Re: [asterisk-users] Investigating international calls fraud
The 25000$ @6.25/min means 4000 minutes of calls (or 66H) Not sure in how many days this has accumulated but i seriously dought this is made from a human accessing the phone. The fact that you get the calls at certain times might have to do with the timezone the calls are going If you phone has an API (most have) and allows for calls to be made, or the web interface allows calls to be placed from there, and there is no password or the default credentials then this is how the calls are made. Check the call velocity (number of calls per minute) from that phone and if you see multiple calls at the same time frame then its probably not a (single) human doing it but some dialler. We start seen this way more often than before. i.e using the phone as an attack surface instead of breaking into to pbx. Also there seem to be a lot of javascript cross-site” scripting attacks on the loose that target voip networks from the inside. I.e using the browser to execute attacks from inside of a secure/firewalled network. Stelios Koroneos Jabber : stel...@soldecom.com On Jan 29, 2015, at 1:19 AM, Steven McCann steven.r.mcc...@gmail.com wrote: Hmm the calls are made during the day (and sometimes very early in the morning). Right now it looks like someone actually made these calls. If that is the case it's somewhat comforting to know the system wasn't compromised. However, the $25,000 phone bill still remains. Yikes. $6.25 per minute to Cambodia seems quite steep to me. On Wed, Jan 28, 2015 at 6:07 PM, Duncan Turnbull dun...@e-simple.co.nz wrote: On 29 Jan 2015, at 11:07, Administrator TOOTAI wrote: Le 28/01/2015 22:03, Steven McCann a écrit : Hello, Hi I'm investigating a situation where there was a hundreds of minutes of calls from an internal SIP extension to an 855 number in Cambodia, resulting in a crazy ($25,000+) bill from the phone company. I'm investigating, but can anyone provide some feedback on what's happened here? I'm investigating how this happened as well as what types of arrangements can be made with the phone company (CenturyLink in Texas). Are you sure the calls weren't actually made internally? Can you see anything to suggest the ip or mac address of the phone changed? Because for someone to take advantage of the calls (assuming they don't get cash out of ringing Cambodia) they needed to proxy through to that phone line, which maybe required them leaving some sort of device on the network. Otherwise I am guessing they got onto your PBX somehow. As suggested logs are important, including DHCP, syslog to see if anything unusual happened. Did the calls run all day or just at night when no one was around? Was there more than one call up at a time? (how many calls does the Mitel phone support?) How long were the calls? Were they varying lengths (more human like) and did they just redial as soon as they were dropped? Or were they automated to trigger as much cost as possible e.g. if the 1st minute is the most expensive then you get a lot of short calls. Good luck Some details: * PBX is located in Texas * Phone carrier is CenturyLink * FreePBX distro running asterisk 1.8.14 * source SIP extension is Mitel 5212, firmware 08.00.00.04, default admin password (argh!). Phone is used by many different people. More PBX setting details: * inbound SIP traffic is not allowed through the firewall * internal network is not accessed by many * FreePBX web interface *Questions I have at this moment:* 1) how were the calls placed? Was the Mitel SIP phone hacked somehow? Asterisk PBX? Check your logs. In the full log with verbosity 3 you can follow how calls were treated. Also the CDR should give you informations like the extension(s) who placed those calls [...] -- Daniel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com
[asterisk-users] Broadsoft - Asterisk interop
Greetings to all. I am not sure of this is a user question or a business so apologies in advance if it should be asked in the business list. A client of mine has a UK branch that is served by a provider that uses the Broadsoft solution. I want to create a sip trunk from a remote asterisk pbx to the client. The provider claims that Broadsoft has a strict protocol on what devices can connect on their network and asterisk is not listed Does anyone know if asterisk (1.8 to be more precise) has passed interoperability tests with Broadsoft ? There seems to be an old (circa 2008) agreement with Broadsoft and Digium for partnership but neither Digium or asterisk are mentioned in the Broadsoft partners page http://www.broadsoft.com/news/2008/digium-and-broadsoft-strengthen-partnership/ I see a device listed in the Partners page of Broadsoft (PIKA's WARP Plus) that as far as i know is using asterisk inside http://www.pikatechnologies.com/english/view.asp?x=1312 Anyone has any info or experience they would like to share with this ? Stelios smime.p7s Description: S/MIME cryptographic signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Tired of dropouts and garbled phone calls - where to go next?
On Mon, 2013-10-28 at 14:29 -0400, Eddie Mikell wrote: All, The users in our organization are well, quite frankly, sick of phone service that is being provided. The choppy phone calls, and drop outs are detrimental to our sales force. I've tried about everything I can think of. Moved the asterisk server from VM machine to dedicated machine More than enough bandwidth Setting 802.1p = 7 Set Dedicated voice traffic 35% of bandwidth. Not sure what option would be the best Put analog lines in the conference room to avoid the dropouts - leave the sip lines in place for day to day use Hire a consultant Ditch the system and buy a pre-packaged system - RingCentral or some such. There are no local asterisk professionals who can help, and we are a little leery of opening up our system to outside consultants. Anyone else face the above, and finally abandoned Asterisk for a commercial system? We have 167 users. I use Grandstream GXP 2100 on the desktop and Polycom ip6000 for the conference rooms. Suggestions welcome. A general rule of thump after several years with voip Voip turns out to be the canary in the coal-mine of a network. The smallest change or problem will manifest itself as a voip issue no matter what. Now to some practical advice Voip was designed for LAN's, The moment voip packets leave your lan and go into a WAN of any sort, it could be the source of frustration for many reasons. 1) Lots of routers/modems are not build to handle intense voip traffic. voip generates lots of small in size UPD packages. In most of the cases the routers/modems bridging your lan with the wan have no problem handling them BUT what i have found is that once you get over a threshold of traffic its possible the routers/modem can not cope with it, mainly because the large number of packets they have to process. In most enterprise grade routers the specs give you 2 numbers for the size of data the router can handle. total throughput and pps (packets per second). Usually total throughput is calculated using a packet size of around 1500bytes and it takes the router the same resources to process a 1500 bytes package as it does a 90bytes packet of a g729 call, as it just looks at the headers and not the payload.So yes your router can handle 60Mbits (of 1500byte frames) which is about 5000 packers per second but for voip that translates to less than 4Mbits of data (5000 packets of 90 bytes) I think you can get the picture 2) Because of 1) its possible that your ISP has issues, especially if its handling lots of voip traffic while its equipment is not optimized for that. 3) QOS and queing in general Whatever you do with QOS to get a better priority/quality, the dirty secret is, you can only control what YOU send, not what you receive. And even that is true till your modem/router. Once the packet is gone you have no control of how it will be handle by all intermediates till it reaches its destination. You have no idea if qos is honored by ALL hops and what kind of queuing they apply (if they do) to that port/service/qos mark That beeing said, its possible that you *might* have much better luck with sip and sip rtp than with iax rtp if your isp and all its interconnects bother to offer qos for rtp. Now for receiving it can be even harder if your isp does not provide correct priority queuing for the rtp stream, as latencies can build fast especially on busy hours (which happen to be the same hours people use their phones the most...) where people download stuff,emails etc. ping.icmp and all the other networking monitoring tools/protocols could be an indicator BUT its most probable that they will be handled by the isp and its interconnects at the higher qos priority The only way to see how rtp traffic is handled is to run rtp traffic. The only way around this is a dedicated circut MPLS or similar between the points of interest (i.e offices), with specific SLA which usually means much much higher costs. Finally my 2 cents for troubleshouting. Check the network first ! Find what triggers the problem. Is it something that happens all time regardless of traffic ? is it periodic ? (when bw goes over X percent, or at a specific time of day ?) Try different qos settings/priority queuing on the router -- Stelios S. Koroneos Phone US : (+1) 347-783-5467 Greece : (+30) 211-800-7655 ext 101 Skype : skoroneos PGP Key fingerprint = DC66 109A 6C3A 2D65 BA52 806E 6122 DAF4 32E7 076A smime.p7s Description: S/MIME cryptographic signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or
Re: [asterisk-users] Performance Asterisk large installation on Vmware/Xen
On Sat, 2013-05-18 at 15:01 -0300, Rafael dos Santos Saraiva wrote: Hi I would like the opinion of you and if anyone has a similar scenario. I have a project for installation of a Asterisk server in a client with about 400 extensions. My question is whether this scenario carry an Asterisk virtualized. Will be used only extensions and trunks sip sip, 1 queue with 2 agents, without call recording. It is best to use XEN or VMware? Which best version of Asterisk for this scenario? If there is no transcoding it could work assuming that this is not a call center where all extensions would pretty much be up and running all the time. i.e this is a setup for a large office installation. Assuming a recent XEON cpu and adequate RAM, main stumbling block is usually network performance so you need a machine with good ethernet chipset and linux driver. Also you will need to twick the dahdi timing source or strange things could start happening I've used KVM and XEN but presonaly i feel much more comfortable with KVM As for asterisk version i would recommend 1.8 which is LTS at this point. Stelios smime.p7s Description: S/MIME cryptographic signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cdr report
On Tue, 2013-04-23 at 17:35 +0300, aristidis tsitras wrote: Hi. i am running asterisk in a low powered machine (alix2d13 from pcengines) without any gui. the machine works fine to route all my calls for the office. the problem is the management of the CDRs. i can see the master.csv file, but it is not very friendly for the secretary of this office to manage the calls. is there a way to have a nice way to see the CDRs?Since the machine is very small on CPU, it has to be as low on CPU/RAM consumption as possible. any ideas? You can have the cdr_odbc or cdr_mysql module loaded and have the cdr in an external database Once in there you can get any report/format you want with minimum programming. The issue is that you need a machine running the database 24/7 Another option is to use the manager interface and an external client to collect the cdr events The manager interface can be setup to output only the cdr events and the resource requirements on the machine running asterisk are minimal. The downside is that you also need an external client running 24/7 to collect them. Stelios smime.p7s Description: S/MIME cryptographic signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk SMS()
On Tuesday 19 February 2013, Nicholas Johnson wrote: Thanks for the help. Right now I'm running asterisk on a raspberry pi using a phone number from flowroute. Is using a company like flowroute the same as connecting to the PSTN? Also i've tried to install smsq but I couldn't find any good documentation to get it setup properly. So no, I'm not using smsq. The bad news: You need a GSM modem to send SMS messages. The good news: It is not so. You can send SMS messages on POTS or ISDN lines See the voip-wiki about it In the US (and other parts of the world) there are SMS gateways the providers offer to reach their subscribers They are free and since you are using a raspberry which means no direct pstn interface might be a good approach with the help of some AGI/bash scripting -- Stelios S. Koroneos -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to connect a POTS robo alert dialer to asterisk with email notification
On Sat, 2013-02-02 at 18:36 -0500, David Smiley wrote: Hello! I'm totally new to the world of Asterisk, and so my apologies in advance if my question has been asked before or is in the manual because I'm too new to even know what to search for. I own a property far-away that isn't inhabited year-round. I want to come up with a low-temperature alert system so I can be notified that there is insufficient heat for whatever reason (e.g. failed boiler, or ... ?). There are some systems in the several hundreds of dollars price range that could either hook up to my WIFI to a monitoring service (sometimes with monthly fees), or a cell-phone based one that sends a text message. Then there are inexpensive ones for about $60 that can hook up to a plain old telephone jack and dial a number with an automated voice to alert the receiver of the problem. But I don't want to buy phone service to this place just for this device. So I'm wondering if I could buy an adapter of some sort with a phone port and ethernet port. An ATA? But then I'm sure it'd need to talk to some sort of VOIP service. Where I live year-round I have an underpowered build-your-own HTPC computer that stays on the internet all the time and occasionally I access it remotely. Perhaps I could install asterisk there. But then I have no idea. Ultimately I want to get notified somehow (e.g. email) that this phone dialer sent an alert. Maybe this is more trouble than its worth :-) ~ David -- Usually the problem with remote locations is the absence of (low cost) internet connections. If you have internet access to the property all year round, then get a usb temperature monitoring dongle (there are several prices around 15-20 $) and find/write a script to query it every X amount of minutes and send you a notification either by email or even sms using a (usually free) email to sms gateway Again if you have internet access you can put the monitor device that has PSTN output to an ATA and have the ata register to an asterisk at your home,so you won't need to pay monthly fees to a Voip provider. That would require to setup a permanent asterisk server at your main house, but you could do it with VM image and avoid setting up a new PC You then need to do some port forwarding in your router for ports 5060 (for the sip signaling) and 1-11000 for the RTP stream Assuming you don't have static ip's in either place you will also need to set up DynDNS so that the phone knows where to find asterisk There are several other options but which one is best depends on budget, knowledge for hooking up the stuff and time it takes to do it :) Stelios -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTP timeout if the asterisk box behind NAT
On Sun, 2013-02-03 at 15:38 -0800, bilal ghayyad wrote: Dears; I am facing a problem in disconnecting the calls, it is related to the rtptimeout (disconnecting if there is no RTP packets from both sides). My Asterisk Box is behind NAT but there is a static real IP address at the ADSL router. We call from the Mobile to the PSTN analogue numbers which are connected to Asterisk Analogue card (the telephone lines are analoge), and then we dial the overseas number, so the asterisk is sending the call to a VoIP service provider which will route the call to the destination. Sometime the destination is connected while ringing !! And this is a problem from the SIP service provider route, then we hangup our mobile (as no one answering our call) but asterisk is not detecting the hangup (it is because the telephone lines are analoge and this problem is common in analoge lines that some hangup are not detected). In that case, the call will stay open and charging and this is a wrong. This problem was not appearing when Asterisk machine was having static real IP address because I was enabling the rtptimeout paramters. But now as the asterisk box IP address is private and it is behind NATing then it is appearing even I enabled the (rtptimeout=50 and rtpholdtimeout=120). What should I do? My advice is to first try to fix your pstn hangup detection problem. Relying on rtptimeout assumes that the voip side has hanged up and the voip provider has also terminated the call and no rtp is coming. Which means that if your pstn caller terminates the call and the voip side does not (for any reason) you will still be charging the pstn caller. To see why rtptimeout does not work get a wireshark capture and see if there is still traffic going on -- Stelios S. Koroneos -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How often to restart Asterisk...
On Fri, 2013-01-11 at 15:09 -0600, Danny Nicholas wrote: The general rule seems to be, don’t restart it unless there’s a problem or you hear of memory leaks. I had a version of 1.4 that I restarted every night because I read about memory leaks, but I hear of 1.2 installs that have been running continuously for 10 years. I can confirm that 1.2 (although ancient) seems to be the version with the least issues in terms of memory leaks/restarts etc (This is based on approx 650 devices i am currently monitoring) Usually we do a yearly shutdown to replace power supplies and some hardware maintenance/cleaning I have a few machines running for 3+ years without any restart/reboot (these are custom powerpc boards in some hard to reach places) That said, if you have an external facing asterisk (i.e accessible public Internet) don't even think about it... Latest 1.4 and 1.6.2 seem decent, although we will be switching to 1.8 on new installs and upgrades this year, due to LTS status The biggest issue i have faced in term of stability is badly written AGI scripts that tend to hog resources and bring systems down in the end. Stelios -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How often to restart Asterisk...
Veering off topic, but still curious :) Since an AGI only exists for part of the life of a single call, how does it accumulate enough resources to be a problem? Call goes away...AGI thread stays back in some zombi state waiting for something (most of the times response from another web service) That leaves you with an interpreter (probably PHP) fully loaded plus whatever buffers/magic it had acquired stuck in memory. If you are *really lucky* it will be stuck in a blocking function bringing the core its running to 100% utilization and you might notice it. if not you are in for a random(15) days crash -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk on arm
On Fri, 2012-08-31 at 17:53 +, Giuseppe Longo wrote: Hi, has anyone tried asterisk on arm processors? how is the performance? have encountered problems in the compilation? Have run asterisk up to 1.4 using openembeded on several arm boards in the past. In general works well with sip/iax calls with minimum or no transcoding. (i think you could do a couple of g729 channels on the RasPi, haven't tested though) For call recording that someone mentioned i think its going to be very difficult regardless of stripping or custom compiling to achieve any good results as it needs lots of resources and heavy i/o which bogs things down. -- Stelios S. Koroneos Digital OPSiS - Embedded Intelligence Tel +30 210 9858296 Ext 100 Fax +30 210 9858298 http://www.digital-opsis.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Reverse phone lookup service
Greetings to all. We have developed an application for asterisk to accept creditcards using the Authorize.net gateway. I am looking at reverse phone lookup services in order to get the *full* address of the caller (not just state,city). This is a very low volume service as you can imagine so i would prefer something with low or no minimums and a pay-as-you-go scheme. An API would be a plus, although not necessary (i.e we can handle web forms). Anyone using or know a service that would like to recommend ? Thanks in advance Stelios S. Koroneos P.S Sorry for the cross-post -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] any working calling card solution open source
On Mon, 2012-07-16 at 20:51 +0100, Goke M Aruna wrote: thank Carlos, Thanks but too big for a demo interms of setup no demo data. I got the astcc working but still looking for alternative A2billing takes less than 30 minutes to setup on an Ubuntu server. It's true it has a steep learning curve, especially to understand the way things are done for customer/card generation but after that its pretty easy to support and a rather robust solution for calling cards. Contact me off-list if you want assistant in setting up a demo system -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX MOS Score measuring solution
Greetings ! Has anyone used any solution for getting the MOS Score on IAX channels using codes like g729. I have found a few but all are measuring sip and/or a-ulaw. Regards Stelios -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] concurrent call tracking
You can use the manager api (interface) and poll that info and then store it in a MYSQL table etc. You can do this outside asterisk,even from a different machine using your preferred dev language as there are manager libraries/bindings for most major dev languages 'Actual' is the key word though. To get the actual concurrent channels you should poll the system, at least every second, and that means 3600 records per hour or 86.400 per day. That would end up taking a alot of time to average using mysql queries. Alternatively you could do N minutes averages and store them in the db i.e read every second but save the average of 60 reads which is 1 minute etc Stelios On Wed, 2011-05-11 at 09:57 -0700, Skyler wrote: Hi all, I would like to track/store concurrent call usage per user by day/week/month and get server totals by day/week/month. Google comes up with mostly info regarding concurrent call limits, though my goal is to calculate actual concurrent channel usage and add it into reporting. I’m using * 1.6.2 + mysql – realtime (no gui). Any suggestions / open-source / AGI on where to start looking into implementing something like this? TIA, Skyler -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk, SIP Firewalls
On Wed, 2011-04-27 at 10:16 -0700, Myles Wakeham wrote: Well there is one 'optimization' that I need to sort out. There seems to be some latency between the Asterisk server (and the SIP Phones) and callers. Depending on the caller's network (ie. POTS, Cell phone, other Voip, etc.) we find about 30% of the time that there is a small delay (about 1/2 a second) between us talking and the caller hearing it, which makes it sound like the caller is talking to an offshore company located in South Asia. I have read numerous posts, discussions, etc. about this sort of thing and it seems that it has something to do with our Firewall, QoS, etc. and I'm entertaining moving the entire Asterisk server outside of our Firewall, and connecting the SIP phones to it on an entirely separate sub-net with a dedicated NAT router. 1/2 second latency i dough it could be attributed to a firewall/qos, unless your Internet connection is saturated with p2p or some other high volume traffic (movie/radio streaming) or your firewall is running on some slow machine with too many rules for packet inspection etc. If that's the case moving asterisk to public ip wan't fix it. As a first indication you could add a qualify=yes in all your sip peers to see how long it takes them to talk to asterisk. It kinda scares me though. I know that SIP is an attractive attack-vector, and that there are scripts out there that target SIP devices. I know I could run Fail2Ban on the server, which is fine (we're doing that anyway now), but before I go down this path, I wanted to get general feedback if we are using our Asterisk system using 'best practices' or whether it should never be sitting behind a Firewall, despite the fact that it is working pretty close to perfect as it is right now. I just want to find a way to reduce the latency. Does anyone have any thoughts about this? 90% of the problems i see with asterisk security has to do with bad configuration, bad dialplans and bad security policies (weak passwords,no monitoring) etc. The other 10% can be protocol or asterisk security issues but usually these get fixed before script-kiddies get a chance to use them. In your case since all your sip traffic would be coming from a single IP address (of your provider) things are a bit easier to setup. IMHO try to avoid as much as you can exposing asterisk to a public ip/network and use it as a last resort method if everything else fails. Stelios -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Vestec for Asterisk
Talk to Vestec. As far as i know they are they ones that can re-issue the license code. Kashif Kahn (kahn at vestec.com) was very helpfull whenever i need it info for my project. Stelios On Tue, 2011-04-05 at 15:36 +0100, Lee Archer wrote: Hi, I installed the Vestec module to one of my development Asterisk servers a few months ago but now I need to move the license to another host. Does anyone know how to do this? I’ve had a look on my Account page on the Digium website but it only shows the Language Pack, and I can’t do anything with this either. Can anyone point me in the right direction please? Thanks Lee -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk on Debian Lenny with timerfd
On Mon, 2011-01-24 at 01:09 -0500, RR wrote: On Mon, Jan 24, 2011 at 12:24 AM, Paul Belanger pabelan...@digium.com wrote: On 11-01-23 10:24 PM, RR wrote: email from Kevin Flemming talking about =2.6.27 so thought I'd ask esp. coz I have 2.6.26-2 yet I don't think I have timerfd on my machine...and I see, the following If you read CHANGES, you will also see you kernel 2.6.25+ *and* glibc 2.8+. Lenny ships with 2.7-1 yep, had read that too, just very new to debian so was fearing I'll have to do a manual install / upgrade of glibcI guess that's what I have to do :( will figure out how to do that. Just an FYI. Be sure to test it to a non production system, trying to replace glibc from source is not an easy task. *MANY* things need tweaking and lots of apps can break with the wrong glibc version. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TTS for asterisk
On Mon, 2010-05-24 at 11:51 -0500, Danny Nicholas wrote: The Cepstral paid version has several languages available and other voices for those 10 people who don't like Allison. At $35.00 a pop, it's not prohibitive (Lumenvox is much more pricey) This is the initial cost, there are also per port licensing costs for usage. -- Stelios S. Koroneos Digital OPSiS - Embedded Intelligence Tel +30 210 9858296 Ext 100 Fax +30 210 9858298 http://www.digital-opsis.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] User on PC?
On Mon, 2010-03-01 at 23:46 +0100, Leif Neland wrote: I'm looking for a way for linux to query a pc if user X is on, and has used the pc recently or the screensaver is not active. If so, I'll route a call for user X to the phone near that PC. Ideas, anyone? 'who' can give you info who is logged in and when for all terminals on a linux machine. Also 'fgconsole' will be usefull. This assumes you got remote access (ssh probably) to the machine and you are able to execute commands as root (for the fgconsole at least) Check also the XDMCP protocol for the X Display Manager XDM, KDM, GDM etc (not sure which one your machines will be running) as it can provide some info also -- Stelios S. Koroneos Digital OPSiS - Embedded Intelligence Tel +30 210 9858296 Ext 100 Fax +30 210 9858298 http://www.digital-opsis.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Fax
On Thu, 2010-02-18 at 19:43 +0800, Gopalakrishnaiyer Venugopal-Q16770 wrote: Hi All I am using a Asterisk 1.6.1.6 and I have Digium cards TE122B for the PRI line and TDM800P cards for connecting the telephone lines.The voice calls are working fine.Now I need to connect FAX machines to this TDM800P cards.Kindly let me know what all changes I need to make to make the normal FAX call work? Got some 'tips' on getting fax (passthrough) to work with asterisk on my blog recently http://skoroneos.blogspot.com/2010/02/solving-fax-issues-in-asterisk.html Hope they are helpful. -- Stelios S. Koroneos Digital OPSiS - Embedded Intelligence Tel +30 210 9858296 Ext 100 Fax +30 210 9858298 http://www.digital-opsis.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Linux/Asterisk on game consoles?
On Fri, 2009-10-16 at 08:16 +0200, Vincent wrote: Hello I don't know much about game consoles, and I was wondering if someone had successfully ported Linux and Asterisk to the current hardware, ie. Nintendo Wii, Sony PS3, or Microsoft XBox360? Thank you. I did it with PS3 and Asterisk 1.2 about a year ago With Yellow Dog linux running on PS3 Was not using any of co-processors though, just the main cpu. -- Stelios S. Koroneos Digital OPSiS - Embedded Intelligence Tel +30 210 9858296 Ext 100 Fax +30 210 9858298 http://www.digital-opsis.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Start asterisk on boot
On Mon, 2009-01-26 at 09:32 +0100, Ralf Träskman wrote: Hi We runs asterisk 1.6 on a ubuntu 8.04 server. How can I get asterisk to start at boot? I have created an file named asterisk in /etc/event.d and put in this # This service maintains Asterisk from the point the system is # started until it is shut down again. description Asterisk daemon start on runlevel-2 stop on shutdown respawn exec //usr/sbin/asterisk –f But it doesn’t work. Regards /ralf Ralf Träskman, IT AdLibris AB, Odengatan 106, 113 22 Stockholm, Sweden Dir: +46-(0)707548074, vxl: +46-(0)85460 60 00, fax: +46-(0)85460 60 99 r...@adlibris.com www.adlibris.com P Please consider the environment before printing this e-mail I am using the /etc/init.d/asterisk startup file and works Just make sure that permissions are set correctly. The default asterisk package from ubuntu reps runs as non-root so if you have installed it prior to installing 1.6 some things will be warped. -- Stelios S. Koroneos Digital OPSiS - Embedded Intelligence Tel +30 210 9858296 Ext 100 Fax +30 210 9858298 http://www.digital-opsis.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which policy for ISDN BRI support in NT/PtMP ?
So my opinion is that these NT-PTMP is really and urgently needed, especially if this TEI management is rather complex and therefore would take a long time to develop and stabilize. The alternative is to keep using those Patton, Quintum, etc ... boxes which is not what we would prefer ;-)) As a quick alternative look at zaphfc and friends, but don't expect it to be trouble-free. -- Stelios S. Koroneos Digital OPSiS - Embedded Intelligence Tel +30 210 9858296 Ext 100 Fax +30 210 9858298 http://www.digital-opsis.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Anyone flying with BA on Monday for Astricon ?
Greetings to all ! I will be flying on Monday with BA from London to Phoenix so i was wondering if anyone else is on the same plane so there will more than inflight movies to pass the time :) If so please contact me off-list and we can arrange to meet. -- Stelios S. Koroneos Digital OPSiS - Embedded Intelligence Tel +30 210 9858296 Ext 100 Fax +30 210 9858298 http://www.digital-opsis.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Anyone flying with BA on Monday for Astricon ?
Greetings to all ! I will be flying on Monday with BA from London to Phoenix so i was wondering if anyone else is on the same plane so there will more than inflight movies to pass the time :) If so please contact me off-list and we can arrange to meet. -- Stelios S. Koroneos Digital OPSiS - Embedded Intelligence Tel +30 210 9858296 Ext 100 Fax +30 210 9858298 http://www.digital-opsis.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] iLBC and G729 codecs
Make sure there is no noload = codec_ilbc.so in the module folder You can also try to manually load the codec from the cli try load codec_ilbc.so For g729 you need to buy a licence from Digium. -- Stelios S. Koroneos Digital OPSiS - Embedded Intelligence Tel +30 210 9858296 Ext 100 Fax +30 210 9858298 http://www.digital-opsis.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Digital-OPSiS is giving you the chance to Win A FREE Astricon Pass
Digital-OPSiS is offering a Free Exhibitors Pass, valued at $695 to one lucky winner. Register at http://www.digital-opsis.com/astricon08 to be part of the Digital-OPSiS team and get all the action of Astricon. For those of you who register we have more offerings * A discount code that entitles you to a 15% discount to all Astricon registrations.* * A free Expo Hall Pass (valued at $50), so you can visit the Astricon Expo and see us at the Digital-OPSiS booth*. The Free Exhibitor Pass includes access to all conference and pre-conference activities. For those who want a bit of background on Asterisk, the Asterisk 123 pre-conference seminar is ideal. Developers won't want to miss the Developer 101 activity, which provides a chance to catch up on the latest additions and changes to the Asterisk code base. Or the Asterisk Ecosystem, where you can learn the latest and greatest of the new products out for Asterisk. The Pass also includes all other conference activities. This Includes * Pre Conference Activities - Tuesday, September 23. Your choice of: * Asterisk 123 * Asterisk Ecosystem * Asterisk Developer 101 * In addition, you will receive access to the following events Tuesday night: * AstriCon Expo Hall Opening Reception * Code Zone Opening Party * All Conference Sessions - Wednesday, Sept. 24 - Thursday, Sept. 25 * All Tutorial Sessions - (Wednesday-Thursday) * All General Session Conference Presentations - (Wednesday-Thursday) * The Asterisk Exhibit Hall - (Tuesday-Thursday) * All Lunches and Breaks * All BOF Sessions * All Panel Discussions The lucky winner will be announced, Friday August 22nd at the Voip Users Conference http://VoipUsersConference.org *Code is valid till September 21st ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] does asterisk 1.4.20 run on a 486 sx
Depending on the gcc version you use you need to set it to produce i486 code. The illegal instructions are probably because the default makefile builds for a later arch. Also without an fpu and fp kernel emulation don't expect things like dtmf to work. Kernel fp emulation is very slow. Stelios S. Koroneos Digital OPSiS - Embedded Intelligence http://www.digital-opsis.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jerry Geis Sent: Tuesday, June 24, 2008 8:40 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] does asterisk 1.4.20 run on a 486 sx I have compiled asterisk 1.4.20 on a 486 (sx) machine. No floating point but math emulation is used in the kernel. When I run asterisk -vc all I get is Illegal instruction. I compiled as normally I do. Whats my next step. this is download source and compiled. Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] time on asterisk
GMT timezone does not have daylight savings, so probably this is why you have the wrong time Select a timezone for a city and usually the correct daylight parameters are used Stelios S. Koroneos Digital OPSiS - Embedded Intelligence http://www.digital-opsis.com http://www.digital-opsis.com/ _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nhadie Ramos Sent: Thursday, June 12, 2008 12:00 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] time on asterisk hi mats, i'm using 64-bit Ubuntu Server Edition 8.04 I just use GMT+0, but i'm on Singapore whcih should be at GMT+8, but if i use GMT+8 the system does not give the correct time. i'm not using ntp, coz when i do i also don't get the correct time. i'm not sure how i can fix this, is this an ubuntu issue? regards, ron --- On Thu, 6/12/08, mkn0014 [EMAIL PROTECTED] wrote: From: mkn0014 [EMAIL PROTECTED] Subject: Re: [asterisk-users] time on asterisk To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Thursday, June 12, 2008, 8:20 AM Nhadie Ramos wrote: Hi Sir, I tried restarting asterisk, but still it has the wrong time. I tried restarting the system, then start asterisk it still uses the wrong time. I also tried recompiling asterisk, checked i have the correct time on the system, then restart the system then start asterisk but still i get the wrong time. My system time (currently) Thu Jun 12 15:12:11 GST 2008 on asterisk i use EPOCH to look at the time, NoOp(SIP/105101-00857e60, DATE: 20080612-081147) i would really appreciate any help. TIA ron --- On *Thu, 6/12/08, Tilghman Lesher /[EMAIL PROTECTED]/* wrote: From: Tilghman Lesher [EMAIL PROTECTED] Subject: Re: [asterisk-users] time on asterisk To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Thursday, June 12, 2008, 1:42 AM On Wednesday 11 June 2008 17:52:15 Nhadie Ramos wrote: I'm using gotoiftime on asterisk, but it seemsnbsp; there is a difference between the asterisk time and the system time. could it be because i adjusted the system timezone on my linux? do asterisk not detect the change of timezone on the system? How can I fix this prob? Yes, that's probably the reason. The system timezone is cached once at startup, for performance reasons. The only way to get it to pick up the new timezone is a restart. -- Tilghman Ron, What OS/Distro are you using ? What timezone are you using ? Do you use NTP for syncing time/date? /Mats ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No-mobo PC for USB Drives Enclosure?
As people have sugested the ATX power supplies can work without a mobo One thing to watch out for your setup is the actual ampere requirments for your disks i.e Your power supply provides 300W but this is partitioned to different voltages (+5, +12, etc) with different amp charecteristics Disks need 2 voltages. One for the logic (+5V) and one for the motors (+12V) and have different current requirments. Most disk (if not all) mention these ratings on the labels they have What you must do, is to see if by adding the current requirments seperatly for +5V and +12V, does not exceed the power supply's amp rating *for that voltage*, allowing also for a 15% -20% margin, as power consumption will be higher than the nomimal mentioned during disk startup (and you will be starting all your disks at the same time) Also make sure your box has sufficient cooling and there is some short of airflow over the disks, as the number 1 enemy of disks is high temperature and stacking so many disks in a box will create large amounts of heat. I would suggest you to get a good (aka expensive) 500W power supply and use 10-12 disks with it to avoid problems in the long run, Also keep in mind that MTBF specs of SATA disks does not make them an ideal candidate for 24/7/365 operations Stelios S. Koroneos Digital OPSiS - Embedded Intelligence http://www.digital-opsis.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Rubenstein Sent: Wednesday, May 14, 2008 7:31 AM To: Col Ferguson Cc: Asterisk -Users Subject: Re: [asterisk-users] No-mobo PC for USB Drives Enclosure? On Wed, 2008-05-14 at 14:06 +1000, Col Ferguson wrote: If I understand right, your problem is that the power supply won't turn on ? ATX power supplies can be told to turn on by jumpering 2 pins on the motherboard power connector. From memory its the Green wire and one of the black wires, I usually use the next one inwards. Pinouts for the connector can be found via Google. If the power supply also has an external on/off switch you can jumper these pins and use the switch to turn the power on or off. Hope this helps, Thanks, that sounds like exactly what I was looking for. Is there any safety risk from jumpering that sensor? Like some kind of extra sensor, like voltage feedback, temperature or somesuch. If this works, it might point to a good way to reduce redundant Asterisk servers, which suck power, by just plugging the drive from each old server into the USB of a single server with a merged dialplan and a few other tweaks to point at several different mounted drives, rather than one per host/IP#. Col - Original Message - From: Matthew Rubenstein [EMAIL PROTECTED] To: Asterisk -Users asterisk-users@lists.digium.com Sent: Wednesday, May 14, 2008 12:22 PM Subject: [asterisk-users] No-mobo PC for USB Drives Enclosure? I have over a half-dozen different SATA hard drives, each with different data (configs, voiceprompts, voicemail, CDRs, AGIs) for each one's different user groups and applications. Each one's load on the Asterisk server is small enough that one server can host them all, accessed easily over USB. But right now, each one is in its own external USB enclosure on a powered USB hub. I want to combine them all into a single large enclosure. I tried to use a single PC chassis, leaving the USB hub inside with the drives screwed into it, and powered from the PC power supply as internal drives on the proper drive power output plugs. But without a PC motherboard plugged into the power supply, too, the power supply won't start up to power the drives. I don't want to add a motherboard: that costs money, and sucks power, and is totally unnecessary. I just want to make this gutted PC chassis power my drives only, and have them connect to the complete PC sitting next to it via the single USB cable coming out of the drive chassis. How do I do that? Is it possible to use the extra, unused floppy power plugs to power more hard drives, with an adapter? Is it possible to split the existing hard drive power plugs to each power multiple drives? How many drives can I split each power plug into? The power supply is a cheap 300W unit, and the drives draw max under 9W each: http://www.wdc.com/en/products/products.asp?driveid=311 . So can I power 25-30 of these drives, or at least 10? -- (C) Matthew Rubenstein ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG. Version: 7.5.524 / Virus Database:
Re: [asterisk-users] Newbie alert: VoIP hardware
Questions: [1] Can I use oslec for echo cancellation? I'll have beefy hardware. Is echo cancellation necessary? Yes you can use oslec provided that either your distribution has a zaptel package with the oslec patch (or you build the zaptel drivers + oslec yourself) Well without echo cancelation you will probably have a number of calls that have either very bad sound quality or are simply annoying With your set i.e 3-4 lines processing requirments are minimal so you should not worry about that.We have been able to run oslec for 4 lines on a 266Mhz (no its not Ghz) powerpc embedded board with very good results Stelios S. Koroneos Digital OPSiS - Embedded Intelligence http://www.digital-opsis.com http://www.digital-opsis.com/ _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Repo Sent: Wednesday, May 07, 2008 8:11 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Newbie alert: VoIP hardware Hello, Please forgive me for i'm not an asterisk user yet. I've done as much research as I can .. and have the following questions. I'm setting up a new office and a home office and i'm shopping for hardware. Office: 2 analog lines Hardware: TDM412B (2 FXO, 1FXO) Link: http://www.voipsupply.com/index.php?cPath=99_555_556 Cost: $303 Home: 1 analog line Hardware: TDM421B (2 FXS, 1 FXO) Link: http://www.voipsupply.com/product_info.php?products_id=3980 Cost: $300 [2] Can I get PCI express x1 cards for the same price? I'm on budget, Any other cards (sangoma? rhino?) that might work well? I'm sure these questions have been asked before.. :-) Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Running Asterisk as root
In general, if your asterisk is accesible from the internet its much better to have it run as a non-root process. (My opinion is that this should be the default out-of-the-makefile ;) asterisk behaviour) This is the norm for more of the servers/services running on a linux system, and can act as a safety-net when things go bad Stelios S. Koroneos Digital OPSiS - Embedded Intelligence http://www.digital-opsis.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Christian Sent: Tuesday, May 06, 2008 3:00 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Running Asterisk as root Hi all, I have seen discussions on this earlier on, but just want to hear some quick thoughts. I am running v1.6 of Asterisk on my Ubuntu installation, I did make config to make it run at boot. Since I've got a firewall and don't have any other servers running I am not worried. I have been htinking about running Asterisk as a seperat user, but haven't done that yet. Everything is working fine. What do you think? Thanks, Christian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hyperthreading and multicore
Asterisk didn't benefit much from having HT enabled on a P4 with HT capability. There are several things that make a difference when optimizing for a specific processor in order to take advantage of its features. Gcc version used to build asterisk (and the system in general) and compile flags can make a big difference A lot of the ready made solutions use very generic optimization as they are trying to be compatible with a wide range of cpu core's and architectures. This has the advantage of having a single binary image to distribute but you pay for it in terms of performance. In most cases the performance penalty is not noticable in small/home installations but you start to notice it when you push the system to its resource limit (i.e cpu, memory,pci bus access etc) either because you handle a lot of calls or your system is resourse limited i.e embedded boards. So in general if you need to get the maximum performance out of a system, make sure you build asterisk tuned for that system and not a generic build. Running code with 486 instruction set, with command scheduling for pentium its not going to give you max performance regardless of the fact that your cpu/core supports HT or not. Stelios S. Koroneos Digital OPSiS - Embedded Intelligence http://www.digital-opsis.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyone have a method of keeping an incrementaltally of calls?
Hi JR !
You could use dbget/dbput to have something like that
i.e
Set(foo=${DB(counter/counter_val)})
Set(foo=${MATH(${foo}+1)}) ;
Set(DB(counter/counter_val)=${foo})
Stelios S. Koroneos
Digital OPSiS - Embedded Intelligence
http://www.digital-opsis.com
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
JR Richardson
Sent: Tuesday, April 08, 2008 5:02 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Anyone have a method of keeping an
incrementaltally of calls?
Hi All,
I thought I read a post a while back of a system call or
something in the dialplan whereby a call count can be
incremented and spit out to a text file.
Not like a group count of active channels.
What I would like to accomplish is have an incremental count
of a specific dialplan routine that gets called, so after a
week or month, I can see how many times a specific dilaplan
action has been used.
Thanks for any advice.
JR
--
JR Richardson
Engineering for the Masses
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Re: [asterisk-users] Finding iaxy's (iaxies?)
I haven't used any Iaxy but from the example it looks like once you ping the ip of the Iaxy it will responde with a udp packet from port So you don't actually ping the port, but again as I said never used it so I could be wrong Stelios S. Koroneos Digital OPSiS - Embedded Intelligence http://www.digital-opsis.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Edwards Sent: Saturday, March 29, 2008 4:01 AM To: Asterisk Users Mailing List Subject: [asterisk-users] Finding iaxy's (iaxies?) According to http://kb.digium.com/entry/12/ The Iaxy will respond to pings on port . You can ping your broadcast IP on your network and listen with tcpdump on your network on port which will show the Iaxy responding and what IP address it is coming from. Ex. ping 192.168.1.255 tcpdump -i eth0 udp port Before I get my karma whacked again, does this work for anybody? 1) Shouldn't ping 192.168.1.255 be ping -b 192.168.1.255 2) Aren't pings ICMP and thus invisible when tcpdump is looking for UDP? 3) How do you set a port on an ICMP ping? 4) How do YOU find an Iaxy on your network? Thanks in advance, -- -- Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyone here using JUNGHANNS.net douBRI 2.0 ISDN ?
For the HFC-4S (4 bri channels) you need to qozap driver not zaphfc Regards Stelios S. Koroneos Digital OPSiS - Embedded Intelligence http://www.digital-opsis.com Quoting Elijah [EMAIL PROTECTED]: Hi, I'm very new to asterisk and managed to set one up in debian, I installed via apt-get the asterisk and asterisk-bristuff packages. I downloaded the bristuff source as well. I managed to get as far as loading the following modules: zaptel199144 4 cwain,zaphfc,zttranscode,ztdummy But the problem now is that all channels that I try to configure gives me an error that that device/address doesn't exist ..and only one unconfigured device is displayed in zttool. Does anyone have any experience with this type of card? care to share some tips? 0c:02.0 ISDN controller: Cologne Chip Designs GmbH ISDN network Controller [HFC-4S] (rev 01) Subsystem: Cologne Chip Designs GmbH Unknown device b556 Flags: medium devsel, IRQ 5 I/O ports at ccf8 [size=8] Memory at d9fff000 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 Regards, Elijah ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Flash IDE
CF flash deviced work fine provided that a) The CF has a wear leveling controller inside (not all do, especially the cheap ones) so even a ext2 filesystem wan't create problems b) You use a distro with read only (or partial write) filesystem .i.e logs to ram or remote server etc Other than that we have deployed a very large number of devices with embedded linux in a CF (not all of them asterisk) with minimal problems Stelios S. Koroneos Digital OPSiS - Embedded Intelligence http://www.digital-opsis.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Juan Sandro Sent: Tuesday, September 11, 2007 12:04 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Flash IDE Hi We have a number offices accommodating 4-6 people each hence it is very important for PBX to be fanless and silent. We have been looking at using IDE flash disks also called DOM. The performance tests we have done so far satisfy our requirements, however we are concerned with DOM durability. We have installed debian and vanilla asterisk on 1GB DOM. All seems to work fine at the moment however will DOM last? How long it will last? Is anyone able to share similar experience? Any other information/tips? Regards, Juan _ News, entertainment and everything you care about at Live.com. Get it now! http://www.live.com/getstarted.aspx ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] bristuff for hfc card on Xscale 80219
Is your system compiled as BE or LE ? bristuff will compile but will not work on BE systems without some patches/endianess fixes as some of the buffer pointers are little endian Also with the existing bristuff you will get invalid data due to the fact that the cache controller of xscale can not snoop into the DMA transfer cycles and update the cache To avoid this you need to allocate a non cachable memory region for buffer. In sort. Although it compiles it does not work :/ Stelios S. Koroneos Digital OPSiS - Embedded Intelligence http://www.digital-opsis.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Thomas Winter Sent: Wednesday, July 18, 2007 1:44 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] bristuff for hfc card on Xscale 80219 Hi, compile and load of modules works fine. After ztcfg I can see . . Changing signalling on channel 1 from Unused to Clear channel Changing signalling on channel 2 from Unused to Clear channel Changing signalling on channel 3 from Unused to HDLC with FCS check and then the board is frozen. Any ideas? regards Thomas ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Scaling Asterisk: Dual-Core CPUs not yieldinggains at high call volumes
Hi, The system seems to be IO bound for some reason. Reading at the older posts you mentioned that there is no significant disc activity so it could be ethernet i/o and/or interrupts that are causing this (old or insuficient ethernet driver maybe ?) Usually this kind of i/o wait is present on machines that have run out of memory and need to swap to disk Also with regard to the higher system usage on multicore systems, its very probable that its due to task migration from core to core Here is something we recently noticed that may explain why the dual-core server is under-performing at high call volumes. The following numbers were collected off both servers while they were in production. Note that while they have similar cumulative idle values, the ratio of system time to user time on the single-core server is roughly 2.3 to 1, but on the dual-core server it is roughly 19.6 to 1. I'm not quite sure what to make of this, but it seems to be very relevant to the problem. Mon Apr 2 12:15:01 EDT 2007 Idle (sar -P ALL 60 14) (60 seconds 14 slices) Linux 2.6.12-1.1376_FC3smp (4core.imminc.com) 04/02/07 12:24:01 CPU %user %nice %system %iowait %idle 12:25:02 all 14.97 0.03 34.25 0.92 49.82 12:25:020 8.83 0.05 33.60 1.28 56.24 12:25:021 17.50 0.02 34.60 0.57 47.32 12:25:022 19.94 0.02 33.52 1.31 45.22 12:25:023 13.62 0.02 35.29 0.52 50.55 Thu May 10 15:30:01 EDT 2007 Idle (sar -P ALL 60 14) (60 seconds 14 slices) Linux 2.6.12-1.1376_FC3smp (8core.imminc.com) 05/10/07 15:38:01 CPU %user %nice %system %iowait %idle 15:39:01 all 2.47 0.01 48.29 0.00 49.23 15:39:010 2.92 0.00 53.17 0.00 43.91 15:39:011 2.98 0.00 48.68 0.02 48.33 15:39:012 2.47 0.02 48.61 0.00 48.91 15:39:013 2.27 0.00 48.35 0.00 49.38 15:39:014 2.38 0.02 47.38 0.00 50.22 15:39:015 2.37 0.02 46.94 0.00 50.67 15:39:016 2.23 0.02 46.63 0.00 51.12 15:39:017 2.17 0.02 46.54 0.00 51.27 Stelios S. Koroneos Digital OPSiS - Embedded Intelligence http://www.digital-opsis.com ___ --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: RE : [asterisk-users] Testing ISDN T1/E1 Bri and Pri etc.
Hello ! For isdn BRI, the zaphfc in NT mode works fine. Some equipment might require voltage on the line (although they don't use it to powerup and it just draws a few mil amps) As for PRI never tested, i would be interested to know how your test goes Stelios S. Koroneos Digital OPSiS - Embedded Intelligence http://www.digital-opsis.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Gavin Henry Sent: Wednesday, May 09, 2007 2:09 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: RE : [asterisk-users] Testing ISDN T1/E1 Bri and Pri etc. On 09/05/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi Gavin, A second Asterisk server replacing the provider (best way), or doing a loop between two different ISDN ports on a same card (worst way) must help you. Thanks for that. Will get a spare * box. Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Gavin Henry Envoyé : mercredi 9 mai 2007 09:40 À : asterisk-users@lists.digium.com Objet : [asterisk-users] Testing ISDN T1/E1 Bri and Pri etc. Hi All, Can anyone recommend any test kit that you can hook up your Pri/Bri cards to without having actual ISDN in your office. For example testing an * system before it goes to a clients office. Thanks, Gavin. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Unable to find a codec translation path from ilbcto ulaw
Oliver, SIP-phone --iLBC-- Asterisk ---ulaw PSTN-Gateway I get the following error: Unable to find a codec translation path from ilbc to ulaw Does your phone support ilbc as a codec ? Is the codec_ilbc loaded on the * box ? Usually you get this kind of error when the codec is not supported Stelios S. Koroneos Digital OPSiS - Embedded Inteligence http://www.digital-opsis.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Oliver Brandt Sent: Friday, April 27, 2007 7:08 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Unable to find a codec translation path from ilbcto ulaw Hi! As the upstream of my DSL-connection is very slow, I'd like my sip-phones to use iLBC to connect to my *. My gateway provider only allows ulaw. Hence, I'd like to use the follwing setup: SIP-phone --iLBC-- Asterisk ---ulaw PSTN-Gateway I get the following error: Unable to find a codec translation path from ilbc to ulaw Setup SIP-phone: disallow=all allow=ilbc Setup PSTN-Gateway: disallow=all allow=ulaw I've googled for overn an houre. But no luck. So I'd really apreciate any help! Thanks! Oliver ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: On Topic: Cheapest Asterisk USB Key? (was: Re: [asterisk-users]Off Topic: Open Source USB Softphone)
Here's a flipside of this subject: what is the absolute cheapest Linux device that can be connected to a PC's USB port? That has just enough power for a minimal Asterisk server running on it. The Asterisk just maintains a CDR database on its Flash memory, which it periodically submits over the PC's network connection with an HTTP hit on a remote full-service Asterisk server? No call handling, DSP or anything really number crunching, no telephony terminal or other services. The lowest-performance device that plugs into the USB, with its own Linux instance. In OEM quantity, under $50? Under $100? When you say devices do you mean an off the self device or a module you can use to build a custom device ? In the first case there are a lot of fisrt generation routers coming into the market at very low prices for example http://www.wirelesslan.gr/product_info.php?cPath=127products_id=866 http://www.wirelesslan.gr/product_info.php?products_id=670 If you are looking for a SoC type device there are several although, the 100$ range looks more realistic There are several devices that could be used. DimmPC comes in my mind - http://www.amctechcorp.com/dimmpci/index.html Digi's Connectcore http://www.digi.com/products/embeddedsolutions/connectcore9u.jsp Check Linux devices for a larger list http://www.linuxdevices.com/articles/AT8498487406.html Hope it helps Stelios Stelios S. Koroneos Digital OPSiS - Embedded Inteligence http://www.digital-opsis.com ___ --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: Two or More Bri Cards
A lot of the problems people are having with ISDN have to do with the service provider setup and not the cards (provided that we are talking about 2-3 HFC cards) For example in Greece the local telephone company uses 3 diffrerent types of ISDN equipment in their centers (Siemens,Ericsson,Alcatel) We have seen a lot of trouble with a certain type and sporadic troubles with the other two which 9 out 10 times had to do with their setup (1 out of 10 had to do with the line itself) IMHO it has to do more with the ISDN timing itself (coming from the telephone company side) and less with the number of interrupts a system can handle. ___ --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] Load balance Asterisk servers?
JR Richardson gave a very nice presentation at Astricon on how to do that with DUNDI check http://www.astricon.net/files/usa06/Friday-General_Conference/JR_Richardson.ppt http://www.astricon.net/files/usa06/Friday-General_Conference/JR_Richardson_Whitepaper.pdf Stelios -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of voiplist Sent: Tuesday, November 14, 2006 4:46 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Load balance Asterisk servers? We are looking to be able to put a device in front of an array of Asterisk systems which would do the job of load balancing them. We would store all the particulars on one or more MySQL servers. What want to accomplish is to have all calls sent to/from a single IP, then push the calls off to another Asterisk server in the array. If one server goes out, we are hoping there will be no effect other than we have reduced capacity until it's fixed. If possible we would like to do this with either a low cost device or an open source solution which can run on a Linux box. Can anyone suggest something that would be reliable in a production environment? We would like to make this solution scale to at least a few hundred simultaneous calls. We have looked at some ready made devices but many of them only support SIP, we need a solution that will support both IAX and SIP. Any advice would be most appreciated. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Embedded Asterisk
Has anybody out there, on non-FPU embedded platorms, made any good use of things like ilbc and Speex? The exisiting implementations of both run very poorly on a non-fpu cpu's, especialy if clock speed 400 Mhz I have run asterisk (and still do) on mips,ixp and powerpc (all without fpu's) and i think that without modifications the codecs are not so usable There are 3 options 1) Get a faster fp library - Been looking into the GoFast fp lib, no definate results yet 2) Convert codecs to fixed point - Although i know a G729 fixed point implementation exists haven't tested and i am not sure that a speex or ilbc implementation exists. 4) Get a cpu with fpu :) - There are mips and powerpc cpu's (i am talking the types used in embedded dev's) that have an fpu I will be also at Astricon and brinking with me a powerpc based embedded asterisk appliance which has support for zaptel also. Maybe we could exchange some ideas on the matter. Stelios ___ --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] Kind of OT : Europeans going to Astricon
Greetings ! Its kind of OT, but if there are any Europeans going to Astricon in Dallas, please send a message of-list. It's possible we will be on the same flight,(i am flying from Frankfurt) ;) so it will be a good way to know it's other and spend some of the 10 hours + flight time . Regards Stelios ___ --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] Annoying Bristuff
Its working for me with no errors. * 1.2.10 bristuff 0.3.0-pre1s with kernel 2.6.15.4. My setup is kind of "special" as its build with Openembedded and runs from a CF on a [EMAIL PROTECTED] Recently i was able to port *+bristuff + zaptel to an embedded powerpc platform and works there also without any major issues. Why don't you trya 2.6 kernel maybe the problem is there (unlikely though) Stelios HiCan anyone confirm a working asterisk 1.2 from bristuff with 1 port PCI, hfc-s based ISDN card (zaphfc driver). If so, could you send your configuration. I mean OS (linux distribution) type, kernel version.Thanks in advanceCheersAndrew ___ --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] Annoying Bristuff
sorry for the html post :( ___ --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: Load balenced (ADSL) network connections, is it possible?
I need to put an Asterisk server in a remote office where only ADSL is available. Maximum of 8meg downstream 646k upstream. Is this an adsl2 line ? If yes ask your provider if it supports channel bonding. You could use 2 adsl lines as one. All load balancing etc is done at the dslam side. Also if annex M is supported you can get up to 3,5 mbits of upload (theoreticaly, usually is close to 2mbits, heavily depends on distance and line conditions) If you need to load balance at your side (ie the office) it can be done but would require setting up 2 * servers connected to each other and using some form of round robing the dns so that requests reach both servers. If the dsl line is also used for other purposes than voip, make sure that you use qos or you will be facing problems with the quality of the calls. Stelios ___ --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] TDMoE question
Greetings ! I am looking into the TDMoE functionality of the Zapata drivers and * and i am kind of confused. Lets say i have 2 linux boxes, one has * running but no fxs/fxo hardware the other has a card (for example an x100p) but does not have * installed. If i just want to use the card (no * reduduncy etc) from the machine that runs *, do i need to have * running on both boxes for this to work ? or loading the appropriate drivers to the second machine will be saficient ? The examples i have seen mention zapata.conf entries which make me think that * should be running on both machines, but i am not sure if this applies in my case. Any ideas, thoughts, links etc are more than welcome regards Stelios S. Koroneos ___ --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
