisk, nonce=2c9c06be Content-Length: 0 to 128.151.224.33:5060 Sip read: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 128.151.224.33:5060 From: 3015321510 sip:[EMAIL PROTECTED];tag=0002b9eb0ef4012c1a228361-11beb479 To: sip:[EMAIL PROTECTED];tag=as74174b76 Call-ID: [EMAIL PROTECTED] Date: Sat, 12 Jul 2003 03:24:34 GMT CSeq: 101 ACK Content-Length: 0 8 headers, 0 lines Sip read: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 128.151.224.33:5060 From: 3015321510 sip:[EMAIL
Re: [Asterisk-Users] Remote Agents
Derek, Screen pops Iin Windows) usually rely on the Microsoft TAPI system. Asterisk doesn't talk to TAPI as far as I know. I am trying to write a VB interface to the management system on Asterisk but the going is slow. We looked at TeleVantage and the cost for 30 agents and full reporting was outraegous. The features are excellent and support is good but we couldn't justify the 50k. We are having the queue app changed for our needs by Digium and will save a bundle. It will allow for remote agents to login and recieve calls from home. Check back often to see the progress on the new queue app. Asterisk scales to many PBX servers and they can all talk to each other and store call info on the same database. If you need reporting you will have to do that yourself with something like Crystal Reports. Good luck. Jim Friedeck Derek Barber wrote: Hi, First a little background: the company that I work at have currently been using a Windows-based PBX solution called Televantage. We are primarily a linux-based development shop but originally went with Televantage on the recomendation of someone who no longer works with us. Suffice to say, we have not been happy with that solution and I have been investigating Asterisk as a replacement to that product. I have recently ordered and recieved the Developers Kit (TDM), and have got a working system up an running with one extension. So far I have really enjoyed working with Asterisk, however I have a few questions about some features. One of the key features we need is the Remote Agent, I am not sure how this works and was wondering if someone could give me some information on that. We would like to have calls routed through Asterisk to remote agents at home and then have a screen-pop on their PCs that would give details of the incoming call. We have an ISDN PRI connection through which the calls will be routed. My other question is concerning the scability of Asterisk - what sort of stats are there on how Asterisk can scale? Also, is there some distributed architecture features? I would like to be able to scale Asterisk over multiple servers and databases, does anyone have any information on whether this can currently be done with Asterisk and if so, how it would be accomplished? If anyone could give some information about this it would be greatly appreciated. Many thanks, Derek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] New budgetone firmware
Hi. Has anyone experienced with the new firmware .77 ? There's Day Light Saving time now, but haven't time to play with it, till now. Matteo. -- Matteo Brancaleoni Espia System Administrator - IT services Website : http://www.espia.it Email : [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Using 2 PhoneJacks with Asterisk for Data calls.
Hi, I have recently discovered the project along with the PhoneJacks produced by quicknet, they could be the answer to something I have been looking into. I would like to be able to test using a dial-in server possibily also a Windows RAS server, however I only have 1 phone line. I was thinking that I create a setup like that illustrated below to solve the problem:- Ext 1000 Ext 2000 // / // /// /Asterix/// Dialin / / Client /-- / PhoneJack / -- / PBX / -- / PhoneJack / --/ or RAS / /// / /// Server / // / // Has anyone used Asterix in a similair problem or perhaps they can see potential problems. I have tried searching the archives for similair posts but did not come up with much of use. Thanks in advance for any help or advice. Regards Lee ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hardware Vendors
[EMAIL PROTECTED] wrote on 07/14/2003 12:37:33 PM: My fantasy machine for this purpose would be along the lines of a mini-itx system with external power supply, dual Ethernet interfaces on board, and one PCI slot available. If it had one real serial port on it, that would be great too. Am I dreaming, or does it exist for a reasonable price? I would be willing to go the 500 MHz 1 GHz range. Something without a fan would be really nice. Im basically looking for a system that someone out there is stamping out in quantities and isnt too outrageous in price. Does it exist, and if so who sells it? www.caseoutlet.com Via Eden 533MHz processor, no fans whatsoever. Runs like a PII 400MHz. They have cases that have 2 PCI slots. That's the biggest limitation: lack of PCI slots. We use these to sell Linux-based firewall computers for clients. They have run for well over a year with exactly zero crashes. With no moving parts (not even hard drives: we use DOM for the firewalls), there isn't a lot to go wrong. Having said all of that, I don't think they'll make good Asterisk boxes. 2 PCI slots isn't much and 400MHz PII-type performance isn't great (though you can get 750MHz or so of PIII performance from the new 1GHz CPU's if you don't mind a CPU fan). But if you can live with that, they're very nice. Don't forget to target the i586 architecture. The VIA CPU's don't have an instruction (CMOV? CMPXCHNG? something like that) that the Intels do and that CGG uses with an i686 target. Unfortunately, the VIA gets detected as an i686... Tim Massey ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk and VMWare
agreed here too. You cannot hook into real hardware interrupts for timing in a VM. A cheap small pentium can run asterisk (I have a dual 200MHz Pentium Pro), but as soon as you add the hardware emulation layer of any VM real/pseudo realtime needs are not met. Even using the USB digium device, the VM cannot handle isosyncronos IO. quote who=Erik Anderson Agreed. Do not try and run Asterisk within VMWare. I use VMWare day in and day out but VMWare (even GSX) is not the place to be running Asterisk. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Dan Sent: Monday, July 14, 2003 1:15 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk and VMWare Hi John, Thanks for your effort to make me buy Call Manager..:-) Maybe a 2K$ server with a couple of 2+ GHZ Xeons and 4GB of RAM will be good enough to run just the Web interface of the Call Manager... If running a maximum of two simultaneous audio calls through Asterisk installed over VMWare is a far too big job for my computer, then you're right. In between I have found an old Compaq Armada notebook who does the job very well, but unfortunately without any possibility to add any Digium hardware to it. Thanks to all of you who have tried to answer me to my question and I consider this issue closed. Dan - Original Message - From: John Laur [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, July 14, 2003 5:49 PM Subject: 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
Re: [Asterisk-Users] SIP call from one extention to another
On Fri, 11 Jul 2003 19:28:38 +, Serge Mankovski [EMAIL PROTECTED] wrote: I am trying to call from Linphone on extention 109 to Xlite on extention 108 and I get this error -- to 216.75.167.18:5068 WARNING[98315]: File pbx.c, Line 1133 (pbx_extension_helper): No application 'Dial ' for extension (sip, 108, 1) == Spawn extension (sip, 108, 1) exited non-zero on 'SIP/sergeXlite-be43' - Can you tell me what might be wrong with my setup? You didn't paste any of the relevant, but it sounds like you might have an extraneous space in there. -rt -- Ryan Tucker Network Engineer NetAccess, Inc. 1159 Pittsford-Victor Road Bldg. 5, Suite 140 Pittsford, New York 14534 585-419-8200 www.netacc.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fwd:[Vocal] Question about Cisco IP hard phones
Interesting notes on the 79xx series. The 7920 is the wireless phone; not mentioned here. For a more complete guide to Cisco's phones, see: http://www.cisco.com/en/US/products/hw/phones/ps379/products_data_sheets_list.html The 7902 is the very inexpensive Cisco phone, and it looks like it will be SCCP (Skinny) only. Twiddling my thumbs here waiting for the chan_sccp to appear. ;-) JT Reply-To: [EMAIL PROTECTED] From: David Kelly [EMAIL PROTECTED] To: Chok Lam [EMAIL PROTECTED], [EMAIL PROTECTED] Org [EMAIL PROTECTED] Subject: RE: [Vocal] Question about Cisco IP hard phones Date: Mon, 14 Jul 2003 11:56:45 -0700 Folks, For the time being, the low-end Cisco IP phones, 7902G and 7912G support SCCP only. The 7905G supports both H.323 and SCCP, but we are not prioritizing new development on the H.323 load. This load is a legacy from the 7905 phone that was released in 2003 and EOL'd last week. This autumn, we will release a SIP image for the 7905G and 7912G. There are no plans to release a SIP image for the 7902G. David [snip] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Remote Agents
Jim, thanks for your reply, that is very helpful. We actually are planning on using Linux for our remote agents so the screen-pop will be a linux application. Do you know if that feature is supported in Asterisk or do you know if it would be easy to implement if it isn't supported yet? Thanks, Derek On Mon, 2003-07-14 at 12:27, Jim Friedeck wrote: Derek, Screen pops Iin Windows) usually rely on the Microsoft TAPI system. Asterisk doesn't talk to TAPI as far as I know. I am trying to write a VB interface to the management system on Asterisk but the going is slow. We looked at TeleVantage and the cost for 30 agents and full reporting was outraegous. The features are excellent and support is good but we couldn't justify the 50k. We are having the queue app changed for our needs by Digium and will save a bundle. It will allow for remote agents to login and recieve calls from home. Check back often to see the progress on the new queue app. Asterisk scales to many PBX servers and they can all talk to each other and store call info on the same database. If you need reporting you will have to do that yourself with something like Crystal Reports. Good luck. Jim Friedeck Derek Barber wrote: Hi, First a little background: the company that I work at have currently been using a Windows-based PBX solution called Televantage. We are primarily a linux-based development shop but originally went with Televantage on the recomendation of someone who no longer works with us. Suffice to say, we have not been happy with that solution and I have been investigating Asterisk as a replacement to that product. I have recently ordered and recieved the Developers Kit (TDM), and have got a working system up an running with one extension. So far I have really enjoyed working with Asterisk, however I have a few questions about some features. One of the key features we need is the Remote Agent, I am not sure how this works and was wondering if someone could give me some information on that. We would like to have calls routed through Asterisk to remote agents at home and then have a screen-pop on their PCs that would give details of the incoming call. We have an ISDN PRI connection through which the calls will be routed. My other question is concerning the scability of Asterisk - what sort of stats are there on how Asterisk can scale? Also, is there some distributed architecture features? I would like to be able to scale Asterisk over multiple servers and databases, does anyone have any information on whether this can currently be done with Asterisk and if so, how it would be accomplished? If anyone could give some information about this it would be greatly appreciated. Many thanks, Derek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ 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.
Don't do it. You will waste your money. Quicknet hardware is not worth even half of what they are charging. Jeremy McNamara Lee W wrote: Hi, I have recently discovered the project along with the PhoneJacks produced by quicknet, they could be the answer to something I have been looking into. I would like to be able to test using a dial-in server possibily also a Windows RAS server, however I only have 1 phone line. I was thinking that I create a setup like that illustrated below to solve the problem:- Ext 1000 Ext 2000 // / // /// /Asterix/// Dialin / / Client /-- / PhoneJack / -- / PBX / -- / PhoneJack / --/ or RAS / /// / /// Server / // / // Has anyone used Asterix in a similair problem or perhaps they can see potential problems. I have tried searching the archives for similair posts but did not come up with much of use. Thanks in advance for any help or advice. Regards Lee ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Remote Agents
you can probably code something rather easily with the management interface On Monday 14 July 2003 21:43, Derek Barber wrote: Jim, thanks for your reply, that is very helpful. We actually are planning on using Linux for our remote agents so the screen-pop will be a linux application. Do you know if that feature is supported in Asterisk or do you know if it would be easy to implement if it isn't supported yet? Thanks, Derek On Mon, 2003-07-14 at 12:27, Jim Friedeck wrote: Derek, Screen pops Iin Windows) usually rely on the Microsoft TAPI system. Asterisk doesn't talk to TAPI as far as I know. I am trying to write a VB interface to the management system on Asterisk but the going is slow. We looked at TeleVantage and the cost for 30 agents and full reporting was outraegous. The features are excellent and support is good but we couldn't justify the 50k. We are having the queue app changed for our needs by Digium and will save a bundle. It will allow for remote agents to login and recieve calls from home. Check back often to see the progress on the new queue app. Asterisk scales to many PBX servers and they can all talk to each other and store call info on the same database. If you need reporting you will have to do that yourself with something like Crystal Reports. Good luck. Jim Friedeck Derek Barber wrote: Hi, First a little background: the company that I work at have currently been using a Windows-based PBX solution called Televantage. We are primarily a linux-based development shop but originally went with Televantage on the recomendation of someone who no longer works with us. Suffice to say, we have not been happy with that solution and I have been investigating Asterisk as a replacement to that product. I have recently ordered and recieved the Developers Kit (TDM), and have got a working system up an running with one extension. So far I have really enjoyed working with Asterisk, however I have a few questions about some features. One of the key features we need is the Remote Agent, I am not sure how this works and was wondering if someone could give me some information on that. We would like to have calls routed through Asterisk to remote agents at home and then have a screen-pop on their PCs that would give details of the incoming call. We have an ISDN PRI connection through which the calls will be routed. My other question is concerning the scability of Asterisk - what sort of stats are there on how Asterisk can scale? Also, is there some distributed architecture features? I would like to be able to scale Asterisk over multiple servers and databases, does anyone have any information on whether this can currently be done with Asterisk and if so, how it would be accomplished? If anyone could give some information about this it would be greatly appreciated. Many thanks, Derek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael Bielicki Managing Director TAAN Consultants Ltd http://www.global-gateway.net/ -- This correspondence is for the named person's use only. It may contain confidential or legally privileged information or both. No confidentiality or privilege is waived or lost by any mistransmission. If you receive this correspondence in error, please immediately delete it from your system and notify the sender. You must not disclose, copy or rely on any part of this correspondence if you are not the intended recipient. Any opinions expressed in this message are those of the individual sender. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] payload framesize
is there any particular reason why there is no option to configure the codec framesizes in iax2 ? It would come rathrer handy to decide if you want less bandwidth or more robustness on the payload side ... -- Michael Bielicki Managing Director TAAN Consultants Ltd http://www.global-gateway.net/ -- This correspondence is for the named person's use only. It may contain confidential or legally privileged information or both. No confidentiality or privilege is waived or lost by any mistransmission. If you receive this correspondence in error, please immediately delete it from your system and notify the sender. You must not disclose, copy or rely on any part of this correspondence if you are not the intended recipient. Any opinions expressed in this message are those of the individual sender. ___ 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.
Jeremy McNamara wrote: Don't do it. You will waste your money. Quicknet hardware is not worth even half of what they are charging. Thanks for the heads-up. Do you know of any alternatives? 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). My main concern is compatibility according to the asterisk docs, the quicknet products are and drivers are also supplied as part of the kernel. Thanks again Regards Leee ___ 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] G729 licensing
Voiceage has a free one. Works good for s/w to s/w but I found it has a problem with s/w to h/w And so far they did not answer me.. I guess u get what u pay for / ? :) ..ya g.729 is a extreme $ play. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jan Rychter Sent: Monday, July 14, 2003 10:31 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] G729 licensing Hi, I'm looking for a good codec to use on a personal VoIP setup. It is strictly for my personal use, I'll never resell it, make money or it, or whatever. It seems a free personal-use G729 codec is available as a WIN32 library. I find it puzzling that at the same time one has to pay license fees to use it under Linux, even non-commercially. I was wondering -- am I missing something? --J. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Making Analog Phones Work
Hi, I have got my TDM400P working.(3 modules), asterisk dials Zap/1 and says Ringing but the analogue phone plugged in, does not ring, or does not have any tone when I pickup the handpiece. Here are by configs: zapata.conf: [channels] signalling = fxo_ks context=internal channel = 1-3 zaptel.conf: fxoks=1-3 Any ideas would be greatly appriciated Thanks Jay ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: X100P mod or USB relay box, RE: [Asterisk-Users] Line Override Device
The Voicetronix Openline6 and Openline12 cards have the functionality you want built in. You can configure (jumpers) which ports are FXO and which are FXS (in groups of 2 IIRC) and 1st FXO goes to 1st FXS etc. in case of power failure. Apparently these cards work with Asterisk (chan_vpb). I think cost is AU$1500 and AU$3000 for 6 and 12. cheers, Wooody On Mon, Jul 14, 2003 at 02:17:59PM -0400, Reed Wade wrote: At 12:57 PM 7/14/2003 -0500, you wrote: This makes me think that you could take this a step further too and incorporate an external power supply and a relay that could interupt mains power so that you could power cycle the PC if the watchdog had power to operate and the PC wasn't responding or generating pings. i like that -reed ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Woody ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] module : cdr_sybase.so
Agreed. I have programmed against FreeTDS a lot. It is a good stable product. It should not take any major effort to put in a cdr_tds.so This way we could connect to Sybase and MS SQL Server. If there is a demand for this kind of feature maybe we should make such a module. Erik -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of John Laur Sent: Monday, July 14, 2003 9:04 AM To: [EMAIL PROTECTED] Subject: 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 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] audio pause/delay problems
John == John Todd [EMAIL PROTECTED] writes: John This happens to me as I mention below, but only rarely. What is John your CVS version? The latest? E.g. I've tested 2 days ago. --J. I'm curious. Isn't anyone else noticing these problems? Or are people simply not using asterisk for VoIP connectivity over wide-area networks this way? Or does it go away with g729 or other proprietary codecs? --J. Jan == Jan Rychter [EMAIL PROTECTED] writes: John == John Todd [EMAIL PROTECTED] writes: John For what it's worth, I have noticed the same problem, but I think John the problem is in IAX2, since my long-haul portions of the John diagram were over IAX2, while my SIP clients are almost always John sitting on the same LAN as the Asterisk server. Jan I have noticed these problems both in this kind of setup and in a Jan SIP call to a remote Asterisk server. John What codec were you testing with over IAX2? Jan GSM. Having investigated this a bit more, it turns out that using alaw instead of gsm on the IAX2 link makes the problem go away. It seems the jitter settings start working then. Any hints? I'd prefer not to be stuck with 80kbps per call... --J. [I have sent a message about SIP problems via gmane, but it seems the list is gatewayed one-way only...] The message was: I've been trying to use Asterisk as a SIP-PSTN gateway. It runs fine when the SIP client is on the local network and there is not packet loss. But now I've tried running a remote client (halfway around the globe) -- this works great until some packets get lost. After that it seems that either my client (linphone) or Asterisk doesn't want to resynchronize -- what gets played back is all voice packets as they have been received. This creates an increasing lag in the conversation and the only way I've found to fix it is to disconnect and reconnect again. Is anyone else seeing this? Is it linphone's fault, or is it expected behavior? Now, I have tried running another * on my side of the link. The setup then becomes: linphone - * - internet (IAX2) - * - PSTN (or echo). I'm testing with the echo application (GSM used everywhere) and I'm getting the same thing: everything seems to work, but sooner or later there is an audio pause and the delay grows. It never gets back to normal. I've had it grow to as much as 10s. What makes it even more surprising is the network performance. I've had ping running in the background, same TOS settings, 10 packets per second. It shows that my RTT is (min/avg/max/mdev) 220/229/287/8.85 with 0% loss! That's a pretty good network. So where do the pauses and delays come from? --J. pgp0.pgp Description: PGP signature
Re: [Asterisk-Users] G729 licensing
Matthew == Matthew Hardeman [EMAIL PROTECTED] writes: Matthew Missing something? No... Matthew So far as I'm aware there is no freely available G729 codec Matthew available that will run under Linux... Kind of funny that Matthew there *is* one for Windows, isn't it? Yes, puzzling. I guess one might go the way the other projects have (like mplayer or xine video players) -- use the Windows DLLs under Linux. This can be done with a bit of glue code. Matthew As an aside, though, what kind of equipment are you using, and Matthew what circumstances are you communicating in? ALAW ULAW make Matthew great codecs for use on a LAN. :) I'm using gnomemeeting (sometimes also linphone, but gnomemeeting is much better), asterisk with oh323 on one end, asterisk with X100P on the other end, doing the bridging to PSTN there. alaw and ulaw are all good and great, but the distance between the two asterisks is 18 hops and 9 hours of time difference, so I'd really like to save on the bandwidth. GSM would actually be fine if it wasn't for the sync problems that I've reported. --J. Hi, I'm looking for a good codec to use on a personal VoIP setup. It is strictly for my personal use, I'll never resell it, make money or it, or whatever. It seems a free personal-use G729 codec is available as a WIN32 library. I find it puzzling that at the same time one has to pay license fees to use it under Linux, even non-commercially. I was wondering -- am I missing something? --J. pgp0.pgp Description: PGP signature
[Asterisk-Users] VXML?
Anyone know of anybody doing VXML with Asterisk and/or Linux? Tia Kevin ___ 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.
John Laur wrote: Quicknet has a big hand in that project too. Quicknet IS the hand in that project. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fwd:[Vocal] Question about Cisco IP hard phones
chan_sccp would be nice :) I've been playing around with the 7960's and have really enjoyed the 7960 as a desktop phone. It's physically well constructed, has a sturdy/heavy handset (a good thing in my book), a very pleasant user interface... And if you're willing to make changes to your network setup to accomodate it's presently finicky firmwarre, you'll be ok... The firmware issue for the 7960 SIP is yet to be resolved, but hopefully it'll come around... I think the entire 79XX lines of phones by Cisco has lots of promise, but we won't really see the others (than the 7960/7940) be much use in Asterisk until there is native support in Asterisk for sccp... I did hear a rumor that someone was working on it... Matt Hardeman PaperSoft - Original Message - From: John Todd [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, July 14, 2003 3:32 PM Subject: [Asterisk-Users] Fwd:[Vocal] Question about Cisco IP hard phones Interesting notes on the 79xx series. The 7920 is the wireless phone; not mentioned here. For a more complete guide to Cisco's phones, see: http://www.cisco.com/en/US/products/hw/phones/ps379/products_data_sheets_list.html The 7902 is the very inexpensive Cisco phone, and it looks like it will be SCCP (Skinny) only. Twiddling my thumbs here waiting for the chan_sccp to appear. ;-) JT Reply-To: [EMAIL PROTECTED] From: David Kelly [EMAIL PROTECTED] To: Chok Lam [EMAIL PROTECTED], [EMAIL PROTECTED] Org [EMAIL PROTECTED] Subject: RE: [Vocal] Question about Cisco IP hard phones Date: Mon, 14 Jul 2003 11:56:45 -0700 Folks, For the time being, the low-end Cisco IP phones, 7902G and 7912G support SCCP only. The 7905G supports both H.323 and SCCP, but we are not prioritizing new development on the H.323 load. This load is a legacy from the 7905 phone that was released in 2003 and EOL'd last week. This autumn, we will release a SIP image for the 7905G and 7912G. There are no plans to release a SIP image for the 7902G. David [snip] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
