Re: [asterisk-users] SIPP problem
On Sat, 2 Sep 2006, Diego Quintana Cruz wrote: Hi everybody, I'm trying to load-test my Asterisk PBX using SIPP, but I always getting errors, I followed the instructions given in [1] which mainly was to create the user sipp in sip.conf and the dialing plan for his context in extensions.conf I'm using Asterisk 1.0.10 Any ideas or tutorial on how using SIP? Here are my notes on the subject: http://lists.digium.com/pipermail/asterisk-dev/2006-June/021162.html -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF between cisco and sipura going through asterisk
On Tue, 29 Aug 2006, Benjamin Lawetz wrote: Hello all, we're having an issue with DTMFs being sent to Sipura's. Calls are originating from a Cisco AS5300 being sent to asterisk which in turn sends it to the Sipura. Connected to the Sipura is a legacy PBX (or actually shows the same problem with a cheap answering machine). The DTMFs sent from the AS5300 aren't recognised by the legacy PBX. - DTMFs are recognised correctly on the asterisk (when we check voicemail) - The cisco is setup with dtmf-relay rtp-nte - in sip.conf the cisco and sipura are set to rfc2833 If I set the cisco in dtmf-relay rtp-cisco it works on the sipura, but not on the asterisk. Unfortunately I can only set one dtmf-relay mode on the cisco. Is there anything I can change on asterisk or sipura to get the sipura to work with the rtp-nte (or to get asterisk to work with the cisco-rtp)? Any hints can help, Ben, What version of Aserisk are you using? If it is the 1.2 series, there are all sorts of RFC-2833 DTMF Relay issues that can crop up. My suggestion is that if you are willing to take the time, it might be worth it to Upgrade to the pre-release version of Asterisk that is currently in TRUNK. This supports the new Variable Length DTMF code that should knock out nearly all of the DTMF issues that Asterisk has had. The 1.2 and earlier RTP stack and RFC-2833 implementation, while not technically wrong according to the RFC, did things a bit differently than the rest of the world has chosen, and therefore can cause DTMF instability. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Performance without RTP?
On Tue, 29 Aug 2006, Nick Hoffman wrote: On Tue August 29 2006 04:39, Greg Boehnlein [EMAIL PROTECTED] wrote: On Mon, 28 Aug 2006, Andrew Kohlsmith wrote: On Monday 28 August 2006 13:02, Greg Boehnlein wrote: I've pushed over 1,000 concurrent calls this way using the SIPP program for SIP performance testing. There was some tuning that needed to be done, but it worked. Never went that far in production, though. May you share some of your tuning with us? What gotchas did you discover? Just making sure your dial-plan as efficient as possible, that you have enough sockets and open file limits in the kernel, not connecting to the CLI console, never, ever using cdr_mysql or cdr_odbc for your CDR records (locking / contention issues) etc... Lots of basic common sense stuff that you often forget about.. :) Hi Greg. What problems/performance issues does cdr_mysql introduce? If the database is unavailable, or performance is slow, it can cause a blocking condition that will stop the entire system from processing anything. It may have been fixed since then, but I thought that cdr_mysql was deprecated.. -- Nick e: [EMAIL PROTECTED] p: +61 7 5591 3588 f: +61 7 5591 6588 If you receive this email by mistake, please notify us and do not make any use of the email. We do not waive any privilege, confidentiality or copyright associated with it. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Performance without RTP?
On Sat, 26 Aug 2006, Kelvin Williams wrote: If Asterisk was used to set up and tear down calls, and using canreinvite allowing the RTP to pass from end-point to end-point, how many calls could Asterisk handle at once? I've pushed over 1,000 concurrent calls this way using the SIPP program for SIP performance testing. There was some tuning that needed to be done, but it worked. Never went that far in production, though. I ask because I have been utilizing OpenSER but find myself constantly needing Asterisk to do this or that, and would like to move OpenSER into more of a Registration server, and letting Asterisk handle all of my calls I understand that the set up and tear down may be a tad slower, but programming (using AGI, etc.) would definitely outweigh the timing IMO. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Performance without RTP?
On Mon, 28 Aug 2006, Andrew Kohlsmith wrote: On Monday 28 August 2006 13:02, Greg Boehnlein wrote: I've pushed over 1,000 concurrent calls this way using the SIPP program for SIP performance testing. There was some tuning that needed to be done, but it worked. Never went that far in production, though. May you share some of your tuning with us? What gotchas did you discover? Just making sure your dial-plan as efficient as possible, that you have enough sockets and open file limits in the kernel, not connecting to the CLI console, never, ever using cdr_mysql or cdr_odbc for your CDR records (locking / contention issues) etc... Lots of basic common sense stuff that you often forget about.. :) -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] High Availability with PRI failover
On Fri, 11 Aug 2006, Senad Jordanovic wrote: [EMAIL PROTECTED] wrote: Hi After a month or so using Asterisk we've had or first downtime period due to a faulty RAM chip on the server, so we're starting to think about the possible high-availability solutions. Hi If you can afford it, below will give you total fault tolerant solution. http://www.bicomsystems.com/products/C/P/319/255_2797/ I think I can afford that. When I go to the link I get: Under Maintenance Please come back later. Bicomsystems.com ;) -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoipNow 1.2.0 Beta
On Tue, 8 Aug 2006, mitcheloc wrote: On Sat, 29 Jul 2006, Tom Vile wrote: Did you look on the site? http://www.4psa.com/products/voipnow/demo.php Ughh.. it's PLESK! Looks like the entire thing is written w/ the PLESK user API. NEXT! Instead of flaming, you could accept that not everyone makes software with just you in mind. It isn't a flame. It is an observation and an opinion. I'm happy that people are creating new interfaces for Asterisk management, but that doesn't mean that I have to like them or the technology that they are based on. People choose to do all sorts of silly things, some more silly than others. PSA knows PLESK, so they have used it to build their UI. I use PLESK for my Web hosting customers, and although it works, it certainly is not pretty nor is it something that I would reccomend as a platform for building an Asterisk management application. Your mileage may vary. Use what works for you. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RE VoipNow 1.2.0 Beta
On Tue, 8 Aug 2006, Matthew Warren wrote: Yes it is an addon of Plesk, thats stating the obvious. But while your complaining about people writing stuff to use what are you doing. If your not a developer don't critisize the developers. I see nothing more than you displaying that you are the Vice President of a 2 man consulting firm. Which means you have to sell other peoples developed products. Wow. Such hostility. Some corrections. 1. I am using a variety of customer built and off the shelf products, both stuff that I've written, commissioned to have written and/or purchased from companies such as PBXware, Switchvox, VoiceRoute etc. My customer's needs drive my decisions as to what should be deployed, not a religious fervor. However, my customers also pay me to make solid decisions on technology based on my (considerable) experience in the field. 2. N2Net, of which I an the Vice President, started 11 years ago as a 1 man consulting company. We are quite a bit larger now, but that isn't important. What is important is that you haven't even taken the time to verify who you are talking to, and/or what their credentials are. Bad move on your part, and probably not the smartest thing to fire off to a list such as this. I'll chalk it up to a mistake on your part. A google search of your name and the term Asterisk doesn't show a whole lot. You might consider doing the same with my name Gregory Boehnlein Asterisk to see what comes up. Not to mention you are being critical of plesk, yet you use to host you websites for your business. Yes.. I host several thousand websites with PLESK, and as such am quite well positioned to voice an opinion about their software and methodologies. As far as Web control panels go, they are the best in the industry, but the industry has pretty low standards. I dislike PLESK for a variety of reasons and personally see that someone using it as a basis for a Web UI for Asterisk is trying to push a square peg through a round hole. That doesn't mean that VoipNow sucks, just means that I'll probably not use it or reccomend it to my clients because I think the architecture and design are fudamentally flawed, and it's my reputation on the line. However, if someone asked me for a reccomendation on a Control Panel software for Web Hosting, I would have no hesitations reccomending PLESK to them, provided they understand the limitations of the platform. Dude, we all have opinions, like crapholes they all stink, your's just stood out. Everyone is entitled to their own opinion, and you have every right to have your own. I don't agree with it, but I'm not going to resort to childish insults and petty comments. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoipNow 1.2.0 Beta
On Sat, 29 Jul 2006, Tom Vile wrote: Did you look on the site? http://www.4psa.com/products/voipnow/demo.php Ughh.. it's PLESK! Looks like the entire thing is written w/ the PLESK user API. NEXT! -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Polycom 1.6.7 Firmware Messages Button
On Sat, 29 Jul 2006, Douglas Garstang wrote: You have a config generator script for the Polycom XML files? What did you build that with? Bash scripts and some sed logic. It isn't pretty, but it works. # This creates the actual SED script that we use to modify the template echo s/reg.1.displayName=\\/reg.1.displayName=\$NAME\/ $$.sed echo s/reg.1.address=\\/reg.1.address=\$EXTEN\/ $$.sed echo s/reg.1.label=\\/reg.1.label=\$EXTEN\/ $$.sed echo s/reg.1.auth.userId=\\/reg.1.auth.userId=\$EXTEN\/ $$.sed echo s/reg.1.auth.password=\\/reg.1.auth.password=\$EXTEN\/ $$.sed echo s/reg.1.server.1.address=\\/reg.1.server.1.address=\$SERVER\/ $$.sed echo s/reg.1.server.1.register=\\/reg.1.server.1.register=\1\/ $$.sed echo s/CONFIG_FILES=\phone1.cfg, sip.cfg\/CONFIG_FILES=\phone$EXTEN.cfg, sip-n2net.cfg\/ $$.sed echo s/LOG_FILE_DIRECTORY=\\/LOG_FILE_DIRECTORY=\$EXTEN\/ $$.sed echo s/OVERRIDES_DIRECTORY=\\/OVERRIDES_DIRECTORY=\$EXTEN\/ $$.sed echo s/CONTACTS_DIRECTORY=\\/CONTACTS_DIRECTORY=\$EXTEN\/ $$.sed # Now, we copy the templates to their new filenames cp phone1.cfg phone$EXTEN.cfg cp .cfg $MAC.cfg # Sed it up sed -i -f $$.sed phone$EXTEN.cfg sed -i -f $$.sed $MAC.cfg # Create Directories and Set Perms mkdir $EXTEN chown -R PlcmSpIp:PlcmSpIp * chmod -R a+rwx * # Voila! rm -f $$.sed phone$EXTEN.cfg~ $MAC.cfg~ -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (OT) DS3 Barrel/T-connector/Adtran MX2800 Problems
On Fri, 4 Aug 2006, Steve Totaro wrote: If your M13 is coming up clean, I'd double check the continuity to those ports from the DSX connections out to the patch panel. That was exactly the issue. The amphenol cable was loose on one end. What is the best way to fasten these things? There is only a screw on one side. I am duct taping these things down for now but what is the standard way of fastening them? The Adtran didn't come with a clip or velcro or anything, neither did the TenorAX boxes I am using. We went the Duct Tape route initially for the side that doesn't have the screw, but eventually moved to some velcro contraption that one of our techs dreamed up while drinking at the bar downstairs. So far, so good. We just stay far away from the connections on the back of the rack! ;) The MX200 is a fantastic product, with great support! I'll never even consider buying a mux from another company after the support that Adtran has given us with ours. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ --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] Polycom 1.6.7 Firmware Messages Button
Hello, I recently updated some Polycom 501 phones to the new 1.6.7 firmware, and have lost the ability to do One Touch voicemail access via the messages button. I've verified that I have the correct XML tags set in the phone config, I.E.: msg.bypassInstantMessage=1 mwi msg.mwi.1.subscribe= msg.mwi.1.callBackMode=contact msg.mwi.1.callBack=85100 I've wiped the phone clean, and re-installed firmware and configs, and it still acts as if the msg.bypassInstantMessage tag is set to 0 and displays the status of the messages in the mailbox. I didn't see anything in the release notes indicating a change in the behavior of these tags. Anyone have any suggestions? -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Polycom 1.6.7 Firmware Messages Button
On Sun, 30 Jul 2006, Peter Johnson wrote: How about up.oneTouchVoiceMail=1 in your sip.cfg Peter Ahhh... that tag wasn't in my config generator script, so I must have set it by hand in the old ones. That does the trick! I owe you a beer! -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ --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] Centos 4.3 Issues
Hello, I was wondering if anyone out there is successfully running Asterisk 1.2 svn w/ Centos 4.3. I had an experience over the last two weeks that has me scratching my head and muttering strange things in the wee hours of the morning. I am going to try and be as descriptive as my brain will allow right now, but if there is something that I do not cover, please do not hesitate to ask and I'll be happy to answer. For the last 2 years, I have been running a mixture of Tao Linux and Centos (both RHEL derivatives) on our production boxes. Asterisk has run flawlessly on all installations. Last week, I updated one of our gateway boxes from Centos 4.2 (under which it ran for 6 months without issue) to the new 4.3 code. Almost immediately, we began to experience problems. Asterisk would core w/ the following: #0 0x004878ab in test_err () from /usr/lib/asterisk/modules/codec_g729a.so The segfaults would happen under very light loads, in some cases with just a single call. Kevin was able to log in to the box, and put a debugging version of codec_g729 on the box. He determined that the problem was that the values that were being returned in that routine were incorrect. I.E. something in the system was returning a non-zero value when multiplying a number by 0. Barring any other explanations, we assumed that there was a hardware issue somewhere, either in the memory, or the FPU on the CPU. So, we replaced the box w/ a brand new Dual-Core system running a Dual-Core Pentium D 920. We loaded the 32 bit version of Centos 4.3 onto the box and proceeded to start testing. BAM.. same problem.. the backtrace showed the failure in the same routine. We scratched our heads, and after many hours of trying various things (backing off the kernel to 2.6.9-22) and even moving to the new development kernel 2.6.9-34.19 (from the testing tree) we could do nothing to solve the issue. Mind you, this is the exact same behavior on two different hardware platforms running the exact same distribution. We even loaded up a third box and could reproduce the behavior on it as well. Three different boxes, one common distribution. As a test, we installed Fedora Core 5 x86_64 on the new Dual Core box and ran extensive tests overnight, simulating 96 channels doing G729 to Ulaw transcoding. The box ran completely stable. No hiccups. So, this morning, we put it back into the cluster, and it's now taking about 200 concurrent calls, doing an insane amount of transcoding and it is working just fine. Before, it would have cored in the first couple of minutes. I'm scratching my head here, because I generally have had excellent experiences with Centos. However, I have NO idea what might be the issue here. Could it be the kernel? (We tried three different ones!). Could it be the libc? Maybe it is the compiler? In any case, if anyone is having success with Centos 4.3 (32 bit), please speak up. I'd like to get to the bottom of it. I generally do not like to run Fedora on production equipment as it is generally bleeding edge. In this case, FC5 is running 2.6.16 something.. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Centos 4.3 Issues
On Mon, 22 May 2006, Greg Oliver wrote: Have you tried compiling statically on CentOS 4.2 and running on 4.3? No. Not really in the plans either. Standard policy w/ Asterisk around here is to compile on the box it is going to be running on, under the distro it's running on. I am assuming you have made sure the dist is up to date with patches. We do not use 729, so I cannot try it out for you, but we do use CentOS. Is it only w/ SVN, or all releases of *? This happens to be with the 1.2 SVN branch. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Centos 4.3 Issues
On Mon, 22 May 2006, alist wrote: Greg, When I upgraded to 4.3 I experienced problems with some non-asterisk RPM's that were compiled on earlier versions of CentOS 4. Once they were recompiled on a fully updated 4.3 system they worked fine. Have you tried recompiling everything? We recompiled Asterisk, libpri and zaptel. The one system was an upgrade from Centos 4.2 to Centos 4.3, but the other two were installed w/ the latest Centos 4.3 ISO downloaded last week. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RE: 5,000 concurrent calls system rollout question
On Thu, 2 Feb 2006, John Todd wrote: [SNIP] 3) Nobody else has thus far taken the bait and made any comments about their systems. I appreciate Signate's comments; they seem to be the only ones to publicly claim large-scale throughput using Asterisk in a public forum. Most other people who claim thousands or even high hundreds of connections do so offhand, without responding to second questions when I raise my figurative eyebrows. John, Per our conversation in San Fransciso, I am starting to push a couple of my Asterisk boxes farther than I've gone before. I'm not yet anywhere near the 5,000 concurrent call level on my boxes, but I am starting to see 150-160 concurrent calls coming through the system. In this case, these are SIP to SIP where Asterisk is staying in the media stream, but rarely transcoding. Approximately 99% of the calls coming through are just pass-through g729, with the occasional gsm conversion. I'm running Asterisk 1.2.4-svn in a completely stock configuration. I.E. no patches whatsoever, and absolutely performance tweaks. In fact, the system is running using MALLOC_DEBUG to catch memory leaks and is built using dont-optimize so we can get backtraces if things go south. My Dial-Plan is highly optimized, with a focus on being as efficient as possible while offering failover options for call completion. 4) There are still no notes on other problems with scale here. I've had systems with several hundred simultaneous SIP connections, but sip show channels sure does start to take a while. What _other_ problems crop up, but don't necessarily cause a failure condition? Well, debugging anything on the console with 160 concurrent calls coming through the system (sometimes 4-5 calls / second) is nearly impossible. Most of the time, I don't even run the console, and simply execute commands from a bash prompt as asterisk -rx 'sip show channels'. I ALWAYS, ALWAYS, ALWAYS issue a set verbose 0 before I reload the box, as a reload causes the box to hiccup slightly while it is printing the data to the console. I had originally opted to write CDRs to disk and then import them into a SQL database, but after I cleaned up my dial-plan, I opted to use cdr_odbc. I am concerned that this could cause a blocking condition if the SQL server is unavailable, but for now I'm taking the risk because I need to have real-time stats on call statistics. 5) I will agree that most SIP testing systems are currently too pricey. I would love to find a well-connected network that rents out a few of the better-known SIP testing tools to beat on Asterisk installations in remote places for short periods of time. But this has always been the case... test gear is a small market, and expensive. Just look at the MSRP of new high-end HP Oscilloscopes if you want to get a picture of price-gouging. I know that Olle spent some time at SipIt w/ Asterisk, and he's been interested in doing some additional compliance and scalability testing. I'd like nothing better than to get a couple of key developers together for a weekend of scalability bashing somewhere, preferably outside of the regular conference circuit (too distracting) to push things to their limits. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T3 Mux and Asterisk Question
On Mon, 23 Jan 2006, Kevin P. Fleming wrote: Greg Boehnlein wrote: (Steve Totaro wrote:) What I would really like to do is have one D channel coming in on the T3 and have it split between each of the T1/PRI or even better one D channel per quad (I know Asterisk can do that). Is it possible? No. Actually, it is, using an Adtran Atlas with a DS3 interface and DS1 interfaces. Not cheap, but possible. Yeah, but he's already stated that he will be using MX2800 muxes. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T3 Mux and Asterisk Question
On Sun, 22 Jan 2006, Steve Totaro wrote: I have a T3 coming from my carrier. From there I want to use an Adtran mx2800 T1 Mux to break the T3 into 28 T1/PRI which feed into seven quad T1/PRI equipped servers. Everything seems very straight forward with the exception of the D channels for the T1/PRI. I am not very familiar with large circuits such as T3s. I know that I can use one D channel per set of quad port on each server. So if each server has a quad port card, I can use one channel as the D channel for all four spans. That gives me seven D channels in my setup. Does anyone know how the Mux handles these D channels onto the T3? My guess is the Mux is simple going to send all of the channels onto the T3 without modifying anything. That's correct. The T1 spans on the DS3 are completely independent of the clocking on the DS3. The D-channel and timing is something that will be handled by your upstream Telco and the switch that you'll be connecting to. Or, your own box.. ;) What I would really like to do is have one D channel coming in on the T3 and have it split between each of the T1/PRI or even better one D channel per quad (I know Asterisk can do that). Is it possible? No. If the Adtran mx2800 cannot do it, is there anther product that can. I have looked at the RAD Optimux T3 product but have had great experience with Adtran products. The price is the same but the Adtran allows for two controller cards so it seems to have more built in redundancy. Any tips would be appreciated. Adtran's MX-2800 is our choice for Muxes. They are solid, reliable and work well. Adtran's technical support is amazing. When you purchase an MX-2800, you are immediately given access to the Adtran Carrier support group, which doesn't even blink about sending out an advance replacement unit overnight if you ask. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] T3 Mux and Asterisk Question
On Sun, 22 Jan 2006, Steve Totaro wrote: Thanks for some answers, that is what I thought. Asterisk is NFAS capable so I am looking at seven D channels on the T3 I guess. I don't want to put a D channel on each T1 or I will lose several channels that could be used for calls. I wonder if there is any way that Asterisk can do NFAS across multiple servers. I would put two cards in a box but they will be doing g729 transcoding so I don't want to push it, so it is one per server. Hehehe.. Ask your Telco if they can provision E1 for you. ;) The Digium cards can handle E1 or T1, and if you go E1 you'll get 30 channels instead of 24 on the span. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ --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] Dialstatus Oddity in 1.2
Hello all,
I am working on a creating some intelligent failover dial-plan
logic and I'm running into something that I'd like some feedback on.
Basically, it appears that if you place a call to an IAX2 peer that
refuses the connection, or is unavailable, a NOANSWER dialstatus is
returned.
Example:
-- Executing Macro(IAX2/cubix-19, nocdial|IAX2/[EMAIL
PROTECTED]/1216410) in new stack
-- Executing Dial(IAX2/cubix-19, IAX2/[EMAIL PROTECTED]/1216410|30)
in new stack
-- Called [EMAIL PROTECTED]/1216410
Jan 21 19:16:07 WARNING[1114]: chan_iax2.c:6970 socket_read: Call rejected by
207.166.192.188: No authority found
-- Hungup 'IAX2/pbx1-21'
== No one is available to answer at this time (1:0/0/0)
-- Executing Goto(IAX2/cubix-19, s-NOANSWER|1) in new stack
-- Goto (macro-nocdial,s-NOANSWER,1)
-- Executing Hangup(IAX2/cubix-19, ) in new stack
-- Hungup 'IAX2/cubix-19'
Shouldn't that return CONGESTION instead? I thought that NOANSWER was
reserved for calls that reach app_dial's timeout limit?
Or am I just missing something simple?
Here is the relevant extensions.conf logic that I am using
[e164]
; Dundi
exten = _1NXXNXX,1,Macro(dundi-e164,${EXTEN})
; Dispatch First Trunk
exten = _1NXXNXX,2,Macro(nocdial,${TRUNK}/${EXTEN})
exten = _1NXXNXX,3,ResetCDR
; On Failure, Dispatch Second Trunk
exten = _1NXXNXX,4,Macro(nocdial,${TRUNK2}/${EXTEN})
exten = _1NXXNXX,5,ResetCDR
; Third time is a charm?
exten = _1NXXNXX,6,Macro(nocdial,${TRUNK3}/${EXTEN})
[macro-nocdial]
exten = s,1,Dial(${ARG1},30)
exten = s,2,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Hangup
exten = s-BUSY,1,Busy(15)
exten = s-BUSY,2,Hangup
exten = s-CONGESTION,1,NoOp
exten = s-CHANUNAVAIL,1,NoOp
exten = s-.,1,Goto(s-NOANSWER,1)
--
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Re: [Asterisk-Users] 1.0.10 to 1.2.1 upgrade..is it worth it?
On Wed, 11 Jan 2006, stevanus wrote: Hi, As I've dealt with asterisk 1.0.10 successfully, I wonder what the benefit I will get from upgrading to 1.2.1.. [Of course I know there're lot of new interesting stuffs in 1.2.1, but are they stable already?] Does the 1.2.1 need more resources, more power hungry? Anyone has success story with asterisk 1.2.1 please share :) Thank you... If it is of any assistance, we just dropped a 90 seat call center with a predictive dialer onto an Asterisk 1.2 box this afternoon w/ a TE405P in it. Over the course of 1/2 hour, they sent 3,000 calls through the box and it is still going strong! ;) It's being hammered without mercy, and this is a single Intel 3Ghz CPU doing SIP - ZAP w/ G729 and SIP - SIP Passthrough. They are averaging about a call a second through the box. We had more problems with our SIP Termination upstream not being able to handle the load... Asterisk didn't blink. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: DTMF Issues With Asterisk 1.2 IVR
On Thu, 12 Jan 2006, Steven wrote: If it is a toll free number, it may be related to http://bugs.digium.com/view.php?id=5266 . You may also want to investigate: http://bugs.digium.com/view.php?id=5970 as well as: http://bugs.digium.com/view.php?id=6027 There are several people that are putting some effort into these issues, and if we can get some input from a wider audience it might help us squash the outstanding issues faster. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] double ringing tone on asterisk 1.2
On Sat, 14 Jan 2006, Brian Capouch wrote: Rich Adamson wrote: Since there does not seem to be anyone else complaining about the same problem, there must be something in your config that is causing it. Without specific copy/paste samples of what you've configured, no one is going to be able to guess at what you are doing. Given the issue is happening with both PRI's and IAX links, I'd have to guess that you've got something wrong in extensions.conf. Actually, I'm not complaining, but I've experienced a similar problem. For me it only happens with IAX calls, and from there only IAX calls with a particular ITSP. I have that problem when using IAX transport as well. I've just ignored it until now, cause it is on a home box. But his description rings very true: there are two rings per time that the far-end phone is ringing, and just like he said, one of them sounds genuine and the other sounds like perhaps my provider is adding it to the audio coming my way. The dinger is that it doesn't happen on all calls; it's more or less random, but frequent. I don't mind it, though, so I'd never complained about it on the list. FWIW. B. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: 1.2.1 Silence suppression is disabled whatthehell?
On Mon, 16 Jan 2006, Koopmann, Jan-Peter wrote: On Sunday, January 15, 2006 12:21 AM Tony Mountifield wrote: In article [EMAIL PROTECTED], Pisac [EMAIL PROTECTED] wrote: I've found something here: http://bugs.digium.com/view.php?id=5374 but I don't understand how this can be connected to my problem :-( It looks like the maintainer of the BRIstuff distribution might have decided that patch was worth including, even though it is not in the standard 1.2.1. That does give scope for confusion though! Look at the CHANGES. I was the one who convinced kapjeod to put that patch in the current bristuff distribution. So yes: It is in bristuff as of 1F: 0.3.0-PRE-1f - THIS IS GETTING CLOSER TO A STABLE RELEASE, USE IN PRODUCTION AT YOUR OWN RISK! - merged patch for bug 5697 (meetme) - merged patch for bug 5374 (asynchronous generation of outgoing frames) - _finally_ fixed sending-nonRFCcompliant-SIP-NOTIFYs bug (asterisk, extension states) - some debug output clean ups in libpri Per other discussions on this issue, that patch breaks a ton of stuff (IAX Jitter Buffer, Music On Hold etc..). Although it does solve some issues, it really needs to be reviewed by They Who Decide Such Things as it has far reaching impact on other areas of the system. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MoH trouble with latest bristuff (0.3.0-PRE-1f)
On Tue, 17 Jan 2006, Kib Eki wrote: Hi Karsten, I have the same problem. MusicOnHold sounds awful. The PRE-1e does not have this problem. I have two identical systems (hard-/software). One system has the problem the other does not. I thought i could be timing problem or interrupt conflict. But we could not find out the problem. Bernd Karsten Wemheuer wrote: Hi, I've installed * 1.2.1 with latest bristuff patches (0.3.0-PRE-1f). When I activate music-on-hold on a SIP-to-SIP connection, the music sounds like in a fast-forward play mode. On the *-console I can see much lines like this: -- Silence suppression is disabled (option_silence_suppression=0 chan-timingfd=18) What's going on? With bristuff 0.3.0-PRE-1d everything works fine (but there was another issue, so I have to upgrade). bristuff 3.0 apparently contains the asynchronous frame generator patch from http://bugs.digium.com/view.php?id=5374 This apparently breaks a LOT of other stuff, MOH being one of them. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk down because of cdr
On Tue, 17 Jan 2006, Jean-Michel Hiver wrote: Dov Bigio a ?crit : Ok.. but I don't use Real Time at all. I just use cdr_mysql. It would be smarter if it simply ignored MySQL outages or at least just logged, but without stopping. What would be even nicer would be for * to buffer it for a while before it starts dropping cdrs... Might want to use cdr_odbc instead. The issue that you are discussing was brought up by BKW last year, and I was under the impression that it had been patched. I.E. that when MySQL died, Asterisk would deadlock. I'm pretty sure that cdr_odbc (which is cake configure) handles this situation properly. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] asterisk down because of cdr
On Tue, 17 Jan 2006, Alexander Lopez wrote: Buffer! For how long? How big of a buffer? If I can buffer 10-20 calls that might work if I have a light use PBX but 100-2000 buffered calls may not hold a busy PBX. OK so make it configurable, With any luck you won't know how much to put so you will allocate more than you need, using more memory for a single senario. My solution, make sure your DB is stable. I would rather put my effort in building a better solution than counting on insurance to bail me out. Hehehe.. it should be a dynamic buffer that should grow / shrink as neccessary and consume as much memory as it needs until the database backend comes online. It should do periodic checks of the server to see if it is available and then spool the buffered calls out to it, but only spool stuff in the event the connection is not available. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] asterisk down because of cdr
On Tue, 17 Jan 2006 [EMAIL PROTECTED] wrote: Buffers don't have to be in memory. My suggestion on the solution would be to buffer the CDR info into a backup file based database (configurable filename/path) on the local filesystem (or NFS mounted system for redundancy) and then when the SQL database connection is restored then it spends a second dumping the buffered CDR info from the file into the database and erases the file (or empties it). Why not just use the astdb instead of a file? Dundi does it.. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ --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] Bugs that Need Your Input!
Hello, I know that Mog was trying to get the bug-tracker cleaned up as the number of bugs has increased substantially over the past few months. I figured that I would do my part to bring attention to a couple of bugs that are interesting and have some wide reaching impact. That being said: http://bugs.digium.com/view.php?id=5970 - This has a patch created by Corydon, but it has segfaulted on three different boxes. Backtraces are attached to help track this down. http://bugs.digium.com/view.php?id=6027 - This is related to the bug above and has a different patch that might be relevant. http://bugs.digium.com/view.php?id=5374 - This is a patch that allows for the asychronous generation of outgoing frames. Very cool. Needs a decision by the Core Developers (Read Kram / KPFleming). BJ has been keeping this in synch w/ Trunk in his own branch. http://bugs.digium.com/view.php?id=5090 - T.38 Support anyone? http://bugs.digium.com/view.php?id=5574 - A nifty new app called Find Me, Follow Me that does exactly what it says. If you can, take a look at these and add your entropy to the mix! -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ --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] O'Reilly's Etel Conference
Hey there, Just wanted to drop a line and let people know that I'll be heading to San Francisco for O'Reilly's Etel. If you are interested in attending, there are some free passes floating around. If anyone is interested in getting together for a beer, let me know! Info on the conference can be found here: http://conferences.oreillynet.com/etel2006/ I'm looking forward to an ejoyable uber-geek experience! -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ipVolution
On Wed, 28 Dec 2005, Goran Skular wrote: Hi, Anybody have some experience and did some testing with ipVolution E1/T1 cards? I chalked these up to VaporWare... -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] app_icd anyone? on 1.2?
On Tue, 22 Nov 2005, Lenz wrote: I also have never found anybody running an Asterisk system using app_icd. Maybe app_queue is now after all flexible enough to be used in most cases. Anybody else using different apps for Asterisk call centre applications? I suspect that since the authors of ICD are no longer really submitting patches to Asterisk, that ICD for Asterisk is probably end of life, unless someone wants to pick up and take on maintenance of the code. As far as I know, ICD was never officially accepted into the mainline tree and was always kept as an outside project. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ --Bandwidth and Colocation sponsored by Easynews.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
Re: [Asterisk-Users] v1-2 install mkdep loop
On Mon, 21 Nov 2005, Bob Knight wrote: Just pulled a v1-2 onto a system that was running a v1-0. Zaptel and libpri, build and install just fine. Building asterisk is fine. But when I try to do a make install on asterisk, it goes into an infinite loop doing on .depend doing: build_tools/mkdep I did the same thing on another box the other day with a different pull and did not have any problems. Do you think this is something related to this box? Hi Bob! Long live the PM3! This is an issue that many many people have been running into, and has been discussed on the dev list. Check the following: http://lists.digium.com/pipermail/asterisk-dev/2005-November/016509.html I'm not sure there is a specific fix, although there are many suggestions in that thread. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ --Bandwidth and Colocation sponsored by Easynews.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
RE: [Asterisk-Users] Asterisk 1.0/1.2 on cobalt Raq2-4
On Mon, 21 Nov 2005, Jonathan k. Creasy wrote: I've thought about doing that as I have a few spare also. I would use the raq4 I think. Let me know if you have any trouble with it. What you may want to do (I have several of these) is see if you can re-install the new Centos + BlueQuartz (GPL'd Raq GUI) ISO onto a drive and get it to boot in a Raq. http://www.nuonce.net/bq-cd.php -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ --Bandwidth and Colocation sponsored by Easynews.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
Re: [Asterisk-Users] Anyone know who is in this picture?
On Fri, 18 Nov 2005 [EMAIL PROTECTED] wrote: I couldn't find his bio on rotten.com http://www.rotten.com/library/bio/hackers/captain-crunch/ -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ --Bandwidth and Colocation sponsored by Easynews.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
Re: [Asterisk-Users] Anyone know who is in this picture?
On Wed, 2 Nov 2005, Matt Darnell wrote: Well that didn't take long! He was a really nice guyI bet it would be a blast to go have a beer with him. We met him at the Internet Telephony Expo. Read his bio on Rotten.Com. I'm surprised to see him posing with Women. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ --Bandwidth and Colocation sponsored by Easynews.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
Re: [Asterisk-Users] Dual PRI fail over
On Tue, 11 Oct 2005, Tom wrote: I currently have a single PRI however we are getting a second PRI, and the provider (qwest) wants to know if our PBX supports GSAS (they say its a redundant d-channel technology but searching on google for GSAS reveals less than nothing). I've set something similar up before on a cisco 5350, where if one of the PRIs fails, all of the calls destined for either PRI will be routed down the one that didn't fail. Basically the 2 PRIs are bonded together, and act as one. During normal operation the calls come down each PRI in a load balanced fashion (IE if I've got 30 calls up, 15 will be on one PRI and 15 on the other). Is there any way to set something similar to this up in Asterisk? Tom They are probably talking about NFAS. And after looking at the date of this message, it's probably a moot point. But I'm not sure if NFAS is supported under Libpri/Zaptel. I'd suspect that if it is, it might not be widely deployed and/or tested with it. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ --Bandwidth and Colocation sponsored by Easynews.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] TDM2420E Availaibility
Hello, Rumor has it that the TDM2400 series cards will be available in the next week or so. If you are a distributor that has pricing / availability information, please contact me offlist. I am putting together a solution for a client that will require a TDM2420E (8 Port FXS w/ Echo Can) and I need to compare it against an Audiocodes 8 port SIP - FXS gateway. Thanks -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ --Bandwidth and Colocation sponsored by Easynews.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
Re: [Asterisk-Users] Polycom Reboot Script
On Mon, 29 Aug 2005, Kristian Kielhofner wrote: Matt, It sure is! You should be testing it! :) Test it and see, but 1.2 will be STABLE pretty soon here... No. No NO NO NO NO! :) 1.2 will never be called Stable, based on the controversy surrounding the naming moniker. Do you ever heard Linux 2.0, 2.2 or 2.4 referred to as stable? No. People tend to confuse Stable as in No Additional Features, Bug Fixes Only with Stable as in It never has any problems. But.. I may be too late on this one.. ;) -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ --Bandwidth and Colocation sponsored by Easynews.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
Re: [Asterisk-Users] Updated Patch to chan_agent.c for PREACKANNOUNCE
On Thu, 18 Aug 2005, Hadar Pedhazur wrote: First, many thanks to Greg Boehnlein for his patch to chan_agent.c for adding a preackannounce option. I am running CVS HEAD from 2005/07/31, and the patch failed in a few hunks, since the code was refactored to add in some CASE statements where there were compound if statements before. Anyway, I have successfully updated the patch to work against head as of 3 weeks ago, and would happily share that with anyone who is interested (just drop me a line off list). If a diff is preferable to the full 70k of C, just let me know what the correct options are for creating a diff suitable for patching the asterisk tree. OK, that said, I have a few questions and comments on this topic. This is my first use of the Queue command (very successfully so far), but I am afraid that expanding my use will require further patches, and I would like to verify that first. 1) If I use the syntax: Member = SIP/100 (rather than member = Agent/100, which maps to SIP/100) Then ackcall isn't used at all. In other words, a hard-wired member seems to ignore the agents.conf file completely. Is this the desired behavior? (It isn't for me...) It is the correct behavior because when you use SIP/100, chan_sip (which has no concept of an agent) is being used instead of chan_agent to deliver the call. Think of chan_agent as an intermediary between the PBX core and the physical endpoint. chan_agent accepts the call, puts the caller on hold and then grabs chan_sip to complete the other side of the call. When all requirements of chan_agent are met, it then proceeds to bridge the two calls together and get out of the way. 2) Since agents.conf is a separate file from queues.conf, having multiple queues does _not_ permit multiple preackannounce messages, each tied to a different queue (this strikes me as having better been patched into the Queue command). Similarly, you can't have one queue that has ackcall=yes, and another with ackcall=no. Right. It's a botched design and chan_agent's design doesn't lend itself to being very helpful in the process, but that is where it had to go. This is the reason that I dropped work on it, as ICD was a much more intelligent design at the time. 3) I have the _exact_ same source version of CVS HEAD (from 2005/07/31) running on different servers (after a cvs co, I tar the source so that I can be sure I'm running _identical_ versions). On one machine, when an Agent logs in, I can see it in the DB, database show shows a key of: //Agents/1001 : [EMAIL PROTECTED];1001 On another machine, the DB shows _nothing_, yet the AgentCallbackLogin application works correctly (logging agents in and out), and shows the correct mapping on the CLI during a login. Still, the DB has _no trace_ of the Agents. I can't explain the difference in behavior, and would _love_ to have someone solve that mystery for me. I'm hoping that I am missing something obvious in the interaction between the Queue command and the Agents channel, and that some kind soul here will educate me. Otherwise, I think I might be off to doing more work in C than I ever though I would again in my life ;-). Don't have an answer for this one. :) -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ --Bandwidth and Colocation sponsored by Easynews.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
Re: [Asterisk-Users] Updated Patch to chan_agent.c for PREACKANNOUNCE
On Fri, 26 Aug 2005, Hadar Pedhazur wrote: Hmmm. I am often surprised when I don't get a response to a post that I think would interest at least _one_ person in the community. This one surprised me a little more, since I offered some code ;-). This morning, I just got a bounce notice that it was undelivered, which might explain it, except that I received the original post back through the list, so I don't understand it at all... Anyway, I solved the one bone-headed problem that I describe below, namely why did the agents show up in one DB and not the other. I didn't set the persistent keyword in the agents.conf file (doh...). All of my other questions still apply, as well as my offer to share the code/patch. I just got your message and responded to it! ;) -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ --Bandwidth and Colocation sponsored by Easynews.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
RE: [Asterisk-Users] yet another Asterisk and VMware question
On Sat, 13 Aug 2005, Lull, Rick wrote: This is one of the main reasons that AstWind has stagnated. The timing granularity of the virtual machines is not acceptable for doing anything IO related. Just since I am curious, what version of VMWare did you use and what kind of box where you running on? 4.5 and 5.0 variants of Vmware Workstation and CoLinux on top of a Windows host environment. Same applies for running Vmware under a Linux host environment as well. Machines where 2.8 Ghz Dell boxes w/ 1.5 gigs of ram. I've just moved my * box to a VM on ESX server and didn't play with voicemail until you mentioned it - now Allison's voice cuts in and out. Sounds like I am going to have to go back to the box I was running on previously. My original box is a P3 500 desktop while my VMWare ESX box is a dual P3 1.4GHz HP Proliant server. You would probably get the BEST performance by using User Mode Linux, w/ the latest SKAS patches. I know someone that does this and only occasionally has a dropped frame in audio here and there. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ 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] IP Cop as a firewall and QOS
On Thu, 18 Aug 2005, Holden Hao wrote: I don't mind buying an appliance to get something solid but IP Cop just looks better than he appliances I see out there. Astaro has been getting good reviews from Linux World. They have an appliance solution or a self-install solution. It features: -Firewall -VPN Gateway -Intrusion Protection -SPAM Filtering -Anti-Virus -Management Platform -Surf and Spyware Protection The details of the features are impressive. For the details visit: http://www.astaro.com You can download a 30-day demo. If cost will be a problem, IP Cop is also a good solution. This is what we have been using. I second the vote on Astaro. They have, without a doubt, fused the best features of a Perimiter Security Device w/ Open Software and an excellent GUI. Version 6.0 now includes a SIP proxy as well! ;) -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ 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] TE405P / TE410P with 2nd generation firmware field upgradable?
On Fri, 12 Aug 2005, Matt Florell wrote: Short answer: NO Long answer: you have to send it to Digium for them to do an upgrade, they don't have an official process for this yet and won't give you a price, I have called and asked them many times. They also mention upgrades from your 405/410 to a 406/411 are available too, but again no specifics. Supposedly if you have a card with the 2nd gen firmware on it you can upgrade to the third gen firmware, whenever it would come out, in the field. Hmm.. That's funny. I called yesterday and talked to someone who told me the upgrade to Second Gen firmware was free, but that if I wanted to add the Echo Cancelling module, it would be $850. Since I do not have any major echo issues that software echo cancelling can't fix, I declined the upgrade. They even offered me an advance replacement option as long as I provided a Credit Card. The RMA process was painless. I spoke with Joy Lister. I should have my new card early next week. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ 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] yet another Asterisk and VMware question
On Fri, 12 Aug 2005, Bruce Leetch wrote: Am I banging my head against at Windows/VMware/Linux/Asterisk incompatibility? Or can this work and I'm just doing something stupid (always a possibility with me). It's not going to work. Vmware presents a complete Virtual PC, so unless EMC / Vmware release drivers to specifically connect the Virtual PC to the real PC hardware, you are in trouble. The do a pass-through w/ USB, Serial and Paralell, but those are a different story. With a great amount of effort, I can drum up a spare machine, but I REALLY don't want to do this and would much prefer the VMware setup. Any advice will be welcomed. I'm afraid that under any Virtual platform (CoLinux, Vmware, MS Virutal PC) you are SOL as far as real access to hardware on the host PCI bus w/out special drivers written specifically for that purpose. On the other hand, I'm sure that Vmware would be happy to help you out if you gave them a couple of million bucks! ;) -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ 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] yet another Asterisk and VMware question
On Fri, 12 Aug 2005, Tom Rymes wrote: VMWare is a virtual machine and has nothing to do with the physical layout of the box (which is why you can migrate vmware images across machines for example). If you want to run Asterisk under Linux setup a box to run it. Agreed. You would be better to grab a used $200 machine and install linux Asterisk on it. Unless you are scaling up to at least 10+ simultaneous calls, I would imagine that something you have lying around would handle it. If you insist on VMWare, I would imagine that you could configure a Sipura SPA-3000 to provide incoming (FXO) and analog extension (FXS) ports This works. I've done it on occasion for testing. However, because virtual PCs rarely operate on a real-time clock, mostly emulating these features, you will find that anything that read/writes to disk will suck badly. For example, it is nearly impossible to use the Voicemail features of Asterisk under Vmware, CoLinux or UserMode Linux. Believe me, I've tried! ;) This is one of the main reasons that AstWind has stagnated. The timing granularity of the virtual machines is not acceptable for doing anything IO related. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ 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 IP301 and 501 with asterisk...
On Thu, 11 Aug 2005, Trevor Peirce wrote: Ing. Marlo R. Beltran G wrote: I am about to buy ip pbx asterisk system but what ip phones do you recommend? Are polycom ip all functional with the ip pbx system??? We just got a Polycom IP501 for testing and have thus far been unsuccessful at getting it to regiser with asterisk. Outgoing cals work fine now (with authentication; verified with ethereal). Then you have something misconfigured. I have nearly 100 of them deployed at customers around the area, and they are bulletproof. Check out http://www.krisk.org/asterisk/pcom/ipmid.cfg for an example. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ 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-Dev] Memory Leak in Stable?
On Wed, 20 Jul 2005, Greg Boehnlein wrote: Hello, I have a client that has a fairly small installation (20 SIP Phones) that is running Stable. Asterisk appears to be consuming large quantities of memory, and growing uncontrollably to the point where after about 6 weeks the box starts to swap itself to death. I've been keeping my eye on it today, and in the last 12 hours, it has grown by about 8 megabytes, and there has been no-one in the office placing any calls. Score one bug fixed for Clue Con! Mark gave me some pointers on how to debug this, and I did some legwork for him. He was able to isolate the memory leak to app_dial, and sumbmitted patches to both Stable and Head branches. Thanks to everyone that sent suggestions! -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ 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-Dev] Memory Leak in Stable?
On Wed, 20 Jul 2005, Greg Boehnlein wrote: Hello, I have a client that has a fairly small installation (20 SIP Phones) that is running Stable. Asterisk appears to be consuming large quantities of memory, and growing uncontrollably to the point where after about 6 weeks the box starts to swap itself to death. I've been keeping my eye on it today, and in the last 12 hours, it has grown by about 8 megabytes, and there has been no-one in the office placing any calls. Score one bug fixed for Clue Con! Mark gave me some pointers on how to debug this, and I did some legwork for him. He was able to isolate the memory leak to app_dial, and sumbmitted patches to both Stable and Head branches. Thanks to everyone that sent suggestions! -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ 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-Dev] Cluecon - Who's going ?
On Mon, 25 Jul 2005, Brian West wrote: I'm going to be speaking about how to use valgrind, gdb and strace to help debug issues... it can be applied to more than just asterisk. Given the following from one of my Client's boxes... pbx*CLI show memory summary [DELETED] 7084 bytes in 435 allocations in file 'res_indications.c' 223 bytes in24 allocations in file 'chanvars.c' 51734730 bytes in 186815 allocations in file 'frame.c' 51993128 bytes allocated 188799 units total I'll be REALLY interested in your talk! Please make sure that you have take-away notes available so it doesn't evaporate into thin air after the conference! :) -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ 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-Dev] Cluecon - Who's going ?
On Mon, 25 Jul 2005, Brian West wrote: I'm going to be speaking about how to use valgrind, gdb and strace to help debug issues... it can be applied to more than just asterisk. Given the following from one of my Client's boxes... pbx*CLI show memory summary [DELETED] 7084 bytes in 435 allocations in file 'res_indications.c' 223 bytes in24 allocations in file 'chanvars.c' 51734730 bytes in 186815 allocations in file 'frame.c' 51993128 bytes allocated 188799 units total I'll be REALLY interested in your talk! Please make sure that you have take-away notes available so it doesn't evaporate into thin air after the conference! :) -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ 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-Dev] Cluecon - Who's going ?
On Mon, 25 Jul 2005, Terry Moore-Read wrote: I'm relatively new to Asterisk and I'm hoping attending Cluecon will be a good way to get up to speed on the project and hear about what others are doing with it. We currently use a Cisco IP phone system at my office, although I just added an asterisk box to provide soft phones to our travelling users. (IAX2 is a lot easier to get through firewalls than cisco's protocols). Terry Moore-Read Lukins Annis, P.S. Spokane, WA I'm going. I'm speaking too. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ 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-Dev] Cluecon - Who's going ?
On Mon, 25 Jul 2005, Terry Moore-Read wrote: I'm relatively new to Asterisk and I'm hoping attending Cluecon will be a good way to get up to speed on the project and hear about what others are doing with it. We currently use a Cisco IP phone system at my office, although I just added an asterisk box to provide soft phones to our travelling users. (IAX2 is a lot easier to get through firewalls than cisco's protocols). Terry Moore-Read Lukins Annis, P.S. Spokane, WA I'm going. I'm speaking too. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ 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] Stupid hold music
On Fri, 22 Jul 2005, Thomas Christie wrote: * If you can get the song from this flash animation converted to MP3, then it might be good (bad): http://www.ebaumsworld.com/flash/spacepeople.html . http://damin.umlcoop.net/spacepeople.mp3 -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ 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] Memory Leak in Stable?
Hello, I have a client that has a fairly small installation (20 SIP Phones) that is running Stable. Asterisk appears to be consuming large quantities of memory, and growing uncontrollably to the point where after about 6 weeks the box starts to swap itself to death. I've been keeping my eye on it today, and in the last 12 hours, it has grown by about 8 megabytes, and there has been no-one in the office placing any calls. [EMAIL PROTECTED] tftpboot]# asterisk -rvc Asterisk CVS-v1-0-07/03/05-05:09:22, Copyright (C) 1999-2004 Digium. Written by Mark Spencer [EMAIL PROTECTED] = Connected to Asterisk CVS-v1-0-07/03/05-05:09:22 currently running on pbx (pid = 731) nection Verbosity is at least 3 [EMAIL PROTECTED] tftpboot]# top 04:57:40 up 16 days, 23:44, 1 user, load average: 0.00, 0.00, 0.00 48 processes: 46 sleeping, 2 running, 0 zombie, 0 stopped CPU states: cpuusernice systemirq softirq iowaitidle total0.2%0.0%0.0% 0.0% 0.0%0.0% 99.8% Mem: 1025700k av, 630604k used, 395096k free, 0k shrd, 84032k buff 406804k active, 76016k inactive Swap: 2096472k av, 0k used, 2096472k free 157732k cached PID USER PRI NI SIZE RSS SHARE STAT %CPU %MEM TIME CPU COMMAND 731 root 15 0 252M 252M 3860 S 0.0 25.2 0:01 0 asterisk 734 root 25 0 252M 252M 3860 S 0.0 25.2 0:00 0 asterisk 743 root 25 0 252M 252M 3860 S 0.0 25.2 0:00 0 asterisk 744 root 15 0 252M 252M 3860 S 0.0 25.2 1:05 0 asterisk 746 root 25 0 252M 252M 3860 S 0.0 25.2 0:00 0 asterisk 750 root 15 0 252M 252M 3860 S 0.1 25.2 0:43 0 asterisk 760 root 15 0 252M 252M 3860 S 0.0 25.2 0:00 0 asterisk 763 root 15 0 252M 252M 3860 S 0.0 25.2 15:43 0 asterisk 766 root 15 0 252M 252M 3860 S 0.0 25.2 0:00 0 asterisk 778 root 25 0 252M 252M 3860 S 0.0 25.2 0:00 0 asterisk 780 root 15 0 252M 252M 3860 S 0.0 25.2 0:00 0 asterisk 781 root 25 0 252M 252M 3860 S 0.0 25.2 0:00 0 asterisk 782 root 15 0 252M 252M 3860 S 0.0 25.2 0:00 0 asterisk 790 root 15 0 252M 252M 3860 S 0.0 25.2 0:00 0 asterisk What can I do to debug this? This is a production system, so during business hours I can't muck about with it, but after-hours I can pretty much do whatever I want to it... I just need some guidance. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ 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] AstLinux creator to speak at Cluecon
On Wed, 20 Jul 2005, Olle E. Johansson wrote: Brian West wrote: Kristian Kielhofner, the lead developer of the AstLinux project, will be speaking at ClueCon. His latest AstLinux Version 0.2.6 is a complete Kristian will also be speaking at Astricon 2005 in California http://www.astricon.net/2005/ And he'll also be speaking at Ohio Linuxfest 2005 in Columbus, Ohio http://www.ohiolinux.org ;) Kristian.. making the rounds... -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ 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] Linksys WRT54GP2-NA settings forperformanceandlow bandwidth?
On Wed, 29 Jun 2005, Paul Fielding wrote: I have indeed already done so - I use G729 quite a bit since I travel alot in adverse conditions. Interesting thing is, I can get less choppy audio sometimes from my Vonage device using (what I suspect to be) Ulaw, while either ulaw or G729 will still give choppy response at that moment from my Linksys Paul Funny thing about codecs.. Sometimes you will get better quality audio and less chop when using ulaw on crappy connections. Why? Very simply because it sends less total audio information per packet, so it can withstand dropped packets a lot better than a highly compressed codec, especially with a good jitter buffer and PLC implementation on the other side. I'd expect that Vonage is running something with a decent PLC and Jitter buffer, which would explain your results. Try using CVS head and enabling the SIP jitter buffer and PLC code w/ Ulaw and see if it improves your results. ___ 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-Dev] ClueCon, Vote?
On Tue, 28 Jun 2005, Brian West wrote: Ok I have to get a vote of all the people that are going to come to Cluecon so we order the beer keg's for the developers board room. Anyone have any preference? (if you haven't registered for ClueCon now is the time to register!) Choices... choices... choices... I want Red Bull on tap! I would suggest that you investigate Great Lakes Brewing (http://www.greatlakesbrewing.com), specifically their Dortmunder Gold. It is the only beer that I drink when given the choice. A second, cheaper option is Killians (AKA Coors). Dortmunder Gold Lager ABV: 5.8% ABW: 4.3% IBU: 30 TYPE/STYLE Dortmunder: During the mid-19th century, seven breweries within the city of Dortmund, Germany began brewing beers in the same manner, resulting in what has come to be known as the Dortmunder style. FOOD COMPLEMENTS Because neither malt nor hops dominate this beer, it complements most foods, especially salads, fish and chicken. BRAND NAME The name of our flagship beer reflects the unprecedented accolades and recognition it has earned in major worldwide beer tasting competitions. AWARDS Silver Medal, 2004 World Beer Championships Gold Medal, 2002 World Beer Championships Gold Medal, 2001 World Beer Championships Gold Medal, 2000 World Beer Championships Gold Medal, 1999 World Beer Championships Gold Medal, 1998 World Beer Championships World Champion, Gold Medal, 1997 World Beer Championships World Champion, Gold Medal, 1996 World Beer Championships World Champion, Gold Medal, 1995 World Beer Championships World Champion, Gold Medal, 1994 World Beer Championships Gold Medal, 1990 Great American Beer Festival -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ 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 Card: Echo, Echo and more Echo
yOn Tue, 21 Jun 2005, Matthew Boehm wrote: We have a TE110P (single span PRI) and are having tons of echo on all calls, both incoming and outgoing. We didn't have any echo at all yesterday and nothing in any of the configs has changed. All of all calls follow this pattern: Cisco 7960 - Asterisk - PRI If it is Near Side echo (I.E. on the 7960 side) try telling your people to lower the handset volume to 50% and see if it goes away. I've found that the 7960's have very sensitive mics in the handset and if the earpiece volume is too high, the mic will pick it up and it shows up as a really annoying echo artifact that will drive you nuts. Nothing you will do on the Asterisk side will take care of it. I've not had any issues w/ my TE100P and TE405P cards using X/O, Easton and Broadwing PRIs. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ 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] Dundi - Multiple Results
Hello,
If one is using Dundi, and it returns multiple, weighted routes to
a destination, how is that applied in the dial-plan?
asterisk*CLI dundi lookup 1588XXX
1. 400 IAX2/dundi:[EMAIL PROTECTED]/1588XXX
(EXISTS|NOUNSLCTD|NOCOMUNSLTD)
from 00:30:48:71:26:70, expires in 1098 s
2. 400 IAX2/dundi:[EMAIL PROTECTED]/1588XXX
(EXISTS|NOUNSLCTD|NOCOMUNSLTD)
from 00:30:48:71:27:51, expires in 1098 s
DUNDi lookup completed in 560 ms
How do I then take that information and cycle through it? According to the
help for the DundiLookup application:
asterisk*CLI show application DUNDiLookup
asterisk*CLI
-= Info about application 'DUNDiLookup' =-
[Synopsis]:
Look up a number with DUNDi
[Description]:
DUNDiLookup(number[|context[|options]])
Looks up a given number in the global context specified or in
the reserved 'e164' context if not specified. Returns -1 if the channel
is hungup during the lookup or 0 otherwise. On completion, the variable
${DUNDTECH} and ${DUNDDEST} will contain the technology and destination
of the appropriate technology and destination to access the number. If no
answer was found, and the priority n + 101 exists, execution will continue
at that location.
But does that return everything? Or just a single result?
--
Vice President of N2Net, a New Age Consulting Service, Inc. Company
http://www.n2net.net Where everything clicks into place!
KP-216-121-ST
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Re: [Asterisk-Users] 1.0.8 Release Candidate
On Wed, 1 Jun 2005, Michael Stearne wrote: On 6/1/05, Russell Bryant [EMAIL PROTECTED] wrote: I am on IRC as drumkilla and also available by email if anyone has any questions or comments. Please test and report any issues on the Asterisk issue tracker, even if it is just a note saying that you have no problems at all! I will release 1.0.8 once I have had enough reports of people successfully running the latest code from the v1-0 branch of CVS. I just downloaded and compile the CVS-HEAD today (asterisk and asterisk-addons). They both compiled and seem to be stable on Fedora Core 3. I am having trouble however compiling add-ons on OS X. What does CVS-HEAD problems have to do with a 1.0.x release candidate? They are totally separate code trees.. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ 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 on Soekris
On Mon, 30 May 2005, Chris Mason (Lists) wrote: I thought I saw a Soekris embedded in the Digium booth photos, can you run Asterisk on one of these? How? I'd be interested in it for a back pbx, given the reliability. In fact, might want to move my home pbx to this also. http://www.kriscompanies.com/modules.php?name=Contentpa=showpagepid=3 Kristian has made it incredibly easy to get Asterisk up and running on a Soekris board. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ 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] Pictures of the Digium booth at ISPCon 2005
On Sun, 29 May 2005, Kristian Kielhofner wrote: Dean Collins wrote: Great booth guys, looks really interesting - can you cull out some of the more lousy photos though. Anything else you've seen at the event that's looks interesting? Dean, I could cut out some of the more lousy photos, but I would rather leave them in case someone finds them of value. You don't have to look at the out of focus/repeat shots. :) As far as other things at ISPCon, I can't really answer that because I didn't make it very far from the Digium booth! Aint that the truth.. We were, for the most part, very very busy. In fact, while breaking down, it was commented to me by several of the surrounding booths that the Digium booth Always seemed busy. From the time that I spent there, I can say that it was. I had to have a constant supply of Cough Drops just to keep my throat in shape. I did take some time on Tuesday and walk the floor, but there wasn't anything interesting really. A lot of lemonade stand, sign here, become a VoIP provider in a box setups. Most of them won't be around long enough. We got lots of questions about the E911 issue, to which I replied Patience. We have 120 days to figure it out and there are already several companies working on it. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ 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-biz] Asterisk at ISPcon
On Thu, 19 May 2005, Tim Simms wrote: Have a link? http://www.ispcon.com If you are interested in manning the booth, talk to Rick Segrest. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ 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 at ISPcon
On Thu, 19 May 2005, John Todd wrote: [DELETED] Err well, that last presentation doesn't have anything to do with Brian or Greg at the moment, but of course SIP CPE is fairly relevant to service providers or PBX vendors/consultants working with Asterisk. Jon, Glad to see that your going to be there this year. It will be nice to finally meet you. Jon Price had asked me to see if I could find someone to do the Barbie-Phone presentation and I briefly thought about doing it myself, but I declined as I've got enough on my plate for next week. However, I'm glad to see that someone as knowledgeable and well-respected as yourself will be handling the topic. I'm making sure to block out the time on my schedule to see your presentation. As Rick pointed out, the Digium booth could use some volunteers to answer questions about Asterisk. If you are in the Baltimore area, and are interested in promoting Asterisk at ISPcon, definitely get in touch with Rick. Aside from getting access to the ISPCon exhibition floor, it will be a good opportunity to meet vendor and client prospects. See you in Baltimore! -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ 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] Connecting 20+ asterisk servers together
On Mon, 9 May 2005, Vikram Rangnekar wrote: [Deleted] Can dundi or the switch statement help me get out of this mess ? Dundi will make this trivial. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ 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] Connecting 20+ asterisk servers together
On Mon, 9 May 2005, David Choo wrote: Actually, this is whats facing me right now. I think Dundi will resolve the problem, but I've never really placed it to the test. Anyone tested Dundi? Best Regards, I run it in production on CVS-Stable and it works without a problem. Our usage of it is to have multiple gateway servers that can dynamically respond to outages without human intervention. I.E. a customer's dial-plan will be setup to return multiple routes and it will try them in order of preference. If one fails, the next available route will be tried and so on. Here is a snippet of some information from one of the working drafts that N2Net is putting together on using Dundi as the core of a Fault-Tolerant network. It is incomplete, and has some stuff that is a bit wrong, but it is a starting point. Much of the really intelligent sounding stuff is ripped right from the Dundi RFC... The two gateways will be reponsible for publishing weighted Dundi routes to their PSTN (PRI) and VoIP Termination carriers. They will also be responsible for recieving inbound PSTN and VoIP traffic and routing the calls back to the customer. DUNDi is designed to facilitate the sharing of resources that can be used to terminate phone numbers by using a peer to peer system, requiring no centralized controlling authority, no single point of failure, and no enforced heirarchy. In this way, systems sharing a dialplan across an enterprise or across the globe can be assembled in an ad-hoc manor, while retaining confidence in the accuracy of the routes that are supplied and the security of both the queries and the answers within the trust group. Route Weights - A Weight is a value indicating the relative cost or indirection of a particular published number. A lower weight represents a route which is more direct to the intended location. The lowest weight value is 0. The generally accepted route weights for the Dundi Peering network are 0,100 and 400. The HAC will publish route weights of 0-99 to internal, private peers, and 100-400 for external Dundi-E.164 and Dundi-Test networks. This will allow routes to be prioritized, such that PSTN will always be used where possible but VoIP routes will be available if PSTN access is not. This will also allow the HAC to preferentially route International traffic using a Least Cost Routing method, preferring VoIP routes, or a particular PRI span if neccessary. A weight of 0-24 indicates a PSTN route to a dedicated carrier. A weight of 25-49 indicates a failover PSTN route to a carrier. A weight of 50-74 indicates a Primary VoIP route. A weight of 75-99 indicates a Seconddary VoIP route. A weight of 100+ indicates a least-preferred route to a private or public Dundi connected network, such as Dundi-E.164, Dundi-Test or FWD-out. For what it's worth.. ___ 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] Zapata.Conf Sanity Check
Hello, Haven't had a chance to test this configuration yet, so I can't really answer my own question, but wanted to get a couple of eyeballs to look at it and verify that I'm going to get the behavior that I expect out of it. Basically, I have a TE405P w/ 3 PRI to the Telco and one going to a legacy PBX, all speaking NI-2 (National ISDN 2). I need to use pri_cpe w/ the Telco and pri_net for the PBX, and I want them in different groups. I hacked the following up last night, while falling asleep, and I'm pretty sure it will do what I want, but I'd like a second opinion. ; Zapata telephony interface ; ; Configuration file [channels] ; Inbound PRI from Telco context=inbound switchtype=national signalling=pri_cpe group=1 channel = 1-23,25-47,49-71 ; Outbound PRI to PBX context=pbx switchtype = national signalling = pri_net group = 2 channel = 73-95 Thanks.. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ 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] TDM users: modified zttest.c for testing
P3 1Ghz under Tao Linux 1.0 (2.4 Kenrnel) cvs-stable w/ X101P --- Results after 66 passes --- Best: 1.024461 -- Worst: 1.024420 -- Average: 1.024447 And on our new gateway box... P4 3.0 Ghz under Tao Linux 4.0 (2.6 Kernel) cvs-stable w/ TE405P --- Results after 106 passes --- Best: 1.023967 -- Worst: 1.023953 -- Average: 1.023960 -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ 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 timing problems: Fax Voice
On Wed, 4 May 2005, Andrew Kohlsmith wrote: When the span was 0, I NEVER got that message. I haven't heard any complaints from the other office mates that use the PRI for voice, but the error just bothers me. What is the real difference between 0 and 1 on the span timing? all that the clock span means is what span * synchronizes to. clock of 0 means do not try to synchronize to the clock on this span 1 means this span is my primary clock sync source 2 means that if the span with '1' is down, use this one 3 means if the spans with 1 and 2 are down, use this one This is very confusing. Most clocking language in the industry refers to either Internal or Recovered clocking. Basically, the span can either use it's own clock, or attempt to pull it from the line. Furthermore, zttool appears to be broken and no matter what the setting of the span (0,1,2,3) it allways reports Internally clocked. I actually opened a bug report on this today, after talking to Digium technical support, who was very apathetic about it. Basically, the Tech said It's a known issue w/ zttool and that if I wanted to report it to the bug tracker, perhaps someone in the community would fix it. So I opened http://bugs.digium.com/view.php?id=4186 -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ 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 Images
On Thu, 5 May 2005, Charlie Watts wrote: Manjit Riat wrote: Out of curiosity what's the reason? Why would they not sell phones to asterisk users? Do they not trust asterisk or their phones to work with each other? My guess: They don't want to compete with the folks that OEM Polycom hardware. Lots of commercial phone system vendors just re-brand Polycom phones, and Polycom doesn't want to hurt their relationship with those businesses. I bet that at some point one of the Asterisk-using Polycom vendors gets the momentum to get a better Polycom-Asterisk user relationship going. The trouble is that most of the Asterisk-using hardware vendors are hardware agnostic, and don't want sign anything that says We'll only sell Polycom equipment. This is not the case. I had a long conversation with Polycom's VP of Product management at VON this year, and asked why they didn't want to support Asterisk. Their response is that Asterisk has not yet completed the Self-Certification program w/ Polycom, and that this should be a relatively easy thing to do. They recognize that they are missing opportunites w/ Asterisk, but they want to make sure that their phones work correctly with it, and that Digium will take some responsibility for certifying their code w/ Polycom. Last I heard, Polycom was supposed to be shipping a couple of cases of phones to Digium for certification w/ Asterisk Business Edition. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ 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 on Fedora Core 2 startup script
On Thu, 5 May 2005, Colin Anderson wrote: I like easy questions! Add to /etc/rc.d: modprobe wctdm ztcfg -vv su yourasteriskuser /usr/sbin/safe_asterisk where yourasteriskuser is the user that you normally run Asterisk under. Omit the su yourasteriskuser if you are running Asterisk as root. safe_asterisk is a watchdog script that restarts Asterisk if it craps out. The Ampersand runs it as a background process, returning control to the tty and allowing you to log into the console. Or, you could just use the pre-existing scripts: /usr/src/zaptel/zaptel.init and /usr/src/asterisk/contrib/init.d/rc.redhat.asterisk They work. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ 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: RE : [Asterisk-Users] Dell PowerEdge SC1425 w/ TE405P?
On Sun, 1 May 2005, David John Walsh wrote: what sort of level of PC is required for 300 concurrent calls? Doing what? Ulaw? That could probably be done on a single P4 2.8 Ghz. If you want to transcode from Ulaw to something else, you need to scale the hardware appropriately. Every case is different. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ 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] TE405P w/ Intel SE7210TP1_E Motherboard
On Sat, 29 Jan 2005, Greg Boehnlein wrote: Hello, I'm looking at building a couple new PRI Gateway boxes using TE405P cards, and was wondering if anyone has had any experiences (good or bad) with the Intel SE7210TP1_E motherboards from Intel. General Technics builds some really nice (and cost effective) 1U servers based on the board: Server: http://www.gtweb.net/gt637.html Specs: http://www.gtweb.net/support/pdf/SE7210TP1-E_Product_Brief.pdf Comments? Per my other costs, I've settled on the following box from General Technics for my PRI gateway boxes. http://www.gtweb.net/gt637.html I had a pretty detailed conversation with Chris from GT about the board and how it is laid out, and it appears that the unit has several PCI busses in it and they are separated nicely; The only stuff on the 5V PCI slot is a 10/100 NIC + the Video. Since there is a Gig-E that shares the same bus w/ the SATA drive I can just use that and keep the bus free for the TE405. On most of my boxes, I disable the Video and use serial console. I'll keep people up to speed on how this works out. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ 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] Best of the best of IP Phones
On Fri, 22 Apr 2005, Chris Coulthurst wrote: Is there a specific SIP or IAX phone that truly shines above the rest where it comes to 'happy' compatibility with Asterisk? I guess I'm talking about feature sets, like early-dial, off hook call announcing, conferencing, echo suppression, etc etc.. I, like many others, bought a Budgetone for early testing, and need some new eye candy! OHCA is a feature that I'd love to integrate, and it seems that not too many phones support it out of the box. I've got Polycom and Cisco phones and quite frankly, I prefer the Polycom Soundpoint IP-500 and 600 to my Cisco's now. All things being equal between the phones, the following are why I prefer the Polycoms: 1. Better speakerphone than the Cisco 7960s. Despite the fact that Cisco licensed Polycom's SpeakerPhone technology, the SoundPoint IP 500 and 600 just sound and work better. 2. Lower price point: $185 for a NEW SoundPoint IP 500 is better than the $225 I see for used 7960s. 3. FTP based provisioning. TFTP is fine, but doesn't work very well through some NAT implementations. The PolyCom's can be centrally provisioned from any FTP server, and NAT doesn't seem to be a problem for it. 4. More intuitive User Interface. My clients require less training and are up and running quicker on the Polycoms. These are my opinions I love BOTH phones, and you can't go wrong with either choice, but for my needs the Polcom's work a lot better. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ 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] Dell PowerEdge SC1425 w/ TE405P?
Hello, I've been asked to build a couple of Gateway servers for a client w/ TE405P hardware, and have been looking around at various 1U options. I've been looking at SuperMicro and Tyan barbones boxes as possible platforms, but then was directed to Dell's SC1425 by a friend. Short story, is that you can purchase a 2x3.0Ghz/1GigDDR400/1xSATA box in a 1U form factor for $1,498.00. This seems almost too good to be true, so I'm asking if anyone has had any experience with this box? I'm not up on my PCI terminology, but as I understand it, the TE405P can only be used in a 32 bit 33Mhz slot at 5.0 Volts. This SC1425 lists a 1x 64-bit/1xxMHz PCI-X slot under it's expandability information. I'd venture to guess this is probably NOT going to work with a TE405P. That being said, if it works, great. If not, what 1U boxes are people using IN PRODUCTION w/ TE405P cards? -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ 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] Dell PowerEdge SC1425 w/ TE405P?
On Fri, 22 Apr 2005, William Boehlke wrote: SC1425 is great value but note it does not have high availablility configurations. In our opinion, telephony requires dual NICs, dual power supplies and RAID 1 to have any hope of achieving five nines. William Boehlke What box would you reccomend for this? -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ 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] ZyXEL Router Terrible Voice Quality
On Sun, 17 Apr 2005, Dave Weis wrote: On Sun, 17 Apr 2005, Greg Boehnlein wrote: On Thu, 14 Apr 2005, Rod Bacon wrote: I have been frustrated by a variety of zyxel issues/products and have found the best solution for all of them lies in a cylindrical receptacle that sits beside my desk... I've had pretty good luck with the Zoom X5V Voice Modem so far. It has a built in DSL modem, and a single FXS port. Decent little router, now that the latest firmware is out, but tcp and udp timeouts through NAT seem to be set a little low, so I lose SSH sessions. I bought a dozen and have had bad luck with them. I couldn't keep an ssh session for more than 15 seconds. Trying to update firmware turned two of them into paperweights. I couldn't get the FXS to ever register. Other than that, it looked like a good idea. Is the NAT timeout configurable now? Not sure. I'm unable to open a ticket with Zoom technical support on the issue. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ 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] SIP Phone Compatability
On Wed, 20 Apr 2005, Daniel Salama wrote: Every once in a while I read messages about people having problems with certain models of SIP phones, some of them being well known models. I'm interested in purchasing new SIP phones for my office and wanted to know which brand/model is most stable with Asterisk, which allows most office features. These features should include: multiple-line appearances (at least 3), call conference, blind and non-blind transfer, memory buttons or speed dials, voice message light indicator, speaker phone, mute, redial, caller-id display. Anything on top of these features is a plus but not really a requirement. I've had very good luck with the Polycom SoundPoint IP 500 phones. They are rock solid, have great speakerphones, all of the features you list and more... Work just great with Asterisk. My clients love them, and at $185 / phone it is hard to beat. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ 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] ZyXEL Router Terrible Voice Quality
On Wed, 20 Apr 2005, snacktime wrote: Not sure. I'm unable to open a ticket with Zoom technical support on the issue. I had an interesting experience with the zoom. Their SIP implementation doesnt' expect to see SIP traffic on the internal lan, and running * on the internal lan would lock up the modem. Completely repeatable, just start up * and the modem locks up hard, lights go solid, etc.. Tech support didn't seem to think that this was an issue, and told me that they only supported voip using their own service. Tried telling them that regardless of what they support, if traffic on the lan locks up the modem, it's a bug. I finally just returned the unit and got my money back. I don't have that problem, and I run SIP asterisk on the inside of my LAN. I have Sipura, Cisco and Grandstream equipment on the same lan segment. However, I use switches so the traffic doesn't hit the Ethernet port on the router. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ 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] SIP Phone Compatability
On Wed, 20 Apr 2005, Daniel Dziubanski wrote: Greg, Are you using AMP? No. And If so, you have any tips and tricks on how to easily manage phones via a amp plugin/fix? No. The Polycom phones will provision themselves via FTP using XML files. It probably wouldn't be hard to write. In fact, SipX is supposed to have a decent Polycom provisioning engine that could be used as a basis. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ 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] ZyXEL Router Terrible Voice Quality
On Wed, 20 Apr 2005, Aza wrote: [DELETED] I seem to have found a solution to the problem I had with a ZyXEL Prestige 2000 series ATA. It looks like these things just can't cope with NAT no matter what you do with Asterisk, STUN servers or SIP Proxies. Specifically the voice quality of the person speaking through the ZyXEL is more chopped up then a diced onion if you're going through NAT. Yeah.. SIP doesn't deal with NAT very well. What a retarded design limitation. If you need to punch through NAT, you can do it by opening ports (as you discovered), using STUN (if your ATA supports it) or by simply avoiding NAT altoghter. Me? I'd get a Digium IAXY and just be done with it. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ 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] ZyXEL Router Terrible Voice Quality
On Thu, 14 Apr 2005, Rod Bacon wrote: I have been frustrated by a variety of zyxel issues/products and have found the best solution for all of them lies in a cylindrical receptacle that sits beside my desk... I've had pretty good luck with the Zoom X5V Voice Modem so far. It has a built in DSL modem, and a single FXS port. Decent little router, now that the latest firmware is out, but tcp and udp timeouts through NAT seem to be set a little low, so I lose SSH sessions. G729 and G711-Ulaw sound great through it, and it supports no-power pass-through to the analog line. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ 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] S100I - competitive price?
On Wed, 13 Apr 2005, Kevin P. Fleming wrote: [ DELETED] Realistically, how cheaply can you put together a box with a T-1 card and a channel bank with 24FXS ports (even disregarding G.729 transcoding, which would add to the cost)? $700? $800? more? I can't say for sure, but if you wanted to use a decent speed machine, I'd expect that the PC+TE110P+channel bank solution would cost at least $900, and that's using a bargain-basement PC and a used channel bank. Or if you didn't need FXS ports, we could take an old PM-3 w/ 50 DSP's in it, and a pair of the Dallas Framers to build a T1 channel bank via Ethernet. ;) Seriously, the PM3 would make an awesome platform for an Ethernet to T1 or PRI channel bank. The core is an AMD 5x86 processor, it can take 16 megs of ram, it has the entire TDM architecture already built into it, and the old Modem cards have Lucent DSPs that could easily implement transcoding and G.168 echo cancellation. Best of all, the boxes are dirt cheap, and.. I know for a fact that the ComOS has been ported to GCC on Linux and can be built. I've talked with the engineer that wrote the drivers for Livingston, and he's been thinking of writing an IAX2 stack for it. Digium.. you listening? ;) -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ 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 phones
On Mon, 11 Apr 2005, Noah Miller wrote: This this may sound ridiculous, but we've had problems with this when the users did not plug the handset cord in completely. 8 out of our 12 employees made the mistake, as the plug on the IPX00's appears to be all the way in when it is actually not. Not ridiculous at all. We had the same problem. In fact, the cord will click into place when it's not really all the way in. I had the same problem.. :) -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ 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 on WRT54GS
On Mon, 4 Apr 2005, Juergen K. Zick wrote: Hi Dalon, I have it running including VMAIL, 3 SIP and one IAX2 account AND OPENPVN ... Incoming and outgoing connections are OK, both in nat'ed local 192.168.x.x and external real IP adresses ... --Juergen Juergen, Did you build this yourself? Or is it one of the other packages out there? I've got a couple of OpenWRT54GS units that I'd like to try this on. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ 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] *-1.0.7 DTFM = Not working
On Wed, 23 Mar 2005, Joseph wrote: My DTFM is not working in current CVS-stable *-1.0.6 and *-1.0.7 but it works in version 1.0.5 (was working with 1.0.3). I'm using SPA-3000 and dtmfmode=inband I noticed the exact same behavior w/ my upgrade to 1.0.7 using Polycom SPIP phones w/ dtmf=inband support. Has anyone opened up a bug-tracker report on this? I had suspected that a change in chan_sip was the culprit, but I have not had the time to back-step through the CVS versions to find which specific patch might cause the behavior change. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ 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 DTMF
On Thu, 24 Mar 2005, David Gomillion wrote: Problem: Polycom SoundPoint IP phones (running SIP) ceases to send DTMF that Asterisk can detect and use. It worked in 1.0.5, but has not worked since. This has been verified on SoundPoint IP 300's and SoundPoint IP 600's. Workaround: It used to be that for DTMF to work, I had to set the mode in sip.conf to inband. Without making any configuration changes on the phones, I changed the DTMF mode to rfc2833. The DTMF is recognized. No reboot to the phone is necessary, and remember that you can reload the sip configuration with a reload in Asterisk, meaning your PBX doesn't have to be restarted either. Discussion: This is probably not the right way to fix this, as Polycom's configurations, by default, will encode DTMF in the active RTP stream. There may have been a change in the sip channel's code that is causing this. Others on the list have indicated that they worked around the problem by reverting the version of the sip app to an older version. As the new code usually fixes other problems, the solution of reverting seemed to be counter-productive, so I tried other DTMF signalling modes. Thankfully, the stock Polycom configs will work with Asterisk's sip.conf rfc2833 DTMF mode, at least as of CVS-v1-0-03/23/05-21:40:48. When I get more time, or if someone else has the time, an examination of what changed to cause this could enable us to fix the heart of the matter. Other users on the Asterisk list (see thread *-1.0.7 DTFM = Not working from 03/23/2005) have reported other UAs not working. Therefore, there may be a bigger problem with the fundamental issue at hand: when do we change DTMF in channels, to ensure compliance with standards, as well as compatibility with older UAs. Hope this helps someone. Sincerely, David Gomillion David, I noticed this in testing the 1.0.7 Release Candidate w/ my Polycom phones. I posted the following in http://bugs.digium.com/bug_view_page.php?bug_id=0003746 03-16-05 16:05 Alright. I updated my Development and Home PBXs to 1.0.7 w/ Slepp's Dundi 1.0.2_diff-4 patch. Both are running solid. One issue that I noticed is that my Polycom SP IP phones had to be changed to use RFC2833 signalling instead of the Inband signalling I had been using earlier. I simply modified the sip.conf to have dtmfmode=rfc2833. This proved to be a slight gotcha for a couple of clients when I updated THEIR boxes. This could be just a fluke, and I may be an idiot for having used inband DTMF in the first place, but it is something to be concious of. Can anyone pinpoint a specific SIP patch that may have been applied where this may have been affected? How should this be handled? Should I add it as a new bug-note? Or should we just chalk it up and slap a notice in the release notes? Should inband signaling be broken in 1.0.7 for Polycom phones? Or should it work? I guess this is the question that needs to be asked.. is it normal behaviour or a bug? I'm not sure if this is related, but I also enabled MMX math routines for Zaptel. I never got any followup on it before 1.0.7 was dropped. I'm going to open up a new bug report on this in Mantis and see what we can get as far as comments. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ 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] Experience with this radius?
On Thu, 24 Mar 2005, Matt wrote: Well this is true.. how reliable is that though? I know even with dialup we SOMETIMES will miss a call accounting packet because they are sent UDP What does it matter? You can't get the information out of just a Start packet. All you'll know is that a call was initiated, but you have no idea how long the call was, or what the rate/billing should be. On Thu, 24 Mar 2005 00:17:31 -0500 (EST), Greg Boehnlein [EMAIL PROTECTED] wrote: On Tue, 22 Mar 2005, Matt wrote: Hi, The reason I didn't look into that is it says: You CAN NOT: Accounting: * generate Start or Alive records, which is doable easily for connected calls, but If I can't generate a start record... what good is it for CDR recording? You don't care about Start records for billing purposes. You care about Stop records. The Stop record will contain all the call stastics such as call time etc.. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ 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] Experience with this radius?
On Tue, 22 Mar 2005, Matt wrote: Hi, The reason I didn't look into that is it says: You CAN NOT: Accounting: * generate Start or Alive records, which is doable easily for connected calls, but If I can't generate a start record... what good is it for CDR recording? You don't care about Start records for billing purposes. You care about Stop records. The Stop record will contain all the call stastics such as call time etc.. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ 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-addons/OS X
On Sun, 20 Mar 2005, Rob Gillan wrote: Hi, Having all sorts of troubles getting mysql cdr support under OS X. Mysql, DBI and DBD all installed and running ok, privileges all set correctly (I think). Latest asterisk-addons checked out of cvs. Keep getting error on make install (implies gcc doesn't support -shared linking, which is wierd): Anyone got mysql cdr support running on OS X? Rob, Is this with Asterisk-Addons 1.0.7? If so, there was a problem w/ the .tgz file that was dropped and Drumkilla has corrected and re-posted the correct one. You'll want to re-download the file again and give it another try. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ 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 vs. Cisco IP Phones
On Thu, 17 Mar 2005, Christopher Jacob wrote: [Deleted] So, the moral of this story While Polycom may not offer configuration type support for asterisk, they stand by their hardware. With Cisco you have to shop around to find a decent deal, and who know how you're going to get support. I don't mind using this list / the wiki / google / etc. for configuration type questions. After all I don't expect Super Micro to help me get FC running on one of their motherboards... Disclaimer: I don't represent Polycom, Cisco or Digium. I represent the community ;) Two comments: 1. Based on VON and the contact I've had w/ Polycom's VP of Product Marketing there, and their ensuing contact w/ Digium, you can be assured that Polycom understands that their stance on the Asterisk community has to change and that they are taking steps to correct their previous behavior. I'm sure that there will be more official announcements about this as things develop, but Polycom is very very very interested in cutting through the corporate red-tape and getting Asterisk certified as an interoperable platform. They have a self certification program that vendors can use to flesh out issues and get on the approved list. Their interest is in ensuring that the products work well together and that their engineers have a clear line of communication with the vendors into any technical difficulties. Previous to VON, the right people were not in the mix. Now, that has been solved and there should be some momentum building. 2. Cisco supports phones that are under maintenance contract. I've never had an issue w/ cisco support on any product that had the requisite Smartnet on it. If your going to use Cisco, make sure that you get phones from a vendor that can provide the appropriate Smartnet. For three years, 8x5 Next Business Day replacement, it cost me about $21 / phone. 7960's may be more difficult to get this on, but 7960G's shouldn't be a problem. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ 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] Realtime Problem = Segmentation faults Seems to be asterisk
On Fri, 18 Mar 2005, Jose R. Ortiz Ubarri wrote: Jose R. Ortiz Ubarri wrote: Hi: I had asterisk with RealTime database working perfectly in a RH 9.0 machine. I used the sip cache so I even had MWI working. The problem is that I decided to move to Fedora Core 3. I installed the lastets cvs version of asterisk and the RealTime addon from asterisk-addons. I at first had the problems with the kernel and the zaptel driver but all that was solved with the configuration from the Asterisk Wiki. Then when I moved my configuration to the new asterisk server and configured the RealTime addon it falls in a Segmentation fault. If I do not load the res_config_mysql.so (edited at modules.conf) then asterisks runs without any problem. But if I load the module from boot or from the asterisk command load res_config_mysql.so then I get the Segmentation fault again. I'm not sure what the problem is. Is it a Fedora Core 3 problem, or an Asterisk latest version problem? I don't think it is a configuration problem because I just used the same configuration I had before. The only diferences may be the OS and probably the asterisk version that is only one week newer than the one I was running in the old asterisk server, so I'm probably even running the same version of asterisk in both machines. Any advise? Someone else have a similar configuration working with Fedora Core 3? Thanks in advance, Debugging the code and as you can see in the backtrace the problem is that it is receiving a Null variable (name) and then making the comparison. Is it an asterisk bug? What asterisk should do if the variable name received is NULL? Has this been entered into the Bug Tracker? If so, what bug number was it assigned? -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ 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 1.0.7 Released
On Sat, 19 Mar 2005, Ronald Wiplinger wrote: Russell Bryant wrote: Hello everyone, Version 1.0.7 of Asterisk, Zaptel, libpri, and Asterisk-addons has now been released. Libpri and -addons have not changed, but have been updated anyway to keep the version numbers consistent. All of the tarballs are available on the ftp site. ftp://ftp.asterisk.org/pub/asterisk/ I have posted the ChangeLogs for easy viewing at the following address. http://dev.asteriskdocs.org/ Russell, I like the info there. It would be also interesting WHEN the release date was, It shows, when the development stops, As a side note, the Changelog should also reflect what Bug Tracker IDs where fixed. If you look at a RedHat Enterprise Linux changelogs, they give a good example of what I consider excellent information. Something like this: * Mon Nov 15 2004 Chris Feist [EMAIL PROTECTED] 6.0.2-0 - Build for new kernel. - Fixed circular dependency in GFS fsck (rbz137256). - gulm_tool prints resources/services registered with lock_gulmd (rbz136220). - init.d/lock_gulmd will not start if quorum is not established after a specified time (rbz135732). - init.d/lock_gulmd will not stop if GFS is mounted (rbz135730). - pool init.d scripts no longer hang on startup until console input is provided (rbz137382). - pool_assemble now contains the override option (-O) (rbz137381). - Support for perl-Crypt-SSLeay added to fence_ilo (rbz137037). - Support added for configurable SSL port locations in fence_ilo (rbz137035). - Panics now print out the whole panic string as part of the panic. Before, the panic message itself didn't contain any information. The useful stuff was printed with printk()s before the panic and might not have been seen in a setup without serial consoles. - A fix for BZ #135684. Fixed a panic caused by incorrect locking of the list of recently used resource groups. Reported and patched by [EMAIL PROTECTED] * Tue Nov 09 2004 Chris Feist [EMAIL PROTECTED] 6.0.0-17 - Make /etc/sysconfig/gfs a config file in the .spec file. - Added fence_bladecenter to the file list (rbz136585). -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ 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: SV: [Asterisk-Users] IPSwitchBoard BETA
On Sat, 19 Mar 2005, Paul Fielding wrote: Make this another vote for Zap and IAX2 monitoring :) Paul Seconded! As cool as FOP is, this looks pretty awesome! ___ 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 1.0.7 -addons doesn't compile
On Sat, 19 Mar 2005, Scott Gruby wrote: On Mar 18, 2005, at 10:47 PM, Russell Bryant wrote: Hello everyone, Version 1.0.7 of Asterisk, Zaptel, libpri, and Asterisk-addons has now been released. Libpri and -addons have not changed, but have been updated anyway to keep the version numbers consistent. All of the tarballs are available on the ftp site. Asterisk-addons doesn't compile. Did you make clean first? I compiled fine.. granted, I'm not using the tarball, and instead using the 1.0 CVS, but it should still work.. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ 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] Pictures from the Asterisk Pavilion at Spring VON 2005
On Thu, 10 Mar 2005, Kevin P. Fleming wrote: http://host-a.starnetworks.us/Members/kpfleming/spring_von/photoalbum_view Enjoy! Anyone have pictures from the Heart show? :( My camera phone just wigged out. I thought I had like 60 pictures right from the stage, but apparently it didn't save anything except for one. If so, please mail me links.. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ 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] Pictures from the Asterisk Pavilion at Spring VON 2005
On Thu, 10 Mar 2005, Kevin P. Fleming wrote: http://host-a.starnetworks.us/Members/kpfleming/spring_von/photoalbum_view If you look closely, you'll see me at the booth doing some troubleshooting for Digium during one of my session breaks. We actually setup an IAX2 connection from the main server to the N2Net Gateway server and made several test calls to flesh out a couple of call completion anomalies. VON was a total blast. Can't wait for Boston! -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ 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: Polycom IP 300/500 Conferencing Behavior
On Fri, 21 Jan 2005, Greg Boehnlein wrote: Hello, I've got a mixture of SPIP 300 and 500 phones in production for various clients. I've got the XML settings configured for local conferencing, but I'm not seeing the expected behavior from the phone when I attempt to conference two calls together. According to the manual, while talking to the first party, you simply hit Conference, dial the second party and then Conference to join them. This is supposed to put the first party on Hold until you bridge them together with the second press of the Conference button. That is all fine and well, but it doesn't quite work the way that the manual describes. Instead of joining the two calls together when the Conference key is pressed for the second time, the first party is taken off hold and hears dead silence. The only way to correctly join the parties is to hit the Hold and then Resume soft key, at which point all three parties can talk to each other. As an illustration Conf - Dial - Conf doesn't work. However, Conf - Dial - Conf - Hold - Resume DOES work. I'm running 1.3.4 firmware on all the phones, and I can't for the life of me figure out what is causing this problem. It is very likely some misconfiguration in the XML files, but I can't find it. Anyone have any suggestions? Hello, I just thought I would follow-up on this post and mention that somwhere between Jan 21st and today, the Conferencing issue that I described below is no longer an issue. Normally, I wouldn't include the entire quoted context of the message for bandwidth reasons, but in this case, since the topic is nearly 2 months old, I figured it would be helpful to keep things consistent and on-thread. I'm not sure what may have been fixed. I.E. I don't know if it was a patch to chan_sip in stable or what. All I can tell you is that I haven't made a single change to either sip.conf or my XML config files, since the original posting in January. However, I have updated asterisk several times from the 1.0 branch. I'm happy that this is fixed, but I am going to do a little more reasearch to see if I can get it to fail again by backdating chan_sip and incrementing it forward. This should be interesting. :) -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ 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
