Re: [asterisk-users] Can't compile on NSLU2 LE
Gunner wrote: Hi, I keep getting the following on my slug no matter what version I try (1.2, 1.4, 1.6, 1.6svn): Do you build this on the NSLU2 machine? Left-over files from an x86 build? Try 'make clean' . This is native on the NSLU2. It was right from svn, but I just tried a make clean and it still has the same error. Which distro are you using on the slug? Have you tried installing from Optware? regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] Call Stealing
Geoff Lane wrote: On Thursday, January 15, 2009, Danny Nicholas wrote: Why not use call-conferencing? If you transferred your call into a conference room, you could join the conference from any extension on your *. When the caller hangs up, just end the conference. Thanks for the reply. AIUI, you need to set up the conference before leaving the extension on which you took the call. If so, call parking would probably be better since that leaves the original extension free for further calls. Thanks again, Would SLA (Shared Line Appearance) work for this? Put call on hold, press button beside flashing light on second handset? regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] top posting again [was: Re: CDR Design]
David fire wrote: you are soamming my mail box whit this useless discution the solution is doble posting (top and bottom) 2008/12/17 Andrew Kohlsmith (lists) akli...@mixdown.ca mailto:akli...@mixdown.ca On December 17, 2008 05:03:00 pm Eric ManxPower Wieling wrote: To me top posting is like people talking about SIP Trunks. There is SNIP- I submit the above as corollary to RE Kushner's point. Is it the spelling that is difficult or the typing? Really, there's nothing like a good, old-fashioned pre-Christmas flame to get you in the Holiday mood, eh? :-) regards, Drew ___ -- 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] Some Good News for VoIP
http://www.theregister.co.uk/2008/12/16/infonetics_enterprise_telephony_numbers_q3_2008/ -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] get first month of trixbox free
Hi Don, Don Kelly wrote: Caution---top posting. It works for me--ignore it if you like. Without forgiving Michael's commercial message to *-users, perhaps we can punish your top posting by highlighting your gross insensitivity to the physically challenged! :-) And (flame follows) we would do it using careful use of English grammar and spelling, especially if we were The only major error I found were threw instead of through. Perhaps that is forgivable for someone using a handle of creepyBLINDy? N'est pas? regards, Drew * From: * asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Babcock, Michael Alex* * hi; threw the end of the year we are running a promo, when ordering any package on http://gwhosting.net including our vps servers and trixbox servers, you can get your first month off. Yes, that's right, enter 30free with out the quote signs into the coupon code field during checkout to get your first month free. Give us a try, you won't be sorry. Your security is our number one priority. GW Hosting, your dedicated home on the web: http://gwhosting.net 30free does truly get you your free month. Stop at any time during your first month and you won't be charged any more, no strings attached. Well, wait there is one string, you have to go to http://gwhosting.net and sign up using 30free to get the free month. thanks Michael Babcock Michael Babcock Owner of GW Hosting, http://www.gwhosting.net http://www.gwhosting.net/ For information on what I may be doing at the moment, please feel free to visit my blog, twitter or brightkite at the following links: Twitter: http://www.twitter.com/creepyblindy Blog: http://www.gwfans.net http://www.gwfans.net/ Brightkite: http://brightkite.com/people/creepyblindy -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] 'dialer' application to trigger call betweenhardphone and number
Have you looked at ADM http://adm.hamnett.org/ ? I don't know how it works on Windows but on Linux you can highlight any number on the screen then leftclick on dialer icon, middleclick into the dialer to make a call. regards, Drew Danny Nicholas wrote: It doesn't matter where the program resides as long as it can interact with your asterisk (FTP, HTTP, etc). You just have to have the right accesses and security. Does anyone know of a small lightweight windows 'dialer' application I can use to trigger a call (via call file or AMI) from any application? (The call would be placed between the target number, and the preconfigured DN of the hardphone at the user's desk) Ideally a phone number would be 'selected' from within any windows application and the call would be triggered via hotkey, or a right-click menu or by clicking a system tray icon. Ideal would be something very 'efficient' with at most two or three discreet actions needed to dial-- (i.e. 1:Select, 2:Hotkey--done!) Any ideas? Any Happy customers? -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] OT: What do you guys think of this?
Alex Balashov wrote: http://www.theregister.co.uk/2008/12/01/richard_bennett_utorrent_udp/ FUD? Interesting? Boring? New news? Old news? Seems the sky isn't falling (yet). The original article didn't have the full story, here's an update... http://www.theregister.co.uk/2008/12/05/richard_bennett_bittorrent_udp/ regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] OT: What do you guys think of this?
Ira wrote: At 12:44 PM 12/2/2008, you wrote: At 04:03 12/2/2008, Benny Amorsen wrote: Doug [EMAIL PROTECTED] writes: Net Neutrality is great in principle. But ISP's need to somehow control those few percentage of users who suck down a huge majority of the bandwidth. It's dollars and cents. Yes, just like the airlines need to somehow control those users who keep showing up to the flight they booked, every single time! It's impossible to do overbooking with customers like that, so we need to find ways of punishing them. What happens if everyone who owns a car drives it at the same time? Owns a telephone and uses it at the same time? If I could get the same plan for my internet as I get for my phones, a few dollars a month plus a bit per minute(megabyte), I'd be all over it, but even better, then the provider wouldn't have to care as they'd be making a fair profit no matter what the user did. You make that sound almost reasonable. I'm sure initial pricing would only be slightly higher for the majority of customers with only the bad users being punished. Six months later, when all the fuss has died down, the price goes up by, say, 0.5c/MB. That's not much is it? Half a penny? After all, they used to raise it by $2/month and customers only grumbled a bit before paying. So 0.5c isn't much, is it? ...and BTW, I'm not a p2p or torrent user but I do enjoy having bandwidth available when the latest Fedora or Ubuntu comes out. regards, Drew ___ -- 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] RTCP too short
Do you have Grandstream phones? I noticed a similar issue last year with Grandstream GXP2000 phones. The phone was sending an empty RTP packet for the keepalive whilst on mute. I reported a bug to Grandstream but nothing happened. regards, Drew Jon Weisman wrote: I get this all the time. Still haven't found a solution but it doesnt seem to affect call quality or server performance. I think there's a way to disable the message, but I lost the link. :( -Jon - Original Message - *From:* michel freiha mailto:[EMAIL PROTECTED] *To:* Asterisk Users Mailing List - Non-Commercial Discussion mailto:asterisk-users@lists.digium.com ; [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] *Sent:* Friday, November 28, 2008 10:07 AM *Subject:* [asterisk-users] RTCP too short Dear Sir, I'm running Asterisk 1.4.21.2 http://1.4.21.2 on a CentOS machineWhen running asterisk -rv I can see a lot of messages about RTCP too short... -- Remote UNIX connection disconnected [Nov 28 13:33:00] WARNING[24863]: rtp.c:891 ast_rtcp_read: RTCP Read too short [Nov 28 13:33:00] WARNING[19803]: rtp.c:891 ast_rtcp_read: RTCP Read too short [Nov 28 13:33:00] WARNING[19803]: rtp.c:891 ast_rtcp_read: RTCP Read too short [Nov 28 13:33:02] WARNING[29804]: rtp.c:891 ast_rtcp_read: RTCP Read too short Can you let me know how to fix this issue? Regards ___ -- 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 -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] SIP to IAX2 with delayed echo
c james wrote: A Polycom 550 and a IAX client (Mozphone and ZoIPer were used) are having a conversation. Call quality is reported as good except for an echo with a 3 second delay. Most of my searches are saying echo happens only on the PSTN piece, but there isn't one here. Which end hears the echo? If it is the Polycom end, try a better quality headset with the softphone. Echo comes from analogue portions of the circuit and is usually caused at the end that doesn't hear it. regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] Recommend Wireless IP Phone
Wilton Helm wrote: Good points. I got an access point instead of a router specifically so I could locate it in the best position. IMO Wi-Fi routers are dumb by definition because where you want a router is probably NOT anywhere close to the best point for the Wi-Fi part. This unit has a particularly sensitive receiver to compliment the higher power. It would have been nice it it had MIMO, too, as that always helps. Repeaters would be a challenge in this case because most of the property is natural wooded (so no power or protection) and I'm trying to cover a road by only own property at one end. naturally wooded does not bode well for WiFi. Trees are much better than walls at absorbing 2.4GHz signals due to their high water content. Mountains block 2.4GHz even better. If the woods are deciduous, it may work well in the winter but fade away come spring. If the road is fairly straight, a directional antenna like a Yagi at one end might give you coverage there. As for the rest of your property, you will have to get an omni antenna up high, say one of your mountains. You may be better off with something that uses lower frequencies. The old analogue cordless phones have much better range than 2.4GHz digital stuff. regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] Dundi Issues
Jeremy Mann wrote: I don’t know if it’s related, but when doing a packet sniff with wireshark, I see UDP checksum incorrect messages: 0.058230 source - destination UDP Source port: 4520 Destination port: 4520 [UDP CHECKSUM INCORRECT] Be careful with this error, some network cards that can do IP Offload processing will show up with bad checksums in Wireshark. Check the specs for your NIC, this may be a Red Herring (or it might not! ). regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] Dundi Issues
Jeremy Mann wrote: I'm not aware of any offloading done on this particular box, it's an HP ML110 G5 using the onboard NIC. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Drew Gibson Sent: Wednesday, November 05, 2008 9:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Dundi Issues Jeremy Mann wrote: I don't know if it's related, but when doing a packet sniff with wireshark, I see UDP checksum incorrect messages: 0.058230 source - destination UDP Source port: 4520 Destination port: 4520 [UDP CHECKSUM INCORRECT] Be careful with this error, some network cards that can do IP Offload processing will show up with bad checksums in Wireshark. Check the specs for your NIC, this may be a Red Herring (or it might not! ). The HP docs indicate a setting for enable/disable checksum offload, try looking at the packets on-the-wire rather than on the server itself. http://h2.www2.hp.com/bc/docs/support/SupportManual/c00846707/c00846707.pdf regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] VoIP traffic shaping
Kristian Kielhofner wrote: On 11/1/08, OCG Technical Support [EMAIL PROTECTED] wrote: This was so interesting I had to move it to its own thread! Is anyone using this script? How does it perform compared to the older WonderShaper script? It was based off Wondershaper originally, enhanced for VoIP traffic and gives the option to use HFSC or HTB. Not only do I use it myself for AstLinux and Star2Star, most of the reports I've (we've) had have been favorable. Try it out! Has anyone tried this on OpenWRT? regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] network design philosophy and practice
Bill Michaelson wrote: I'm wondering how prevalent the practice of physically segregating voice and data networks is in the Real World. What are the factors that typically lead to such a decision? DIscussions of pros and cons are most welcome by me. Experiences, anybody? We chose to go with a segregated network and certainly don't regret the choice. Voice and data are on separate ports at the desk, avoiding QoS issues completely and reducing confision amongst users who still expect separate Phone and Computer plugs on the wall. The traffic does run through the same switches and inter-switch trunks but always on distinct VLANs. My experience with connecting the desktop computer through the phone has been very poor. Audio breaks up when the computer does large data transfers. Yes, Sir. I'll just look that up in our datab...baba.bas.ss..ss..se In addition our users require gigabit to the desktop. The phones are 100Mb. Worst part is the few Cisco phones we have insist on searching for VLAN (which doesn't exist) for 5 minutes on startup. Hopefully they will be replaced through attrition but despite being over-priced, over-featured and proprietary, Cisco do build robust kit. Sigh. regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] network design philosophy and practice
David Gibbons wrote: Two separate networks? Did I miss something? I feel like I'm taking crazy pills! Two separate physical networks means twice the hassle, twice the maintenance, twice the cost, twice the headache. Not to mention the fact that the whole idea of VOIP is to simplify IT and focus on converging data and voice networks. This is what VLANs and QOS do best. I dare say it's what they were designed foe. I can't think of any reason that I would ever recommend two ports per desk to support telephony -- ever. It's ludicrous to think that two ports will be better than one if we're setting up our VLANs and QOS properly. A phone takes very, very little bandwidth away from the desktop and a decent one will support tagging its frames for the alternate voice VLAN. --snip-- In almost all cases it is much better to have two seperate networks. This may be impractical in some smaller installs, but in any office setting we always do this. The only reason I can think of not to is to eliminate the cost of the second cable. --snip-- That's two _logically_ separate networks. The key point is that the last yard cable to the phone is not shared with the computer. The issue is not a lack of bandwidth but that the phone has to try and get its little packets inserted between the massive packets of a database lookup or file transfer in a timely manner (latency and jitter). You might get away with a single logical network on a smaller site or a larger one with very light traffic. QoS is not required on lightly loaded links and will do nothing for you on over loaded ones. I only use it on WAN links where bandwidth is more expensive. regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] network design philosophy and practice
Kristian Kielhofner wrote: On Wed, Oct 29, 2008 at 10:48 AM, Drew Gibson [EMAIL PROTECTED] wrote: Worst part is the few Cisco phones we have insist on searching for VLAN (which doesn't exist) for 5 minutes on startup. Hopefully they Drew, Disable CDP on the phone and that will go away. I know you said you're not using VLANs but... You can use CDP and set your voice-vlan on Cisco switches. Or... you can install cdp-tools on a Linux box and have it advertise a voice vlan for you! http://gpl.internetconnection.net/ I added the voice vlan support to cdp-tools. ;) I tried out the cdp-tools some time ago (it may have been on your recommendation, Kristian) but with no success. Is it possible to disable CDP on the 7940 (image_version : P0S3-08-2-00)? regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] Emerging dilema? DID forwarding meets SMS
Gordon Henderson wrote: On Thu, 23 Oct 2008, Karl Fife wrote: We have a number of DID's that do the standard VoIP tricks: ringing multiple locations, findme-followme etc. What is happening more and more is that customers call those DID numbers, and draw the reasonable conclusion that they are calling mobile numbers because they literally can HEAR that the called party is on a mobile. Consequently many of those customers draw the conclusion that they can safely send SMS's to those DID numbers. Naturally the SMS messages disappear into the ether. Er, they don't dissapear for me. I send a TXT to a landline, the phone rings and there is a text to speech robot which reads it out to you, or, Don't you have that facility? Maybe it depends on country and telco. Err, Gordon, you must be in a country from the 21st century. North America is just beginning to emerge from the mobile Stone Age. Some people have heard of text messaging but most think you have to pay Blackberry to send emails. I ran into the issues Karl mentions when trying to txt our ISP contact during our office move. Can anyone clarify how SMS to non-mobile numbers are generally handled in North America? Is it possible to have SMS delivered direct to your landline DIDs? Then have Asterisk relay it to the actual mobile DID. regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] anoyingly answers already in use pstn line
Tzafrir Cohen wrote: On Fri, Oct 17, 2008 at 05:04:32PM -0400, Gleim, Jason wrote: I am using Asterisk and an X101P card as a glorified answering machine. We have a residential PSTN line with about six phones connected to it. Like an answering machine, I want Asterisk answer the line *only* when an incoming call is not answered after four rings. This mostly works. My extensions.conf is at the end of this message. The problem is that Asterisk will sometimes answer the line when someone is already talking on one of the six phones connected to it. Sometimes Asterisk will answer the line and start playing the greeting in the middle of a conversation! This is especially a problem when I am talking Others may wish to chime in and confirm or deny this but the card is probably getting confused by you loading the line with the other phones. I know most of the analog cards I've worked with (which does not include the X101P) really get cranky if there is anything else hanging off that line. The only solution I've seen to the problem is to change things around so that the card is the only thing on the line. The cranky card here is not the issue. It would be the same with any other card. In know you said you haven't switched to IP or FXS but is there a reason why? That would require rewiring. I swapped my X101P for a Linsys SPA3102 some time back. Calls come in to the SPA on its FXO port and get forwarded to Asterisk which then rings the legacy phones on the SPA's FXS port, as the other members of the household (read wife) are used to. All of Asterisk's features are available. If any issues arise, you just pull the power to the SPA and calls just pass through directly to the legacy phones. In addition, the SPA is supposed to pass calls through to the FXS if it loses its registration with Asterisk (eg Asterisk crashes) but I never got this working. One day, I would like to teach Asterisk to behave like an old-fashioned answering machine to allow a second chance to catch calls after the fourth ring but I haven't had the time yet. -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] is there a way
Steve Totaro wrote: My only wish is that Linux had a facility like XP to bridge NICs without running all sorts of commands for brctl. Just a GUI like XP. Last time I setup a bridge in Linux, I had to change many kernel options and rebuild the entire kernel to get bridging working properly. With XP, you just select the NICS, right click and select add to bridge. I've always seen this as (YA) flaw in Windows. Bridging seems to be the lazy man's way of avoiding configuring Windows networking properly (admittedly, this is not a simple task), especially with OpenVPN. Bridging should be left to switches, they're the professionals! For linux, I find that running firestarter, ICS/Firewall is fine, my end game is to get all of my traffic to go over an OpenVPN tunnel at my colo which is the default gateway over OpenVPN. Windows seems to have the easiest method of getting this done. OpenVPN has an option to do just this, redirect-gateway, that will push (almost) all the traffic out through the tunnel. Works the same with any supported OS regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] Text messaging and Asterisk
C. Savinovich wrote: Can somebody please give a pointer to a complete neophyte (like me) on text messaging, what product can I use to send and automatic text message to a cell phone from within the asterisk dialplan? (the part of the dialplan I have down, the part of the text message no) Thanks C. Savinovich I don't use it but on my Asterisk 1.4 slug there was a file /etc/asterisk/fastsms.conf which had info about connecting to SMS services for about 4c per txt. regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] PoE switch recommendations?
Ken D'Ambrosio wrote: Hey, all. We're rolling out VoIP, and I'm wondering about PoE recommendations, as we're going to have to replace our current network equipment. My first inclination would be to just plunk down the cash and do a Cisco system, but I'm relatively certain that would get shot down by finance. Any recommendations for a couple-hundred-port solution with VLANs, PoE, and QoS? Don't care much if it's in a single chassis or not, so long as it has Gbit uplinks. We tried a Linksys SRW208P about 2 years ago but couldn't use it due to the noise from the fans (think jet engines) We settled on the Netgear 16/8 port (8 PoE + 8 non-PoE) switches for the call centre pods, each supporting 5 phones. None have caused problems to date. In the server room, we have Dell PowerConnect 3548P switches which are fully managed driving a mix of phones, access points cameras. They are excellent value, Cisco are way overpriced. regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] OT: Do You Know What the Problem With CDMA is?
Andrew Kohlsmith (lists) wrote: On September 25, 2008 09:01:52 am Dean Collins wrote: Yep you got it world coverage includes all the countries of the world like USA, Canada and Mexico, and not something like USA and 212 other countries globally. BTW I hear that Iraq also now uses CDMA (some senator shoe-horned it into a funding bill for the war that they had to use CDMA to 'support usa businesses'), of course that means that they now use a different handset type to all of their neighbours... though I hear Iran will also be forced to implement CDMA once they are 'liberated' which should be any day now :) That doesn't mean that GSM towers won't be built, it just means that the CDMA towers will be there first. I dunno; GSM 3G is all CDMA tech anyway. Yes, but it is a standard agreed upon by a large number of carriers around the world. Once CDMA has gone the way of the dodo in North America, I really will miss one of my favourite scenes:- Visiting Brit steps off plane and checks phone for messages... Puzzled look appears as they ask Why doesn't my phone work? It worked fine in France/Italy/Germany/Timbuktu. You start to explain about CDMA and their eyes open wide as they realize they have just stepped back into the cellular stone age... regards, Drew ___ -- 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] OT: Do You Know What the Problem With CDMA is?
Andrew Kohlsmith (lists) wrote: On September 25, 2008 10:41:45 am Drew Gibson wrote: Once CDMA has gone the way of the dodo in North America, I really will miss one of my favourite scenes:- Visiting Brit steps off plane and checks phone for messages... Puzzled look appears as they ask Why doesn't my phone work? It worked fine in France/Italy/Germany/Timbuktu. You start to explain about CDMA and their eyes open wide as they realize they have just stepped back into the cellular stone age... You don't have ATT towers near airports? -A Nope, no ATT north of Buffalo. To be honest, it happened a few years ago (~2002). We now have Rogers' towers near airports (and 3G iPhones in stores). Bell Canada and Telus are moving to GSM 3G (side-stepping standard GSM so they don't have to admit their mistakes) regards, Drew ___ -- 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] sip conversations overlapping!!!!
RoLaNd RoLaNd wrote: i appologize for not making myself clear.. i have my asterisk box, connexted to 4 sipura3102.. these sipuras has 4 PSTN lines connected to them through one cable, which has 8 lines inside of it (2 connected to an RJ11 and plugged into its respecitve fxs port in the sipura) Lines plugged into the fxS ports? I hope you have them in the LINE ports (fxO) Are there any telephones plugged directly into the Sipuras? Into the PHONE ports (FXS) my prob is as such: when i call from softphone#1 to sipura #1, sound is pretty good and everything is working perfectly.. though if asterisk recieves a call from another sipura.. lets say its sipura #2, then! i could hear the attendnat answering the incoming phone in my current conversation, and i could hear some1 picking up and answerinfg the call..! if i ask them to hang up! my line breaks as well.. I would double check the wiring of the 8 line cable. 4 POTS lines = 4 pairs of tip ring. Are there some of the tipring pairs mixed up? eg tip from line 1 mixed with ring from line 2, etc. This is the most likely scenario since I can't imagine Asterisk bridging the calls without being asked to. Otherwise, are we still missing something in the topology here? regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] Newbie: Queue and CDR Reporter and Analyser
We use http://www.areski.net/asterisk-stat-v2/about.php http://www.micpc.com/qloganalyzer/ on Asterisk 1.2, don't know how well they work with later versions regards, Drew Mark Hamilton wrote: Doesn't Queuemetrics run on a license basis? Anything else that's probably open source and free? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Faraz Khan Sent: August 13, 2008 8:20 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Newbie: Queue and CDR Reporter and Analyser queuemetrics Lee, John (Sydney) wrote: I am trying to look for a software (open source or proprietory) that could do reporting on both queue and CDR in Asterisk 1.4.* Could someone give me some suggestions? ___ -- 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 -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] sip conversations overlapping!!!!
RoLaNd RoLaNd wrote: Hi all, i'm facing this weird prob...my topology is as such: softphone --- asterisk sipura 3102 sipura 3102 -sipura 3102 -sipura 3102 when am on a call, sometimes when some1 else tries to call out.. i hear the actual tones which ends up preventing the other end from talking to me.. moroever, when some1 calls me through one sipura, while im talking on another... i can hear the attendant welcoming message, then i hear the voice of whoever have picked tht line up..! and if i ask that person to hang up... my line breaks as well..! can any1 help me with this issue! below is my config: How are the analogue phones wired? One phone plugged directly to one 3102 FXS port? or is there common wiring ? Are all the FXO ports connected to telco lines? regards, Drew NOTE: Holding the SHIFT key down whilst typing the first person, singular, pronoun will produce stunningly readable results. Either SHIFT key will do, you can even use the CAPS LOCK key if both of those are broken/can't locate them. You can also use this procedure for the first letter of each sentence, it makes everything much easier to read. -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] implementing an intercom with asterisk
Jonathan Disher wrote: I am looking to replace the phone system at my father's shop with an Asterisk box and some Cisco phones, but one piece of the implementation is tripping me up. He has two buildings (the office, and the shop proper), separated by about 3-400 yards. Currently with the ancient Meridian system installed, there is a paging intercom (to page employees, etc) on a dedicated extension - play a loud tone, then set up a 2 way channel. Anyone got any ideas, hardware wise, on how I might implement this with an Asterisk system? Thanks, and if this isn't appropriate for this list, if anyone has a better destination for the question, Id be quite appreciative. Hi Jon, how is the existing intercom implemented? Is it on the phones or separate speakers? If it is a paging system wired into the Nortel, you may be able to reuse it via an ATA, If it's done through the phones, try http://www.voip-info.org/wiki/index.php?page=Asterisk+Paging+and+Intercom for ideas That is, once you've bridged the ethernet across the 3-400 yards. You can use fibre, wireless or Thicknet. Fibre is the most robust, wireless is the cheapest and Thicknet, well, good luck getting the parts! :-) regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] Limit to the length of string ?
Gordon Henderson wrote: On Tue, 26 Aug 2008, Tilghman Lesher wrote: On Tuesday 26 August 2008 06:49:54 Gordon Henderson wrote: A-Ha... That string is 256 characters long... Now there's a fishy number if ever there was one. So, if this a real limitation? This is 1.2.30 if that makes a difference... Did this limit go away in 1.4 ? Yes, it did. OK. Thanks. Now I guess I have to play the 1.4 lottery :) (or will it get fixed in 1.2.31 ;-) Is there a maximum string length for use with the legacy interface chan_string? Does it depend on the type of cup used? Does styrofoam give better range than paper? regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] Two peers, same IP and port
Chris Hastie wrote: Is it possible to have two peers register to Asterisk from the same IP/port combination? I have a Zoom 5821 two port ATA that can support up to 4 VOIP accounts. I want to use it to provide two different extensions on an Asterisk system. In the past I have configured two port ATAs to use a different SIP local port for each account, but the Zoom unit does not appear to allow the SIP local port to be specified on an account by account basis. Can I get the unit to register two separate accounts on Asterisk from the same port and IP? Hi Chris, from testing I did a year ago with 1.2, I would say that his is not possible. Asterisk was tracking the registration by IP and could not differentiate the accounts by port number alone. regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] Siemens Gigaset IP in USA (S685 IP in particular)
Paul Chambers wrote: For some unfathomable reason, Siemens USA doesn't offer the Gigaset IP range in the U.S. I'm particularly interested in the Gigaset S685 IP. Since it's DECT 6.0, and there's an English (UK) version, I'm thinking it should work just fine, after dealing with the walwart issue (and maybe caller ID signalling). Anyone imported one from the UK and using it in the US? for how long? impressions? anything not working? Have you purchased additional US-spec handsets and used them with the UK basestation? Thanks in advance, Paul The original DECT standard uses 1880-1900MHz, as implemented in Europe. The US FCC designated 1920-1930MHz. This is marketed as DECT 6.0. The FCC might get angry at you for using regular DECT phones in the US. And your neighbours with iPhones (GSM) might also get angry... regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] Asterisk might be dropping RTP packets before reaching eth int?
Do you have a firewall enabled? If so, disable it. Else run /etc/init.d/iptables stop (or equivalent) and try again. regards, Drew Jonathan Miller wrote: We put a 3c509 in, just for posterity, and it did not help the issue. I verified that the NIC is not sharing any interrupts, that there is no excessive disk wait, and that asterisk thinks it is sending the packets. They are simply not making it to the physical interface. Is there somewhere else I can look or something else I can do? How would I go about prioritizing the asterisk process? There's not a lot of processes running besides asterisk itself, dhcpd, tftpd and postfix. Really just kernel stuff after that... This is killing me. Voice drops out for various periods (between 1/2 and 5 seconds) and lost packets do not show up with a rtp rtcp stats... This is weird. Any help you can offer would be appreciated. We spent 6 hours on phone with Digium support yesterday and could not locate an issue within asterisk itself. -Jonathan On Wed, Aug 13, 2008 at 9:08 PM, Steve Totaro [EMAIL PROTECTED] wrote: On Wed, Aug 13, 2008 at 7:40 PM, Jonathan Miller [EMAIL PROTECTED] wrote: From what I can determine while troubleshooting a voice-dropping issue, the Asterisk server in my organization has been dropping RTP packets between the asterisk server process and the network interface. I determined this from an RTP debug that showed packets sent to the phone and packets received from the phone during the entire call. A tcpdump done on the server for the interface that would deliver the packet to the wire does not show the packets. Is there somewhere I can look to resolve this? Something anybody has come across? It is happening frequently and with great discomfort to many users. I had upgraded from 1.2.x to latest 1.4.x in attempts to resolve this. I also disabled a lot of COM/LPT and USB devices in the BIOS to free up some IRQ's. no devices are sharing IRQ's at this point, with I thought might have been part of the issue, but has proved to at least not be directly related. These calls are from a PRI to a Cisco 7940 using SIP. There is a Juniper EX switch between the two. Both sides negotiate at 100Mbps/Full Duplex. I have ruled the switch out of the problem as it's not seeing the packets on the wire when the issue is occuring. Please help or point me to someone that can. -Jonathan [EMAIL PROTECTED] Jonathan, It sounds hardware specific to me. Is this a new install or a new problem? If it is a new problem, then what has changed? Is the NIC in question onboard? What hardware are you using? Brands, MoBo, NIC, etc... If I were you, I would remove or disable the NIC and stick a tried and true old school 3Com NIC in the server and try that. Thanks, Steve Totaro -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] Problem with Aastra 480ci and qualify=yes
James Lamanna wrote: Hi, We have a few Aastra 480ci phones and we've noticed that in order to get the phone to receive a call, qualify must be = no. Apparently the Aastras do not respond to the qualify message (or respond in a way Asterisk doesn't understand) and Asterisk thinks the phone is unreachable. However, this now prevents MWI from working properly on the phones. Does anyone know how to get MWI working without qualify? Or how to get qualify working again with the Aastras? We have a number of 480i and one 480ct all setup with qualify=yes (Asterisk 1.2.24) Our inbound call centre seems to be pretty busy and my own MWI lights up far too often. Never had a problem with either. Which version of firmware? Which version of Asterisk? What's in your sip.conf? What error messages show on the console? Anything relevant in the logs? regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] TDM400P FXO not seeing ringing after software update
Kevin P. Fleming wrote: Drew Gibson wrote: Did anyone find a fix for this (besides down-grading to 1.2.24)? It has been fixed in Subversion already, but a new Zaptel release hasn't been made. That will be done in the next few days. Thanks Kevin, sorry I am unable to test the svn version because the site is (literally) on the other side of the world. On-site visits for screw-ups are rather pricey! :-) No rush for the update, I will keep my eyes open for the new release. regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] Multi-homed Asterisk
Jon Weisman wrote: Hey guys its me again! So I need to setup our Asterisk server with multiple IP providers. The server has two NICs, we have two providers, now we want redundancy! Any guides on how to set this up? We're running Fedora Core 5 w/ Asterisk 1.4. What I would like is that incase one link goes down, or even if the link is showing up, but for some reason its not routing traffic the other link should take over. This is for both inbound and outbound SIP. Now the hard part... I need this to be reliable. Your help is greatly appreciated, Jon I assume it is the Internet connection that you want to make redundant. If you have any inbound connections, you will have to have a router running BGP and your two providers will have to co-operate to get this setup and working properly. We have two connections from the same provider and it did not cost us anything to add BGP support. The router can be a dedicated device such as from Cisco, et al or can be a Linux-based machine running Zebra. There are other budget methods to create a form of redundancy but they have drawbacks (see post from GH) regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] OT: Asterisk on fitPC
You can use an external ATA. I have a Linksys SPA3102 which has 1 FXO (Bell line) port, 1 FXS (phone) port and one Ethernet port to connect via SIP to Asterisk. Alternatively, there are USB connected adapters eg. Xorcom.com but I haven't used them. regards, Drew Mark Hamilton wrote: I love the thin client stuff. It probably looks as big as the Samsung SWA-4000. But in terms of hardware, don't I need a PCI card to get it working? How would that work? Sorry, I have no idea about Asterisk working for home, but just SIP related stuff. :( -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gordon Henderson Sent: August 12, 2008 4:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] OT: Asterisk on fitPC On Tue, 12 Aug 2008, mail-lists wrote: I can't see why not. You should easily have enough power for asterisk. You can probably also run it as your firewall in a home environment thanks to the dual RJ45's I don't know whether or not you can use the built in RJ11 to interface with your POTS line though - maybe someone else could speak to this? Looks like it's rs232, so I suspect not :) However if the AMD Geode 500MHz processor is any good, and I'd expect it to be better than the old 500MHz Via processor I use as my test/development system then you'll be able to run well over a dozen concurrent calls (not transoding) without any issues.. Hm. $300 in the US and the UK disty is selling them for just short of £240, so they can go stuff themselves, low-power or not. (I buy 1GHz systems with 1GB of RAM, running at 15W for half that. No drive though) Gordon Hi, I?d like to install Asterisk at home. But don?t want to use a full blown PC to host it. I was thinking of using fitPC www.fit-pc.com http://www.fit-pc.com to do all the Asterisk work, interfacing with the local Bell Canada line, and using a SIP VoIP line as well. What do you experts think of it? Thanks, Mark. - --- ___ -- 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 ___ -- 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 -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] TDM400P FXO not seeing ringing after software update
Kevin P. Fleming wrote: Tony Mountifield wrote: My guess is that 1.2.24 will work, which was at revision 3842 of the tree (rev 3741 of wctdm.c). The next change to wctdm.c (rev 4126) looks innocuous enough, but the follwing two (revs 4128 and 4132) look likely culprits, from looking at the areas of code that they affect. It is very likely 4128, based on the code it affects and the behavior that is being reported. Please let us know as soon as you can (anyone who has this problem), if reverting r4128 from current Zaptel branch 1.2 SVN solves the problem. Did anyone find a fix for this (besides down-grading to 1.2.24)? regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] comparing pots solutions for asterisk
Eric Fort wrote: I've been looking at various solutions for getting FXS and FXO lines in and out of asterisk. one solution is using TDM-400 cards. Another solution is using the grandstream GXW400x and GXW410x gateways. Cost per port seems lower on the gateways and no pci slot is required. Why would one choose to use the TDM-400 cards? what would be the advantages and disadvantages of each approach? The internal card should give you higher reliability as there are fewer parts and cables although the external gateways could allow you to have redundant servers. External gateways would also be easier to scale when you need more lines. Does anyone have experience with the Grandstream gateways? Are they more reliable than their GXP2000 phones? Can they actually provide support yet? regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] AA50 Failover
Tzafrir Cohen wrote: On Thu, Jul 31, 2008 at 02:10:34PM -0400, Dave Welsh wrote: If I buy two AA50s can I set them up so that everything runs through the first one, but the second one will take over if the first one goes down? I can see the extensions recovering, because they use ethernet, but what about the FSO lines? Is there a way they can be spliced to both AA50s so that no one need to do any emergency rewiring? Thinking aloud: what happens if you just connect th two units on the same phone line? Use one of those fax/modem/phone sharing boxes that blocks the other devices when one is off hook. You basically need a way to prevent the slave unit from answering calls if the master is alive. Set the primary unit to pick up on, say, 2 rings and the secondary unit to pick up on 4. If the primary fails, it won't pick up the line and the secondary takes over. You could try having your phones register to both but I'm not sure how that would work regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] AA50 Failover
Brent Torrenga wrote: Set the primary unit to pick up on, say, 2 rings and the secondary unit to pick up on 4. If the primary fails, it won't pick up the line and the secondary takes over. You could try having your phones register to both but I'm not sure how that would work regards, Drew That's exactly how I have things setup. Phones register to both servers. The backup server doesn't do much - only answers incoming calls if the ring goes on for too long. Keep in mind that you may want to configure the backup server to not use the lines for placing outbound calls - maybe setup a SIP trunk for that. Otherwise, when the primary server comes back up, the two will be talking over each other trying to use the same lines. --Brent That's where the fax/modem sharing box comes in, if one device port has the line, all other device ports are blocked. Downside is that this will also cause dead air on an outbound call or two until things settle down again. regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] OT - How to test tftp for phones provisioning
Olivier wrote: tftp is the client, do you have it installed ?... example: # tftp hostname tftp get /srv/tftp/foo.txt tftp ^D # cat foo.txt ... That was exactly what I was after : I installed tftp on my Ubuntu system and checked Debian tftp server Things to check: is /srv/tftp the tftp directory or is it the os filesystem directory where the tftp root resides ? Also, the tftp daemon in CentOS is started by xinetd and can be invoked with extra -v flags so as to increase logging verbosity. Check your dist. This may help... Yes, that's the next step. I could see a tftp service is running ok on my server and I need to increase its logs to pinpoint root cause. Olivier, also check that you don't have any firewalls in the way, especially if there is nothing in the logs. Turn them off for testing, on both server and CLIENT. I found out about the client f/w blocking tftp the hard way! :-) regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] TDM400P FXO not seeing ringing after software update
Tony Mountifield wrote: A customer has an Asterisk box with two TDM400P cards, running [EMAIL PROTECTED] 2.8, which has Asterisk 1.2.7.1 and Zaptel 1.2.5. This has been running ok for a while, with only some small issues. They have FXO ports going to analogue POTS (UK standard) lines, and SIP phones for extensions. I just tried updating Zaptel to 1.2.26 and Asterisk to 1.2.30.1, and although the update appeared to go ok, the system would no longer detect incoming calls. It appeared not to see the ringing signal. Outgoing calls over the PSTN lines still worked fine. Does anyone know if there have been changes in the 1.2 series that affects ring detection on the TDM400P FXO ports? Any critical settings in zaptel.conf or zapata.conf? For now, I've had to revert to the original versions, in which ringing once again works fine. Cheers Tony YES! I have a 4xFXO board in Singapore. Having tried many various settings, etc., I had assumed it was the I18N of the ringtones and signalling. Does anyone know at which rev the issue appeared? regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] TDM400P FXO not seeing ringing after software update
Drew Gibson wrote: Tony Mountifield wrote: A customer has an Asterisk box with two TDM400P cards, running [EMAIL PROTECTED] 2.8, which has Asterisk 1.2.7.1 and Zaptel 1.2.5. This has been running ok for a while, with only some small issues. They have FXO ports going to analogue POTS (UK standard) lines, and SIP phones for extensions. I just tried updating Zaptel to 1.2.26 and Asterisk to 1.2.30.1, and although the update appeared to go ok, the system would no longer detect incoming calls. It appeared not to see the ringing signal. Outgoing calls over the PSTN lines still worked fine. Does anyone know if there have been changes in the 1.2 series that affects ring detection on the TDM400P FXO ports? Any critical settings in zaptel.conf or zapata.conf? For now, I've had to revert to the original versions, in which ringing once again works fine. Cheers Tony YES! I have a 4xFXO board in Singapore. Having tried many various settings, etc., I had assumed it was the I18N of the ringtones and signalling. Does anyone know at which rev the issue appeared? Tzafrir, Tony tried Zaptel 1.2.26, and I was at 1.2.25, both were failing to pickup. I just downgraded to 1.2.24 and the system is now picking up incoming calls. Asterisk remained at 1.2.28. From the Changelog file I found a possibly relevant change... 2008-04-04 04:29 + [r4126-4132] sruffell [EMAIL PROTECTED]: ... involved wctdm.c and fxo_modes.h. Merges with 1.4 code? Any use? regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] need ata suggestion
randulo wrote: On Wed, Jun 18, 2008 at 5:05 PM, Tilghman Lesher [EMAIL PROTECTED] wrote: those devices, a tinfoil hat would block the interference (as long as it contains no holes). How can you put holes in interference? ... and how many would it take to fill the Albert Hall? -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] fxotune question
Drew Gibson wrote: Tilghman Lesher wrote: My fxotune.conf:- 13=7,255,251,251,2,255,255,1,255 14=9,254,251,255,2,0,1,0,0 15=9,254,251,255,2,0,1,0,0 16=5,0,0,0,0,0,0,0,0 17=9,254,251,255,2,0,1,0,0 18=9,254,251,255,2,0,1,0,0 19=9,254,251,255,2,0,1,0,0 20=9,254,251,255,2,0,1,0,0 21=9,254,251,255,2,0,1,0,0 22=9,254,251,255,2,0,1,0,0 23=9,254,251,255,2,0,1,0,0 24=9,254,251,255,2,0,1,0,0 regards, Drew Thanks Tilghman, I will try flipping one of the lines at the weekend, can't touch it during the week. Our rxgain was raised from 0.0 to 2.0 at the end of February after complaints that CC staff couldn't hear customers but the 'static' issues pre-date that change. regards, Drew Hi Tilghman, I thought I had found something, all of the lines were patched in with Cat 5 patch cords except Port 16 which had a telephone cable (which would flip the polarity). After changing all the patch cables to telephone type, I re-ran fxotune but the results were almost identical to the original settings. new fxotune.conf:- 13=7,255,251,251,2,255,255,1,255 14=9,254,251,255,2,0,1,0,0 15=6,1,254,253,0,255,0,0,0 16=5,0,0,0,0,0,0,0,0 17=9,254,251,255,2,0,1,0,0 18=9,254,251,255,2,0,1,0,0 19=9,254,251,255,2,0,1,0,0 20=7,255,251,251,2,255,255,1,255 21=9,254,251,255,2,0,1,0,0 22=9,254,251,255,2,0,1,0,0 23=6,1,254,253,0,255,0,0,0 24=9,254,251,255,2,0,1,0,0 These results are very similar to John Morey's, eg. not much changed after reversing polarity. The one line that shows zero's all the way across is the line with our DSL backup line on it. Is it the filter giving these results or does the telco take more care balancing lines for DSL? Shouldn't I have seen a major change of settings after changing polarity for tipring? regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] Asterisk not picking up incoming calls from TDM400P
Hi, I am having some issues with a new server install in Singapore. Outbound calls work fine. Inbound calls are not picked up by Asterisk. Zaptel 1.2.25 and Asterisk 1.2.28 both built from source. libpri installed wctdm and zaptel load without error Jun 6 23:34:03 fs01 kernel: [211138.372933] Zapata Telephony Interface Registered on major 196 Jun 6 23:34:03 fs01 kernel: [211138.372937] Zaptel Version: 1.2.25 Jun 6 23:34:03 fs01 kernel: [211138.372943] Zaptel Echo Canceller: KB1 Jun 6 23:34:03 fs01 kernel: [211138.383639] Freshmaker version: 73 Jun 6 23:34:03 fs01 kernel: [211138.384053] Freshmaker passed register test Jun 6 23:34:04 fs01 kernel: [211139.076180] Module 0: Installed -- AUTO FXO (FCC mode) Jun 6 23:34:04 fs01 kernel: [211139.275847] Module 1: Installed -- AUTO FXO (FCC mode) Jun 6 23:34:04 fs01 kernel: [211139.475514] Module 2: Installed -- AUTO FXO (FCC mode) Jun 6 23:34:05 fs01 kernel: [211139.675182] Module 3: Installed -- AUTO FXO (FCC mode) Jun 6 23:34:05 fs01 kernel: [211139.682518] Found a Wildcard TDM: Wildcard TDM400P REV I (4 modules) Jun 6 23:34:08 fs01 kernel: [211142.686305] Registered tone zone 18 (Singapore) Jun 6 23:34:14 fs01 kernel: [211149.412565] -- Setting echo registers: Jun 6 23:34:14 fs01 kernel: [211149.412990] -- Set echo registers successfully Jun 6 23:34:14 fs01 kernel: [211149.413005] -- Setting echo registers: Jun 6 23:34:14 fs01 kernel: [211149.413426] -- Set echo registers successfully Jun 6 23:34:14 fs01 kernel: [211149.413435] -- Setting echo registers: Jun 6 23:34:14 fs01 kernel: [211149.413848] -- Set echo registers successfully Jun 6 23:34:14 fs01 kernel: [211149.413861] -- Setting echo registers: Jun 6 23:34:14 fs01 kernel: [211149.414276] -- Set echo registers successfully Jun 6 23:34:28 fs01 kernel: [211163.107241] Zaptel Transcoder support loaded The one and only POTS line has been tuned with fxotune and fxotune -s has been run. ztmonitor shows incoming ring (volume peaks at 3-4000) No, nothing, nadda, zero response from Asterisk. Can anyone suggest a tool to help find the gap between zaptel and Asterisk? regards, Drew zaptel.conf:- fxsks=1-4 ;(have tried fxsls=1-4 but no difference) loadzone=sg defaultzone=sg zapata.conf:- [channels] group=1 context=incoming callprogress=no rxgain=2.0 txgain=0.0 immediate=no usecallerid=yes callerid=asreceived signalling=fxs_ks relaxdtmf=yes pickupgroup=1 faxdetect=incoming channel = 1 indications.conf:- [general] country=sg Other countries snipped --- [sg] ; Singapore section borrowed from http://csusap.csu.edu.au/~whaase01/itc308/asterisk/indications.conf description = Singapore ; Singapore ; Reference: http://www.ida.gov.sg/idaweb/doc/download/I397/ida_ts_pstn1_i4r2.pdf ; Frequency specs are: 425 Hz +/- 20Hz; 24 Hz +/- 2Hz; modulation depth 100%; SIT +/- 50Hz ringcadence = 400,200,400,2000 dial= 425 ring= 425*24/400,0/200,425*24/400,0/2000 ; modulation should be 100%, not 90% busy= 425/750,0/750 congestion = 425/250,0/250 callwaiting = 425*24/300,0/200,425*24/300,0/3200 stutter = !425/200,!0/200,!425/600,!0/200,!425/200,!0/200,!425/600,!0/200,!425/200,!0/200,!425/600,!0/200,!425/200,!0/200,!425/600,!0/200,425 info= 950/330,1400/330,1800/330,0/1000 ; not currently in use acc. to reference dialrecall = 425*24/500,0/500,425/500,0/2500; unspecified in IDA reference, use repeating Holding Tone A,B record = 1400/500,0/15000 ; unspecified in IDA reference, use 0.5s tone every 15s ; additionally defined in reference nutone = 425/2500,0/500 intrusion = 425/250,0/2000 warning = 425/624,0/4376 ; end of period tone, warning acceptance = 425/125,0/125 holdinga= !425*24/500,!0/500 ; followed by holdingb holdingb= !425/500,!0/2500 [incoming] ; Added Answer statement for troubleshooting exten = s,1,Answer() include = office-incoming include = internal [office-incoming] ; OANDA Office incoming calls ignorepat = 9 exten = s,1,Wait,1 ; Waiting a little longer for CID exten = s,n,Answer ; Answer the line exten = s,n,Set(TIMEOUT(digit)=2) exten = s,n,Set(TIMEOUT(response)=10) exten = s,n,Set(TIMEOUT(absolute)=14400) ;exten = s,n,Goto(ivr_menu) - rest of extensions.conf snipped - -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] Asterisk not picking up incoming calls from TDM400P
Nope, didn't help. Doesn't the context declaration come *before* the channel assignment in zapata.conf? It's working that way in our main Asterisk server. regards, Drew Tim Nelson wrote: It looks like you may be missing a context declaration right after your channel = 1 line. Try adding context=incoming right after that. Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105 - Original Message - From: Drew Gibson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, June 6, 2008 12:37:47 PM GMT -06:00 US/Canada Central Subject: [asterisk-users] Asterisk not picking up incoming calls from TDM400P Hi, I am having some issues with a new server install in Singapore. Outbound calls work fine. Inbound calls are not picked up by Asterisk. Zaptel 1.2.25 and Asterisk 1.2.28 both built from source. libpri installed wctdm and zaptel load without error Jun 6 23:34:03 fs01 kernel: [211138.372933] Zapata Telephony Interface Registered on major 196 Jun 6 23:34:03 fs01 kernel: [211138.372937] Zaptel Version: 1.2.25 Jun 6 23:34:03 fs01 kernel: [211138.372943] Zaptel Echo Canceller: KB1 Jun 6 23:34:03 fs01 kernel: [211138.383639] Freshmaker version: 73 Jun 6 23:34:03 fs01 kernel: [211138.384053] Freshmaker passed register test Jun 6 23:34:04 fs01 kernel: [211139.076180] Module 0: Installed -- AUTO FXO (FCC mode) Jun 6 23:34:04 fs01 kernel: [211139.275847] Module 1: Installed -- AUTO FXO (FCC mode) Jun 6 23:34:04 fs01 kernel: [211139.475514] Module 2: Installed -- AUTO FXO (FCC mode) Jun 6 23:34:05 fs01 kernel: [211139.675182] Module 3: Installed -- AUTO FXO (FCC mode) Jun 6 23:34:05 fs01 kernel: [211139.682518] Found a Wildcard TDM: Wildcard TDM400P REV I (4 modules) Jun 6 23:34:08 fs01 kernel: [211142.686305] Registered tone zone 18 (Singapore) Jun 6 23:34:14 fs01 kernel: [211149.412565] -- Setting echo registers: Jun 6 23:34:14 fs01 kernel: [211149.412990] -- Set echo registers successfully Jun 6 23:34:14 fs01 kernel: [211149.413005] -- Setting echo registers: Jun 6 23:34:14 fs01 kernel: [211149.413426] -- Set echo registers successfully Jun 6 23:34:14 fs01 kernel: [211149.413435] -- Setting echo registers: Jun 6 23:34:14 fs01 kernel: [211149.413848] -- Set echo registers successfully Jun 6 23:34:14 fs01 kernel: [211149.413861] -- Setting echo registers: Jun 6 23:34:14 fs01 kernel: [211149.414276] -- Set echo registers successfully Jun 6 23:34:28 fs01 kernel: [211163.107241] Zaptel Transcoder support loaded The one and only POTS line has been tuned with fxotune and fxotune -s has been run. ztmonitor shows incoming ring (volume peaks at 3-4000) No, nothing, nadda, zero response from Asterisk. Can anyone suggest a tool to help find the gap between zaptel and Asterisk? regards, Drew zaptel.conf:- fxsks=1-4 ;(have tried fxsls=1-4 but no difference) loadzone=sg defaultzone=sg zapata.conf:- [channels] group=1 context=incoming callprogress=no rxgain=2.0 txgain=0.0 immediate=no usecallerid=yes callerid=asreceived signalling=fxs_ks relaxdtmf=yes pickupgroup=1 faxdetect=incoming channel = 1 indications.conf:- [general] country=sg Other countries snipped --- [sg] ; Singapore section borrowed from http://csusap.csu.edu.au/~whaase01/itc308/asterisk/indications.conf description = Singapore ; Singapore ; Reference: http://www.ida.gov.sg/idaweb/doc/download/I397/ida_ts_pstn1_i4r2.pdf ; Frequency specs are: 425 Hz +/- 20Hz; 24 Hz +/- 2Hz; modulation depth 100%; SIT +/- 50Hz ringcadence = 400,200,400,2000 dial= 425 ring= 425*24/400,0/200,425*24/400,0/2000 ; modulation should be 100%, not 90% busy= 425/750,0/750 congestion = 425/250,0/250 callwaiting = 425*24/300,0/200,425*24/300,0/3200 stutter = !425/200,!0/200,!425/600,!0/200,!425/200,!0/200,!425/600,!0/200,!425/200,!0/200,!425/600,!0/200,!425/200,!0/200,!425/600,!0/200,425 info= 950/330,1400/330,1800/330,0/1000 ; not currently in use acc. to reference dialrecall = 425*24/500,0/500,425/500,0/2500; unspecified in IDA reference, use repeating Holding Tone A,B record = 1400/500,0/15000 ; unspecified in IDA reference, use 0.5s tone every 15s ; additionally defined in reference nutone = 425/2500,0/500 intrusion = 425/250,0/2000 warning = 425/624,0/4376 ; end of period tone, warning acceptance = 425/125,0/125 holdinga= !425*24/500,!0/500 ; followed by holdingb holdingb= !425/500,!0/2500 [incoming] ; Added Answer statement for troubleshooting exten = s,1,Answer() include = office-incoming include = internal [office-incoming] ; OANDA Office incoming calls ignorepat = 9 exten = s,1,Wait,1 ; Waiting a little longer for CID exten = s
Re: [asterisk-users] Asterisk not picking up incoming calls from TDM400P
It's the thought that counts! :-) Tim Nelson wrote: You are correct... my mistake. :-/ Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105 - Original Message - From: Drew Gibson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, June 6, 2008 1:36:13 PM GMT -06:00 US/Canada Central Subject: Re: [asterisk-users] Asterisk not picking up incoming calls from TDM400P Nope, didn't help. Doesn't the context declaration come *before* the channel assignment in zapata.conf? It's working that way in our main Asterisk server. regards, Drew Tim Nelson wrote: It looks like you may be missing a context declaration right after your channel = 1 line. Try adding context=incoming right after that. Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105 - Original Message - From: Drew Gibson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, June 6, 2008 12:37:47 PM GMT -06:00 US/Canada Central Subject: [asterisk-users] Asterisk not picking up incoming calls from TDM400P Hi, I am having some issues with a new server install in Singapore. Outbound calls work fine. Inbound calls are not picked up by Asterisk. Zaptel 1.2.25 and Asterisk 1.2.28 both built from source. libpri installed wctdm and zaptel load without error Jun 6 23:34:03 fs01 kernel: [211138.372933] Zapata Telephony Interface Registered on major 196 Jun 6 23:34:03 fs01 kernel: [211138.372937] Zaptel Version: 1.2.25 Jun 6 23:34:03 fs01 kernel: [211138.372943] Zaptel Echo Canceller: KB1 Jun 6 23:34:03 fs01 kernel: [211138.383639] Freshmaker version: 73 Jun 6 23:34:03 fs01 kernel: [211138.384053] Freshmaker passed register test Jun 6 23:34:04 fs01 kernel: [211139.076180] Module 0: Installed -- AUTO FXO (FCC mode) Jun 6 23:34:04 fs01 kernel: [211139.275847] Module 1: Installed -- AUTO FXO (FCC mode) Jun 6 23:34:04 fs01 kernel: [211139.475514] Module 2: Installed -- AUTO FXO (FCC mode) Jun 6 23:34:05 fs01 kernel: [211139.675182] Module 3: Installed -- AUTO FXO (FCC mode) Jun 6 23:34:05 fs01 kernel: [211139.682518] Found a Wildcard TDM: Wildcard TDM400P REV I (4 modules) Jun 6 23:34:08 fs01 kernel: [211142.686305] Registered tone zone 18 (Singapore) Jun 6 23:34:14 fs01 kernel: [211149.412565] -- Setting echo registers: Jun 6 23:34:14 fs01 kernel: [211149.412990] -- Set echo registers successfully Jun 6 23:34:14 fs01 kernel: [211149.413005] -- Setting echo registers: Jun 6 23:34:14 fs01 kernel: [211149.413426] -- Set echo registers successfully Jun 6 23:34:14 fs01 kernel: [211149.413435] -- Setting echo registers: Jun 6 23:34:14 fs01 kernel: [211149.413848] -- Set echo registers successfully Jun 6 23:34:14 fs01 kernel: [211149.413861] -- Setting echo registers: Jun 6 23:34:14 fs01 kernel: [211149.414276] -- Set echo registers successfully Jun 6 23:34:28 fs01 kernel: [211163.107241] Zaptel Transcoder support loaded The one and only POTS line has been tuned with fxotune and fxotune -s has been run. ztmonitor shows incoming ring (volume peaks at 3-4000) No, nothing, nadda, zero response from Asterisk. Can anyone suggest a tool to help find the gap between zaptel and Asterisk? regards, Drew zaptel.conf:- fxsks=1-4 ;(have tried fxsls=1-4 but no difference) loadzone=sg defaultzone=sg zapata.conf:- [channels] group=1 context=incoming callprogress=no rxgain=2.0 txgain=0.0 immediate=no usecallerid=yes callerid=asreceived signalling=fxs_ks relaxdtmf=yes pickupgroup=1 faxdetect=incoming channel = 1 indications.conf:- [general] country=sg Other countries snipped --- [sg] ; Singapore section borrowed from http://csusap.csu.edu.au/~whaase01/itc308/asterisk/indications.conf description = Singapore ; Singapore ; Reference: http://www.ida.gov.sg/idaweb/doc/download/I397/ida_ts_pstn1_i4r2.pdf ; Frequency specs are: 425 Hz +/- 20Hz; 24 Hz +/- 2Hz; modulation depth 100%; SIT +/- 50Hz ringcadence = 400,200,400,2000 dial= 425 ring= 425*24/400,0/200,425*24/400,0/2000 ; modulation should be 100%, not 90% busy= 425/750,0/750 congestion = 425/250,0/250 callwaiting = 425*24/300,0/200,425*24/300,0/3200 stutter = !425/200,!0/200,!425/600,!0/200,!425/200,!0/200,!425/600,!0/200,!425/200,!0/200,!425/600,!0/200,!425/200,!0/200,!425/600,!0/200,425 info= 950/330,1400/330,1800/330,0/1000 ; not currently in use acc. to reference dialrecall = 425*24/500,0/500,425/500,0/2500; unspecified in IDA reference, use repeating Holding Tone A,B record = 1400/500,0/15000 ; unspecified in IDA reference, use 0.5s tone every 15s ; additionally defined in reference nutone = 425/2500,0/500 intrusion = 425/250,0/2000 warning
[asterisk-users] Anyone using zaptel analogue hardware in Singapore?
If anyone is using or have experience of Asterisk with zaptel hardware on a POTS line in Singapore? If so, would you mind sharing your zaptel and zapata configs? I'm having a little trouble getting my new server to answer calls (outbound is working, see thread Asterisk not picking up incoming calls...) regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] fxotune question
Tilghman Lesher wrote: On Wednesday 04 June 2008 22:02:19 John Morey wrote: Hello, I've run fxotune at different times but continue to get what seem to be strange numbers in /etc/fxotune.conf. It ends up with: 5=7,255,251,251,2,255,255,1,255 6=7,255,251,251,2,255,255,1,255 7=7,255,251,251,2,255,255,1,255 8=9,2,250,253,4,252,0,255,255 9=4,0,0,0,0,0,0,0,0 10=5,0,0,0,0,0,0,0,0 11=0,0,0,0,0,0,0,0,0 12=0,0,0,0,0,0,0,0,0 ports 5-10 have lines hooked up to them. The first four lines seem strange when compaired to what others have posted and what ports 9 and 10 have. Also if I'm reading things right my echo ratios seem to be very high. Running fxotune -d -b 5 -w 1004 gives the following: Dumping module /dev/zap/5 echo ratio = 0.1759 (1960.0 / 11145.0) Which I read to be over 17%. This seems crazy. Am I reading this right? Where should I start to look for problems? You might check to see if the tip and ring are reversed in your wiring. That can frequently cause weird echo problems. Which ports would you expect to be reversed? 5-8 or 9-10? I have similar settings in my fxotune.conf for a TDM2400P and I'm getting complaints of static on the line that I suspect are related to an overtaxed h/w echo canceller My fxotune.conf:- 13=7,255,251,251,2,255,255,1,255 14=9,254,251,255,2,0,1,0,0 15=9,254,251,255,2,0,1,0,0 16=5,0,0,0,0,0,0,0,0 17=9,254,251,255,2,0,1,0,0 18=9,254,251,255,2,0,1,0,0 19=9,254,251,255,2,0,1,0,0 20=9,254,251,255,2,0,1,0,0 21=9,254,251,255,2,0,1,0,0 22=9,254,251,255,2,0,1,0,0 23=9,254,251,255,2,0,1,0,0 24=9,254,251,255,2,0,1,0,0 regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] fxotune question
Tilghman Lesher wrote: On Thursday 05 June 2008 09:50:05 Eric ManxPower Wieling wrote: Echo Canceler Freak Out, this happens when the rxgain is too high and the echo canceler freaks out. Some users describe it as screeching, feedback, static, or other useless terms. If users report static on a system where there cannot be static (all digital, PRI, SIP, etc), you might be experiencing ECFO. Considering he's using an analog card, that last statement does not apply. Drew Gibson wrote: Tilghman Lesher wrote: On Wednesday 04 June 2008 22:02:19 John Morey wrote: Hello, I've run fxotune at different times but continue to get what seem to be strange numbers in /etc/fxotune.conf. It ends up with: 5=7,255,251,251,2,255,255,1,255 6=7,255,251,251,2,255,255,1,255 7=7,255,251,251,2,255,255,1,255 8=9,2,250,253,4,252,0,255,255 9=4,0,0,0,0,0,0,0,0 10=5,0,0,0,0,0,0,0,0 11=0,0,0,0,0,0,0,0,0 12=0,0,0,0,0,0,0,0,0 ports 5-10 have lines hooked up to them. The first four lines seem strange when compaired to what others have posted and what ports 9 and 10 have. Also if I'm reading things right my echo ratios seem to be very high. Running fxotune -d -b 5 -w 1004 gives the following: Dumping module /dev/zap/5 echo ratio = 0.1759 (1960.0 / 11145.0) Which I read to be over 17%. This seems crazy. Am I reading this right? Where should I start to look for problems? You might check to see if the tip and ring are reversed in your wiring. That can frequently cause weird echo problems. Which ports would you expect to be reversed? 5-8 or 9-10? I have similar settings in my fxotune.conf for a TDM2400P and I'm getting complaints of static on the line that I suspect are related to an overtaxed h/w echo canceller My fxotune.conf:- 13=7,255,251,251,2,255,255,1,255 14=9,254,251,255,2,0,1,0,0 15=9,254,251,255,2,0,1,0,0 16=5,0,0,0,0,0,0,0,0 17=9,254,251,255,2,0,1,0,0 18=9,254,251,255,2,0,1,0,0 19=9,254,251,255,2,0,1,0,0 20=9,254,251,255,2,0,1,0,0 21=9,254,251,255,2,0,1,0,0 22=9,254,251,255,2,0,1,0,0 23=9,254,251,255,2,0,1,0,0 24=9,254,251,255,2,0,1,0,0 regards, Drew Thanks Tilghman, I will try flipping one of the lines at the weekend, can't touch it during the week. Our rxgain was raised from 0.0 to 2.0 at the end of February after complaints that CC staff couldn't hear customers but the 'static' issues pre-date that change. regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] Best Linux distribution to use in Asterisk server
Tzafrir Cohen wrote: On Fri, May 09, 2008 at 11:50:59AM -0400, Drew Gibson wrote: equis software wrote: Hi, I allways use Gentoo y my Asterisk servers and work well, but what do you think about to use Ubuntu or another distibution?? Thanks I have run Asterisk on several Fedora versions, Debian, Unslung (on the NSLU2 or Slug) and recently Ubuntu. My most critical servers are on Debian and new ones will be on Ubuntu LTS. I have had very few OS specific issues. I have always built from source except on the Slug but I noticed that Ubuntu has it in the apt repository which would be a great convenience if you are new to Linux or manage a lot of servers. Now that you mention it, are packages of that distribution really maintained? The recent volnurability of AST-2008-006 is a good test case for that. If affects both 1.2 and 1.4 . The annoncement by Digium: http://downloads.digium.com/pub/asa/AST-2008-006.html As with the previous ones, the text is quite clear about the fix. backporting that patch to a slightly older version is not that tricky (and it is something that a distribution package maintainer is used to doing anyway). So what about updates? The optware feed for Unslung (on the NSLU2) is up-to-date on Asterisk 1.4 with 1.4.19.1 but a rev or two behind on 1.2 with 1.2.24 The LWN page for this advisory only lists Fedora and Debian: http://lwn.net/Articles/280318/ Response ime in both was quite reasonable. LWN also tracks adsisories from various other distributions. You can see the list in http://lwn.net/Alerts/ . The following other distributions have 'asterisk' packages: * Gentoo * Mandriva (??? - probably only in contrib and is unsupported) * rPath (Not sure. See below about AstriskNOW) * SUSE * Ubuntu (the package is in 'universe', and not officially supported) The issue is listed as corrected in AsteriskNOW 1.0.3, but the latest version available for download is 1.0.2 . If I read rpath's repository page correctly, then the most recently released version of Asterisk is 1.4.17-2 , from Feb-2008 and thus does not contain this fix. To see the versions of packages i Ubuntu: http://packages.ubuntu.com/asterisk As you can see, both Hardy and the development distribution (Interpid) include the same version of the package. As you can see from following the changelog link: http://changelogs.ubuntu.com/changelogs/pool/universe/a/asterisk/asterisk_1.4.17~dfsg-2ubuntu1/changelog The security issues of 1.4.18.1 were backported to that 1.4.17 package. But nothing about the recent advisory. The Gentoo port is basically where the Ubuntu package is: missing only the last one: http://packages.gentoo.org/package/asterisk The FreeBSD port has not been updated yet. It is still at 1.4.18, and no sign of backported fixes: http://www.freebsd.org/cgi/cvsweb.cgi/ports/net/asterisk/ OpenBSD port was updated pretty fast (by upgrading to asterisk 1.4.19.1) http://www.openbsd.org/cgi-bin/cvsweb/ports/telephony/asterisk/ I don't know where to look for in other distributions. -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] Zap Channels Collide (Incoming Outgoing)
I think the scary thing is that, for most people, basic knowledge of telephony was almost impossible to come by outside the opaque and secretive world of telco. That is until Asterisk came along! Perhaps there should be a regulatory requirement to read The Future of Telephony, cover to cover, before installing any Asterisk system! :-) http://www.asteriskdocs.org/ regards, Drew Al Baker wrote: I know that everyone has gaps in their knowledge, but I am just staggered that systems are being sold/deployed with such fundamental TELCO workings not being understood. Frightening. C. Chad Wallace wrote: At 5:22 PM on 08 May 2008, Forrest Beck wrote: I have a client that is using the Sangoma A200DE with two phone lines attached. The problem is: They use their phone (Grandstream GXP2020) to dial out of the system. Instead of getting ringing, there is someone on the other end of the line that happened to dial in at the exact same moment. So now they are stuck talking with this person, instead of the one the originally called. The ZAP channels are in a dial plan context that instructs it to just dial the office phones. [zap1] exten = s,1,Dial(SIP/1001SIP/1002SIP/1003) exten = s,n,Voicemail([EMAIL PROTECTED]) Anyone know how to get around this? This is known in the telephony world as glare, and there's not much you can do about it, especially if you only have one line. If you have multiple lines on an over-ring (or hunt group or whatever you call it), the best thing to do is find out which way the telco assigns calls to those lines wrt how they are assigned to the Asterisk box. And then allocate outgoing calls in the other direction. On our installation, the calls are allocated from the first FXO port (Zap/25) up. So we set Asterisk to dial out starting from the last FXO port in the group by calling Dial(Zap/G2) (capital G means dial down from last, lowercase g means dial up from first). That minimizes glare. But, as I said before, if you only have one line, you can't do that... ___ -- 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 -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] Zap Channels Collide (Incoming Outgoing)
Eric Wieling wrote: Drew Gibson wrote: I think the scary thing is that, for most people, basic knowledge of telephony was almost impossible to come by outside the opaque and secretive world of telco. That is until Asterisk came along! Perhaps there should be a regulatory requirement to read The Future of Telephony, cover to cover, before installing any Asterisk system! :-) http://www.asteriskdocs.org/ People that try to wing it and install Asterisk when they don't know telecom just gives people a bad impression of Asterisk and VoIP in general. This helps nobody except the pocketbook of the consultant. but how else do they learn? -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] Best Linux distribution to use in Asterisk server
equis software wrote: Hi, I allways use Gentoo y my Asterisk servers and work well, but what do you think about to use Ubuntu or another distibution?? Thanks I have run Asterisk on several Fedora versions, Debian, Unslung (on the NSLU2 or Slug) and recently Ubuntu. My most critical servers are on Debian and new ones will be on Ubuntu LTS. I have had very few OS specific issues. I have always built from source except on the Slug but I noticed that Ubuntu has it in the apt repository which would be a great convenience if you are new to Linux or manage a lot of servers. For a newbie, I would recommend an Ubuntu LTS release. Pick the distro you are most comfortable with. regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] VOICEMAIL OPTIONS help needed
Steve Johnson wrote: Hi everyone, We have a particular user on our Asterisk 1.4.x system who always listens to his voicemail messages via email. - Is there some way to send the voicemail ONLY to email and not retain them on the phone? - Alternatively, can the voicemail system only keep, say, just the last 10 messages (as backup in case of email delivery failure or a message getting deleted in email accidentally before it is heard), purging out the oldest when a new one is received? (If we set the option maxmsg=10 on his mailbox in voicemail.conf, I think it will stop accepting voicemails after 10 messages, not turf the oldest one and accept a new one in its place). Everyone else uses the normal voicemail options on their phones, so the solution should be just for this single user. Thanks for any suggestions. S. from voicemail.conf (1.2.24):- ; delete=yes; After notification, the voicemail is deleted from the server. [per-mailbox only] regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] New generic sounds
Tilghman Lesher wrote: We're about to do another batch of sounds, and I see by my word count that we have some extra time left over. So, suggestions will be entertained for additional prompts in English, Spanish, or French. The only rules are: 1) the prompts have to be generic to Asterisk. No Welcome to so-and-so's business unless the business is fake and the prompt is funny. 2) The prompt may not be profane. Our professional speakers do have a sense of humor, but there are some things they just will not say. I'll open it to the floor now, with the caveat that since Digium is paying for the recording session, it maintains final editorial approval over which sounds are selected. For those providers that want to show they are a SERIOUS phone company. We don't care. We don't have to. (snort) We're the Phone Company! No, no, no, you're dealing with the telephone company. We are not subject to city, state, or federal legislation. We are omnipotent. Sorry?... What was that Ernestine? Oh, it seems they've already been done! regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] Simple queue announcements
Doug Lytle wrote: Chris Bagnall wrote: Greetings list, I've been playing around with queues on an old asterisk 1.2 box at a customer's site. They want to be able to add really simple queue announcements every minute, along the following lines: sorry for the delay, someone will be with you shortly. I believe you'll need to migrate them to 1.4. Doug Works fine for me on 1.2.24 ... from queues.conf:- periodic-announce = Custom/periodic-fxqueue01 periodic-announce-frequency=90 plays a brief sorry,etc, press any key to leave a message or continue to hold for next available etc, etc regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] queue logging
Flash Operator Panel http://www.asternic.org/ regards, Drew Arjan Kroon | Mobillion wrote: Hi, I’m not looking for a programma that show the queue logging. But is there a way to check during a call, which member is connected to the caller. Kind Regard, Arjan Kroon * From: * [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] ] *On Behalf Of *Scott Wolfe *Sent:* woensdag 9 april 2008 17:19 *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] queue logging You could ASTassistant to see this. Its Freeware. www.astassistant.com http://www.astassistant.com - Original Message - * From: * Arjan Kroon | Mobillion mailto:[EMAIL PROTECTED] * To: * Asterisk Users Mailing List - Non-Commercial Discussion mailto:asterisk-users@lists.digium.com * Sent: * Wednesday, April 09, 2008 1:01 AM * Subject: * [asterisk-users] queue logging Hi, I’ using with asterisk a queue with tree members and round robin. When a caller enters this queue and it is connecting to one of the members, is there a possibility to see which member the caller is connected to? And is there a way to see in de application to see if the connection from the caller to one of the members was successful of not successful? I know you can see it in de queue. log. But I want to know if I can see it also in the hangup (h) in de application? Kind Regards ___ -- 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 -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] question about queue
BJ Weschke wrote: Rilawich Ango wrote: Thanks. I have checked that the queue.conf. I keep the default setting as autofill=yes in my tests. That's mean even autofill=yes, the 1st caller will still stick the whole queue. asterisk version : 1.4.18 --queue.conf-- ; AutoFill Behavior ;The old/current behavior of the queue has a serial type behavior ;in that the queue will make all waiting callers wait in the queue ;even if there is more than one available member ready to take ;calls until the head caller is connected with the member they ;were trying to get to. The next waiting caller in line then ;becomes the head caller, and they are then connected with the ;next available member and all available members and waiting callers ;waits while this happens. The new behavior, enabled by setting ;autofill=yes makes sure that when the waiting callers are connecting ;with available members in a parallel fashion until there are ;no more available members or no more waiting callers. This is ;probably more along the lines of how a queue should work and ;in most cases, you will want to enable this behavior. If you ;do not specify or comment out this option, it will default to no ;to keep backward compatibility with the old behavior. ; autofill = yes This was something I put in a long while back on 1.2 branch because we really needed it for 1.2 to bug fix the behavior, but also needed to prevent the change in behavior for those that didn't want it to change. Is this option active in 1.2.24? I thought it was only in 1.4 It's not mentioned in the queues.conf.sample. regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] RTCP not being sent when on hold
Adrian A wrote: Hello, When I receive a call to my CounterPath Bria from Asterisk 1.4.18.1 http://1.4.18.1 and I place the call on hold, the call is dropped after 30 seconds. It looks like there is no RTCP/RTP sent to the client from Asterisk while on hold (music on hold playing to caller) thus client disconnects the call. During this time, I get the following messages in the CLI: NOTICE[24194] rtp.c: Unknown RTP codec 126 received from '0.0.0.0 http://0.0.0.0' In sip.conf I have rtpkeepalive=15 but that does not seem to help. Does anyone know what I can do to fix this, other than increase the timeout on Bria? Thanks, Adrian Is it not up to the phone to send the keep-alive packets? Sounds like Asterisk does not understand the keep-alive packets coming from the phone. Try setting rtptimeout=300 in sip.conf to test this. It should now hangup after 5 minutes. regards, Drew regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] FW: [newtech-1] Skype 24 Hour Rolling Analytics
Dean Collins wrote: http://deancollinsblog.blogspot.com/2008/04/skype-24-hour-rolling-analytics.html http://bp3.blogger.com/_jmYevHrBr6M/R_Pq-vgIjvI/Af0/PgE_8gFqrY8/s1600-h/World%2Bpopulation%2Bawake.png Totally stumbled across this really interesting post http://skypejournal.com/blog/2008/04/world_online_or_asleep.html So, it seems that people make more calls on Skype when they are not sleeping and that people who don't have computers don't make so many Skype calls, even when they are awake. Fascinating! regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] FW: [newtech-1] Skype 24 Hour Rolling Analytics
Yes Dean, we do use Asterisk at OANDA. We've been running our office on it for 2 years now and the call centre's 1st Asterisk anniversary was April 1! Glad you like our site, we've just launched FXGame Mobile (http://fxlabs.oanda.com) for the gadget lover and FXGlobal Transfer for ex-pat Aussies to send their money home. :-) Perhaps I should have added a smiley for my fellow North Americans but I think my analysis is valid. Perhaps the interesting points is that % of Skype users seems to be fairly uniform across cultures around the world although perhaps a little higher in Europe (a.m. usage while North Americans are still sleeping). I suspect that this is due to the call billing structure in Europe. They make the North American telcos look positively philanthropic. regards, Drew Dean Collins wrote: Wow Drew, I had no idea someone from Oanda was a subscriber to the Asterisk list (and therefore an asterisk user company?). Just wanted to say you guys run a fantastic sight and I've been a long time user for at least the last 2 years. Now for the irony part of your email. I found it interesting with regards to global penetration and world population. I'd also be interested in seeing some comparisons where this goes against the trend. Eg would love to see a transaction per hour chart for Oanda on a global basis (though I don't know what your customer spread is). I've already sent this email to the creator of twitter vision to see if he can do something similar against tweets per hour on a rolling basis. I'd also love to see something truly global like SMS use which is pretty much a global application. Something like number of SMS's sent per hour across the various countries over a 24 hour period would be hugely interesting (well to me anyway). Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 +61-2-9016-5642 (Sydney in-dial). -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Drew Gibson Sent: Thursday, 3 April 2008 9:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] FW: [newtech-1] Skype 24 Hour Rolling Analytics Dean Collins wrote: http://deancollinsblog.blogspot.com/2008/04/skype-24-hour-rolling-analyt ics.html http://bp3.blogger.com/_jmYevHrBr6M/R_Pq- vgIjvI/Af0/PgE_8gFqrY8/s1600- h/World%2Bpopulation%2Bawake.png Totally stumbled across this really interesting post http://skypejournal.com/blog/2008/04/world_online_or_asleep.html So, it seems that people make more calls on Skype when they are not sleeping and that people who don't have computers don't make so many Skype calls, even when they are awake. Fascinating! regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] Had it with Dell Garbage - HP Question
My own fear from hell is having to call tech support! My personal experience is that I get better support, faster, from mailing lists than from a paid for tech support. If it's a common failure, Google will get you there faster than you can dial 1-800. If it's more unusual, the paid for techie won't have it in his script and you will have to wait to be escalated. If it actually requires some real troubleshooting skills and/or brain power, your mailing list buddies will be far more interested in helping you than an underpaid, under-challenged help line dweeb who just wishes it was Friday already. The idea that paid for or at least there's a neck to choke support is better is just FUD. BTW the original IBM phrase was Fear, Uncertainty and DEATH (not the P.C. Doubt, which is redundant). regards, Drew Al Baker wrote: Helps a bunch !!! One follow up question - out of all of your possible choices for the OS how did you pick *Debian*. I 'm not saying is bad, I just know nothing about the particular disto. and and very curious what it brought to the table that made you pick over say *RedHat* - where you can *buy support *or *SUSE* - where you can *buy support*. My fear from hell is that I' get 50 or 60 of these boxes in, start having kernel panics, and have no damn body to help except the folks on mailing lists. Mind you these are often really smart people, very generously giving of their time, but not quite the say as a manned/paid support organization. Thx for sharing !!! Michiel van Baak wrote: On 08:02, Thu 27 Mar 08, Al Baker wrote: How do you get notifications ? Is this thru one of the add on packages HP sells for the box ? Which One ? Could you be more specific about what you mean by a recovery CD and hod do you get console access below multi used to do recovery ?? What is integrated ILO BIOS Access sounds cool. What O/S you usin and what made you pick it ? What kind and how many RAIDS are you using. The HP site gave like 8 different RAID controllers and like 20 CPUs to chose from. How did you chose ? Thx for sharing !!! I'm not the op, but sending a reply anyways. The notifications come from the HP tools you can download for free from their website. The recovery cd is probably a selfmade installer for their setup. At least that's what we have. the ILO stuff is to give you access to the box like you were sitting right in front of it with a physical keyboard and monitor, but over IP. You can boot the machine, access the cd in your local machine etc, even if the box is on the other side of the moon. We use Debian. HP even supports it on their DL380 boxen. We use the P400 raid controller. Setup RAID5 with 3 disks. CPU we use right now is the Intel E5405 Hope this helps a bit. ___ -- 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 -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] FYI about my Mona Vie business venture
Does it work with Asterisk 1.2? BerkHolz, Steven wrote: Asterisk work does not pay all of my bills, so I have joined up with a company that has a very good payment plan. I have recently become a Mona Vie Independent Distributor. I am not going to go into a sales pitch. This is just an FYI to this opportunity. The company has grown into a Billion dollar company in just 2 years. This company's compensation plan is the best and quickest that I have seen. My brother-in-law has only been in the business for a month and is already making a profit. The first thing that I noticed when researching the opportunity, was that I could find no negative statements about it. The product itself has many health benefits. So far: My knees no longer ache. We are both sleeping better. I literally do not stir once asleep. My restless leg syndrome has not been noticed. I seem to have more energy. The main ingredient is the acai berry. Here is a list of what it is supposed to do: Boosts energy levels Improves digestive function Improves mental clarity/focus Promotes sound sleep Provides all vital vitamins Contains several important minerals Is an extremely powerful free radical fighter Acai has very high levels of fibers Cleanses and Detoxifies the body of infectious toxins Strengthens your immune system Enhances sexual desire and performance Fights cancerous cells Slows down the aging process Promotes healthier and younger-looking skin Alleviates diabetes Normalizes and regulates cholesterol levels Helps maintain healthy heart function Minimizes inflammation Improves circulation Prevents artherosclerosis Enhances visual acuity The income can be made in two ways (actually more, but two primary ways) 1. Reselling the product at a marked up price. This is something that I have no interest in, and do not personally know anyone doing this. 2. Team Commissions. a. You make back 5 percent of the sales that occur below you in your tree. b. You only have to personally sign 2 people. Other people above you will be adding to your tree. c. They call it a binary system, where you only have 2 people directly under you, and any other people that you add go down to the bottom and benefit others as well as yourself. d. I already have two people underneath me and have not personally signed anyone yet, so it is a quick growing tree, even for people that may not be as motivated. e. After a month, My brother-in-law has NO more out of pocket expenses to stay in this system. The money he is earning is paying for his Minimum requirements. The rest is profit. To sign up to be a distributor , which is required to make money, is $54 A case of Mona Vie is $120. A case will last 2 people a month. (you only take 2 ounces a day) This may seem like a lot, but: 1. You will not need to buy any vitamins. 2. My brother-in-law is already making $200 a month, after being in the system for a month, So his cost for the Mona Vie is covered and he is making $80 a month. 3. As more people sign up, the amount he gets back will increase. Anyway, I am not intending this to be pushy or salesy, I just wanted to let my associates, that may be looking for additional income, know about this. Here is the Website, if you are interested in researching this: http://teamvie.blogspot.com/ http://www.monavie.com Also, feel free to Google it. I am very excited with this, both in the health benefits I am already seeing, and the income potential. Please feel free to let me know if this is something that you may be interested in, and I can get you more information. Thank You, Steven B [EMAIL PROTECTED] Please visit us on the web at www.hirotecamerica.com HIROTEC AMERICA Ph. 248-836-5100 Fx. 248-836-5101 Please only print this email if it is necessary. Help spread environmental awareness. This e-mail and any files transmitted with it are intended only for the person or entity to which it is addressed and may contain confidential material and/or material protected by law. Any retransmission or use of this information may be a violation of that law. If you received this in error, please contact the sender and delete the material. ___ -- 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 -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] Hardphone SIP phone costs
Anciso, Roy wrote: I ’ m trying to understand something that just doesn’t seem to compute. How can companies like Cisco justify selling their hard phones for as much as they do? I know there is a matter of recouping RD costs but when you look at the iPhone with all its amazing features for less than $500.00 it just doesn ’ t make sense. Am I the only one that thinks this? Hi Roy, Although Cisco generally make good gear, it does tend to be over-featured and over-priced. It's due to two main factors, branding and lock-in. Branding:- NoLogo by Naomi Klein is a good primer (regardless of which side of the fence you sit), see also H. C. Andersen's The Emperor's New Clothes http://en.wikipedia.org/wiki/The_Emperor's_New_Clothes Lock-in:- If you've already bought into The Brand (eg Cisco's Call Manager software) you don't have a choice if you want all the features. You _could_ try telling the boss that the $100k+ that was spent on The Brand was a waste of money, that it could have been done for $20k with a mix of Brands, with more relevant features and that you're _sure_ the VP would understand but then, no one got fired for buying IBM. regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] British Summertime Grandstream Phones
Gordon Henderson wrote: On Sun, 25 Mar 2007, Gordon Henderson wrote: and change the Optional Rule to: 3,3,7,1,0;10,4,7,2,0;60 someone correct me if I've goofed! Well, it's been a year since I wrote this, and of-course I goofed and no-one corrected me ;-) Bother. So I've looked again. In the UK, We Spring Forward on the last Sunday of March, and we Autumn Back on the last Sunday in October. Not the 3rd as I put there, which worked last year. So there's still 1.5 weeks to go... So it needs to be: 3,-1,7,1,0,10,-1,7,1,0;60 I think... Well, I'll give this a go and see what happens... Gordon Numbers look OK but I think you have a typo, try 3,-1,7,1,0;10,-1,7,1,0;60 (with 2 semicolons) Check the f/w rev too. Grandstream fscked up the time change parsing in the f/w last year while trying to deal with the new US/Canada changes. It worked in the Spring 2007 and didn't work in the Autumn. We upgraded to 1.1.4.18 and it did, to our amazement, change at the right time (for Ontario) this Spring! regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] capacity
Having ventured high enough and far enough to view the curvature of the Earth and having stayed up late enough long enough (why do disks only fail at the weekend?) to rebuild and restore RAID 5 sets, I proffer the following (not so) Humble Opinion . Dual power supplies, two thumbs up but RAID 5 is only good for reducing storage costs on large volumes of data. It reduces performance and reliability over RAID 1. Don't put the OS on RAID 5 unless you like rebuilding servers from bare metal. It's much easier to rebuild and restore the data on RAID 5 sets if the OS is already up and running. Your OS and other system critical files (Asterisk) should be on RAID 1 for performance, redundancy and cost reasons. More disks = higher cost and higher chance of failure. Asterisk in general does not need much disk storage. The minimum drive size available in a new server tends to be overkill. Two drives as RAID 1 gives you redundancy and performance. Adding a third drive for RAID 5 adds cost, increases complexity and reduces reliability just to add storage capacity that you don't really need. (but the reseller WILL make more money and impress you with their command of the big words and acronyms on the spec sheet.) If and only if you need to store many hundreds of gigs of data (eg. recording a very large volume of calls) then RAID 5 becomes useful (or RAID 10 or RAID n). You should add this bulk storage IN ADDITION TO the mirrored pair holding the OS. regards, Drew Steve Totaro wrote: And I can post a link that shows a bunch of guys think the earth is flat with a 5/10 google ranking also (like the barf guys). http://www.alaska.net/~clund/e_djublonskopf/Flatearthsociety.htm I usually just call my guy at CDW and give him my needs, he is a former techie gone sales. He puts together a quote and emails it to me for approval. I find HP server are very robust and rock solid at a decent price point (IBM as well). I like the 380 because you get six hot swap scsi bays and redundant power supplies in a 2u profile, also, Digium and Sangoma T1 cards have never given me an issue. Many on this list love Supermicro, I have yet to try them but I will in the near future. I have not heard a single complaint, only rave reviews. I guess my original point was going for redundancy as far as storage and power supplies with your dollar, not the fastest proc or maxed out RAM that will not be needed. Regardless of the actual hardware or RAID setup, that is the angle I suggest you take. 4k - 6k students will require quite a bit of storage. Thanks, Steve Totaro On Wed, Mar 19, 2008 at 9:38 AM, Ron Joffe [EMAIL PROTECTED] wrote: On Tuesday 18 March 2008 22:12, Steve Totaro wrote: For your use, I would go for a RAID 5 I would highly recommend against a raid 5 set. I can give you more details if you are interested, but these guys have most if it down : www.baarf.com see the link on the left on why should I not use Raid 5 Ron ___ -- 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 -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] capacity
Our office averages around 1.5MB / mailbox, call it 10MB for rounding. 6,000 x 10MB = 60GB (n'est pas?) 2 x 250GB drives, mirrored, should cover that and the system quite nicely. regards, Drew Disclaimer: Most of our employees are programmers so probably don't have any friends to call and leave messages! :-) Steve Totaro wrote: RAID arguments (preference really) aside, 4k - 6k worth of student voicemails is going to require quite a bit of storage space. Thanks, Steve Totaro On Wed, Mar 19, 2008 at 12:01 PM, Drew Gibson [EMAIL PROTECTED] wrote: Having ventured high enough and far enough to view the curvature of the Earth and having stayed up late enough long enough (why do disks only fail at the weekend?) to rebuild and restore RAID 5 sets, I proffer the following (not so) Humble Opinion . Dual power supplies, two thumbs up but RAID 5 is only good for reducing storage costs on large volumes of data. It reduces performance and reliability over RAID 1. Don't put the OS on RAID 5 unless you like rebuilding servers from bare metal. It's much easier to rebuild and restore the data on RAID 5 sets if the OS is already up and running. Your OS and other system critical files (Asterisk) should be on RAID 1 for performance, redundancy and cost reasons. More disks = higher cost and higher chance of failure. Asterisk in general does not need much disk storage. The minimum drive size available in a new server tends to be overkill. Two drives as RAID 1 gives you redundancy and performance. Adding a third drive for RAID 5 adds cost, increases complexity and reduces reliability just to add storage capacity that you don't really need. (but the reseller WILL make more money and impress you with their command of the big words and acronyms on the spec sheet.) If and only if you need to store many hundreds of gigs of data (eg. recording a very large volume of calls) then RAID 5 becomes useful (or RAID 10 or RAID n). You should add this bulk storage IN ADDITION TO the mirrored pair holding the OS. regards, Drew Steve Totaro wrote: And I can post a link that shows a bunch of guys think the earth is flat with a 5/10 google ranking also (like the barf guys). http://www.alaska.net/~clund/e_djublonskopf/Flatearthsociety.htm I usually just call my guy at CDW and give him my needs, he is a former techie gone sales. He puts together a quote and emails it to me for approval. I find HP server are very robust and rock solid at a decent price point (IBM as well). I like the 380 because you get six hot swap scsi bays and redundant power supplies in a 2u profile, also, Digium and Sangoma T1 cards have never given me an issue. Many on this list love Supermicro, I have yet to try them but I will in the near future. I have not heard a single complaint, only rave reviews. I guess my original point was going for redundancy as far as storage and power supplies with your dollar, not the fastest proc or maxed out RAM that will not be needed. Regardless of the actual hardware or RAID setup, that is the angle I suggest you take. 4k - 6k students will require quite a bit of storage. Thanks, Steve Totaro On Wed, Mar 19, 2008 at 9:38 AM, Ron Joffe [EMAIL PROTECTED] wrote: On Tuesday 18 March 2008 22:12, Steve Totaro wrote: For your use, I would go for a RAID 5 I would highly recommend against a raid 5 set. I can give you more details if you are interested, but these guys have most if it down : www.baarf.com see the link on the left on why should I not use Raid 5 Ron ___ -- 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 -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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 ___ -- 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 -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ -- Bandwidth
Re: [asterisk-users] Is Asterisk ready for Prime-Time?
Bill Andersen wrote: This is not a troll. I've used my real email because I want this taken seriously. I'm not trying to make anyone mad, I just want some real discussion on this issue. Please bare with me... I'm a USER of Asterisk. We purchased 3 commercially available Asterisk Based PBXs a little over a year ago. (I won't mention which one at this point - I don't want to bad mouth them - yet!) Two of the systems are very small (5 SIP lines/6 Polycom phones). The third is on a PRI with 30 Polycom phones. My smaller sites work pretty good. I've only had to restart Asterisk every month or so. However, my 30 station system is a continuous headache. I average a restart at least once a week. Sometimes a couple of times in the week. I'm always being called to fix something that just stopped working. I DON'T WANT TO GET INTO A Well, don't just complain, tell us your setup and we can help you get it working. This list HAS helped me figure out some of the issues. THANK YOU! But the purpose of this post is more of a fact finding mission. 1) Was choosing Asterisk for our company the wrong decision... a) IF... I expect a phone system to just work. Once it is configured, a phone system should just work with very little attention. My previous system was a Comdial with external voice mail on a DOS based PC. I LITERALLY WENT OVER 4 YEARS WITHOUT HAVING TO REMOVE POWER TO THE COMDIAL CONTROL OR RE-BOOT THE VOICE MAIL PC. b) IF... I really only need a phone system that allows an operator to answer each call and transfer them to the appropriate person. I need voice mail, but very little auto attendant features (mostly after hours). All the bells and whistles that Asterisk offers are cool, but don't bring that much to the table for our purpose. c) IF... Stability is more of an issue than high end features? 2) Are there any users out there that really DO have an Asterisk system that just works like clockwork? I'm saying, once setup, run for a year (or more) without any issues? 3) If SO, Should I simply consider a different vendor? 4) If NOT, and if my expectations are that a system SHOULD just run and run without any problems. Is Asterisk simply not my solution. Is Asterisk not REALLY ready for production. Because in my mind (as a user of phone services), dealing with the phone system, even on a MONTHLY basis, means that the system is NOT really production ready... Before we installed an Asterisk based PBX, I spent maybe 4 hours per YEAR with phone issues (setting up a new station?). Since we moved to an Asterisk based PBX, I spend 4 hours (or more) every WEEK! Am I expecting too much? Bill I don't think you are expecting too much. We have:- 130 physical extensions including 24x7 inbound call centre Debian on Dell server [EMAIL PROTECTED]:~# uptime 13:15:31 up 192 days, 23:49, 2 users, load average: 0.00, 0.01, 0.00 (Power was removed to switch to new UPS) asterisk*CLI show version Asterisk 1.2.24 built by root @ asterisk on a i686 running Linux on 2007-09-08 17:17:07 UTC asterisk*CLI show uptime System uptime: 63 days, 4 hours, 26 minutes, 40 seconds (Asterisk was restarted after queue config changes) We had a single power supply and single drive fail in one incident in Feb 2007 (one drive of RAID 1). System stayed up but was taken down for 15 minutes to swap the drive. PS was hot-swapped when it arrived later. regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] Is Asterisk ready for Prime-Time?
The box has been up since we upgraded the UPS, time before was for the disk failure in Feb 2007. Asterisk has now been up for 5 hours, 44 minutes (yes, by Murphy's Law, I'm troubleshooting a problem butrestart when convenient does not impact real uptime) but yesterday it had been up for 63+ days (last restart was for queue config changes) This is stock code on stock OS on stock hardware. We don't tweak it, poke at it, fiddle with it, update it unless necessary. We do OS and Asterisk updates on planned maintenance days infrequently) KISS and don't fsck with it! regards, Drew Mojo with Horan Company, LLC wrote: An off-the-shelf 5+ year old MSI MS-6378X-L motherboard, 1.6GHz AMD, 512 RAM, 10 extensions, no more than three concurrent calls: [EMAIL PROTECTED] ~]$ uptime 11:31:45 up 103 days, 1:00, 2 users, load average: 0.00, 0.00, 0.00 But: [EMAIL PROTECTED] ~]$ sudo asterisk -rx 'core show uptime' System uptime: 9 hours, 32 minutes, 25 seconds I reboot every evening :) Drew, what's the uptime on your asterisk process on that box that's been up for 193 days? Drew Gibson wrote: Bill Andersen wrote: This is not a troll. I've used my real email because I want this taken seriously. I'm not trying to make anyone mad, I just want some real discussion on this issue. Please bare with me... I'm a USER of Asterisk. We purchased 3 commercially available Asterisk Based PBXs a little over a year ago. (I won't mention which one at this point - I don't want to bad mouth them - yet!) Two of the systems are very small (5 SIP lines/6 Polycom phones). The third is on a PRI with 30 Polycom phones. My smaller sites work pretty good. I've only had to restart Asterisk every month or so. However, my 30 station system is a continuous headache. I average a restart at least once a week. Sometimes a couple of times in the week. I'm always being called to fix something that just stopped working. I DON'T WANT TO GET INTO A Well, don't just complain, tell us your setup and we can help you get it working. This list HAS helped me figure out some of the issues. THANK YOU! But the purpose of this post is more of a fact finding mission. 1) Was choosing Asterisk for our company the wrong decision... a) IF... I expect a phone system to just work. Once it is configured, a phone system should just work with very little attention. My previous system was a Comdial with external voice mail on a DOS based PC. I LITERALLY WENT OVER 4 YEARS WITHOUT HAVING TO REMOVE POWER TO THE COMDIAL CONTROL OR RE-BOOT THE VOICE MAIL PC. b) IF... I really only need a phone system that allows an operator to answer each call and transfer them to the appropriate person. I need voice mail, but very little auto attendant features (mostly after hours). All the bells and whistles that Asterisk offers are cool, but don't bring that much to the table for our purpose. c) IF... Stability is more of an issue than high end features? 2) Are there any users out there that really DO have an Asterisk system that just works like clockwork? I'm saying, once setup, run for a year (or more) without any issues? 3) If SO, Should I simply consider a different vendor? 4) If NOT, and if my expectations are that a system SHOULD just run and run without any problems. Is Asterisk simply not my solution. Is Asterisk not REALLY ready for production. Because in my mind (as a user of phone services), dealing with the phone system, even on a MONTHLY basis, means that the system is NOT really production ready... Before we installed an Asterisk based PBX, I spent maybe 4 hours per YEAR with phone issues (setting up a new station?). Since we moved to an Asterisk based PBX, I spend 4 hours (or more) every WEEK! Am I expecting too much? Bill I don't think you are expecting too much. We have:- 130 physical extensions including 24x7 inbound call centre Debian on Dell server [EMAIL PROTECTED]:~# uptime 13:15:31 up 192 days, 23:49, 2 users, load average: 0.00, 0.01, 0.00 (Power was removed to switch to new UPS) asterisk*CLI show version Asterisk 1.2.24 built by root @ asterisk on a i686 running Linux on 2007-09-08 17:17:07 UTC asterisk*CLI show uptime System uptime: 63 days, 4 hours, 26 minutes, 40 seconds (Asterisk was restarted after queue config changes) We had a single power supply and single drive fail in one incident in Feb 2007 (one drive of RAID 1). System stayed up but was taken down for 15 minutes to swap the drive. PS was hot-swapped when it arrived later. regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE
Re: [asterisk-users] Telemarketer Torture....
James Finstrom wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Anyone have the telemarketer torture prompts? I would seriously like to revive this. - -- James Finstrom -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.6 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFH3I8qdloC7YyaIOoRAlAjAJ9Hp+3SS2Z8179HecWIETp4RVDzWQCeMizp fW2JPZdYl/uxG1ziUwYnHGo= =QPbv -END PGP SIGNATURE- ___ -- 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 Hi James, I have a copy of the prompts. Will the list accept attachments? regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] SNOM on Do Not Call list????
Some light relief SNOM say Please note that you will not be able to reach us by phone. http://www.theregister.co.uk/2008/03/13/dont_call_us/ regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] FXS channel banks
Chris Bagnall wrote: Greetings list, I've been asked to provide a system for 200 extensions, most of which will be existing analogue POTS handsets, not IP handsets. I've not really had any experience with large channel banks in the past (since most of our deployments are strictly IP-only to the desk), so I'm at a loss as to which ones are worth looking at. If anyone's had experience using channel banks on reasonably sizeable installs I'd be interested to hear what device(s) you used, how simple or complex they were to configure, and whether there'd be any issues attaching multiple units to a single server. This install would be in the UK, so we do need to factor in the different conditions expected by UK POTS handsets (line impedance, etc.). Are most channel banks country-neutral, or do specific models need to be purchased for different line conditions in each country? Thanks in advance. Regards, Chris www.citel.com I used them a few years back in a pilot install with legacy Nortel phones and it worked well. I gather they have grown tremendously from there. I'm in North America, don't know how well they support UK stuff. regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] Automon reliability issue
Jaap Winius wrote: Hi list, Can someone please explain how to get one touch recording (automon) to work reliably? I'm using Asterisk 1.4.14 on a Debian etch system. My We made this function reliable by including the word quickly in our instructions for pressing the keycode to start the recording. (Asterisk 1.2.24 on Debian Etch) Although a private confirmation beep to the initiator of the recording would be handy, this is the way things have to be in order to use the features.conf codes and still allow the use of * and # when calling outside IVR and voicemail systems. eg. Enter your password followed by the pound key... regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] How to hookup to cell phone for outbound calls?
Ed W wrote: Hi I need a small PBX for use on the move. This means that outbound calls will need to be made over the cell phone network. Assuming a small hardware PBX with a spare mini-PCI slot or a USB slot then what hardware options do I have to get an outbound cellular channel? Options need to be rock solid, so no bluetooth to a cell phone kind of solutions need apply. Can any of the 3G usb devices out there offer outbound analogue calls (ie other than via voip)? Cheers Ed W ___ -- 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 How about http://www.mgamble.ca/oss/iphone_asterisk/ ? regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] How to hookup to cell phone for outbound calls?
Erik Anderson wrote: On Feb 5, 2008 2:37 PM, Drew Gibson [EMAIL PROTECTED] wrote: How about http://www.mgamble.ca/oss/iphone_asterisk/ ? Hah! Cool, but quite ridiculous. :-) I have a Linksys NSLU2 (Slug) at home running Asterisk (see http://www.nslu2-linux.org/ ) It's small, relatively cheap and runs Asterisk very well. You could slip it into a pocket. I haven't tried yet but I have done a little reading and hope to connect the Slug to a mobile network when I get the time to play. I know you said no bluetooth, Ed but if you're in North America and your cellular network is CDMA, AFAIK option 1 is the only one possible. These carriers generally won't allow devices on their networks unless they are purchased from the carrier. If your cellular network is GSM then there are two approaches to try, 1. Slug, 4GB USB stick, USB Bluetooth dongle, dedicated bluetooth mobile phone + Asterisk 1.4 with a chan_mobile. Unfortunately, I doubt that chan_mobile is packaged for the slug (it's in 1.4 trunk) and you would have to build it. Cost ~$150 + phone + your time 2. Slug, 4GB USB stick + SIP-GSM gateway. Much easier to configure but it's a second box so less portable and more expensive than a BT dongle and an old phone. Probably more robust. Cost $250-$400 You could also substitute a Linksys WRT54GL ( http://openwrt.org/ ) for the Slug which would give you ethernet ports and wireless too. Hope this gives you some ideas regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] pulling my hair out over voicemail
John Von Essen wrote: Any ideas what could be going on? I tried tweaking the extension 1000 so it looks like: exten = 1000,3,VoicemailMain,s6000 It may be your syntax, try :- exten = 1000,3,VoicemailMain(6000|s) regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] POE draw on Aastra 480i
Octavio Ruiz wrote: Allen Casteran wrote: Anyone know what the POE draw is for the Aastra 480i phones? We have switches that will do 15 watts on 12 ports but only do 7.7 watts on all 24 ports. A Cisco 3560 switch will do 15.6 watts on all 24 ports. Just trying to find out if we need that much power. Drew wrote: According to Aastra tech support, 5 watts (peak) per 480i. We are testing five phones running on a Linksys SRW208P that will only support full 15W on up to 4 of 8 ports. I can power up the switch while all phones are connected without any issues. I would expect your lower power switch will provide ample power. But, PoE class does not matter? Did you plug five Aastra phones? I'm suspicious about how that scenario worked, I mean, as far as i know Aastra phones should register as a zero PoE class, that means it would reserve up to 12.94 watts no matter how many watts uses. So, my guess here is even if the phone use only 5 watts, the switch already reserved 12.94 watts for it. I would love to see what happens if you plug a sixth phone or figure out if you used an Aastra phone. Can you tell us what model/brand you used? Dimensioning PoE devices over capable switches has been a new issue which involves many factors like those described before. Regards, PD. Sorry about the original thread break off, I've been unable to find the original one. Hi Octavio, We are using Aastra 480i phones powered by PoE. During the testing we decided to drop the Linksys SRW208P as it is extremely noisy due to the fans, not suitable for an office environment. We continued testing with Netgear FS116P switches. These have 8 PoE ports plus 8 non-powered ports. Completely silent, no fans. Our testing was done with 5 phones per switch as our call centre is laid out in pods with 5 seats per pod and will not require more devices powered by PoE. I did not look into the issues of PoE class as there were no problems meeting our requirements with the equipment we had. Perhaps the Cisco switches are more particular about PoE class, they are much more complex devices. I like Cisco gear, they make good products but they tend to be over-featured and over-priced for many applications. We have had 5 of the Netgear switches in production for almost a year now, each powering 5 Aastra 480i phones without any issues whatsoever. regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] Peak number of calls?
Gordon Henderson wrote: Is there any way to find-out the peak number of calls that an asterisk system has had? Not the total number of calls, but the maximum number of simultaneous calls. I know I can porobably go through the CDR logs and look for calls which have overlapped in time, but I'm wondering if there's some counter somewhere I could access... (I'm looking for evidence for an ISDN client who wants to know if he's spent too much on the number of ISDN lines he has installed!) Cheers, Gordon We use Asterisk-stat from Areski (GPL). It will show peak number of calls by the hour. Select Daily Load, scroll down and choose the hour you want and Fluctuation Graph. Lots of other goodies too. http://areski.net/areski/index.php?option=com_contenttask=viewid=22Itemid=54 regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] [Fwd: Re: Large issue - having trouble diagnosing.]
Hi Cameron, I think this paragraph goes to the heart of the matter... Cameron Hissey wrote: are connected to the standard switch, however the cabling was a bit of a rush job and consequently the PoE has proven unstable on many of the points, with some of them not even supplying data packets. Benny Amorsen suggested trying a store-bought cable direct to a phone, how did that go? I would have a very close look at the cabling. Were the R45 plugs put straight onto the cable or terminated in a wall jack? There are two types of plug. The one that is most commonly available is not intended for the solid copper core cable that is most commonly available in bulk boxes and WILL cause flakey connections. Check the cables for correct pinout (see link below), not just continuity. The price of a reasonable cable tester will save you a fortune here. Nothing fancy but it must be able to check for split pairs. Split pairs could cause the symptoms you describe. http://www.ertyu.org/steven_nikkel/ethernetcables.html regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] OT: Traffic Shaping
Dovid B wrote: - Original Message - From: Matt Riddell [EMAIL PROTECTED] -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Erik Anderson wrote: On Jan 9, 2008 9:40 PM, Matt Riddell [EMAIL PROTECTED] wrote: Heh yeah that's what I was thinking of doing. What's the traffic shaping like? Can I specify max bandwidth etc or use hfsc shaping? DD-WRT will do both HTB and HFSC shaping, though I've only ever used HTB. Sweet. We had been using HTB but upgraded all our CPE to use HFSC when AstLinux did and found it great. Here's the dd-wrt wiki page on its QoS implementation: http://www.dd-wrt.com/wiki/index.php/Quality_of_Service Cool, looks good, although I would have thought Default Bandwidth Level would have been pretty self explanatory if it works as expected. I guess I'll have to try it out and update the wiki if it does. Looks like they don't recommend HFSC currently due to some lag issues. That might have been fixed, though, in the more recent firmware builds. Will try both out and see how they go. Thanks for the pointers. - -- Kind Regards, Matt Riddell Director ___ Matt, The WRT54GL is not dual port. Chorus from the audience:- Oh, yes it is! It has 5 ports! Although the ports are labeled as 1 Internet port and 4 LAN ports, each can be assigned to a VLAN of your choosing and you can use them as you please (at least you can under openWRT). So you could allocate 2 WAN ports on distinct networks and leave 3 for the LAN. I haven't done it myself but a far more clever gentleman at http://garycourt.com/blog/post/openwrt-advanced-firewall/ seems to know how. regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] [OT] black dogs
Philipp Kempgen wrote: Drew Gibson wrote: A well-written application should attempt to minimize the amount of 'conversion' the user/programmer has to do. Therefore the command structure SHOULD be in a form that is natural for the user/programmer, NOT to the machine. Personally, I would vote for show dogs colour black but maybe I've spent too much time with Cisco's IOS! :-) show me all the colored dogs now. hurry up! don't spent any time in those other threads. and while you're at it would you please fix that stupid mistake i made in the config file Is this in the schedule for 1.6? regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] Upgrade to Asterisk 1.4 - it's one year's old!
Since we're WAY OT anyway Tony Plack wrote: That being said, ordering in a command structure should make sense to the application (less intelligent entity), not to the programmer (hopefully more intelligent). If that were true then we really should be writing our dialplans in binary machine code, that is what that dumb computers REALLY understand. Fortunately, it's not true. We can take advantage of a GOOD programmer's skill to have the computer do the grunt work of converting something real people understand into machine code. We call the product of this process a High-Level Programming Language. A well-written application should attempt to minimize the amount of 'conversion' the user/programmer has to do. Therefore the command structure SHOULD be in a form that is natural for the user/programmer, NOT to the machine. Personally, I would vote for show dogs colour black but maybe I've spent too much time with Cisco's IOS! :-) regards, Drew PS. There does seem to be an assumption that programmers are intelligent, I'm not sure that this is a defensible position. ;-) -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] X100P Fxo card headaches
Chris Boczko wrote: Hello List, Im just dipping my feet into the asterisk world, and im having major fxo problems Im running Asterisk (from svn) + libpri (from svn) + asterisk-addons (from svn) + asterisk gui (svn 1.4 branch) + zaptel (svn 1.4) on a Debian Etch box, with 1gb ram, running all of the services for my home server (web / db / music server etc), and i would like to run my PSTN line from Kingston Comms, but i can't get this box the recoginsie this line! The X100p is a cheap clone i got off ebay for a tenner, so im not expecting much, i know they have echo issues, but im going to upgrade to a SPA3012 / TDM400B when i have the cash. Ztcfg -vv reports orange:~# ztcfg -vv Zaptel Version: SVN-branch-1.4-r3374 Echo Canceller: MG2 Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) 1 channels to configure. orange:~# and my zaptel.conf file contains orange:~# more /etc/zaptel.conf fxsks=1 loadzone=uk defaultzone=uk orange:~# but zap show status in the command line shows orange*CLI zap show status Description Alarms IRQbpviol CRC4 Fra Codi Options LBO Wildcard X100P Board 1 OK 0 0 0 CAS Unk YEL 0 db (CSU)/0-133 feet (DSX-1) orange*CLI and i think the Unk means unknown...? Does anyone have any ideas on how to get this line to work, ive followed every howto i can find, and google seems to be comming up short, as far as i can see, ztcfg should report the card as configured, but it isn't, and ive no idea why. Hope you can help Chris Try adding channels=1 to the end of your zaptel.conf to assign the settings you have made to a channel. regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] [asterisk-dev] trunk working under windows!
+=-I. -Iael -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] RTP Read too short
I saw this with Grandstream GXP2000. When the phone is on a call and on mute, the phone sends SIP keepalive packets that are, indeed, too short. So asterisk is correct in this case. Grandstream said that they were just warnings and to ignore them. We have chosen to ignore Grandstream and move to a different phone vendor. regards, Drew PS. I missed your question earlier because it was a reply to an existing thread. If you want to be seen, start a new thread, don't hijack an old one. John Faubion wrote: Am I the *ONLY* one that has this issue? John Faubion -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of John Faubion Sent: Thursday, November 01, 2007 11:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] RTP Read too short Hello, I'm getting the following logs: [Nov 1 10:54:37] WARNING[20878]: rtp.c:1138 ast_rtp_read: RTP Read too short [Nov 1 10:54:39] WARNING[20878]: rtp.c:1138 ast_rtp_read: RTP Read too short [Nov 1 10:54:40] WARNING[20878]: rtp.c:1138 ast_rtp_read: RTP Read too short Anyone know how to correct this? I'm using SIPConnect from CBeyond and this appears on incoming calls. I haven't had any complaints about voice quality and I haven't seen any dropped calls. Should I be concerned? Thanks, John Faubion -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] (no subject)
[EMAIL PROTECTED] wrote: Hi all, We have a client that needs to setup about 80 desk phones (about 50 in one location and about another 30 in 5 different locations). Which brand/model would you recommend. We were personally thinking in recommending either Cisco, Aastra, Polycom, or Snom, for we've heard great things about them. However, having no real experience with them makes it hard in recommending one to our customer. The only experience we've had is a very frustrating one trying to load the IP software on a Cisco 7970G and so we assume that if we have to go through that for all 80 phones, we'll probably commit suicide :) Thanks We have used Cisco and Aastra, can't comment on Polycom or Snom. I cannot recommend Cisco, good sound quality but that's it. Ridiculously overpriced, too few usable features, incredibly awkward to manage. Aastra have good sound quality, reasonable price, configs are plain text and not to hard to work with. We have the 9133i as our basic phone and 480i in the Call Centre for the soft buttons. Both can be fed from the same config templates. We used to use Grandstream but quality and support issues have driven us away. regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] FXO ATA Options?
Conall O'Brien wrote: Hello, I'm currently looking at FXO options to provide a POTS line to Asterisk to trunk calls with. Does anyone have any experience using the Linksys Sipura 3201 as an FXO device for Asterisk? I use one at home and can recommend it as functional and reliable. It has an unbelievable number of configuration options. Linksys docs are a bit sparse, try the Sipura site under SPA3000. regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] Stuck Voicemails?
Matt wrote: This question is about 1.2.x asterisk. Please no flames, or you should upgrade to 1.4. Does anyone know what might be the cause for 'stuck voicemail's in 1.2.6 asterisk? By stuck, I mean the phones show a voicemail, and if you log in you get you have 1 new voicemail, and if you delete it it says 'deleted', however it remains. Going into the mail directory reveals that there is either a msg0001.txt.tmp or a msg0001.txt file, but no associated wav file. It happens very randomly, not often, and so far has eluded me being able to figure out what causes it. Why does this happen? I don't know why it happens but we have run a very early 1.2 svn before jumping to 1.2.17 (fixed several issues) and incrementally through to 1.2.24 and we have not seen this issue. Any related errors in asterisk logs or system logs? Has fsck been run recently on the relevant filesystem? Is your mail stored on a local filesystem? Would upgrading to 1.2.24 be an acceptable upgrade? regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] Grandstream GXP-2000's and Asterisk.
Thomas Kenyon wrote: Thomas Kenyon wrote: I have 5 GXP 2000's with firmware 1.1.4.25 running with Asterisk 1.4.13. Is anyone else getting the following error in the asterisk console: [Oct 22 11:39:01] WARNING[7100]: rtp.c:1142 ast_rtp_read: RTP Read too short every couple of seconds when a handset is in a call? I didn't notice this happening when I was using an older GXP2000 with the same firmware (doesn't mean that it didn't happen). The Call in question is using G.729. TIA for any help with this. I will hopefully get a bit more time to play with this today. (When I'm in the office in question). Changing codec doesn't appear to matter. I gather that the cause is that the GXP-2000 sends empty udp packets as keep-alives. (which is all well and good, but even with a handful of handsets with light call volume the logs fill up with notices, at the moment there is only 1 call going through the server and this is generating 2 notices/second. Is there any way to make asterisk ignore the empty packets from certain peers? Hi Thomas, I have tried to work through these (and other) issues with Grandstream but they seem to have a short attention span. We now buy Aastra phones. regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] Refrigerator Alarms
Hi Balu, http://www.digitemp.com/ has all the info you need. Cost of hardware around US$70. Will give you pretty fridge temperature graphs too! You can easily hack some of Brian's scripts to do different levels of temp alarm and trigger calls in Asterisk. regards, Drew Balu Raman wrote: Omar, I am hoping that there may be some temp sensor interface that can be routed to a pc and if the temp falls out of a range, I can have this event call someone. I know what to do in asterisk to make a call. I have to do some research. may be, someone has already done a similar thing. Has to be event driven. Thanks, balu raman On 10/17/07, Omar A. Sabek [EMAIL PROTECTED] wrote: Balu, Do you want events passed to Asterisk from the refrigerator? Or does a reminder type phone call need to be placed on an interval? Please be more specific, since this sounds like a special purpose refrigerator, does it have any way of passing events to an external device? Omar A. Sabek On 10/17/07, Balu Raman [EMAIL PROTECTED] wrote: Hi, I want asterisk to call a person on the phone for monitoring the refrigerator storing vaccines. I am clueless where to look. Can someone clue me in ? Thanks, balu raman ___ --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 ___ --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 -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] BBC on Atserix
Tzafrir Cohen wrote: On Thu, Oct 18, 2007 at 01:04:06PM +0100, Cartwright, Dave wrote: Just for fun. http://news.bbc.co.uk/1/hi/magazine/7049642.stm It's Asterix != Asterisk. Though named after *. In Britain, it's called humour :-) regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] BBC on Atserix
SIP wrote: Drew Gibson wrote: Tzafrir Cohen wrote: On Thu, Oct 18, 2007 at 01:04:06PM +0100, Cartwright, Dave wrote: Just for fun. http://news.bbc.co.uk/1/hi/magazine/7049642.stm It's Asterix != Asterisk. Though named after *. In Britain, it's called humour :-) regards, Drew Only if it's actually humourous. ;) N. ___ --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 It's looking good ... http://news.bbc.co.uk/2/hi/entertainment/4343264.stm regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] Is there real benefits on a SMP machine for Asterisk?
Hi Raul, I think that Bacula is going to cause you many headaches with Asterisk. Backups are (surprise!) I/O intensive and when streaming data from the network to disk or tape it will saturate the available bandwidth. This will cause the I/O wait time on the CPU to run high and effectively block other processing (eg. voice traffic) for seconds at a time. Try running a full backup during office hours and see what your users have to say! My recommendation is to move either Asterisk or Bacula to a separate machine. regards, Drew Raúl Gómez C. wrote: Well, this has become a hot topic! :p Thinking about my original post, I was reluctant of installing my PBX on a shared system, is a Dell PowerEdge 2950 with 2 Intel Xeon Dual Core CPUs @2GHz (4 totals cores) and 4GB RAM which serves as Domain Controller and File Server (Samba), central backup server (Bacula with a LTO2 external tape drive), it has dual NIC in a bonding alb mode and redundant PSU (each one connected to a different UPS). It has a PCI slots in which I can install my Sangoma Remora A400D card. But now I think the PBX will work just fine in this system, maybe breaking the channel bonding and dedicating a NIC for the PBX and the other NIC for the remaining task, what do you think? Or its better to install the PBX on a dedicated system? Let me know your opinions! Regards... Raul On 10/12/07, *Mojo with Horan Company, LLC* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: You're correct, it IS 100 -- but 100%. Expressed in decimal format, this is of course 1.0 -- and as each cpu has this average, 4.0 indicates that no threads regularly wait for execution. This worldcommunitygrid you mentioned binds your cpu by design it sounds like. ___ --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 -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] Grandstream GXP2020 / 2000
Erik Wartusch wrote: Hi, Has somebody experiences with the Grandstream GXP2020 / 2000 phones in a business graded installation (with really traffic on not 3 calls a day ;-) ) Of course with Asterisk PBX (1.4.1 or 1.4.11 or 1.4 in generall) Hi Erik, we have about 75 Grandstream GXP2000 phones running with Asterisk 1.2. We also have about 25 Aastra 480i phones in our call centre and 10 Cisco phones in meeting rooms. The Grandstreams work fine on the whole, they are probably good value for money in smaller installations. We use some simple bash scripts to manage the configurations so the need to convert the config files from text to a binary format is not an issue. Sound quality is OK, the speaker phone is not great. Most users are happy with the Grandstreams but we give Aastras to anyone who spends a lot of time on the phone. The reason I cannot recommend these devices for larger installations is the mediocre response from Grandstream's technical support. Tech support will acknowledge your initial problem report but then ignore you if they don't have an immediate fix. This is a pattern repeated over several reported incidents. Our latest issue is with the GXP2000's running f/w = 1.1.1.14. The phone does not send a keep-alive packet when the mute function is used, despite this bug being documented as fixed in a much earlier release. This results in a disconnect after 5 minutes of being on mute. Very annoying when on a conference call or on hold to tech support. This is fixed in 1.1.4.18 but this release introduces an issue with very loud (for our environment) ring tones rendering the GXP2000 unusable in our office. We no longer purchase Grandstream phones but will consider them again in the future should the support issues be resolved. regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ 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] sip.conf best practices?
Erik Anderson wrote: All - I've been wrestling with how to best structure the sip device accounts on a new asterisk server I'm deploying. All of the sip devices (currently only Linksys SPA941s) will reside on the same subnet as the server, and I have already set up a decent automatic provisioning system for the phones. When the rollout is complete, there will be about 100 SIP devices authenticating and routing calls through this server. The question is what to use for the username portion of the SIP account. Part of me says that I should standardize on using each phone's MAC address as the sip account UID, like so: ; Joe Smith, x123 [000E08DA0409] secret = blahblah ... and so on and so forth Doing it that way is nice for standardization's sake, but it makes the dialplan quite a bit more complex. The obvious alternative is to use the extension as the sip UID: ; Joe Smith, x123 [123] secret = blahblah ... This makes the dialplan *much* more simple, but when looking through sip.conf, it's not as immediately obvious what device should be authenticating with that account. Since this is my first large-ish asterisk deployment, I'm seeking the advice of those who have gone before me. What tactic (one of the above options or otherwise) is best to keep your sip.conf sane? Thanks! -Erik Hi Erik, We have around 100 devices and most of our changes are adds for new users/devices with occasional re-assignment of devices. We manage our users and devices with some simple scripts and good old vi for exceptions. Our extensions.conf has a list of global vars that tie an extension to a sip (or iax, or whatever) device. (I think this is straight from TFOT v1) eg EXT_100=SIP/100 This allows us to redirect extensions to different devices or make extensions ring on multiple devices by changing that var alone (no need to alter macros or other dialplan elements) eg EXT_100=SIP/101SIP/102 The device-specific hardware and the SIP configurations are generated from a master map that contains a line per device including technology and extension, MAC, user display name and email address. Scripts create the phone hardware configs with device type determined by MAC address (eg. Aastra, Grandstream or Cisco) from the map file and add the user to sip.conf and voicemail.conf. sip.conf ... [grandstream] ; Aastra 480i phones for general office ... (general SIP settings) context=office-dial [aastra-cc] ; Aastra 480i phones for Call Centre only ... (general SIP settings) context=cc-dial [100](grandstream) username=100 secret=*** mailbox=100 callerid=Joe Bloggs 100 [101](aastra-cc) username=101 secret=*** mailbox=101 callerid=Agent 99 101 Initially we only had one class of user, general office types using Grandstreams. When we migrated the Call Centre to Asterisk, they were the only users with Aastra phones, apart from one or two in the general office. So each class of user had a different hardware type and was easy to automate, the exceptions are currently handled with manual edits. This system is working well and is stable but not sufficiently flexible for the future. Our company is growing rapidly and since we will no longer be buying Grandstream devices, more Aastras are appearing in the general office environment. This means we now have two classes of users that require different configs for the same device type, general office users and Call Centre agents. This means a choice will have to be made between updating the scripts to cope with the two user classes or moving to realtime. Where's my Magic 8-ball... :-) regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ 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] how to route outgoing calls on IP-level
Kate Kretz wrote: Dear Sirs, out asterisk server has multiple network cards. I want some outgoing calls (from several extensions) to use one IP address, and others to go through another address. is there a way to achive that using asterisk ? Cheers, Kate This is the job of your network, not Asterisk. Policy-based routing is not much fun (unless you think the Cisco CLI is really cool) but it can be done. regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ 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] Cisco UC 500
Jeremy Mann wrote: Is the Cisco UC 500 able to integrate with Asterisk? Specifically does it work via SIP? Just curious, as the Cold Call Cisco sales rep had no clue what SIP even was, and this device looks interesting. Google cisco UC500, hit #2 = http://www.cisco.com/en/US/products/ps7293/products_data_sheet0900aecd8061fb06.html Quotes: Core components of the Cisco Unified Communications 500 Series include:Cisco Unified IP phones, including wireless handsets and Session Initiation Protocol (SIP) phones PSTN interfaces and features: SIP trunks and RFC 2833 support Does that help? I'll bet Asterisk is cheaper though. :-) regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ 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] understanding queues
Elliot Finley wrote:
Hello,
I feel like I understand how the dial plan works pretty well with one
exception. It seems like queues are using the stdexen macro to ring
the agents/extensions. Is this normal? Is there anyway to configure
this differently?
Hi Elliot,
The queue will use the dial method in the current (or default) context.
If you want a different dial method for the queue, put the queue in a
new context along with the code for the new dial method.
For example, assuming call centre phones are on extensions 100 through
199 ...
[office]
... regular office dialplan here ...
; Call any extension
exten = _XXX,1,Macro(stdexten,${EXT_${EXTEN}})
[call-centre]
...IVR, etc here
; Only call call centre extensions
exten = _1XX,1,macro(ccexten,${EXT_${EXTEN}})
[macro-stdexten]
... handle dialing this way
[macro-ccexten]
... handle dialing that way ...
regards,
Drew
--
Drew Gibson
Systems Administrator
OANDA Corporation
416-593-6767 x322
www.oanda.com
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