[Asterisk-Users] Re: FXS or VOIP
Hi Jim, My decision had more to do with the infrastructure of the existing wiring more than anything else. I really *wanted* to go with voip but I couldn't justify the extra cost since our office is wired for analog. I ended up going with the TE410P Quad span T1 card, 2 PRIs and an adit-600 channel bank for the FXS ports. I really had to do very little to tune the FXS ports other than setting tx and rx gain on the channel bank. We have 5 other branch offices that we are connected to via WAN and we have * servers at each of those locations, doing voip between those and also the larger install that I describe above. So just because you have FXS ports does not mean that you cannot do voip. There's always services like nufone for long distance that you can connect * to. For your smaller setup just evaluate what's there already in terms of network infrastructure then decide what fits best for both your budget and your growth. Best Regards, Jason Stewart On 11/01/06 15:06 -0600, Jim Freeze wrote: Hi I am setting up a phone system for a small office. The office will have 5-8 phones and a fax line. There are 4 hunt lines coming into the office. We have made no hardware purchase yet. Being an asterisk newbie, before I suscribed to this list I just assumed that I would buy voip phones and connect all the phones to a private ethernet network. However, I see many people inquiring about FXS cards. Is there any reason why I would need to consider using analog phones and FXS cards? Seems to me the cheapest way is with voip phones and voice quality should be good since the phones are on a private network that only has voice traffic. Thanks -- Jim Freeze ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Disable Console Audio
On 22/07/05 02:49 +0900, Kuniyoshi Murata wrote: Hi, Now, I think I want to disable Asterisk's access to console audio device based on the logic above. How can I do that? Make sure the following is in your modules.conf file: noload = chan_alsa.so noload = chan_oss.so ___ 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
[Asterisk-Users] Re: Busy Extensions.
On 21/07/05 15:22 -0400, Tim King wrote: I seem to have almost everything working now. The only problem is all of my extensions seem to be busy. I can call out, but not in. Can someone point me to the settings in the extensions file that could cause this. Hi Tim, Nice to see a fellow Grand Rapidian on the list :) It looks like you're using AMP, which makes the troubleshooting process hard since we cannot ask you for your extensions.conf. Check that the extension that you're dialling is set up correctly. If you can dial out then this is probably the problem. What kind of hardware are you using for FXO? Jason Stewart ___ 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
[Asterisk-Users] Re: IAX over HTTP
HTTP uses TCP. Too much overhead. Add SSL on to that and you have a huge amount of overhead. The end result would be poor and choppy sound quality. Jason On 21/07/05 21:58 +0200, Rob Scott wrote: For work environments where you only get HTTP or HTTPS access, what is the feasibility of doing IAX over HTTP? I have heard of projects such as stunnel. Has anyone tried something like this? I did a quick search but didn't come up with much. ___ 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 ___ 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
[Asterisk-Users] Re: So you all think VoIP sypply is warm and fuzzy
On 18/07/05 17:06 -0700, Michael D Schelin wrote: I was waiting for everyone to reply so here is mine.. Check out the Mediatrix web site. There are no downloads or lists of resellers who might have this provisioning software that is normally included with purchase. You may be right that it is a refurb but every indication points that it is not. I have contacted both companies and I'm waiting for replys. I'm on the west coast and it took over 7 days to get here. I am a little pissed when all other ATA's are configurable from their built in web server. And Yes, I'm self serving as well as mostly everybody I've ran into in this business. This unit was purchased for testing. Because of the timezone problem, When I get the product from UPS it's too late to call Canada or FL. when all I need is a simple download to correct the problem. Is it too much to expect everything in the box when you purchase it? Or have a web site with these free included software so if this happens we don't wast our valuable time. By the way I did get an email from VOip Supply asking me to wait until morning so they could find the software. This is at 2:30 PST. This complaint was to hear from others about VoIP Supply and their business practices. I wanted to get feedback ether way, or maybe a contact name so I can get this paper weight working and tested. Has anyone used the 2102? Please let me know. Obviously you have a misunderstanding. Why not assume that there is a misunderstanding, with voipsupply then work from there instead of dumping your anger out on all of us? I don't doubt that there is a CD or there was once a CD that shipped with the 2102, but - According to the Medaitrix Web Site... --- Copy and paste from mediatrix web site --- With the Mediatrix 2102, service providers get the product characteristics allowing them to successfully deploy residential IP telephony applications. The Mediatrix 2102 provides a web interface, giving users a convenient access to the unit for initial set-up. The Mediatrix 2102 can auto-provision by fetching its encrypted configuration file from a TFTP or HTTP server making installation transparent to end-users. To further facilitate deployments, factory loaded configurations are possible. Automatic firmware and configuration file downloads ensure that the 2102 is always up-to-date. --- end --- You are supposed to use a web interface for initial set up. Jason ___ 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
[Asterisk-Users] Re: SS7
On 07/06/05 11:30 -0400, Matt wrote: Hi, Has anyone used the SS7 link from Digium? If so, how did it work for you? Any issues? Anything to be aware of? Do I just need a T1 card like the PRI card I have now from Digium? Hi Matt, There are some links to user reports on the wiki: http://www.voip-info.org/wiki-Asterisk+SS7 It also looks like your Digium PRI card will work too. If you're in doubt call Digium, I'm sure they would answer your questions. Jason ___ 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 ___ 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 13:00 -0500, Giudice, Salvatore wrote: MySQL: Speed, Power and Precision _ Speed, yes. Anyone can write an SQL layer over a flat file and make it fast. If you want real speed (faster than MySQL with the same level of reliability choose SQLite. Power - I agree here too. There are lots of great tools for MySQL due to it's ubiquity. Precision - No Way! see- http://sql-info.de/mysql/gotchas.html MySQL is free. It can be installed in less than 59 minutes from source for light use by a first time user AND there is no need for extravagant tuning. and if you are particularly keen on undertaking elaborate tuning projects to squeeze every last drop of life from a database, you can even write your own database engine for MySQL. So a beginner user can install MySQL in less than an hour from source with no need for tuning, but if they feel the need to tune their database other than what's out of the box a newbie can write their own database engine? I'd much rather mess with a few config options that write a database engine. For the record PgSQL can be installed in the same amount of time as MySQL. For the extreme noob who knows nothing about databases and is still learning then tuning will not be a factor. For anyone else the first thing that they'll do is look at the manual for the tuning section. It's not rocket science. If you are so keen on paying for something, try buying support - MySQL AB. With PostgreSQL, you could get support from a mom and pop shop... However, either way you will save tons of money over Oracle. You could also get enterprise level support through Pervasive, a company much larger and older than MySQL AB. http://crn.com/sections/breakingnews/breakingnews.jhtml?articleId=57700307 For benchmark information comparing MySQl with several DB's on various OS's (yes Oracle and PostgreSQL are included) see the following link: http://ftp.iranscience.net/pub/databases/mysql/information/benchmarks.ht ml Hmm... More benchmarks, eh? I've see benchmarks swing both ways with MySQL being faster and others with PGSQL being faster. In my experience Postgres has handled our multi-gigabyte database much more smoothly than MySQL. Larger, complex queries seem to return much more quickly with Postgres. My mantra is pick the right tool for the job. For smaller webapps I use MySQL. For huge enterprise databases I use PostgreSQL. Regards, -- Jason Stewart | Tel: 616-532-2300 Systems Administrator/ | Fax: 616-532-3461 Programmer | Email: [EMAIL PROTECTED] Right to Life of Michigan | Web: http://www.rtl.org ___ 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
[Asterisk-Users] Re: asterisk@home scary log
On 10/02/05 15:10 +0100, Jean-Louis curty wrote: so I stopped asterisk, type mail and got a strange mail saying that user [EMAIL PROTECTED] could not be reached and body was like if it was the result of commands ifconfig etc unfortunally I don't have the message anymore but I went to the log Feb 9 20:30:17 asterisk1 sendmail[10093]: j1A1U7mf010088: to=[EMAIL PROTECTED], ctladdr=[EMAIL PROTECTED] (0/0), delay=00:00:10, xdelay=00:00:10, mailer=esmtp, pri=30329, relay=gsmtp171.google.com. [64.233.171.27], dsn=2.0.0, stat=Sent (OK 1107998984) the thing is i did not send any message to [EMAIL PROTECTED] nor to somebody at yahoo, anybody got the same ? what can I do ?? There's a chance that you may have been hacked, but the logs you post look more like your mailserver is an open relay. What OS/Distro are you using, what version, and do you have the latest patches applied? What services are you running? Look for strange entries with uid 0 in your passwd file. Also check for root kits with a rootkit checker (chkrootkit.org). If everything pans out security-wise then the only problem is that you MTA is configured to be an open relay. If that's the case, then you need to fix it right away before you get on umpteen million blackhole lists. Consult the docs and/or community for the MTA that you're using to fix that. Jason ___ 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
[Asterisk-Users] Re: VOIP Phone Suggestions
On 15/12/04 22:53 -0600, Kevin Curtis wrote: I would recommend Uniden UIP200 phones. Great sound quality with inbuilt phone book, call logs etc works great with asterisk. I recently purchased from [1]www.qualvoip.com (they also provided me sample configuration files for asterisk). Kevin One gripe about these guys - They clearly use * for their PBX product, which looks like it's not much more than * with a web based config interface. There's not one mention of * on their site! No, there's nothing wrong with that legally but they should be giving props to * instead of promoting it as their PBX software. Instead of calling the product The Asterisk Based PBX System they call it The Open System Based PBX System. Are they afraid that potential customers will discover * and try to do it on their own? Jason ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] [OT] Adit 600 Question
Hi, I'm using an Adit 600 Channel Bank with *. I love it and it works really great for my FXS lines. One problem that I have with it (It's really not a problem yet, but it's a potential one) is that I've scoured the manaual for the Adit to see if there's a way to dump out a config file from the bank so in the event of a power and battery failure I don't have to type in the configuration commands, just load a file. Is there a way to get a config from the Adit 600 and load it back in again? Thanks, Jason Stewart ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Old Dialogic Hardware Questions
On 09/11/04 16:13 -0500, Matt Gibson wrote: Hi Everybody, I have a quick question regarding some old Dialogic hardware. We have an old Artisoft PBX (http://www.artisoft.com/PBXPhoneSystems.html). In this box are some older ISA Dialogic cards. My question is, does anyone know if the following Dialogic cards work with asterisk or in Linux at all? They are not mentioned on the digium site as supported, nor could I find anything specific to these cards on the mailing list archives. Dialogic D/80SC-4LS and Dialogic MSI/240SC-Global Thanks in advance, Matt Hi Matt, There's not many people using Dialogic cards with *. The best way to know if the Dialogic card has * drivers is to call Digium since they wrote the drivers. Be prepared to pay money for the drivers since Digium had to pay Intel to develop the drivers. I've worked with the MSI boards and I do know that you can use them with SR5.1 of the Dialogic SDK. 5.1 is the last release that Intel released for free. The MSI board is legacy and IIRC it's not supported in the newer releases. I would assume that the D/80SC-4LS is also supported. If you decide to use the Dialogic SDK be prepared to be locked into a redhat 7.2/7.3 only configuration with lots of back-assward kernel patches and STREAMS subsystem add-ons. Getting frustrated with the lack of support and rising prices on the part of Intel, I sold all my Dialogic equipment on ebay and bought a TE405P. Regards, Jason ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: IAXTel and Telesthetic
On 23/09/04 16:57 -0700, Dan Clark wrote: I'm trying to run some inbound test to my Asterisk box using Telesthetic's gateway in MI to my GNU/IAXtel account. Am I missing something? I set up my user account on the GNUPhonne site, configured Asterisk to talk to IAXTel. * registers fine. In fact I can make calls to other test users. I haven't tried having someone call my number. When I call into Telesthetic's exchange it answers, tells me transferring to VOIP, I enter my number 1-700-, 1 second later I'm back at the VOIP prompt. If I leave of the 700 I get a response stating the user is offline. I tried a few other numbers that people had publish and the results were similar. thanks in advance, Dan Hi Dan, I had the same problem when I tried calling myslef via IAX also. I contacted telesthetic and they said they would look into it (they thought that it was iaxtel's fault). I waited a couple of weeks and tried again to no avail. I gave up trying telesthetic/Iaxtel. Telesthetic does work with FWD. I've tried this myself and it does work even using FWD's IAX services. Cheers, Jason Stewart ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: X100P Kernel Panic
On 05/08/04 15:24 +0100, Tom Lawrence wrote: snip 0Kernel panic: fatal exception in interrupt i have had to rebuild the kernel to get the modules in but they seemed to go in ok after that. If I run ztcfg I can see both of the cards working. Could it be something to do with the IRQ numbers being used? Both cards are on same IRQ as 2 other devices. I don't know if that's what's causing your panics, but sharing an interrupt on an X100P is a no-no. Sharing interrupts can cause all sorts of headaches. Jason ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Shady dial anyone??
On 08/07/04 19:04 +0500, Nauman Farooq wrote: wondering if anybody knows this..does shady dial work only with a zap interface or can it be configured to be used with SIP or IAX. Nauman --- Unecessary reply to asterisk-users digest snipped out--- It should work with any type of channel, seeing as the only files modified are app_queue.c and chan_agent.c I cannot vouch for this, as I've never used shady dial, but I will be and I'll be sure to give my review of it in the near future. Cheers, Jason P.S. Please do not reply to an existing message; this wastes bandwidth and screws up threaded mail user agents such as pine and mutt. just send a new mail to [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: FYI House bill exports analog phone regs to VoIP
On 06/07/04 15:17 -0400, Joe Baptista wrote: -- Forwarded message -- Date: Wed, 07 Jul 2004 00:31:21 -0400 From: Declan McCullagh [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: [Politech] House bill exports analog phone regs to VoIP http://www.politechbot.com/docs/boucher.voip.bill.070604.pdf There's a new bill in the House of Representatives to regulate phone calls made over the Internet. It was introduced this evening by Reps. Rick Boucher, D-Va., and Cliff Stearns, R-Fla., and it's called the Advanced Internet Communications Services Act of 2004. I've placed the text online and have a summary and commentary at News.com. The AICS bill takes a more regulatory approach than a competing proposal from Rep. Chip Pickering, R-Miss. Boucher and Stearns say VoIP servcies shall be subject to access charges and universal service taxes -- which is a really big deal. More on this later. snip Makes sense for these guys. They're telecom industry puppets. Lets hope it gets shot down. See Boucher: http://www.opensecrets.org/politicians/contrib.asp?CID=N2171cycle=2004 See Stearns: http://www.opensecrets.org/politicians/contrib.asp?CID=N2782cycle=2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: OMG THE SKY IS FALLING!! NOT!!!
On 14/05/04 10:36 -0400, Joseph Finley wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of brian k. west Sent: Friday, May 14, 2004 11:24 AM To: [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Subject: [Asterisk-Users] OMG THE SKY IS FALLING!! NOT!!! http://www.eweek.com/article2/0,1759,1591131,00.asp bkw Funny how he references buddy, pal told himShouldn't this type of security information come from a security person and not a typical OVERHYPED journalist? It's funny how most of these alarmist security articles are written by clueless journalists and cite Marketing Officers as sources for their information instead of real security people. Use encryption or tunneling (if you don't mind a bit of latency) if you're in need of extreme security. Anything on the Internet is as secure as you (the admin) make it. Anything insecure is not the fault of the service itself, but the fault of the expert in charge of making it secure. There was one good source in the article, which was the MCI tech. He's not too worried, since MCI encrypts all of their voip traffic. Jason ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: T100P + Zap Errors
On 21/04/04 08:37 -0500, Sean Bruton wrote: I am having some difficulty getting a T100P card to work with my PRI. When I attempt to make an outbound call via: exten = 1004,1,Dial(Zap/g1/NPANXX) I see the following on the asterisk console: -- Executing Dial(SIP/sbruton-b8ce, Zap/g1/NPANXX) in new stack Apr 21 08:18:48 NOTICE[16401]: app_dial.c:554 dial_exec: Unable to create channel of type 'Zap' == Everyone is busy at this time What is the output of the command zap show channels in the console? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: 10 day old email, virus already received
On 22/03/04 17:58 +0100, randulo wrote: For info, I receive the mailing list on a brand new account that is not used for anything else. Just received, a virus (*apparently*) From: [EMAIL PROTECTED] I suppose there may be 8,000 people getting it but just in case. No, not necessarily. A virus could just match up addresses randomly from the user's address book and files laying around on the hard drive. Cheers, Jason ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Grandstream G726-32 now working properly with *
On 19/03/04 14:11 +1100, Master Abi wrote: Hi, G726-32 codec from beta firmware 1.0.4.54 now works fine with *. Tested on BT101 and HT286 over a 64K DSL line. Some progress but iLBC still has not surfaced. Great news. This fw update breaks NTP sync in the phone for me, but your milage may vary. Get it from http://www.grandstream.com/BETATEST/ Master ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
