Re: [Asterisk-Users] Command Reference
On Tue, 2005-04-05 at 11:27 -0500, Chris wrote: Is there a good place to get the command references for the Extensions.conf? The Wiki and other documentation seem to be rather limited and don't explain all the parameters. Souce code is great. From the CLI, show applications This returns a list of apps show application fooreturns foo's information -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dynamic Zap/{channel} allocation for out going possible?
On Sat, 2005-04-02 at 18:39 +0200, Etienne Pretorius wrote: Never Mind. oops. I just needed to play around with some syntax. Zap/1,2,3,4/$EXTEN Ps: Is there a better santax because 1-4 doesn't work. Look at groups in the /etc/asterisk/zaptel.conf Once you define your groups, you can just exten = _0,1,Dial(Zap/g1/$EXTEN) And asterisk will pick some available channel out of the channels defined in group 1 to use for dialing out. Hi All * users... Question: In extensions.conf - I am awaire that you can use macro's but what I am wondering about.. is that can you create a macro to do dynamic Zap channel allocation for a out going call? I don't want to reserve a channel/port in the TDM400P card for Out break calls, so i was just wandering if some1 could help me a bit over here. [outgoing] ;Dial 0 on the phone for external line exten = _0,1,Dial(Zap/4/$EXTEN) ;=== statically allocated to Zap/4 needs to be dynamic exten = _0,2,Goto(102) exten = _0,102,Congestion exten = _0,103,Hangup I'll apreciate any help in this regard. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Packetization
On Sat, 2005-04-02 at 21:16 -0500, Matt wrote: I'm aware that asterisk only supports 20ms packetization rates. Due to the fact that I will be using some voip devices on a wireless network which is highly sensative to framerate.. is there any way I can hard code the packetization rate at say 30 or 40ms and then compile astrisk? If so, can anyone in the know tell me what variables I need to look at to change? Are you sure your other devices support different packet sizes? Are you sure the added delay in audio delivery can be handled decently and not cause added echo? Have you considered what IAX trunking can do for you? It will reduce frame rate as you add channels since each packet will then hold the frames for each of the consecutive calls. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SuperMicro X5DE8-GG Motherboard Goes Kaput afterInstalling TE410P Card - Yikes!
On Thu, 2005-03-31 at 11:05 -0500, Tim Bass wrote: We installed one Digium TE410P in the PCIX slot and put the power cable back on. The machine tried to come up, but the TE410P card flashed red lights in all four ports and there was no video output, no motherboard beeps or anything. This was a very simple (1) shutdown, (2) remove power supply (3) install riser card and TE410P, and (4) reconnect power cord. Not sure, but I didn't think any of the Digium cards where PCIX compatible. The TE410P was compatible with a 64bit slot but nothing more. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SuperMicro X5DE8-GG Motherboard Goes Kaput after Installing TE410P Card - Yikes!
On Thu, 2005-03-31 at 00:32 -0500, Tim Bass wrote: Hello All, This is my first post. Sorry to post under such sad circumstances. Here is the situation: We installed a TE410P (today) in a SuperMicro 1U server today (Motherboard X5DE8-GG), which was running great until installing this card. After installing the card, the motherboard will not boot (no beeps or indicators) and there is no video output. The fans sign and some of the motherboard lights blink, but like a city with no nightlife, the board is, for all practical purposes, dead.We took the TE410P out and have tried just about very thing under the sun, including clearing the CMOS and, sad to say, the motherboard is still dead with no video out and no beeps. I have had something similar happen with a different card. Does the PSU fan pulse when you attempt to boot? If it is pulsing, you have a short. I have seen standoffs under a motherboard finally touch something that wasn't intended to and the system won't boot. The super micro 1u cases are not very stiff. It is an interesting engineering problem to make something that is hollow stiff with out being able to cross brace. So my suggestion is to pull the board out and maybe hook up to a different PSU with no chance of it shorting out to verify it is ok. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sangoma VS. Digium
On Thu, 2005-03-31 at 10:00 -0500, David Brodbeck wrote: -Original Message- From: Brian Capouch [mailto:[EMAIL PROTECTED] My understanding is that to an extent when we buy Sangoma we're putting the dagger to Digium. If anything puts the dagger to Digium it'll be their own inability to engineer reliable hardware. I appreciate what Digium has done for Asterisk, but reliability expectations for phone equipment are extremely high. I sympathize with people who need hardware that doesn't need to be restarted once a week just to do its job properly. If Digium can't deliver on those reliability expectations, and do it soon, people are going to switch to companies that can. And you know what? I don't blame them. Funny how I have 2 production machines that handle fairly large call loads without any stability issues. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AGI STREAM FILE command
On Mon, 2005-03-28 at 08:30 -0600, Bill Kervaski wrote: Has anyone had success with the AGI STREAM FILE command with the CVS? I can't get it to work with the debian 1.0.5 package or the CVS on Redhat or Debian. It's not syntax, I'm doing that right. It doesn't give me an error when I use AGI DEBUG, it doesn't even give a response, just goes right on to the next command. I put a SAY NUMBER 123 # before and after the STREAM FILE and they both work fine, returning 200 OK, etc. DO NOT SEND A DIGEST TO THE MAILING LIST You do not mention if you followed the suggestion someone else made about making sure you do not have the extension on the file. Without providing the line you are trying to make work, you can not make most of us believe you haven't made a mistake. There are way too many people using Asterisk AGI successfully for it to likely be a bug in the Asterisk code if your command is simple. With as much as I love Debian, the distribution is not the problem. Do not use the very old Debian asterisk packages though. They are so old as to have well known bugs. If you want to participate better in the mailing list, maybe you would be better off to remove the digest option from the mailing list and use a proper mail filter to split the list mail to a folder other than your inbox. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Which analog phones to use and why?
On Mon, 2005-03-28 at 12:29 -0500, [EMAIL PROTECTED] wrote: Hello! Now that I finally have my TDM board working, I want to move forward with using PBX functions. However, it seems cumbersome to use standard POTS telephones with Asterisk. I know that there are many of you installing even large systems based on channel banks and analog telephones. What phones are you using? How do you simulate phone system features on a phone that doesn't have extra buttons? Or are you all using ADSI telephones? It seems that for the price of a ADSI telephone (never mind the cost per channel of a channel bank and T1 card), you can get a good quality IP telephone. In that case, what is the appeal of analog? Depends on what functions you are trying to implement. Hold isn't hard on a regular phone. Transfer isn't hard. Voicemail access isn't hard. Beyond that, there isn't a lot that needs to be done. If you find that you need more functions, then you may need to move up to a SIP phone. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Verizon ISDN
On Mon, 2005-03-28 at 12:48 -0700, Kevin P. Fleming wrote: Brian G wrote: I'm looking to use Asterisk with Verizon ISDN centex service in the US. I'd be connecting to an NT1 so I'd need an S/T interface. Users would have SIP phones registered with Asterisk and sharing the ISDN lines. ISDN BRI interfacing into a PC is hard to do in the US... there are just not many (if any) cards available to do it. The few cards that are available only have firmware for Euro-ISDN, not NI-2 (and US BRI is _not_ the same as anywhere else in the world). There are other options, though: Adtran (and others) make boxes that can cross-convert multiple BRIs into a PRI, which could then be connected to Asterisk via a T-1 card. Not an inexpensive way to go, though. I looked at the Adit cards for that, and didn't see it as a simple conversion. The ADIT cards support putting the 2b+1d into 3 B channels on a T1. Then you have to deal with the signaling for the D channel speed differences. In BRI the D channel is 16k and the B channel it gets stuck into on PRI is 64k. There was another method of encoding that allowed multiple D channels to be combined into a B channel to get better density, but again, it wasn't a direct fit with the zapata libraries and didn't look easy to do. Does the Adtran way differ significantly enough to make this become easy? -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call files run at certain times
On Mon, 2005-03-28 at 15:55 -0600, Anton Krall wrote: Im checking the wiki for call files info and seems somebody has a wake up script that runs call files at certain times. Do you know if its possible to run a call file by using some other methods different from cron jobs or at? The wiki mentions that it might be possible to do this is you modify the files creation date and time by using touch -f.. Is it true? The wakeup script creates call files with the date and time on the filename but I don't see how asterisk can then now when to run it since it runs all the files on the outgoing dir when it sees them. Without looking at code, I don't know for certain. If you modify the creation time and then 'mv' it into the outgoing dir, asterisk will see it and ignore it till the creation time is older than current time. You will find a lot of suggestions to not create files in the outgoing directory due to race conditions on your creation time and when you are done writing the file. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] call files run at certain times
On Mon, 2005-03-28 at 16:37 -0600, Anton Krall wrote: BTW, can you use the same call file to make 2 calls in order or just 1 call per call file? 1 call per file What I want to do is first make a call to a sip phone and playback some file and then make another call to the same sip phone but this time connect the call to a zap line. This has to be in order since the first call is kind of an announce and the second is the actual connect. While does this have to be 2 calls. Why not be one call that is scripted. Call the SIP phone then connect to a generic wakeup script that plays the canned audio then connects to the appropriate Zap channel. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] call files run at certain times
On Mon, 2005-03-28 at 23:00 -0600, Anton Krall wrote: But this doesn't work in an environment where multiple person are using it.. For example, multiple call files with multiple announcements.. IT can easily enough. Go read about Macros. You will see easily enough how to put both a prerecorded file name and a phone number to call in the data section. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can't get format_mp3 to work for music on hold
On Sun, 2005-03-27 at 13:41 +0100, Umar Sear wrote: Hi Guys, I am having trouble trying to get format_mp3 working to play music on hold. I have followed the instructions in the read-me and the wiki however it seems after un-installing mpg123, asterisk is not even attempting to play MOH. Executing Answer(SIP/-56f4, ) in new stack -- Executing MusicOnHold(SIP/-56f4, default) in new stack Mar 27 13:40:05 WARNING[13181]: res_musiconhold.c:354 moh0_exec: Unable to start music on hold (class 'default') on channel SIP/-56f4 == Spawn extension (default, , 2) exited non-zero on 'SIP Please tell me I am doing something stupid ! as I would love to know, before I pull all my hair out :-) It can't be installed because you have uninstalled mpg123. Why did you uninstall mpg123? Reinstall mpg123 and stop messing with your hair. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Music on Hold Broken??
On Sun, 2005-03-27 at 09:58 -0800, Noah Silverman wrote: Hi, I am having some trouble with music on hold. Here is the situation. Asterisk Server. Polycom IP500 phone. Everything is configured and works perfectly for incoming and outgoing calls. 1) If I use the hold button on the IP500 phone to place a caller on hold, they just get silence. 2) I made a test entry in extensions.cfg: exten = 6000,1,Answer exten = 6000,2,MusicOnHold() This works fine and I can hear music when I dial 6000. What do I have to do to get the MOH working when a caller is placed on hold from the phone?? Let asterisk do the hold not the phone. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM01B
On Sun, 2005-03-27 at 16:00 -0600, Rich Adamson wrote: Might try modprobe zaptel then modprobe wcfxo (or wctdm). The order makes a difference and I don't remember exactly which one comes first. Modprobe doesn't care as it will load dependencies also. Only insmod needs to be loaded in a certain order. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk with Winmodem
On Sat, 2005-03-26 at 16:42 -0700, Bellows, Jared wrote: I thought that the phone.conf file was for using Linux Telephony modems. I'm not looking at using the zaptel drivers but in my case the pctel driver and using the channel type phone. There isn't much documentation on this channel though. Please learn to trim the messages. No need to bounce 3 copies of the list added footer. The phone channel you speak of is meant to use hayes compatible commands and results in half duplex sound. If you can make it work, it would be like a cb radio. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Any 24 (or 30) way FXS PCI cards?
On Fri, 2005-03-25 at 10:53 -0700, Max W Blackmer Jr wrote: Just found a 12 port single card with opensource drivers 12 user configurable FX0/FXS analogue ports for $1,680 at asterisk mall ( http://www.asteriskmall.com ). I am not sure how well this card works with asterisk. Has anyone used these cards? Compared to a Channel bank(Rhino from asterisk mall, $1249) and T1 card($500) to get 24 FXS ports? $1749 - $1680 = $69 to have a more expandable system. Of course you could look around ebay for cheaper channel banks too. Voip supply has a few 24 port gateways that are FXS based. The biggest one for FXO is 10 ports. They are not cheap the both cost about $2000 USD. a Channel bank with a T1 card will cost you about the same at least with a FXS ports. FXO costs more usually because that is typically the Office station side that has much lager power requirements. Where FXS is the phone/customer side of the Communications. . -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] re: Square Key system
On Fri, 2005-03-25 at 10:27 -0800, Mark W Wood wrote: Most PBX systems have the ability to accommodate this type of configuration. Square Key in this context, refers to the set configuration (every C.O. line appears on every phone) Can I program a specific C.O. line directly to a button? First turn off the HTML email, please. Next spend a bit of time thinking about phones you have seen that use standard analog ports. It isn't a normal option to have more than 1 line, and unless you are getting into cheap shared signal phone setups. you usually don't see more than 2 lines. If you go the SIP route, you might get a little further down the road, but you also take a chance of integrating echo into the system. The answers are there for someone with the requisite background knowledge of the hardware available. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk compare with Skype
On Fri, 2005-03-25 at 13:18 -0800, Sys Admin wrote: if some one was to create a open source IAX client as good/better then skype, even then a asterisk IAX based network will not be able to compete with skype. Since asterix is a centralized server regitration network it can not grow as big as a skype P2P network can grow, You think way to small. You don't have to be centralized with asterisk. Administration of a secure network is easier being centralized, but you could easily run several asterisk machines in a network much like IRC is. A few or even several asterisk boxes peered to each other to know who is where and the users spread over the many asterisk machines. With IAX attempting to handoff calls if possible, it is possible to create true P2P calls as well as routed calls to handle NAT or anything else. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P FXO card-No Dial Tone
On Fri, 2005-03-25 at 12:41 -0800, Jay Ray wrote: Hi I have the X100P card which as to sockets (LINE - for fxo line ) and PHONE (to connect a n analog line) This card is setup as fxs_ks I was getting dial tone but suddenly no Dial ToneHelp appreciated When I try to route the Call using - Dial Zap/1) to this FXO Line I get this error: The X100P is just like a modem. When it picks up the line to dial out, it will disconnect anything plugged into the phone port on it. It is a single channel FXO device. It will not provide a dialtone to another device. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Where to put the modules to start on boot?
On Wed, 2005-03-23 at 17:15 +0100, Remco Barende wrote: Sorry for this kinda n00b question but I've been looking through the wikis but didn't find the answer. All info pages tell you how to load modules from the commandline but what is the `proper' way to do this at boot time? My gentoo box has a /etc/modules.autoload.d/kernel-2.6 but there is no such thing on RedHat boxes. Where do you put the module load and init commands on a RHEL 4 box and where to put it on a RHEL 3 box? See, this isn't an asterisk specific questions. It becomes a distro specific question. Continue looking for a /etc/modules.conf or /etc/modules or even /etc/conf.modules -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] zaptel.o undefined references
On Wed, 2005-03-23 at 20:58 +, Filipe Abrantes wrote: Hi all, I've just got asterisk and i was trying to compile it for my amdk6 running 2.4.27 debian patched kernel. However I got some some undefined references when trying to load the zaptel module: # modprobe zaptel snip /lib/modules/2.4.27-1-386/misc/zaptel.o failed /lib/modules/2.4.27-1-386/misc/zaptel.o: insmod zaptel failed Do you have any idea of what this may be? You will do well to compile a kernel from kernel.org sources and not a distro specific version. Even debian includes things that may cause you trouble. It is better to tune your kernel well and know it is working at the most efficient you can for telephony apps to work well. Next, you need to learn about very generic module debugging. The unresolved symbols that you are having trouble with need to be known. Use depmod -ae to tell you what symbols are missing. Then from the names you may well know what it is you are missing. Since you are using debian specific kernel, I am betting it has symbol versions turned on, and it would require a specific .h file created at compile time to have made the linking of the zaptel module know the precise name of the symbol. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk ChangeLog
On Wed, 2005-03-23 at 22:54 +0100, David Hajek wrote: Hello, is it possible Asterisk's ChangeLog will contain a reference to appropriate bug number in bugzilla? This can be very handy. It isn't bugzilla, it is mantis. There is a cvs list that would be more appropriate for you to watch. Many of the commits will have bug number references. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Spandsp question ( re: compiling )
On Wed, 2005-03-23 at 17:50 -0800, Sean Kennedy wrote: I am trying to compile spandsp on my asterisk server, and it keeps failing out with the following t4.c:38:21: tiffiop.h: No such file or directory In file included from t4.c:41: spandsp/t4.h:62: error: syntax error before TIFF spandsp/t4.h:62: warning: no semicolon at end of struct or union spandsp/t4.h:63: warning: data definition has no type or storage class spandsp/t4.h:64: error: syntax error before '*' token spandsp/t4.h:64: warning: data definition has no type or storage class spandsp/t4.h:87: error: syntax error before '}' token Using my amazing powers of comrehension, I'm getting that I'm missing the TIFF lib. I'm on fc2, so I do a simple yum search tiff. I then installed libtiff-devel, the only thing I didn't have installed, and I get the same error. Is that not the correct lib? The almighty google didn't help much, so I'm sorta stuck not knowing where to go next. You did most of the right work. Follow the link below to get the proper google search string. It returns 3 links for me and the first one I checked had the proper answer for you. http://tinyurl.com/6ec25 -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] bottlenecks
On Tue, 2005-03-22 at 12:07 +0100, [EMAIL PROTECTED] wrote: Hi I must to estimate the* performance. I am try to understand which can be the eventual bottlenecks. Have you some suggestion? Can you to signal to me some problems? Are you going to share your telecom engineering degree with us all if we answer your homework questions? How did you get to this point in your education when you haven't learned that YOUR homework is YOUR responsibility not ours? Why haven't you learned to not cross post to every asterisk list? -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Asterisk and X [was: Re: [Asterisk-Users] zaptel PRI drivers]
On Mon, 2005-03-21 at 08:57 -0700, Tom wrote: This box never was primarily an * box, it is a server that people have used VNC from windows desktops to run a couple of apps that are X11 only that we need in house. We just have been trying to get off of our old PBX, and onto * as our primary system, and it's been working fine with the wctdm haven't seen any degredation of voice quality, call quality, anything previous to this. We run the GDM system so that users can sign on with their same username/password, and they get their same groups/restrictions etc all through LDAP, this has been working for 2 years now. We don't want to set up 45 user accounts locally on the box, set up separate passwords, have the users manually keep those passwords in sync, and then have separate passwords (again!) for vnc, which is what we have to do if we can't get GDM/xdm/kdm and XDMCP to work. There are never more than 3-5 people logged in at once, and as I said previously this was all working just fine with wctdm cards, its just the wcte110p that has issues, and those are that it can't keep the timing right (according to our provider) when X is enabled. Our provider and our asterisk box get flooded with HDLC Abort(6) errors. We don't want to have to spend an extra 3 grand for another server just to take up more space when we have this box that is sitting here idle 99% of the time, and as it has worked spectacularly well with the wctdm cards, I don't see why it can't with the wcte110p/PRI. Tom Christensen Maybe you need to look at an inexpensive dell 1u machine. You shouldn't have to spend more than $1k for a machine to dedicate to asterisk. As you have seen, asterisk needs realtime speeds and when other apps get in it's way something gets dropped. If you don't need a rack mount server, you can find even cheaper machines around to dedicate to it. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PRI Question
On Mon, 2005-03-21 at 21:16 -0700, Tim Chandler wrote: Let me further clarify this. I am looking to buy the TE110P. The website says that The TE110P with Asterisk will route voice and data traffic, and eliminate the need for an external router. How does this work? How is the data transferred - as a pass-through like a NAT to the server's network card? What kind of network slowdown are we looking at? How does this affect the processor? I would appreciate some more information on how this works. Look here http://www.voip-info.org/wiki-Asterisk+cmd+ZapRAS for one option. or here for the other http://www.google.com/search?q=hdlc+zapata Hi Everyone, Thanks for all the input you add to the list. This seems to be a very good list. I am still new to Asterisk. If I run a PRI integrated T1 line into my office, do I need to split the line between the data and voice before plugging it into the asterisk box or is there some other way to do that? What are some good options for splitting the line? Thanks for any input. Tim BTW - Giving everyone a hug is an expression in Brazil. Everyone says it... it's like saying have a good one or good to see you. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] virus
On Sun, 2005-03-20 at 08:17 -0500, dean collins wrote: Great, just received 9 virus emails in the past 24 hours from the asterisk list where people have had my address in their address book. Heads up people, its an attachment, the text looks a little jinglish why would you open it? This begs the question why any intelligent sane person would ever read email on a windows computer? Don't you know by now it leads to virus infections and spam. Funny that you complain about the virus when my spam levels go up considerably every time another slashdot article goes out about asterisk. Stupid users who haven't learned safe computing littering the net with all their trash. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Any 24 (or 30) way FXS PCI cards?
On Sun, 2005-03-20 at 09:54 -0600, Rich Adamson wrote: It seems to me silly to have a T1/E1 card to connect to a channel bank when you could just have a 24/30 way FXS card in the slot in the first place. Does such a thing exist? Wouldn't Digium have a lot of customers if they could produce one for say $1000 retail? Trouble is power. Unless there is more power made available, you may not be able to drive the ring voltage of several consecutive lines at once. Take for instance the Adit 600, it has a 130w power supply for just 25 ren capability. Think of the troubles that would cause trying to be regulated through your standard PC PSU of 300w. Won't you just love trying to diagnose random reboots right after a phone call comes in and over draws your PSU capacity and it goes into short protection where it begins pulsing power. The entire telephony system relies heavily on statistical probabilities, including central office switches, inter-office trunking, and ringing. Applying the above thought process to a central office, there is no way a single central office could possibly ring all telephones associated with that office simultanously. The ring generators could never handle it. Likewise for a pc card supporting 24 fxs lines. The probability of three or more lines ringing at exactly the same time are very small. With at least a little engineering forethought, its not that difficult to create ring cycles where ports 1 through 6 ring during some period, followed by 7 through 12, etc. (That's actually how most central offices handle ringing now with a couple of exceptions.) Maybe, but is that something you would expect in a PBX? And do you do that in drivers or in circuits? A bigger issue is really oriented around how many ringers exist on a single fxs port, and that _could_ be limited by specifically limiting user implementation (via specs) to one phone per port (or whatever), etc. Couple that with the fact that current ringer designs require substantially less power then the old electomechanical ringers, it certainly isn't that difficult to design a PC board to support 24 fxs ports. You are right, but you still hit the same problem just at a lower probability of major problems. I still contend that any design is going to need it's own external powersupply so as to not over draw the PC's PSU and cause it to either fail or pulse causing system stability problems. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] zaptel PRI drivers
I am not flaming you even if it may seem so below. On Sun, 2005-03-20 at 23:12 -0700, Tom wrote: I have a quick question. I know that running X on an asterisk server is not officially supported, however, I've never had any trouble with it until now (8 months, using wctdm cards with fxo and fxs ports, IAX trunks, SIP phones, everything except a PRI card). Now I just installed my first asterisk box that terminates a PRI, and bam, HDLC errors up the wazoo if X is running, if its not, everything is fine, I assume this is because the timing parameters for the PRI are so much more strick. IT isn't just not supported, as you see, it just doesn't work reliably. Stop doing that. I don't mind if X is a little less responsive (even alot less responsive), but I would really like to be able to run X on a server with a PRI. Is there any way to reduce X11's priority so that it doesn't interfere with the zaptel driver for the PRI... I've tried renicing X as far down as I can and renicing Asterisk up as far as I can, however I fear this won't ever fix the problem since I think the actual kernel module that is running the pri card needs to get higher priority (ie, the kernel itself needs higher priority). Is there any way to do this? Am I correct in my analysis? I really don't understand why on a system that averages less than 3% CPU usage with X running, why it can't handle the PRI. I know for whatever reason X always gets a really high priority (although top doesn't show X getting any special treatment its PR 15 NICE 0 by default, lower than most other processes on the system). Your problems our beyond the reach of nice. Bad graphics chips use lots of CPU time when doing updates and therefore will cause the zaptel drivers to miss interupts. Missed interupts will cause your PRI errors. You can't fix broken graphics cards, nor fix the drivers. Another idea is that right now the system is only a single proc, but it is dual proc capable. Would this somehow help if we added the second proc? My thinking is it won't because it's a kernel module we are dealing with, and because of that I can't control the affinity of the driver (I was thinking at one point put X11 on 1 proc and Asterisk on the second, but it's not Asterisk that has the problem I don't think.) I doubt it will make a difference. As I mention above, it ends up being a kernel level problem and you probably can not get the drivers to operate on different CPUs. It would just be better to spend the money on a decent workstation and teach yourself how to manage the asterisk machine from the separate workstation. My final idea is that currently the system has an onboard 8mb ati graphic card that leaves almost all actual graphics processing to the CPU, could adding a better graphics card possibly help X use less cpu and not get in the way so much? Not really, graphics sucks time away from a CPU no matter. Anyway, I know this isn't a supported setup, so if thats your answer don't bother replying, I'm know this will be a kludge/hack to get working (if I can get it working at all). I'm just trying to do something that will be convienient for me and my users, there are other systems running on the server that I don't want to manage through the CLI, and the users don't know how to manage through the CLI, and there is no web management for them. Ahh, so you are breaking many cardinal rules here. You have too much stuff on your asterisk machine. You are begging for troubles. Move those services off of the asterisk machine. Does anyone have success running X on an asterisk box that terminates a PRI? If so what hardware (video card, cpu, ram, mobo, etc)? Thanks as I know this setup isn't supported, and I'm probably asking alot, don't think I'm just relying on the list for bizarre things, I've been trying various ways of doing this for the last 3 weeks, I can successfully run a vnc server on the box (without X running) and everything works, so for whatever reason it is getting a lower priority or something. I really need to run GDM though as managing VNC passwords/usernames/desktop settings is quite cumbersome and if we can just get GDM running, we can use our ldap authentication server for logins to this box (which is what we were doing previously when we didn't have a PRI terminated on this box). X and graphics drivers are big hogs on memory and CPU. VNC moves the graphical portion over to the client machine. If you need GDM, why not get X servers for your other machines and let GDM broadcast. This should mean your X server run from whatever other machine should be able to be configured to use the GDM and login. While the app will run on the asterisk machine and be bad, the real CPU drain will not happen on the asterisk machine. Just to recap, I am not flaming you no matter how it seemed in the message. -- Steven Critchfield [EMAIL PROTECTED
Re: [Asterisk-Users] Any 24 (or 30) way FXS PCI cards?
On Sat, 2005-03-19 at 17:53 +0100, Rob Scott wrote: It seems to me silly to have a T1/E1 card to connect to a channel bank when you could just have a 24/30 way FXS card in the slot in the first place. Does such a thing exist? Wouldn't Digium have a lot of customers if they could produce one for say $1000 retail? Trouble is power. Unless there is more power made available, you may not be able to drive the ring voltage of several consecutive lines at once. Take for instance the Adit 600, it has a 130w power supply for just 25 ren capability. Think of the troubles that would cause trying to be regulated through your standard PC PSU of 300w. Won't you just love trying to diagnose random reboots right after a phone call comes in and over draws your PSU capacity and it goes into short protection where it begins pulsing power. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CAC Access Bank Manual
On Fri, 2005-03-18 at 14:05 +0545, Vicky Shrestha wrote: Hi, The asterisk configuration and the channel bank configuration are both set to esf and b8zs. Howerver I am still getting the framing Error Red and blinking. zttool shows there are no alarms. According to the manual, Framing Error (Red and Blinking )means Network T1 is out of frame (received signal cannot be framed to ESF or D5 as configured by T1 Option switch 4) I tried with both DIP switch on and off, but no help. /etc/zaptel.conf = span=1,1,0,esf,b8zs Do you have the CAC set to provide timing to the line? If not, you need to set your timing to 0 here so the TE410P card will provide timing. Also, as a precaution, It is helpful to power cycle the machine when you change timing. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Voice getting cutoff
On Fri, 2005-03-18 at 11:48 -0600, Anton Krall wrote: What do you think? Haven't you read about sharing interupts is problematic? CPU0 0: 16148159 XT-PIC timer 1: 4 XT-PIC keyboard 2: 0 XT-PIC cascade 5: 0 XT-PIC usb-uhci 8: 1 XT-PIC rtc 10: 161351663 XT-PIC usb-uhci, wcfxo This isn't good. 11:1276097 XT-PIC usb-uhci, eth0 12: 161350551 XT-PIC ehci-hcd, PS/2 Mouse, wcfxo Also not good. 14: 138574 XT-PIC ide0 15: 33 XT-PIC ide1 NMI: 0 ERR: 0 Any problems here? BTW, please, at the least trim the footers off of the emails. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Phone ringing and not going to voicemail?
On Thu, 2005-03-17 at 14:16 -0500, Matt wrote: Hi, I have one phone on my network that just keeps ringing (when I call it) and does not go to voicemail. If the person there is on the phone, and someone calls it they get the busy message, but they never seem to get the 'unavailable' message... instead it will just ring and ring and ring... any ideas? Any configs? -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: PC sound hardware for voice recording
On Thu, 2005-03-17 at 11:30 -0800, snacktime wrote: On Thu, 17 Mar 2005 11:14:01 -0700, Wiley Siler [EMAIL PROTECTED] wrote: I recorded my last set of prompts over my Plantronics DSP 500 USB Headset. I have also used a Logitech USB Headset. These and similar are easiest to use along with X-lite or similar softphone. I used the suggested method of dialing an extension on the PBX and letting Asterisk record for me direct via the phone. The quality is just as good as using a dedicated soundcard and mic IMHO. At $50-75 for the USB headset, it is a good cheap investment. Thanks for the tip. I was worried someone was going to say it couldn't be done on a budget.. Those look pretty nice from what I was reading on them. Don't forget that on a reasonable budget you can get Alison to do the recordings for you. It should match up well with other pre-recorded prompts as well. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AGI kill
On Wed, 2005-03-16 at 11:20 +0100, Pepe Aracil wrote: Hello. When the caller hangup the phone, asterisk kills my AGI python script without notification. I caught all signals, but none was trigered. How can i trap this event to resume some operations. Asterisk doesn't send any signal upon hangup. Asterisk closes the pipes that show up as STDIN and STDOUT for your AGI app. You need to deal with it gracefully. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem starting Asterisk - libssl.so.4 cannot be found
On Wed, 2005-03-16 at 15:04 +, Andy and Jayne Slim wrote: I'm sure this is a pretty basic problem, unfortunately I am a telecomms rather than a Linux person so any suggestions would be most appreciated. I have successfully downloaded and installed the various Asterisk packages. However, when I try to start Asterisk, I immediately get a message saying module 'libssl.so.4' cannot be found and the startup is halted. I don't have this file anywhere on my system but I read on some articles that this was a symbolic link to libssl.so.0.9 so I did an 'ln -s' to point the offending module there. This made no difference. I therefore upgraded my Open SSL version to 0.9.7d and then re-installed Asterisk. Still no joy. I have moved the module and its symbolic link to the same folder as the Asterisk executable, and checked the path statements in ld.conf, but the program still will not start. Please can someone advise me on what else I should try to resolve this? Where did you get your version of asterisk? It sounds like you are having dependency problems. It sounds like you downloaded a binary copy of asterisk and the vendor of that package didn't put in proper dependency information to stop you from installing it till all the required packages are installed. You really should download the source, compile, and install. This will mean that asterisk will be linked to your libraries. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to determine the voicemail file name for an AGI script
On Tue, 2005-03-15 at 07:03 -0600, Bryan Boatright wrote: I've read several of the Wiki sections on Voicemail and Asterisk variables but could not find an obvious answer to this question. I would like to run a script that post-processes the voicemail after the Voicemail application returns (with AGI or DeadAGI), but I cannot figure out how to easily determine the name of the file written by the Voicemail application. Does anyone know of an efficient way to access that information from within the dialplan so that it may be passed to the AGI script? Well, you could store the extension/context of the voicemail box that you sent the call to. From there it is just a matter of looking in the appropriate directory and dealing with the way voicemail write the information out. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Setting up Security Groups
On Tue, 2005-03-15 at 07:21 -0800, PA wrote: Right now here is how I have it structured in extensions.conf. What am I missing? Why would a sip-basic member be able to make toll calls? [default] include = sip-basic include = sip-operator include = sip-superuser You probably want to remove those 3 entries. I can't remember for sure if you can inherit includes, but I do remember that unregistered sip phones could have access to the default context. Guessing without the benefit of the logs from your machine, your phones may be entering the default context and getting access that they don't deserve. [sip-superuser] include = outbound-local include = outbound-longdistance include = outbound-tollfree include = outbound-toll --- sip users info follows here [sip-operator] include = outbound-local include = outbound-longdistance include = outbound-tollfree --- sip users info follows here [sip-basic] include = outbound-local include = outbound-tollfree --- sip users info follows here [outbound-local] --- outbound calling info follows here [outbound-longdistance] --- outbound calling info follows here [outbound-tollfree] --- outbound calling info follows here [outbound-toll] --- outbound calling info follows here Without the details of these outbound sections, we can't tell if you have a pattern matching problem that is causing your troubles. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Learning the Ropes of *
On Tue, 2005-03-15 at 17:36 +, Barry FAWTHROP wrote: In having configured my first * server there are a few questions I could not understand or find answers to 1) How does one use ztmonitor to adjust the rxgain and txgain. I have set mine to -1.0 each to get rid of echo on std phones connected on the TDM10B FXS module I'll pass on this one. I haven't used analog lines in a long time. I do seem to remember a write up on it but I don't remember the URL. 2) Is it best to use a TDM card with an TFX and FXO module or is the the FXO = X100p and FXS on TDM a good method? Depends on port density needs. A FXO daughter card on an already deployed TDM400 card is fine if you have the space needed to install it. An X100P card though will introduce more interupts and load to your machine when used in conjunction with the TDM card. If you start running out of space, you should look into a T1 card and channel bank. 3) How do I disable mgcp, skinny etc that I will not be using ? Asterisk loads / scrolls a lot when you start it. In the modules.conf, use the noload option to not load the specific modules you are not using. Consider this a must as it reduces security concerns. While asterisk developers strive for error free code, there is always a chance of some bug biting you before you can patch. If you turn off all the unneeded modules, you will limit your exposure to new potential exploits. 4) How do I calculate/assess the Load * is placing on the server? I'm using a 2.0 Ghz Pentium III, 1 GB Memory. top shows good stats. Top isn't a real good tool. It should however show that your system still has resources available. 5) Why do I get pixelating (sorry I'm a video guy) on a call? Every now and again, you hear these strange beeps, and tweets in the middle of your call. at random times. not sure about this one. 6) Where can one post thier configs to check / validate that they are good, and no overloading the server ? It is unlikely you will overload a server because of your configs. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FW: AntiSpam Alert from Rusten McKenzie
On Tue, 2005-03-15 at 10:17 -0800, Robert Goodyear wrote: __ On Mar 15, 2005, at 10:04 AM, dean collins wrote: Is there anyway we can get this shit off the asterisk list apart from posting their email address [EMAIL PROTECTED] here for the spambots to pick up? You could send an email asking him to whitelist the list. Won't help as the software he is using specifically checks the from: header. Just another example of broken methods to fix this problem. Also it is unlikely he can fix the software as it is a commercial solution. I'm pretty close to writing a rule to bounce a copy back to him and the company who sold him the software. Anyone want to provide a spam list to add him to for being annoying? -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] OT: Best DB
On Tue, 2005-03-15 at 14:21 -0500, Giudice, Salvatore wrote: Sticks and stone still break my bones, but PostgreSQL is still a dog. Market share: According to CD Times magazine dated July 1, 2004 Top Deployed Databases poll shows following databases in use: SQL Server with 78%, Oracle - 55%, MySQL - 33% and PostgreSQL - 8%. I see they created this with Mysql, 78 + 55 + 44 + 8 = 185% I'm sure if you add in the others we would get to something around 300% deployment. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: Best DB
On Tue, 2005-03-15 at 19:04 -0600, Jon Gabrielson wrote: On Tuesday 15 March 2005 06:34 pm, Robert Hajime Lanning wrote: quote who=Giudice, Salvatore So, let me see if I am right. You run a support shop? You want your database to validate your data for you instead of leaving that logic to your application? Usually, a database is considered to be an asset worth protecting from unvalidated user input. Also, do you routinely try to insert text strings into fields, which are not created large enough to accept these strings? This is somewhat disturbing. Data validation should be done at all levels. Period. Validating the SAME data at each level greatly decreases your speed. It is much simpler and easier to just validate it first. Of course most of us want to follow DRY (Don't repeat yourself). In doing so, you try and let one place be an authoritative source. The DB should be authoritative as to what is correct. You shouldn't have to babysit the DB to make sure it is doing the correct thing. If you have to babysit it, it isn't worth it. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Skype - Bandwidth
On Mon, 2005-03-14 at 14:30 -0300, César Davi Ávila do Nascimento wrote: Hi All, Does anyone know the amount of memory used by skype? Did you think about the best venue to ask this question. We are not a skype support forum. And BTW, TURN OFF HTML EMAIL. Exercise some thought about those who will read your message before assuming we all need to see blue. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] School design question
On Mon, 2005-03-14 at 10:46 -0800, Chris Hobbs wrote: My school district will be building a new elementary school in 2006. We were about to go to bid with a traditional intercom system for the campus but I would like implement Asterisk at the campus. My question is, do we build in a traditional intercom/paging system and tie that into the Asterisk PBX, the way such intercoms have been connected to other PBX's in our district in the past, or do we put IP phones in the classrooms and tie that into a PA system for paging? Are there IP based paging systems that could be used instead of the traditional PA/loudspeaker systems in most schools? You may want to continue having a general access PA that is just like your currently installed system. Your PA probably needs to be easily understood in the hallways as well as the classrooms. Cost of wire and speakers are much lower than an IP phone and the extra power requirements and possibly buggy firmware. We will be writing a spec shortly (and I will be seeking a consultant on asterisk-biz soon to assist us), but I need to know whether I need to get our engineering consultant to redraw the cabling to reflect a data jack at the location of the classroom phone instead of the telco jack that is currently on the plans. If you aren't planning on data to the classroom, you probably are already behind. You probably should plan on running 2 cat5 cables to every room. At worse, you use 1 cat5 for plain old telecom. You at least have options at that point. See about running them all to nice patch panels so that you just make jumpers from the kind of network you want over to the port that needs it. Do consider that you don't have to purchase fancy phones for the classrooms. You could use analog telephones that are cheap to replace and use a group of channel banks to support the phones. Maybe a bit more expensive than the IP phones, but it is tried and proven technology. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Skype - Bandwidth
On Mon, 2005-03-14 at 16:36 -0300, César Davi Ávila do Nascimento wrote: Ok... take it easy... But do you know skype, don't you? I know of it, I don't use it as I have no use for it and it isn't open source. Next lesson is to learn how to trim the bottoms of your messages. You might want to learn in-line quoting as well and configure your mail app for proper quoting as well. - Original Message - From: Steven Critchfield [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, March 14, 2005 3:56 PM Subject: Re: [Asterisk-Users] Skype - Bandwidth On Mon, 2005-03-14 at 14:30 -0300, César Davi Ávila do Nascimento wrote: Hi All, Does anyone know the amount of memory used by skype? Did you think about the best venue to ask this question. We are not a skype support forum. And BTW, TURN OFF HTML EMAIL. Exercise some thought about those who will read your message before assuming we all need to see blue. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] School design question
On Mon, 2005-03-14 at 11:33 -0800, Chris Hobbs wrote: I guess the question comes down to which of three possible network designs to use (once wiring is assumed to be CAT5 across the board): A) Analog classroom phone - Paging/Intercom System - Paging/Telco gateway - Asterisk * Limits number of clasroom calls that can be made to capacity of gateway Obviously a bad idea for the denoted reason. B) IP Classroom Phone - Asterisk - Paging/Telco Gateway - Paging System C) Analog classroom phone - Channel Bank - Asterisk - Paging/Telco Gateway - Paging System B and C differ only in where do you place complexity and cost. B pushes more complexity to the classrooms and _possibly_ more support time. C keeps the complexity in the wiring closet and requires fewer logins to fix any problem. Now you need to also think about power. Unless the entire school is on a backup generator that you can trust, I would suggest option C as you can consolidate your UPSs in the wiring closet and make sure your internal phone network is still functional in an emergency. With option B, you would either have to do POE and the added cost of the POE injectors to get the same functionality. Keeping the idea of IP phones open lets you possibly add IP phones to the network as needed. Think of the possibility to add a phone for a student teacher/observer with private extension while leaving the main classroom phone available as usual. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Skype - Bandwidth
On Mon, 2005-03-14 at 17:31 -0300, César Davi Ávila do Nascimento wrote: Talk about skype is forbidden, but to be impolite is allowed... Great list! Did you ask about skype in asterisk? Did you ask an asterisk related question? On both of those it was a no. So you came in here asking questions that are not on topic for this list. It is high enough traffic as it is without talking about something completely unrelated. And you still haven't learned to trim your messages. - Original Message - From: Steven Critchfield [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, March 14, 2005 5:16 PM Subject: Re: [Asterisk-Users] Skype - Bandwidth On Mon, 2005-03-14 at 16:36 -0300, César Davi Ávila do Nascimento wrote: Ok... take it easy... But do you know skype, don't you? I know of it, I don't use it as I have no use for it and it isn't open source. Next lesson is to learn how to trim the bottoms of your messages. You might want to learn in-line quoting as well and configure your mail app for proper quoting as well. - Original Message - From: Steven Critchfield [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, March 14, 2005 3:56 PM Subject: Re: [Asterisk-Users] Skype - Bandwidth On Mon, 2005-03-14 at 14:30 -0300, César Davi Ávila do Nascimento wrote: Hi All, Does anyone know the amount of memory used by skype? Did you think about the best venue to ask this question. We are not a skype support forum. And BTW, TURN OFF HTML EMAIL. Exercise some thought about those who will read your message before assuming we all need to see blue. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Skype - Bandwidth
On Mon, 2005-03-14 at 16:46 -0600, Jay Milk wrote: Yes yes, it's totally hypocritical to post the following, but I think everyone should read it (yes, I think I'm just THAT important). Here's a perfect example of a mis-post that was blown out of proportion. When I first saw Cesar's post, I responded off-list to the effect that this would be the wrong forum to ask this question. If others had done the same -- responded off-list -- or simply ignored that message, that would have been the end of the thread. Instead, 16 more messages clogged the list. Yes, it's necessary to explain to posters why certain messages won't get answered. But I don't think it's necessary to do it in front of 10,000 people. Yes, it's annoying that some folks don't understand or respect the rules of this list -- but in absence of a moderator, there aren't really any rules here. It's also annoying that some don't make the effort to find an answer before asking, and then get upset that nobody will explain to them what a telephone is, or what VoIP means. But if you look back at the last week or two, the majority of the noise on this list comes from those who are responding and then enter a drawn out argument as to why the initial post bothered 10,000 people. I believe the list volume would decrease by at least 50% if we all simply took our disagreements off-list, or ignored messages that don't belong. And in the process, the signal-to-noise ratio here would greatly improve. If you don't agree, please ignore this message or respond off-list. If you do agree, great. The problem of taking these messages off list is exactly that no one else sees the answer or at least the reason for no answer. There are many lists that I come across that seem to think that answering via private email is a good idea. Trouble is that the information then isn't archived for later searchers to reap the benefit of the exchange. If the occasional user doesn't get reminded publicly that what they did was wrong, you would have more people assuming it is okay to do the same thing. If you didn't see the occasional person pulled over for speeding, would you eventually let your speed creep up while driving? You are right though that the extra responses are usually not necessary. I understand that some feel the need to smooth the waters. I equate it to a child getting spanked by one parent and running crying to the other parent for some consoling. I am quick to spank users who misbehave. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem Compiling Spandsp
On Mon, 2005-03-14 at 22:14 -0300, Juanjo Portela wrote:
Yes, I do.
I know it seems a problem of libtiff not installed, but i have
libtiff-3.5.7 installed :(
It was asked if your had the SOURCE installed. At the least you should
have a -dev or -devel package installed for the libtiff package. We KNOW
you don't have it installed in the normal location at least because of
the last error message I left in the trimmed message below.
Everyone should remember to TRIM unnecessary portions of their email.
On Mon, 14 Mar 2005 18:01:50 -0700, Max W Blackmer Jr
[EMAIL PROTECTED] wrote:
do you have libtiff source files installed?
Original Message
Subject: [Asterisk-Users] Problem Compiling Spandsp
From: Juanjo Portela [EMAIL PROTECTED]
Date: Mon, March 14, 2005 6:44 pm
To: Lista Asterisk asterisk-users@lists.digium.com
Sirs,
I can't compile the source spandsp-0.0.2pre10; when i try to do the
make sentence the following errors appear:
bert.c:43:20: tiffio.h: No such file or directory
--
Steven Critchfield [EMAIL PROTECTED]
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Re: [Asterisk-Users] chat line
On Sun, 2005-03-13 at 11:15 -0600, James Taylor wrote: Yes, the meetme can be part of it. I was thinking more of a classified ad chat line, you know the male-female thing: ...If you are a man looking for a woman, press one... ...If you are a woman looking for a man, press two... ...If you are not sure, press three... ...If you don't care, press four... ...If you are a dog looking for his master, press five... I'll share my notes if someone wants to team up and work on this. James Learn the wonderful and very successful unix way of doing things. You should use many small tools that do exactly one thing and do it very well. String a few of these tools together and you have a well tested larger tool. Use one of the calling card apps for your billing, just know it doesn't have to dial out of your system. Use an IVR menu system to direct the person to the chat room you want them in. Use meetme to conference them all in. Ohh my, looks like I just put enough work for you to need to send you an invoice for you or your customer to pay me Do your own damn homework more often. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chat line
On Sat, 2005-03-12 at 21:35 -0600, James Taylor wrote: Anyone done a chat line app? Any reason why the meetme app doesn't fullfuill your needs or did you not bother to look? -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk and USB ISDN controllers ...
On Fri, 2005-03-11 at 11:31 +0100, Vledder, Hans wrote: Hi Steve, Since you don't mention what USB ISDN adapter specifically you are thinking about, what do you think we will be able to tell you. All I know about the adapter is what I've told you. It's a USB Colognechip based ISDN controller - probably HCF-USB based. It's supported by Linux, but there's no info on access to B and D channels. Okay, this shows where you should do some research. Just to point out how easy it is to find out the answer, watch the steps. 1. use google to look up cologne isdn usb linux 2. follow link that points to the actual manufacturer 3. Notice that the usb driver is the hisax driver 4. use google to look up hisax site:lists.digium.com 5. notice how many people are already discussing the use of it. 6. follow a couple of posts. 7. conclude that it is possible to use. optional steps 8. think about how little time and effort it took to follow the above pattern to quickly answer questions on your own. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Steven Critchfield Sent: Thursday, March 10, 2005 6:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk and USB ISDN controllers ... On Thu, 2005-03-10 at 18:13 +0100, Vledder, Hans wrote: Guys, I am planning on building a small SIP PBX with a single ISDN line. Currently I am looking into the specs of a very tiny barebone system that has an option Colognechip base ISDN controller. The only thing is that the ISDN module that comes with this barebone hooks up to the motherboard using USB. My intention is to allow incoming and outgoing calls from SIP to ISDN. Is this setup in any way supported by *? Since you don't mention what USB ISDN adapter specifically you are thinking about, what do you think we will be able to tell you. The first step would really be to ask if your specific ISDN adapter can be used under linux. After that, can that specific ISDN adapter give access to voice channels. What method is used to get access to the audio and the signaling. It may well be usable if the drivers for it implements the same API as the current ISDN cards in use support. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: Best DB
On Fri, 2005-03-11 at 08:51 -0600, Matthew Boehm wrote: If you're a VoIP provider, and are trying to provide a near carrier-grade service, postgres shines. I'm not disagreeing with you, but we are a CLEC and we do provide 'carrier-grade' service and we use MySQL everywhere. IMHO, MySQL is just so much more easy to use, install and maintain. phpMyAdmin makes it even easier. If that is a deciding reason, you should check out phppgadmin sometime. Very similar interface but for postgres. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is it an AGI bug in 1.06? IAX Calls going to wrong extension with AGI.
On Fri, 2005-03-11 at 12:26 -0500, Kanuri, Seshu (Company IT) wrote: I am using PBXware for configuring users and extensions. Pbxware uses Internal script called init.sh to process the calls based on its own version of extensions.conf defined in the GUI. I have IAX2 Extensions 56 and 101 and SIP extensions 50 and 51. I have used IAX2 extension 101 and dialed SIP Extension 51 But the PBXWare's Init.sh AGI command identifies the DNIS as another IAX Extension - extension 56, instead of SIP Extension 51 and sends the call there. Just a quick thought here, as the vast majority doesn't have access or at the minimal don't use the software you are using to do config and as it is an agi script outside of asterisk, you should go to the vendor of PBXWare and see what they say. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] OT: Best DB
On Thu, 2005-03-10 at 08:57 -0600, Jay Milk wrote: IB/FB stores the DB in one file, but the file can span multiple drives if needed. However, you can't select which table goes into which file. Personally, I don't think that's very feasible, nor is it required -- if a table is accessed often enough to be mission critical, large parts of it will reside in memory due to caching anyway. Maybe I work in an odd environment where writes(updates and inserts) are probably equal to or more than the reads. Caching isn't real helpful at making the data get to disk faster. Caching helps for reads only. I'll admit I haven't had to use this feature yet, but I see where some people could really need it. -Original Message- From: Steven Critchfield [mailto:[EMAIL PROTECTED] Sent: Thursday, March 10, 2005 1:00 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] OT: Best DB If it stores the entire DB in 1 file, it can not scale as well as other DBs. Postgres 8 supports splitting a single DB up so you can put portions of it on different media if needed. If you have to tune for absolute speed, you can purchase one of the solid state drives for the tables that need that kind of speed while using much less expensive harddrives for the rest of the DB. While I do not remember mysql supporting it this directly, I think I remember the file structure being not to difficult to figure out and split and symlink back together if need be. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: Best DB
On Thu, 2005-03-10 at 09:17 -0500, Andrew Kohlsmith wrote: On March 10, 2005 08:44 am, Walt Reed wrote: Now one of our lead engineers has done some performance testing last night for our app and found MySQL to be 8 to 100 times faster for all but one of our operations (combination of ~80% reads, 20% writes on the InnoDB table type.) His testing basically increased the load until performance was unacceptable. I'd *love* to see the particulars of that test. It's been shown time and time again that postgres' speed CLOBBERS mysql for anything but the simplest selects, and that it can handle far more concurrent connections without slowing down. This brings back the question of testing methodology. If the tester that posted here only tested sequential queries, I could see MySQL showing faster. A test that would probably show less of a gap is running whatever testing app multiple time simultaneously as it will start showing the ability to handle concurrent users. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Delay on outgoing calls
On Thu, 2005-03-10 at 15:20 +0100, Stefano Arata wrote: Hi, I've a wildcard TDM400P card with 2 fxo and 2 fxs modules. I've set this extension in my extensions.conf for obtain the external line: exten = 0,1,Dial(Zap/g2,10) The dial application is executed immediatly but next this there is a delay before I can hear the tone. This is the output in the CLI: -- Starting simple switch on 'Zap/1-1' -- Executing Dial(Zap/1-1, Zap/g2|10) in new stack -- Called g2 -- Zap/3-1 answered Zap/1-1 -- Attempting native bridge of Zap/1-1 and Zap/3-1 The delay is about 2 seconds and it is between Called g2 and Zap/3-1 answered Zap/1-1. Where can I reduce this delay? I have tried to find it in zaptel.h, but I'm not sure if it is in the driver sources or in the asterisk sources. Is there some decent reason you felt the need to post this exact question again a day later than the original? Could you not look at the suggestions provided already and work from there or provide feedback as to why you didn't like the suggestions? -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and USB ISDN controllers ...
On Thu, 2005-03-10 at 18:13 +0100, Vledder, Hans wrote: Guys, I am planning on building a small SIP PBX with a single ISDN line. Currently I am looking into the specs of a very tiny barebone system that has an option Colognechip base ISDN controller. The only thing is that the ISDN module that comes with this barebone hooks up to the motherboard using USB. My intention is to allow incoming and outgoing calls from SIP to ISDN. Is this setup in any way supported by *? Since you don't mention what USB ISDN adapter specifically you are thinking about, what do you think we will be able to tell you. The first step would really be to ask if your specific ISDN adapter can be used under linux. After that, can that specific ISDN adapter give access to voice channels. What method is used to get access to the audio and the signaling. It may well be usable if the drivers for it implements the same API as the current ISDN cards in use support. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Delay on outgoing calls
On Thu, 2005-03-10 at 23:10 +0100, Stefano Arata wrote:
On Thu, 2005-03-10 at 11:07 -0600, Steven Critchfield wrote:
Is there some decent reason you felt the need to post this exact
question again a day later than the original? Could you not look at the
suggestions provided already and work from there or provide feedback as
to why you didn't like the suggestions?
Sorry, I've looked for delay in the posts and I've not seen the others.
I have tried to pass the number directly to the Dial command with this
extension:
exten = _0.,1,Dial(Zap/g2/${EXTEN:1})
But the delay between the Dial and the answer of the Zap channel is the
same one. It seems that the fxo interface wait one ring before answer,
this don't happen on internal calls.
Your internal calls probably don't have to wait for the pattern match
like above.
--
Steven Critchfield [EMAIL PROTECTED]
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Re: [Asterisk-Users] TDM400P slow getting line tone
On Wed, 2005-03-09 at 14:34 +0100, Fabrizio Mazzoni wrote: Hello all, I just installed a TDM400P with 2 FXO modules on my asterisk server. The card works perfectly. To get users to ring out from my SIP phones i setup an extension with 0 that basically does something like this: extension = 0,1,Dial(ZAP/g1) where g1 is the group of the two FXO channels extension = 0,2,Hangup This works exactly as i want so users basically can dial 0, wait for the dialtone and then dial the requested number. The only problem that i have is that from when a user dial 0 to when i get the dialtone from the telephone line, something like 5 seconds pass... is it possible to pull this wait time down to about 1 second? or even less?? Unless you are doing something odd that requires the user to listen to the dialtone and validate there is one, why don't you just go ahead and capture the number and dial it out. The benefit is that asterisk then logs the outgoing number and the times in CDR. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Print-to-Fax client
On Wed, 2005-03-09 at 11:31 -0500, [EMAIL PROTECTED] wrote: Hi, Does anyone know of a Print-to-Fax client that works with asterisk spandsp? Astfax is a partial solution but that only lets us email the fax in, we'ld like to set it up so the user can hit the print button and send the fax (even if all it does is email - transparently to the user - the fax to astfax). Turn off HTML. Research cups. I know people who use cups to create PDFs for them. Shouldn't be any really big effort to get cups and windows to talk to each other and create a print job that creates the .call file and the ,ps file as well. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: Best DB
On Wed, 2005-03-09 at 14:50 -0500, [EMAIL PROTECTED] wrote: I know this is a bit off topic but we are using Asterisk :) Since this list is full of tech gurus w/ all different sorts of backgrounds, I thought I would get the best opinions here. We have several different switches and other telecom equipment at our facilities which all have their own proprietary cdr platforms, which are rather limited. The company I work for is looking to develop their own in-house billing system that would combine cdr from all platforms and bring it into one big db, so we can do whatever we like w/ the data...billing, invoices, reports, asr...etc... So my question is this What's the most stable, fastest reliable database for this project? Call volume is about 8 to 10 million minutes per month, and we want to have 12 months of cdr available at any given time, anything older can be archived on tape. So what's the best db...oracle, ms sql, informix, mysql or something else? Best is subjective. It really depends on what needs you have to satisfy and what you are willing to compromise on. You can drop all the commercial DBs if your group is willing to develop/work a bit more. If you need replication, you need to look into what supports it. I understand there are tools for Mysql and Postgres to do replication, but postgres may not be as well polished there yet. My experience has been that postgres is better at scaling than mysql without having to jump through the hoops to make it scale. Of course my experience was about as much writing(inserts, updates) as reads. For stability, I don't think there is any problems with mysql or postgres. I only had problems with mysql under heavy load. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: Best DB
On Wed, 2005-03-09 at 15:43 -0500, [EMAIL PROTECTED] wrote: For some reason I didn't think PostgreSQL was for mission critical apps. I don't think I have any reasoning behind it, just didn't think it was hardcore...sounds like i might be wrong...i'll have to look into it more. Open source advantages are obvious, but aside from licensing and cost factors, I believe speed, security, and stability are going to be the key factors for us, whether open source or not. Postgres is probably more developed than mysql. Mysql gets a lot of press though as being an easy to install and config database. As for stability/scalability, the .org registry is on postgres. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] can I use an external modem such as USR robotics V92
On Wed, 2005-03-09 at 18:15 -0300, Jose Jorge Masdeu wrote: Hi, can I use an external modem such as Usr or other une attached at ttyS0 port on my linux box whith asterisk for routing incoming and outgoing calls? Simple search would have yielded many examples of why you can't. External modems must use the Hayes AT command set. The command set is not designed to allow for bidirectional comms traffic. So you would get into a cb radio situation where one side could talk but not the other. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digium : no lead time!
On Wed, 2005-03-09 at 15:18 -0600, Matthew Boehm wrote: Again, may be off topic but are there any cards out there supported by asterisk that have on-board DSPs to do better 729-711 or 729-PRI conversion? Not yet, and I don't know if anyone is working on the drivers for such a card. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] OT: Best DB
On Thu, 2005-03-10 at 00:37 -0600, Jay Milk wrote: So much for the history. As for usability, IB is your typical (almost) ANSI SQL-92 compliant database engine. It supports RI, triggers, stored procs, just like we all like'em. Its engine is touted for the superserver architecture but in years of working with it professionally, I still couldn't quite tell you what that is. I saw a few benchmarks floating around in the mid-90s were Interbase just rocked the heck out of Oracle, MSSQL and Sybase. I'm not sure if it's still in the lead on performance, but IB has supported SMP since version 7 -- firebird trails a bit on this. In practice, I've seen performance issues in databases of a certain size. If it stores the entire DB in 1 file, it can not scale as well as other DBs. Postgres 8 supports splitting a single DB up so you can put portions of it on different media if needed. If you have to tune for absolute speed, you can purchase one of the solid state drives for the tables that need that kind of speed while using much less expensive harddrives for the rest of the DB. While I do not remember mysql supporting it this directly, I think I remember the file structure being not to difficult to figure out and split and symlink back together if need be. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] using the i extension
On Tue, 2005-03-08 at 18:19 +0100, Florian Effenberger wrote: Hello everyone, what is the secret about using the i extension? I tried exten = i,1,Playback(invalid) but nothing happens when I dial a nonassigned number, it is completely ignored. Do you have wildcarded extensions in the context or in contexts that are included? -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] using the i extension
On Tue, 2005-03-08 at 18:52 +0100, Florian Effenberger wrote: Hi Steven, Do you have wildcarded extensions in the context or in contexts that are included? no, I don't. Then next step is to include your config files so we aren't just guessing at your problem. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Wildcard X100P or TDM400P?
On Tue, 2005-03-08 at 09:56 -0800, Spencer Nassar wrote: I'm looking to add a single FXO port to my Asterisk box. It looks like my options are a Digium Wildcard X100P off eBay for $6.99, or a Wildcard TDM400P with an FXO Module from Digium for $125. Can anyone explain the tradeoffs (other than the ability to put 4 FXO/FSO modules on the TDM400P). What about RTC for the system - I know the TDM400P provides it. Does the X100P? The X100P you are seeing on Ebay are not from Digium. They do not come with support time from Digium. They seem to only support US style analog lines(600ohm). If you have trouble, you are most likely on your own. TDM400P with FXO daughter card includes 1 hour of Digium support. It is supposed to support other line types. If you have trouble, it is likely you will get direct support from Digium and from the community here. TDM400P card is also capable of adding more ports without increasing interupts on your server. If you need to add another X100P later on, that will double the interupt load on the machine. With no one really making much money on the X100P card and I don't think anyone is making them anymore, you may not get new features added to the driver. The TDM400P card will probably be developed for a while to come as it the current option. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Wildcard X100P or TDM400P?
On Tue, 2005-03-08 at 13:46 -0500, Steve Prior wrote: Steven Critchfield wrote: TDM400P with FXO daughter card includes 1 hour of Digium support. It is supposed to support other line types. If you have trouble, it is likely you will get direct support from Digium and from the community here. It should be noted that several people including myself are having voicemail volume problems with the TDM400P (is anyone having it with the X100P?) which for us makes the card unusable for what was intended (a basic home PBX/answering machine). This is documented in bug #2023: http://bugs.digium.com/bug_view_page.php?bug_id=0002023 You'll also notice that it has been dormant for quite some time. Voicemail volume is not related to the card you use. If it was related to the card used, you wouldn't have the separation needed to make all the interfaces work. Voicemail volume is usually complained about by people not using standard wav format. It is due to the volume of wav files being manipulated at write time where as all other formats are as they come off of the line. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Wildcard X100P or TDM400P?
On Tue, 2005-03-08 at 13:24 -0600, Rich Adamson wrote: TDM400P with FXO daughter card includes 1 hour of Digium support. It is supposed to support other line types. If you have trouble, it is likely you will get direct support from Digium and from the community here. It should be noted that several people including myself are having voicemail volume problems with the TDM400P (is anyone having it with the X100P?) which for us makes the card unusable for what was intended (a basic home PBX/answering machine). This is documented in bug #2023: http://bugs.digium.com/bug_view_page.php?bug_id=0002023 You'll also notice that it has been dormant for quite some time. Voicemail volume is not related to the card you use. If it was related to the card used, you wouldn't have the separation needed to make all the interfaces work. Voicemail volume is usually complained about by people not using standard wav format. It is due to the volume of wav files being manipulated at write time where as all other formats are as they come off of the line. Steve, help me understand exactly what you said above. Having problems with to the card you use and related to the card used. By the time any audio makes it to voicemail, it has been normalized into ast_frames. The source of audio is not relevant to the voicemail app. Also, please clearify not using standard wav format. Does that truly mean recording voicemail messages in gsm format is the cause for the additional 10db of loss measured and noted in bug 2023? I don't know or care how you are measuring a 10db loss. I know for a fact that in format_wav.c there is a section of code that effectively doubles the volume as it is saving and removes the doubling on playback. No other audio format is given the same treatment. If that's the case, then why has bug 2023 been lurking without any such comments for many many months? If it is lurking for 10 months it is because no one cares to read it. I promise you I don't look at any bug unless it is mentioned on a list as is pertaining to code I am running. I have so little time now, I don't go looking for needles in haystacks that I don't need. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Adit 600 for asterisk
On Tue, 2005-03-08 at 14:07 -0600, Dennis Webb wrote: Ok, I've pretty much decided to try the Adit route. Somebody who has experience with these tell me if I'm missing something. I have 15 incoming PSTN lines. T1 is not an option at current location. I want to put in an Adit 600 with 2 8-port FXO boards. The adit will then connect to * via a digium t1 board. I configure zaptel.conf for the T1. What other parts would be needed? How do the PSTN lines connect to the Adit, standard rj11 jacks? It looks to be about a $2500 investment and I need to know if there is anything special I am missing. The Adit has a 50 pin D connector that is like the old SCSI connectors. To connect your phone lines, you either want to get a punch down block that has a 50 pin connector on it, or possibly a device with RJ11 style jacks and a 50 pin connector. Then you connect it with a 25 pair cable with 50 pin connectors on either side. Go to any reputable supplier near you and they should be able to help you look at and find what you are comfortable with to use for installation. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Wildcard X100P or TDM400P?
On Tue, 2005-03-08 at 14:27 -0600, Rich Adamson wrote:
On Tue, 2005-03-08 at 13:24 -0600, Rich Adamson wrote:
If that's the case, then why has bug 2023 been lurking without any such
comments for many many months?
If it is lurking for 10 months it is because no one cares to read it. I
promise you I don't look at any bug unless it is mentioned on a list as
is pertaining to code I am running. I have so little time now, I don't
go looking for needles in haystacks that I don't need.
The only reason for questioning the above is that Mark (and several others)
have contributed various comments to bug 2022 and 2023, but at no time
have any of them ever mentioned comments relative to wav vs other file
formats, and effectively have left the bug unresolved.
Go look at the code for format_wav.c. I have code in there and have
actually looked at it myself.
This is at line 64 of my code base.
#define GAIN 2 /* 2^GAIN is the multiple to increase the volume by */
And this around line 480 in the wav_write function.
if (fs-buf) {
tmpi = f-data;
/* Volume adjust here to accomodate */
for (x=0;xf-datalen/2;x++) {
tmpf = ((float)tmpi[x]) * ((float)(1 GAIN));
if (tmpf 32767.0)
tmpf = 32767.0;
if (tmpf -32768.0)
tmpf = -32768.0;
tmp[x] = tmpf;
tmp[x] = ~((1 GAIN) - 1);
f is the frame to be written. f-data is the audio data to be written.
The for loop just uses pointer math to traverse each sample of the
buffer and increase the volume of the sample with clipping control.
--
Steven Critchfield [EMAIL PROTECTED]
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Re: [Asterisk-Users] Asterisk MySQL Blobs
On Mon, 2005-03-07 at 16:06 -0500, Eric wrote: Hi Vinko, MySQL blobs will store binary data, so you should be OK there. I'd focus on whether or not storing the data in a variable is a good idea. Typically, with any programming language, it's good practice to keep variable lengths short so you aren't passing the variable itself between functions. I can't say if that could cause performance issues under higher load. I'd love to hear how you make out, as well as anyone else's input. With as much as I despise the idea of the audio being stored in a DB much less MySQL, it shouldn't be much different from it being in NFS as long as MySQL can keep pace. Ideally you would stream the results up fom the MySQL driver and read the appropriate chunk size and send it through asterisk. The downside is the need to essentially rewrite formats/* to deal with whatever file formats you want to store in the DB or be incomplete. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk MySQL Blobs
On Mon, 2005-03-07 at 13:30 -0800, beonice wrote: --- Colin Anderson [EMAIL PROTECTED] wrote: IMO, use filesystem for files. Use DB for DB. Put a pointer in a field to the file. Your DB will love you for it. The problem I suspect will arise is the number of inodes allowed by the file system. I don't know the exact size of the typical inode-max, but this will also presumably become an issue when the user tries to scale to really large amounts of faxes or voicemail! Would it help to split the db off to a separate server (that should reduce the CPU load on the asterisk server)? Any other alternatives? Anyone verified whether the BLOB storage solution breaks down first or the number of inodes runs out first? :) On a 40gig drive I have, there are 4.7million inodes. On an 80gig drive I have there are 9.4million inodes. Block size is (I think) 32k on these machines. At 32k I could only use 2.6 million inodes pointing to minimum sized files on the 80gig drive. If you are worried about inodes, I believe it is xfs that dynamically creates inodes as needed and won't run out. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Question about AGI vs. FastAGI vs. straight C/DB development
On Mon, 2005-03-07 at 14:21 -0800, beonice wrote: Folks, I want to build a custom IVR for my setup. I've got it working (well, the bells and whistles are not there yet, but the basic stuff works) using AGI, but I'm worried about how well this will scale. I've seen references to FastAGI, and presumably this will be more efficient. Question, though: how well do either of these (AGI or FastAGI) scale if my system is handling a large number of simultaneous calls? I'm assuming that even with FastAGI, there's going to be some CPU overhead that will increase as the number of simultaneous calls increases. Would it make more sense to write a custom application in C instead, designing it to work sort-of like the built-in app_voicemail.c and others? I do know C, but is it worth the effort in terms of ROI? :) I'd like to figure this out in advance rather than after all my customers (hopefully there _will_ be customers) start yelling at me about performance issues. :) You said a magic word there, customers. You need to become versed in the ins and outs of licensing. If you write it in C and distribute it, you must also offer to those you distribute asterisk your modifications and linked in code. For AGI and FastAGI, it isn't linked to GPL code in asterisk and therefore is sheltered from the GPL license of asterisk. But then you need to verify you aren't using any libraries that are distributed under a GPL license or you would also have to open source your code upon distribution. FastAGI offers faster initial response times than AGI as it will most likely already be running and ready to accept a new call. FastAGI should also be capable of being load balanced if that process is doing too much work to be responsive upon scaling. AGI is not capable of being run from a different machine. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Question about AGI vs. FastAGI vs. straight C/DB development
On Mon, 2005-03-07 at 19:19 -0500, Karl H. Putz wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Steven Critchfield Sent: Monday, March 07, 2005 6:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Question about AGI vs. FastAGI vs. straight C/DB development snip You said a magic word there, customers. You need to become versed in the ins and outs of licensing. If you write it in C and distribute it, you must also offer to those you distribute asterisk your modifications and linked in code. For AGI and FastAGI, it isn't linked to GPL code in asterisk and therefore is sheltered from the GPL license of asterisk. But then you need to verify you aren't using any libraries that are distributed under a GPL license or you would also have to open source your code upon distribution. Please correct me if I'm wrong here but, the script/process called by the AGI could be any executable adhering to the AGI standard. This could be a compiled C or any language executable that simply parses the startup info provided by the AGI and communicates with * via stdin and stdout. This means that the executable would be standalone and not need to be linked to the GPL code. I think this means that the source of any AGI application would not be subject to the GPL licensing if the author would so choose. My first paragraph was incomplete. When I said to write it in C, I meant a linked to asterisk application. I didn't say it explicitly, and it wasn't clear to assume it either. Yes you can write an AGI or FastAGI in any language including C. At that point you are only possible limited by the libraries you choose to include with your application as to whether or not the GPL applies. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P Clone, Which one?
On Sat, 2005-03-05 at 19:30 +, Mike Dent wrote: And if nobody's going to educate the newbies, then how will they ever learn? Do you believe in letting your children do whatever they want, too? There are 'defacto' rules for any system. No, I don't have my shiny ListCoptm There is a difference when you are that childs father or mother but you are neither? Seems shop owners don't mind telling a person how to behave or leave. Any time you are in an enclosed space, people attempt to enforce rules. Do you stop people in the street when you see them doing things wrong and try and tell them you shouldn't be doing it like that, this is how you should do it. I hope not :) Essentially this list is the same. While you may not stop people who haven't asked for help, but once you enter a public forum and request the help you need to be following the rules or at the least be respectful of those willing to give the help. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems with g729 codec
On Fri, 2005-03-04 at 13:29 +0100, [EMAIL PROTECTED] wrote: Hello, I´m trying the g729 codec for testing pourpose. Whe I try to make a SIP call from a phone using g729 codec to another phone using another codec, when the destination phone answer, the call hangs up. this happend in both ways. In the asterisk console I get. Mar 4 13:11:35 NOTICE[24572]: channel.c:1724 ast_set_write_format: Unable to find a path from gsm to g729 What does it mean? Could this occur cause I am using the g729 without licence? If i buy a licence could solve my problem? G729 will not work without a license. The error message above told you that asterisk couldn't find a valid path to convert from gsm audio to g729 audio data. Seems that should have been very obvious from the error. It is well documented had you even decided to search. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem getting Voice Contract script to work
On Fri, 2005-03-04 at 22:31 +0800, mechaman wrote: Hi, wondering if anyone can help me with my problem. I can't get the verify.agi script to work in Asterisk This script is available for download at http://www.sineapps.com/downloads.php The agi script works for recording and playback when accessing it directly at it's extension, but will not record anything when doing the flashhook procedure during a call. Recording is cut off after the flashhook What interfaces are you using? Did you realize that it is making a three way call to get the recorder in the mix? So do you have three-way calling enabled on whatever interfaces you are using? -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems with g729 codec
On Fri, 2005-03-04 at 12:02 -0500, Erick Perez wrote: sorry to ask, but what does it mean in passthrough mode ? data, in this case audio, passes from one side through to the other with no need for modification. A standard serial cable is a passthrough cable. Same for standard network patch cables. The software here behaves much the same way, it picks the audio data out of the packet and passes it through to the other side of the communication. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Placing a call from command line and passing it to an extension if connected - Is it possible?
On Fri, 2005-03-04 at 13:41 -0700, Joseph wrote: Is it possible to dial number from the command line and passing the connection to one of my extension (or speakerphone) if the other party answers the call? I was thinking of implementing this sort of feature with and accounting application. The customer phone number is in the database, so clicking and icon asterisk would dial the number and connected to my speakephone when the connection goes through. lookup sample.call -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voice over Frame Relay Asterisk
On Fri, 2005-03-04 at 12:44 -0800, asterisk phones wrote: Has anyone done Voice Over Frame Relay with Asterisk. With Frame Relay work reliably with Asterisk? Any experiences? Doesn't look like you visited google first. Nor did you bother to look at the code. channels/adtranvofr.h [EMAIL PROTECTED]:/usr/src/asterisk/configs$ more adtranvofr.conf.sample ; ; Voice over Frame Relay (Adtran style) ; ; Configuration file [interfaces] ; ; Default language ; ;language=en ; ; Lines for which we are the user termination. They accept incoming ; and outgoing calls. We use the default context on the first 8 lines ; used by internal phones. ; context=default ;user = voice00 ;user = voice01 -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Placing a call from command line and passing it to an extension if connected - Is it possible?
On Fri, 2005-03-04 at 13:58 -0700, Joseph wrote: On Fri, 2005-03-04 at 14:45 -0600, Steven Critchfield wrote: On Fri, 2005-03-04 at 13:41 -0700, Joseph wrote: Is it possible to dial number from the command line and passing the connection to one of my extension (or speakerphone) if the other party answers the call? I was thinking of implementing this sort of feature with and accounting application. The customer phone number is in the database, so clicking and icon asterisk would dial the number and connected to my speakephone when the connection goes through. lookup sample.call Lookup where? on Wiki web-page. If you can't figure that out, you are unlikely to implement anything. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PRI HDLC Abort (6) Errors
On Fri, 2005-03-04 at 15:27 -0700, Tom wrote: Hello, I have searched and searched, and come up with nothing. I am running Asterisk with a wcte110p configured for t1. Our PRI is staying up, and we can make calls however our service provider's logs are flooding with errors and we are getting lots of HDLC Abort (6) on Primary D-Channel Errors. Our provider says it looks like our box is trying to be the master timer on the circuit (which is not correct they are providing the timing) we have tried both span=1,1,0,esf,b8zs and span=1,0,0,esf,b8zs in zaptel.conf both produce the same problems. The problem is not in Asterisk per se as the errors start happening as soon as I modprobe the driver and run ztcfg. As soon as the circuit comes up the errors start on the provider's end. Did you make sure to power cycle afterwords? Sometimes the zap cards don't change critical settings like timing once configured. We are running CVS Asterisk/zaptel/libpri from March 2nd 2005 on Fedora Core 3 fully patched as of last night, I was thinking the problem was with the 2.6 kernel getting preempted and therefore the driver not being able to do its timings right, however fc3's kernels have preemption disabled by default. Does Digium hardware really need/expect a real time OS to run properly? Like I said previously I think the problem is in the driver itself not in asterisk. Any help would be appreciated, and I can code a bit in c so if someone can point me in the right direction I might be able to fix it myself... You probably want to dump the FC kernel like a bad habit. Get a plain vanilla kernel and see if that fixes your problems. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MGCP to Inter Tel system
On Thu, 2005-03-03 at 12:23 -0800, Dustin Moore wrote: I've been trying to figure out if it's possible to connect Asterisk to a parent Inter Tel Axxess system through the MGCP protocol. The archives for this list aren't searchable and I'm wondering if anyone has a simple answer... The archives are searchable. Use google and learn about the advanced options such as limiting to a site. Leo answered the other part, MGCP is modeled much like normal phone networks where there are 2 distinct roles and they talk only to the other role. Asterisk and the InterTel play the same role and therefore can't talk to each other. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help for studying Asterisk source code
On Thu, 2005-03-03 at 16:20 -0500, Amit wrote: Hi Everyone, I am student and I have to study about the source code of Asterisk. I have downloaded asterisk and was able to install it on Red Hat Linux. My study is to go into the source code of asterisk and see how it works, how the asterisk works when anybody calls to it or when it makes a call which functions are called. I have to make a chart like which can explain how everything works. Is there anyone who can help me with that? Any kind of help would be great. Homework questions shouldn't be community work. If your homework is to read and understand the source, then you have a lot of reading to do. You may want to look at the progdocs generated when you do a make progdocs -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [OT] - [Asterisk-Users] Why should I answer a Newbie question,therethick!
On Thu, 2005-03-03 at 17:59 -0700, Paul Fielding wrote: - Original Message - Look, don't answer lame questions if you don't want to. Flaming a newb for being a newb is just mean. (they will eventually RTFM or STFW or they will fail). This is the way of the open source community. Here Here, I'm with you. I find it a constant source of amazement how, in all the various lists I've followed, people find it necessary to beat on the new guy. Even the 'if you don't want to get flamed then do some research first' attitude i'm not a fan of. Sometimes newbies are also newbies to the concept of lists, etc, as well as the topic of the list. Frankly, I agree. If you don't like the question, feel it's lame or dumb, or don't like that someone hasn't done their research, then delete the message. If you think they're wasting your time by writing a message, then don't waste any more of your own time by responding to it. I find the pummelling of newbies more annoying than the newbie question itself. As I have told others before. This list is a valuable resource even for those of us who know a lot about asterisk already. The users who would rather come here than do any work on their own become pollution to this list. They are the repetitive spam that has no benefit to the ones receiving it. This type of behavior has run many of the gurus off of this list. They are unlikely to be replaced. BTW, telling a user they haven't done the prerequisite home work before asking a question is no where near pummelling. Pummelling would be calling them names and making personal attacks. Life is full of venues where you need to meet specific criteria before you are considered worthy of interacting. All that to lay the ground work to say that when we send a user back out to the search engines to do their homework, we do so as a jealous protecting of this forum and what value we receive from it. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [OT] - [Asterisk-Users] Why should I answer a Newbie question, therethick!
On Thu, 2005-03-03 at 17:09 -0800, Jeff Busch wrote: As someone who is new to Asterisk and Linux (I guess I am a newbie), but who has been doing a ton of research, Google searches, and is getting to intimately know the wiki, I take offense to Steven Critchfield's commentary about newbies. 1. Before asking a question, do a Google search 2. After a general Google search, do a specific search on this group 3. After a Google search, look at http://www.voip-info.org/wiki-Asterisk the information contained in these pages will answer 95% of your startup questions. 4. If you have done 1, 2, 3 - feel free to email the list. 5. Please do not email the list asking people to hold your hand. That is not what the list is for, it is for help if you run into an implementation problem, not to teach you the basics by using 1, 2, 3. After trimming 4 or so email messages out of this that didn't seem to be relevant to your reply other than being in the same thread, I am left wondering what exactly you took offense to. It seems the portion left below matches pretty much with your comment above. We both seem to feel it is important to do the searching first and not expect hand holding. You and I are much more the same than not. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Critchfield Sent: Wednesday, March 02, 2005 11:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Why should I answer a Newbie question,therethick! The payment is not always monetary. Sometimes the payment is just a showing of sufficient effort. Back to your snow driver analogy, if the driver in the ditch is just waiting in the car for you to come over and push them out without even attempting anything on their own, you would be less inclined to bother. You would be even less inclined to continue exerting your own effort if the driver was not cooperating or wasn't even interested in getting out to help push. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Why should I answer a Newbie question, there thick!
On Wed, 2005-03-02 at 09:38 +, BCS Support wrote: It would be nice just for once to actually use a mailing list with people who are a little more sympathetic to the fact that your not a rocket scentist or molecular biologist and that you might actually need some help, without being made to feel like your completely useless and should be cleaning toilets for a living. But someone has to clean toilets. Microsoft isn't the only one to blame for the reported millions of infected computers on the internet. Some people really shouldn't be tasked with administration of a machine let alone the provisioning of a phone system. Ahhh man not another stupid newbie question! are these people completely lazy and thick? lets postup some sarcastic comment! --- really usefull! Yes I have spent hours researching on Google, but what may take me 3 days to workout, wading through pages of out of date information, can normally be answered by some with a little experience in seconds. Sounds like either you haven't made it up the first part of the learning curve and/or your search/research skills still need work. The latter part is not within the scope of this mailing list. If you can do basic research, you should be able to then ask direct questions with the proper amount of details that it becomes an interesting question to those who you would want to answer your question. Opensource is about a freindly, helpfull community of people who instead of choosing the large corporate companies, decide to give the little guy a chance. As others have pointed out, this is not true. No where is friendly a requirement. No where is helpful a requirement. No where is opensource a anti corporate tool. Examples, Go look up the reputations of some of the biggest names in opensource software. ESR went to the trouble to write a fairly long paper just so you could learn how not to get quickly flamed even by himself. There are some well known BSD developers that are well known for their treatment of people who don't agree with them let alone just act like newbies. Don't put people off just because their not the next Albert Einstein, otherwise the likes of Microsoft etc. have already won! Microsoft did win, we are just about to take the trophy away. Maybe you should use that whiny line on your next employer and see if it helps you get the job. Of course you might find that whiny adults actually generate a repulsive reaction by other adults and you are actually more likely to receive more of the same treatment. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dual Asterisk Servers
On Wed, 2005-03-02 at 09:22 -0600, Nik Martin wrote: The relatively new switch dialplan command seems like it will assist in accomplishing this, but does anyone have a simple IAX config and dialplan that will help me understand how this all works? switch is OLD. I have been using switch now for over 3 years. switch is easy, you just include it in the appropriate context and it will contact the remote machines for dialplan completion. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Why should I answer a Newbie question, therethick!
On Wed, 2005-03-02 at 11:01 -0500, Race Vanderdecken wrote: This list is owned by no one. Actually it is owned by Digium. It has many contributers though. When someone drives off the road into a ditch in a snow storm they last thing they need is someone telling them they should have invested in snow chains and defensive driving lessons before leaving the house. Newbies need help getting out of the ditch so traffic can continue to flow and the rubber neckers can be abated. If you are not willing to pull off to the side of the road and help the fool by pushing their car out of the ditch you have no right to give him the finger as you drive past. As a person who spent 9 hours in traffic last winter just to drive 15 miles due to idiots who should have just stayed home, I think your analogy breaks down. At some point, you either need to learn to drive or you pay someone else to transport you or your stuff. Same applies to computer work, either you can do the work yourself or you pay someone else to do it. Even your snow driver analogy works here, you either get yourself out of the ditch or you pay someone to do it for you. The payment is not always monetary. Sometimes the payment is just a showing of sufficient effort. Back to your snow driver analogy, if the driver in the ditch is just waiting in the car for you to come over and push them out without even attempting anything on their own, you would be less inclined to bother. You would be even less inclined to continue exerting your own effort if the driver was not cooperating or wasn't even interested in getting out to help push. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] newbie questions
First tip, use a descriptive subject line. On Wed, 2005-03-02 at 15:14 -0500, Jean-Francois Theroux wrote: Hello, At the office we have a Lucent PBX, which has 3 lines coming from the CO. 2 are used for phones, 1 for fax. In the office, we have 16 phones. All those are connected in the PBX. We do not have an automated system nor voicemail system for now. But this is something we would like to have now. Since we do a lot of work with Linux, I was asked to look into asterisk to deplace our PBX. Software-wise, I don't have any problems yet, doesn't look too bad hard to configure. Now, I know I would need a quad-port FXO card for our lines coming in from the CO in that PC. What would be the best way to connect all those 16 digital phones to the Asterisk box? I could always buy quad-ports FXS cards for now, as we don't use the 16 phones, but I don't think that's going to work well in the future when the company grows and we require more phones. Unless the phones are IP phones, you won't be able to use them directly. You may wish to look at what is possible with your current PBX to put asterisk to the side of it to handle your IVR and/or your voicemail. You could even use it as a VoIP gateway. It all depends on what features your current PBX will allow you to implement. If you would rather replace your PBX completely with asterisk, you really need to look at a T1 and a channel bank. Adtran and CAC Adit 600 units are modular. The Adit units use 8 port cards to get service. So you could have an 8 port FXO card and 2 8 port FXS port cards. This would let you hook up 8 incoming lines and 16 extensions. Of course since all the channels are programmable, you get a third FXS port and you can mix and match ports 1-8 FXO and 16-24 FXS ports as long as you don't exceed 24 ports. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [OT] stupid firmware question...
On Wed, 2005-03-02 at 14:24 -0600, Chris Wade wrote: I know this is a really stupid question, but I just have to ask... Where would I start if I wanted to try and develop my own firmware for a particular phone. Namely, I want to try and 're-write' the SIP firmware for Cisco 7940's. Any ideas? -Chris PS: [* put on flame suit *] why won't any of the phone manufacturer's just open-source the firmware for their phones? [* ducks head back inside gopher hole just before a giant fireball hits *] Simple, it is likely that they put the code for the codecs in the firmware. They have to ensure you pay your patent license fees. Granted it is easy enough to get around once you have a firmware image for the ciscos, but it doesn't let them off the hook on how they must guard the code. If Cisco didn't have access to G729, most people would laugh at their offerings and quickly jump to another product. If you where going to rewrite the firmware, you should try and make an IAX2 firmware for it. In fact, you should try and contact Cisco and see if you could partner with them to write the code for an IAX2 firmware and get help from them. They would be able to incorporate the codecs they have licenses for and charge money for the product. Might be interesting. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Has anyone seen this before
On Wed, 2005-03-02 at 16:19 -0500, Kevin Collins wrote: During an answered call I'm seeing these stream by in the CLI every ten seconds or so. Mar 2 16:16:13 NOTICE[16453]: rtp.c:505 ast_rtp_read: Unknown RTP codec 72 received Thanks in advance for any idea. Simple search would have told you that it had been seen and discussed before. http://www.google.com/search?q=Unknown+RTP+codec+72+received+site%3Alists.digium.com -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Has anyone seen this before
On Wed, 2005-03-02 at 19:03 -0500, Andrew Kohlsmith wrote: On March 2, 2005 05:58 pm, Steven Critchfield wrote: Simple search would have told you that it had been seen and discussed before. http://www.google.com/search?q=Unknown+RTP+codec+72+received+site%3Alists.d igium.com Actually, Steven, I was poised to reply the exact same way but I actually checked out the links that Google provided. They were wholly unhelpful. Next case would be to examine the source and see wtf it's printing that for, but this *is* -users. I left the question thinking someone else would fall for the trap (that, and I do not use SIP at all), but I didn't imagine it'd be you. :-) My answer was more about the subject than the rest of it. As the link above and your comments suggest, it took near no effort at all to prove the answer to the subject line. One more of the fun google tricks is to use -site: to trim out a specific site that ranks highly in your google searches. If you remove lists.digium.com from your search for rtp codec 72, you find a bug report on mantis pretty easily. http://bugs.digium.com/print_bug_page.php?bug_id=2945 It would hopefully point out a bit more. Either way it demonstrated lack of effort. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] in calling
On Tue, 2005-03-01 at 10:53 +0100, Satchid wrote: Dear Users, Thank you all for a great Forum. In the past with my analog PBX I had 6 separate telephones configured auto pickup and they where connected to the meeting room's microphone. When we had a meeting, it was then possible for 6 members simultaneously to call in with a normal landline and be transferred to one of the phones. In this way he could be part of the meeting. Is it possible to configure that in the Asterisk? And how would I go to do this? Would this be possible with only one telephone in the house and 6 members to be able to connect simultaneously? Or is there something in the * that can do this? Go read the docs. Read them one or two more times after that. You need meetme. Read about it. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with Asterisk Unable to allocate channel
On Tue, 2005-03-01 at 07:52 +0100, Edgar de Leon wrote: Hello, yesterday when i wasnt in the office the asterisk server stop working, it was registering the sip terminals but cant make calls, because im not in the office i told the people to reboot the server to make the server works again but today i found this lines in the full log, can anybody tell me what was happend? Feb 28 17:30:07 WARNING[6351]: Unable to allocate channel structure Feb 28 17:30:07 DEBUG[6351]: build_route: Contact hop: luiscarlos sip:[EMAIL PROTECTED]:5060 Feb 28 17:30:07 NOTICE[6351]: Unable to create/find channel First, You need to know about how email messages are threaded. You replied to a message in the How does the g.729 registration program work thread, written by Kristian Kielhofner. Your message has nothing to do with the message you responded to. Next. You need to provide critical information such as what version of asterisk you are using. If it is from CVS, you MUST provide the date it was last updated. Most likely you are running an old version that had a memory leak with variables not being freed properly. I say that because the first message above looks like you have run out of ram and swap. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] agi RECORD FILE with offset
On Tue, 2005-03-01 at 13:42 -0500, John Hammen wrote: Hi All, I've been playing about with the RECORD FILE agi function and am finding two distinct problems with the resulting wav file when using a non zero sample offset. Specifically, I call the function with a zero offset and a given filename (the original recording), and then later call it with the same filename and a non-zero offset (the overdub). When I do this, I experience the following: What file format are you using and what version of asterisk are you using? There had been a bug for all file formats with a header(wavs) that would mess up the recording. A bug was filed with a patch that was accepted during the 1.0.5 deployed time. If you are below 1.0.5 then you are sure to not have the patch. If your 1.0.5 is older than Feb-06, you don't have the patch. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
