RE: [Asterisk-Users] Source for Sangoma or Digum 2+ port T1 Card near NH??
http://www.voipsupply.com or call at 1-800-398-VOIP they can rush deliver if you need it. Original Message Subject: [Asterisk-Users] Source for Sangoma or Digum 2+ port T1 Card near NH?? From: Tom Rymes [EMAIL PROTECTED] Date: Fri, July 01, 2005 8:31 am To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com I need a Digium or Sangoma T1 card that has at least 2 spans on it fairly quickly. Does anyone know of a vendor for either of these in NH or Northern MA? Please let me know! Tom ___ 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] Asterisk server with remote monitoring capabilities
Original Message Subject: Re: [Asterisk-Users] Asterisk server with remote monitoring capabilities From: beonice [EMAIL PROTECTED] Date: Thu, June 23, 2005 7:52 pm --- Michael Welter [EMAIL PROTECTED] wrote: William Boehlke wrote: Dell sells a remote management card for under $400 that enables remote reboots. I know there are others out there but have no experience with them. William Boehlke Signate -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of beonice I'm tired of having to drive out to the colocation facility each time my dedicated asterisk server craps out, just to press the button to do a hard reboot. (I'm running 1.05 stable at present, no telephony hardware, as this is mainly a system that receives calls, no dial-out ability is needed.) APC makes a power strip with a web server. Each socket has its own IP address. All you have to do to power cycle is access the IP address from your web browser and give the power cycle command. It is sooo cool. Thanks for your responses, folks. Okay, so what makes more sense: 1) a remote management card that will let me actually log in to the machine to monitor it as well as to reboot it vs. 2) a remote-accessible powerstrip that will allow me to remotely reboot the server? A little note. make sure your server motherboard/bios supports power on after power loss to use the remote control power strip. Secondly make sure the power strip control uses SSH and NOT telnet to control it. Telnet is too insecure because passwords are sent plain text. Another possibility is to write a reboot script and set up a cron job to automatically reboot every night until you solve the bigger problem of why is the server having problems? With Linux their is little need to reboot Linux. There is only one time that you have to reboot Linux. When you upgrade the kernel or its modules. Kernel modules do not always need a reboot. Kernel module that do require a reboot are critical to operation of your system for example RAID# . The best way is to have a script that uses the init script to restart the applications that are questionable on a cron job schedule for low usage. With a good script you could also check on the status of the service and perform functional test of the service. Then the script would perform the necessary tasks to recover from application failure. This wont help with a total system failure as the script will not work. Some of the remote monitoring cards can detect a system lockup and preform a system reboot automatically. When all of these fail you can use remote control power strips or a KVM (Keyboard Video Mouse) over IP to remotely control the hardware as if you are there. Cyclades (www.cyclades.com) sells both KVM and Remote Power management solutions that are secure. They even have RSA authentication tokens and a Biometric/RSA token authentications for secure management of the remote locations. Cheers, Max W. Blackmer, Jr. Consultant, Knowledge Power IT ___ 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] howto write CDRs on two mysql servers
Original Message Subject: Re: [Asterisk-Users] howto write CDRs on two mysql servers From: Matthew Boehm [EMAIL PROTECTED] Date: Thu, June 09, 2005 1:47 pm To: Mark Musone [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Mark Musone wrote: why not just use mysql replication to the second one? Replication does not solve the issue of redundancy. If the master goes down, so do the updates to the slave. -Matthew Replication can be set up to go bothways in a 2 server scenario for MySQL. This would solve your replication issue for 2 servers. More servers can be set up in a loop configuration for 3 or more servers. There is one problem with this method if one MySQL goes down updates are not passed around the loop. but as soon as the loop is reestablished the updates are passed around the loop to bring the other servers up to date. You might also consider DNS round robin to distribute initial connections to MySQL. A better solution is to use a load balancer that tracks to see if the servers are alive and balance against the load on each server. Cheers Max W. Blackmer, Jr. ___ 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 G729 licensing - is it worth the trouble?
If you have licenses or experimenting with g729 you might want to look at http://www.readytechnology.co.uk/open/g729/ it requires Intel C/C++ compiler and IPP libraries. Original Message Subject: Re: [Asterisk-Users] Digium G729 licensing - is it worth the trouble? From: Gavin Hamill [EMAIL PROTECTED] Date: Sun, June 05, 2005 11:08 am To: asterisk-users@lists.digium.com On Sunday 05 June 2005 16:31, Chris Mason (Lists) wrote: I have purchased 50 licenses at $10 each from Digium, Cool, so you have satisfied yourself that you are licensed to use the G.729 codec and not get your ass sued by the IP holders. Now you can simply use the no-license-required codecs from here... http://kvin.lv/pub/Linux/Asterisk/ I'm probably massively over-simplifying the problem.. but I see this as the same problem as having installed the same MS Office 2000 CD + key on 50 PCs, but as long as you have 50 unique certificates of authenticity for Office 2000, it doesn't matter a crap (in the real world) what actual key got used on each PC. gdh ___ 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] Asterisk @ Home 1.1 Released
[EMAIL PROTECTED] 1.1 Released and can be downloaded from Sourceforge. http://sourceforge.net/project/showfiles.php?group_id=123387package_id=135368 Cheers, Max W. Blackmer ___ 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] UPS rating for SOHO asterisk box
Hello, APC has a nice selector tool on their website. http://www.apc.com/tools/ups_selector/index.cfm It asks several questions to recommend an APC solution. it even gives you a percentage of the capacity of the ups systems capacity. Original Message From: Wilson Pickett [EMAIL PROTECTED] Date: Tue, May 31, 2005 3:00 am Slightly OT, but I think this is of possible interest to many of you, I need to get a UPS for my asterisk box. They are rated in VA but I can't quite figure out how that converts to real life. I have a PIII-800 box with two X100P and one TDM400P plus graphics adapter, an IDE hard drive etc. Will a small 400VA box be enough for this? tia ___ 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] CRM integration (was RE: CallerID)
hello everyone, I just had a thought on this subject why not create a daemon process on the Client PC That registers its self and What phone the user is connected. An AGI script could monitor the progress and when answered could send a push to the registered daemon which would push a link to the registered daemon on the telephone operators on the desk top. this would not waist resource as much as polling? perhaps the daemon could be written in something portable like java or even as a small applet that is launched first and minimized to launch the CRM App in a browser and could be written into other CRM Applications. What are your thoughts on this? Max W. Blackmer Jr. ___ 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] G729 codec
Original Message Subject: Re: [Asterisk-Users] G729 codec From: todd [EMAIL PROTECTED] Date: Wed, May 25, 2005 11:59 am To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Steve Trying to understand the floating point vs fixed, forgive my ignorance. By floating you mean it can very depending on usage from 6.4kbps to 11.8kbps; were as the fixed will be constant 8kbps? Thanks Todd No Fixed point calculation and floating point executions are diffrent only in the way calculations are preformed. Floating point executions use to take more time than useing fixed point calculations. With todays math coprocessors especially with SSE floating point calculations actually preform well on Intel x86 platform. Fixed point calculations use a fixed decimal place to perform calculations with some performance increases with less precision and accuracy. Cheers, Max ___ 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] POE hub
Original Message Subject: [Asterisk-Users] POE hub From: Chris Mason [EMAIL PROTECTED] Date: Sun, May 15, 2005 9:19 pm To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com I need to connect up to sixteen phones per building, I can use a cheap hub, but POE would be useful. Is there a cheap POE hub available? Everything I have seen has been expensive. Well for inexpensive you could look at D-Link They have 2 switches that provide POE. DES-1316 Web-Smart 16-port Switch Web-Smart Switch with (8) Ports 10/100Mbps and (8) PoE 802.3af 10/100 ports List price is $499.99 USD. I can get them for $420 USD. DES-1526 Web-Smart 24-port PoE 10/100 + 2 Combo Gigabit Copper/SFP ports Switch. List price is $999. I can get them for $850. Cheers, Max W. Blackmer, Jr. ___ 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] Voicepulse down?
it works here in Chicago. you might want to check with your provider their dns may be out.. that happened with comcast about 3 weeks ago. Original Message Subject: [Asterisk-Users] Voicepulse down? From: Trevor Harrison [EMAIL PROTECTED] Date: Wed, May 11, 2005 8:57 am To: asterisk-users@lists.digium.com Anyone else using Voicepulse? This morning I noticed that they seem to be doa... no dns resolution, no ping, etc. -Trevor ___ 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] Satellite Providers
Satellite delays are always bad. It is more a delay because of the time it takes a signal to travel to the satellite and back to a receiving station. You might want to check into ground station to station microwave communications stations. The best is to have a tap to a phone company that may have cell towers in the area. Cheers, Max Original Message Subject: [Asterisk-Users] Satellite Providers From: Yiannis Costopoulos [EMAIL PROTECTED] Date: Wed, May 11, 2005 12:23 pm To: asterisk-users@lists.digium.com Hi All, I am investigating the deployment of VoIP/* in Eastern European areas where there is no PSTN infrastructure. As you can understand DSL/Cable connections are a dream. The only option is satellite. Does anyone know of any satellite providers that have low enough/acceptable delays for VoIP? Thanks, Yiannis. ___ 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] Echo from a mail loop in list
Every time I post to the list I get the following response indicating that there is a mail loop. the recipient [EMAIL PROTECTED] has some problems with looping. Hi. This is the qmail-send program at smtp.register.it. I'm afraid I wasn't able to deliver your message to the following addresses. This is a permanent error; I've given up. Sorry it didn't work out. [EMAIL PROTECTED]: This message is looping: it already has my Delivered-To line. (#5.4.6) --- Below this line is a copy of the message. Return-Path: [EMAIL PROTECTED] Received: (qmail 3082 invoked from network); 11 May 2005 19:39:07 - Received: from unknown (HELO mail-relay-4.tiscali.it) (213.205.33.44) by smtp.register.it with SMTP; 11 May 2005 19:39:07 - Received: from digital-system.it (84.222.72.69) by mail-relay-4.tiscali.it (7.1.021.3) id 4202035700C6932B for [EMAIL PROTECTED]; Wed, 11 May 2005 21:39:07 +0200 Received: from pop.digital-system.it ([195.110.124.132]) by digital-system.it (digital-system.it [192.168.200.1]) (MDaemon.PRO.v7.2.3.R) with MultiPOP id md5080353.msg for [EMAIL PROTECTED]; Wed, 11 May 2005 21:29:09 +0200 Delivered-To: [EMAIL PROTECTED] Received: (qmail 17684 invoked from network); 11 May 2005 19:33:42 - Received: from unknown (HELO lists.digium.com) (69.16.138.164) by smtp.register.it with SMTP; 11 May 2005 19:33:42 - Received: from [69.16.138.164] (localhost [127.0.0.1]) by lists.digium.com (Postfix) with ESMTP id 33BDF2FF58A; Wed, 11 May 2005 14:23:36 -0500 (CDT) X-Original-To: asterisk-users@lists.digium.com Delivered-To: asterisk-users@lists.digium.com Received: from psmtp.com (exprod5mx31.postini.com [64.18.0.186]) by lists.digium.com (Postfix) with SMTP id 2B1D12FE2D7 for asterisk-users@lists.digium.com; Wed, 11 May 2005 13:40:01 -0500 (CDT) Received: from source ([64.202.165.193]) by exprod5mx31.postini.com ([64.18.4.10]) with SMTP; Wed, 11 May 2005 13:40:07 CDT Received: (qmail 31981 invoked from network); 11 May 2005 18:40:07 - Received: from unknown (HELO gem-wbe05.mesa1.secureserver.net) (64.202.189.37) by smtpout02-03.prod.mesa1.secureserver.net with SMTP; 11 May 2005 18:40:07 - Received: (qmail 952 invoked by uid 99); 11 May 2005 18:40:07 - Message-ID: [EMAIL PROTECTED] Date: Wed, 11 May 2005 11:40:07 -0700 From: Max W Blackmer Jr [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Satellite Providers To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com MIME-Version: 1.0 Content-Type: TEXT/plain; CHARSET=US-ASCII X-pstn-levels: (S:99.9/99.9 ) X-pstn-settings: 1 (0.1500:0.1500) gt3 gt2 gt1 X-pstn-addresses: from [EMAIL PROTECTED] [90/4] X-BeenThere: asterisk-users@lists.digium.com X-Mailman-Version: 2.1.5 Precedence: list Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com List-Id: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users.lists.digium.com List-Unsubscribe: http://lists.digium.com/mailman/listinfo/asterisk-users, mailto:[EMAIL PROTECTED] List-Archive: http://lists.digium.com/pipermail/asterisk-users List-Post: mailto:asterisk-users@lists.digium.com List-Help: mailto:[EMAIL PROTECTED] List-Subscribe: http://lists.digium.com/mailman/listinfo/asterisk-users, mailto:[EMAIL PROTECTED] Sender: [EMAIL PROTECTED] Errors-To: [EMAIL PROTECTED] X-MDMultiPOP: [EMAIL PROTECTED]@pop.digital-system.it X-MDRcpt-To: [EMAIL PROTECTED] X-Rcpt-To: [EMAIL PROTECTED] X-MDRemoteIP: 195.110.124.132 X-Return-Path: [EMAIL PROTECTED] X-Spam-Checker-Version: SpamAssassin 2.64 (2004-01-11) on bazumbis.digisys.local X-Spam-Status: No, hits=-4.9 required=7.0 tests=BAYES_00 autolearn=ham version=2.64 X-Spam-Level: X-Spam-Processed: digital-system.it, Wed, 11 May 2005 21:29:10 +0200 X-MDAV-Processed: digital-system.it, Wed, 11 May 2005 21:29:15 +0200 X-MDRedirect: 1 X-MDaemon-Deliver-To: [EMAIL PROTECTED] ---content of my post to the list. ___ 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] A good SIP receptionist phone
Take a look at the Polycom 360 if you only nee 12 lines. otherwise look at the Snom 220 with a sidecar (up to a total of 3 side cars may be added for a total of 65 lines in the extreme need.) Max W . Blackmer, Jr. ___ 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] Diffrence bewteen FXO and FXS
Here is an excellent document explaining the differences between FXO and FXS. http://www.google.com/url?sa=Ustart=5q=http://www.patton.com/technotes/fxs_fxo.pdfe=7385 Also you can look at Digium's site for their description, which describes it from a stand point of Asterisk as the PBX. http://www.digium.com/index.php?menu=fxsvfxo Why are there FXO cards, and FXS cards? What's the difference, and why is it needed? Modem cards, seem to be able to dial out, and receive calls, so why are these cards different? ___ 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] 4 - 8 port w/QOS switch for Asterisk
Linksys has a low end router with an 8 port switch that does QoS model BEFSR81. It can be gotten for under $100 USD. For more information http://www.linksys.com/products/product.asp?prid=604scid=29 Max W. Blackmer, Jr. Original Message Subject: [Asterisk-Users] 4 - 8 port w/QOS switch for Asterisk From: Joseph [EMAIL PROTECTED] Date: Sun, May 01, 2005 1:05 pm To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Can anybody recommend Switch 4 to 8 ports with QOS for Asterisk? -- #Joseph ___ 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] BIND VoIP anyone?
Don't forget Dundi is such a system that is already integrated into Asterisk. http://www.dundi.info/ Original Message Subject: [Asterisk-Users] BIND VoIP anyone? From: Andres Paglayan [EMAIL PROTECTED] Date: Thu, April 28, 2005 11:39 am To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Hi List, I was looking for, but I couldn't find any product or project like BIND that works with VoIP in an homologous way. I mean, is there anybody working in a way to register user-ids or domain name-like information so VoIP calls can be dialed in a number string format from any IP phone? Any clue? Thanks all, ___ 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] Polycom IP4000 Conference Phone
look here http://www.polycom.com/resource_center/1,1454,pw-6812-9192,00.html Original Message Subject: RE: [Asterisk-Users] Polycom IP4000 Conference Phone From: Wiley Siler [EMAIL PROTECTED] Date: Tue, April 26, 2005 8:39 am To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com ___ 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] Slack 10 install - THANK YOU - Cisco Reseller Help
Now for my present delima. - Actually this one's been racking my brain since about March. I need to find a Cisco Reseller. very good and will pre configure the Cisco phones for SIP. http://www.voipsupply.com/home.php --Snip-- Last month, I purchased a Cisco-7905G IP Phone from a vendor in the States paid a pile of money to FedEx to get the thing sent to me. The phone DID NOT include the SIP firmware. - Since I have a 'BroadVoice' account, I'm stuck until I can SIP into this phone. It is a real pain to upgrade to sip on a Cisco. Voip Supply charges an extra $100 to do it for you I'd also like to use the 7940/7960 phones but, again, there's the SIP Firmware problem. I've been all over the 'wiki' pages have read what's necessary to get this firmware get it installed. - Basically I need a 'service contract' for each phone. you need a sip license for each phone. Wading through the pages search engines on Cisco's web site is a true excercise in futility! - A google search is more accurate. But I've yet to find a Cisco Reseller who knows what I need is willing to talk about it. - Heck! - Even finding a Cisco Reseller with a valid phone number where you can talk to a 'PEOPLE' is a feat in itself. So - If anybody knows a Cisco Reseller, or if you ARE a Cisco Reseller and know what I need, please contact me. - I'm sure there are others in this ML that are in the same boat as I or who have been there themselves. You may also want to look at Polycom phones they are less of a headache than the cisco phone. They are both good on quality, look and feel of a business class phone. Cheers, Max ___ 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
spending over $A10,000 in the process. The cards are more expensive than the server they're going into (Dell poweredge 750's). When a GPL'd hardware It is obvious that you have never experienced high end servers. We have had a single server cost as much as $20,000 and that is nothing but high performance hardware(Raid, REG ECC memory[mirrored for redundancy], Dual Xeon Server). Then you add in any specialized hardware that can easily up the cost to $30,000. and that is just one machine. when you need performance you pay for it one way or another. A lot of times it is better to pay more for reliability and performance. Max ___ 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: Polycom sound quality problems
There's no transcoding going on. It's ulaw on IAX with Sixtel and ulaw on SIP to the phone. I considered that as a possibility originally, and even tried using GSM with Sixtel to force it to do transcoding, but had the exact same problem. The Asterisk box is a 2.4ghz P4 with 512MB RAM, doing nothing but Asterisk. I have only 9 extensions. That should not be a problem with those specs. I would think there's a possibility of packet loss on the IAX channel, except the other SIP phones (SJPhone softphone) work flawlessly. Also, OUTBOUND calls are just fine on the Polycoms. Only incoming calls are messed up. There is one other possibility. Are you using Switches or hubs? if you are using hubs you could have collisions that cause data loss. Switches with store and forward are the best especially if they have QoS, ToS and CoS management features in the switch. Switches typically do not loose packets but they do expire on rare occasions due to high traffic volume. Max ___ 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: Polycom sound quality problems
I don't see any way to tell the Polycom to ignore QoS. It's mainly routers and switches that pay attention to QoS, the phone would just set QoS on its outgoing packets. Anyway, here's what's in the QoS section- it all seems to be related to sending packets: It is not in the transport if it is sounding bad look and see if there is any transcoding occuring from the IAX to the SIP. What codecs are accepted on the AIX should be the Same codecs accepted on the SIP channel ... and what codects are being used on each phone. This sounds like a transcoding issue. ___ 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] Open Source Billing Software
I am just beginning work on Trabas now. nothing as of yet. I just liked the features that it currently offers, but it does definitely need allot of work yet. I am looking at adapting this one or take the concepts and rewrite for PHP. Some features I am looking for that are not in the current system. 1. Better ability to pull in records from asterisks CDR using billing codes. 2. Dynamic reports for CDR according to Clients requirements. 3. Allow clients to look at the current state of their account to integrate to End user web site through SOAP calls. 4. Make PDF bills and Reports with the capability of emailing or generate on demand for web download for clients. Any other Ideas anyone might need in addition to trabas features? Thanks, Max W. Blackmer, Jr. ___ 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] VoIP Provider problems
We recently configure an asterisk server to use with an VoIP provider to make calls to a PSTN. We use (voipjet, nufone, diamond) We feel that we haven't got the quality that we hope. Sometimes our calls gets mute, or we feel communication cuts on our phone calls. We have got an QOS router (Draytek) reserving 1/2 of our wideband to the SIP an IAX2 protocols, and an ADSL line about 2 Mb. ADSL has slower upload speeds than download speeds (your 2Mbps is download). so you may have problems with your outgoing packets of sound. g.711 codec (the default codec for most voip providers because there is virtually no sound quality loss) uses about 84Kbps per channel or simultaneous connection. For example if you have an Upload speed of 128Kbps. and you try to have 2 phone conversations you would need 168kbps transfer speed. That is 40kbps more than your upload speed. This is a major problem with ADSL the upload and download speeds are not equal. Another potential problem is that your provider is over subscribed for the available bandwidth. What this means is that when allot of people are using their connection to your provider. The provider may not be able to handle all those users at once and packets get dropped or delayed. Dropping or delaying packets is very bad for VoIP especially if they do not do QoS or ToS routing which most providers do not. What is your upload speed? Some other possibilities are to use some compression codecs which will cause some sound quality loss like gsm or iLibc and g.729 to pack more calls in the limited bandwidth limitations. Another option is to use SDSL where the speeds of both the upload and download are the same. We feel our quality decrease when in US are about 9:00 or 10:00 in the morning. This time is when businesses in the us are opening and starting to do business In the united states. Both for phones and Data. We do not know if this is it correct or all the people using VoIp provider feel the same quality? This may mostly be in relation to you Internet provider and how many hops you have to take to get to the VoIP provider and if they oversubscribe their bandwidth capacity. One provider may be good for one person with one person in a different ISP than an ISP you have. And you are even right next door to each other. This is as a result of how the internet is connected and may not nessessarly be geographic. For example you may be connecting to a server in your own city lets say Chicago but you are actually routed to San Francisco then back to Chicago. But it will not always take the same path the next time you may be routed through New York. This is a simplification of how it works. The closer you are to a Tier 1 provider(they own the major trunks interconnects) the less time it will take to get to your target. Anyone knows any provider without this kind of problems? I have seen many Providers have both Good and bad connection links. It is best to have a provider that routes with QoS and/or ToS within their routers and have only one or two hops between your provider and a tear 1 provider. Witch provider do you use to get the best sounds quality? It is not that simple. But you can begin by doing a traceroute to the many providers at different times of the day. This will see the route changes and time delays between hops to get to VoIP Providers gateways. Hope this helps in understanding the problems involved with choosing a provider. Thanks, Max ___ 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] Erratic CPU load
Snip During tests with a IAX2/PSTN gateway I've been getting strange results for processor idle time and load. I (re)search(ed) this issue for a while, but I didn't get any good explainations. Can somebody help me? Yes, Speex is pretty cpu intensive compared to other codecs. The best perfoming codecs are g.711u/a, gsm and g.729 in that order. I have several sites that rely on a central server for connection to the PSTN. Calls to the PSTN are routed over the Internet to this PSTN gateway using IAX2 in trunk mode. To minimize bandwith usage, the Speex codec is used. The central PSTN gateway is a P4 3.0GHz, 1GByte mem, has a TE110P card supporting ISDN30 and runs Asterisk version 1.0.3 on Debian Sarge. While sustaining 5 connections dialed in through the TE110P (terminated at remote sites through IAX) and running top on the PSTN gateway, I see 98% CPU idle time most of the time. I also see short (around 10sec) bursts of high CPU usage (40-50%) by one of the asterisk processes supporting the connection. The bursts happen in irregular intervals, ranging from 30 to 60 sec. Meanwhile, the reported average load jumps up and down between 0.1 to 0.7. What's happening here? Is the processor load really this erratic, or am I looking at an artefact in cpu usage measurement? Maybe there is an aliasing effect caused by the periodic cpu load (20ms, default trunk frequency) and the cpu usage measurement (also periodic?), but I don't know how to check this. If this top reading is an artefact, is there a way to check the actual (realtime) load? With top you can increase the number of seconds beteween refreshes. this will give you a better useage over time. but if your cpu is spiking too 100% or even over 75% using the default update of 3 seconds your codec or your free memory may be the problem. Codec transcoding can take some serious hits on memory. Regarding the actual processor usage for speex encoding: this report suggests my processor is indeed quite busy encoding a few speex channels: http://astertest.com/astricon_performance.ppt. Given the results in this report, I doubt the PSTN gateway will support more than 10 speex encodings. At the same time, the same processor encodes 756x756 PAL television to mpeg-4 on my mythtv box at home. Twice, leaving room for scheduled jobs. Has anyone some references to documentation to put these figures into perspective? ___ 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] Open Source Billing Software
You Might want to look at Trabas ( http://www.trabas.com/opensource/index.html ) it is by far the most complete billing system. ___ 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] Open Source Billing Software
:) trabas and asterisk have big misunderstanding the don't thing to work like it should be :) Just needs some programming to translate asterisks logs and import them into the database tables. :) that is the good thing about opensource Max ___ 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] Voicepulse connect has doubled their rates
It's going from 7.95 ( 8 bucks ) a month to 11 bucks, an increase of 72%. That's hardly what I'd call doubling ( unless you're using that new math I've heard so much about ). h, actually it is only a 28% increase. you want to see outrageous you should see my gas bill. ___ 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] High Availability on Asterisk
Hello, Yes, there is high availability, Clustering and Load Balancing. Each one has its own advantage and disadvantages. First option is one that you mention High Availability. This option you have a second machine watching heartbeats from the primary machine. when the heartbeats stop the machine takes over the IP address and functionality of the primary machine. This is usually limited to 2 machines. The second option is Clustering. This option makes multiple computers(nodes) to work as one virtual system. when one system fails the node is removed from the cluster and continues operation as normal. Clusters can be from 2 and up nodes. Your Third option is load balancing. This option sets up a machine to direct traffic to multiple servers based on several factors round robin, server load, least connections and availability. My personal preference is Load Balancing and clustering because there is virtually no transition time and you can take machines down for maintenance and upgrades without interrupting the service. My suggestion is google High-Availability Linux HOWTO, Clustering Linux HOWTO and load balancing Linux HOWTO for more information. Also, you might want to look up Carrier Grade Linux( http://tinyurl.com/5c6vy ) this provides information and specifications. Max Hi, I would like to know if Asterisk (installed on Linux or Free BSD) have any possibility of high availability (such as, if one box down, the other one get all configuration)? If yes: 1 - how can I do that? 2 - Who is using that? 3 - How long is using? 4 - How Many SIP phones is using on that Asterisk? Thanks in advanced, Otto ___ 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] Any 24 (or 30) way FXS PCI cards?
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? 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. . ___ 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?
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. . ___ 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] Global Intercom on SIP phones
Thank you John, Max Blackmer I would like to create an Intercom extension that will dial a group of extensions which are connected to SIP phones. The SIP phones are setup to auto answer a particular extension assigned to one of the lines in the phone. All phones must answer and broadcast the page message at the same time. Has anyone done this? Or should I install an overhead speaker system using the oss/alsa console as a broadcast. Can the local port be set to auto answer calls? . Yes, it's been done. http://www.voip-info.org/wiki-Cisco+7940-7960+auto-answer+config JT ___ 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] Global Intercom on SIP phones
Hello Everyone, I would like to create an Intercom extension that will dial a group of extensions which are connected to SIP phones. The SIP phones are setup to auto answer a particular extension assigned to one of the lines in the phone. All phones must answer and broadcast the page message at the same time. Has anyone done this? Or should I install an overhead speaker system using the oss/alsa console as a broadcast. Can the local port be set to auto answer calls? Thank you, Max W. Blackmer, Jr. ___ 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
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: # make Making all in src make[1]: Entering directory `/export/usr/src/spandsp-0.0.2/src' make all-am make[2]: Entering directory `/export/usr/src/spandsp-0.0.2/src' if /bin/sh ../libtool --mode=compile gcc -DHAVE_CONFIG_H -I. -I. -I. -I-g -O2 -MT adsi.lo -MD -MP -MF .deps/adsi.Tpo -c -o adsi.lo adsi.c; \ then mv -f .deps/adsi.Tpo .deps/adsi.Plo; else rm -f .deps/adsi.Tpo; exit 1; fi mkdir .libs gcc -DHAVE_CONFIG_H -I. -I. -I. -I -g -O2 -MT adsi.lo -MD -MP -MF .deps/adsi.Tpo -c adsi.c -fPIC -DPIC -o .libs/adsi.o gcc -DHAVE_CONFIG_H -I. -I. -I. -I -g -O2 -MT adsi.lo -MD -MP -MF .deps/adsi.Tpo -c adsi.c -o adsi.o /dev/null 21 if /bin/sh ../libtool --mode=compile gcc -DHAVE_CONFIG_H -I. -I. -I. -I-g -O2 -MT awgn.lo -MD -MP -MF .deps/awgn.Tpo -c -o awgn.lo awgn.c; \ then mv -f .deps/awgn.Tpo .deps/awgn.Plo; else rm -f .deps/awgn.Tpo; exit 1; fi gcc -DHAVE_CONFIG_H -I. -I. -I. -I -g -O2 -MT awgn.lo -MD -MP -MF .deps/awgn.Tpo -c awgn.c -fPIC -DPIC -o .libs/awgn.o gcc -DHAVE_CONFIG_H -I. -I. -I. -I -g -O2 -MT awgn.lo -MD -MP -MF .deps/awgn.Tpo -c awgn.c -o awgn.o /dev/null 21 if /bin/sh ../libtool --mode=compile gcc -DHAVE_CONFIG_H -I. -I. -I. -I-g -O2 -MT bert.lo -MD -MP -MF .deps/bert.Tpo -c -o bert.lo bert.c; \ then mv -f .deps/bert.Tpo .deps/bert.Plo; else rm -f .deps/bert.Tpo; exit 1; fi gcc -DHAVE_CONFIG_H -I. -I. -I. -I -g -O2 -MT bert.lo -MD -MP -MF .deps/bert.Tpo -c bert.c -fPIC -DPIC -o .libs/bert.o bert.c:43:20: tiffio.h: No such file or directory In file included from spandsp.h:75, from bert.c:45: spandsp/t4.h:74: error: syntax error before TIFF spandsp/t4.h:74: warning: no semicolon at end of struct or union spandsp/t4.h:147: error: syntax error before '}' token spandsp/t4.h:147: warning: data definition has no type or storage class spandsp/t4.h:181: error: syntax error before '*' token spandsp/t4.h:181: warning: data definition has no type or storage class spandsp/t4.h:189: error: syntax error before '*' token spandsp/t4.h:195: error: syntax error before '*' token spandsp/t4.h:202: error: syntax error before '*' token spandsp/t4.h:208: error: syntax error before '*' token spandsp/t4.h:216: error: syntax error before '*' token spandsp/t4.h:225: error: syntax error before '*' token spandsp/t4.h:231: error: syntax error before '*' token spandsp/t4.h:237: error: syntax error before '*' token spandsp/t4.h:243: error: syntax error before '*' token spandsp/t4.h:249: error: syntax error before '*' token spandsp/t4.h:255: error: syntax error before '*' token spandsp/t4.h:261: error: syntax error before '*' token spandsp/t4.h:267: error: syntax error before '*' token spandsp/t4.h:273: error: syntax error before '*' token spandsp/t4.h:281: error: syntax error before '*' token spandsp/t4.h:281: warning: data definition has no type or storage class spandsp/t4.h:288: error: syntax error before '*' token spandsp/t4.h:294: error: syntax error before '*' token spandsp/t4.h:300: error: syntax error before '*' token spandsp/t4.h:306: error: syntax error before '*' token spandsp/t4.h:316: error: syntax error before '*' token spandsp/t4.h:324: error: syntax error before '*' token spandsp/t4.h:331: error: syntax error before '*' token spandsp/t4.h:337: error: syntax error before '*' token spandsp/t4.h:345: error: syntax error before '*' token spandsp/t4.h:351: error: syntax error before '*' token spandsp/t4.h:361: error: syntax error before '*' token spandsp/t4.h:367: error: syntax error before '*' token spandsp/t4.h:373: error: syntax error before '*' token spandsp/t4.h:379: error: syntax error before '*' token spandsp/t4.h:385: error: syntax error before '*' token spandsp/t4.h:392: error: syntax error before '*' token In file included from spandsp.h:76, from bert.c:45: spandsp/t30.h:195: error: syntax error before t4_state_t spandsp/t30.h:195: warning: no semicolon at end of struct or union spandsp/t30.h:198: error: syntax error before '}' token make[2]: *** [bert.lo] Error 1 make[2]: Leaving directory `/export/usr/src/spandsp-0.0.2/src' make[1]: *** [all] Error 2 make[1]: Leaving directory `/export/usr/src/spandsp-0.0.2/src' make: *** [all-recursive] Error I'm using Fedora Core 2 and Asterisk 1.0.5 Can you help me? Thank you in advance, Juanjo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:
RE: [Asterisk-Users] I changed some minor things, but how can I contribute it?
Ronlald, Did you use any options with the diff command? The usual option for producing patch files is -u the unified format. Example: diff -u originalfile newfile patchfile Hope this helps Max It bothered me, so I changed it to my need, how can I contribute this changes back to the community? I copied both *.cgi of ASTCC into new ones, added the start time into the table I tried diff oldfile newfile, but it does not look like then normal patches, ... how should I make it? Where to post the diff? ___ 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
