[asterisk-users] Re: Call center manager for Asterisk (Release 0.3)
just to let you know that i've started a mailing list on sourceforge [EMAIL PROTECTED] You can subscribe here https://lists.sourceforge.net/lists/listinfo/ccmanager-users Other news regarding ccmanager will be posted on this mailing list, i invite interested people to subscribe. Thanks On 3/14/07, nik600 [EMAIL PROTECTED] wrote: Hi i just want to let you know that is available a new release of ccmanager. I've added the possibility to import queue_log information in a mysql database and to generate reports using this information. The software is in a beta state and provides this functionality: - users management - call generation (making a GET or POST request on a certain URL) - queue management (LOGIN / LOGOUT / QUEUE STATUS) - pickup a call from a queue even if the user isn't logged in the queue - outbound call in customizable context - queue stats import from queue_log - queue reports creation (using an open xml format) Please note, i think that the xml definition of a report is very important, if many people share each other their reports there is the possibility to build a reports-repository, so the final user can use many reports and, if the user know what he is doing, he can customize the reports. I am looking for people to improve this project, any help would be appreciated. - developers (php / mysql / postgres / ajax ) - tester - graphics (div css) Here there are some screenshots https://sourceforge.net/dbimage.php?id=115442 https://sourceforge.net/dbimage.php?id=115440 https://sourceforge.net/dbimage.php?id=114381 And here there is the sourceforge project. https://sourceforge.net/projects/ccmanager Thanks, nik ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cost of Branded Equipment for Voip Provider Implementation
Hi Guys, Im looking for the pricelist of big scale pbx like nortel and avaya.Because im going to make a presentation of cost against cost of open source implementation for voip provider. Anyone there could help me? Thanks so much. eduard eng'r.eduard - No need to miss a message. Get email on-the-go with Yahoo! Mail for Mobile. Get started.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] voip-info.org status update
On Wed, 14 Mar 2007, shadowym wrote: Hard to expect the business community to take Asterisk seriously when this sort of stuff happens IMHO. I think you hit the nail on the head with one word: community. Asterisk is free, community supported, and the voip-info site has been provided for free - with the support of the community. The site would appear to be financially supported by a small number of quite unobtrusive google ads, and therein lies the problem... Hosting isn't free. If you can't/won't pay for hosting, then you have to support it by advertising. I can sell you web space/servers/co-lo facilities with full disk/server/location redundancy, backups and so on, but would you be willing to pay for it? Probably not. So you takes your chances with a popular hosting company, put in a small number of google ads to pay for a basic hosting package and go with it. After-all, there are millions of websites hosted on millions of servers throughout the world - it's a highly competitive business - there are offers of hosting for £1 a month or even less, but do you think it's a sustainable model? I don't. Well, maybe it is when you have 1000s of clients with 10s of 1000s of websites (spread over 100s of servers!) but with scale comes more issues. I can't understand how 3 of 4 hard drives could just suddenly fail simultaneously. There must be more too it. No UPS? Someone spilled their coffee into it? Something! That does strike me as odd, but I've seen it myself with a bad batch of disks. (IBM DeathStar, Hitachi, etc.) You usually get warnings, but if you're employing monkeys paying them peanuts, then they usually just treat them as fire forget once installed in the rack and plumbed into their automated selling/billing system. Either way, it's amateur hour! It's the way 99% of all co-lo facilities work. Buy big, sell cheap with little or no SLA - hope that the hardware/premises/internet is reliable enough, employ monkeys, pay peanuts. If you want quality, then be prepared to pay for it, and £1 a month does not give you quality IMO, and in my experience as someone who runs a small co-lo facility, people will not pay for quality hosting. A quality server costs me £650, more if the client insists on a Dull. Sure, I can put together something with pair of disks for under £300, but I know (from experience!) it won't last the 4+ years I want it to last, nor deliver the preformance my clients (who are willing to pay for such a service) demand. I'm not blaming James here because that's the way it is! I bet he's spent 100s of hours (unpaid) setting it up, running it and maintaining it, and resorted to google ads. purely to fund it. I don't envy him at all. If I can't be confident enough in an important source of information like this then I can't be confident enough to provide an Asterisk solution to businesses. That's the way I see it. Yea, it's a wiki but it's the best source of info out there. So how much are you willing to pay to support such a service? Gordon___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: While the VoIP-Info.org site is down...
SB == Stephen Bosch [EMAIL PROTECTED] writes: SB As somebody else has already pointed out -- There must be more to SB it. Let's say three of four drives failed -- the odds of them SB failing at the same time are vanishingly slim; Not as slim as manufacturers want to make you believe. RAID drives tend to be purchased at the same time, so they are often from the same batch. They are then subjected to exactly the same load in exactly the same environment. Is it any surprise that they fail at the same time? Especially if they are kept running for a very long time and then shut down by a power failure. /Benny ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MP3Player
Hi All, I'm having problem with MP3Player app. I want the caller to hear mp3 when he is waiting until I answer my phone. -- from extentions.conf -- exten = 200,1,Answer() exten = 200,2,MP3Player(/home/user200/mp3/hanna-hais.mp3) exten = 200,3,Dial(SIP/200|20|tTrR) exten = 200,4,Hangup() -- end -- here is debug from CLI: -- Executing Answer(SIP/200-08a64d98, ) in new stack -- Executing MP3Player(SIP/200-08a64d98, /home/user200/mp3/hanna-hais.mp3) in new stack Mar 15 11:25:32 NOTICE[4991]: app_mp3.c:121 timed_read: Poll timed out/errored out with 0 -- Executing Dial(SIP/200-08a64d98, SIP/200|20|tTrR) in new stack -- Called 200 -- SIP/200-08a6a2d8 is ringing Asterisk 1.2.16 and mpg123 installed. Any ideas? Thank you in advance, Dominik ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Meetme variables
Hi, anybody who has a complete list of variables used by meetme conferencing application in asterisk, plz share. -- Regards Rizwan Hisham Software Engineer ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] busy/hangup/answer detection in PRI E1 channels
Hi, Please discribe me how we define busy/hang/answer detection with PRI E1 channels. Since busydetect, callprogress, busycount giving falts hangup and call drops what is the solution on PRI channels? -- Thanks Regards, Vidura B. Senadeera. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: While the VoIP-Info.org site is down...
On Thu, 2007-03-15 at 10:15 +0100, Benny Amorsen wrote: Not as slim as manufacturers want to make you believe. RAID drives tend to be purchased at the same time, so they are often from the same batch. They are then subjected to exactly the same load in exactly the same environment. Is it any surprise that they fail at the same time? Especially if they are kept running for a very long time and then shut down by a power failure. It all comes back to perceived security. As you rightly say the disks in a RAID will have a good chance of failing at the same time because not only being out of the same batch but probably constructed one after the other. This normally will not be a problem for the manufacturer because they would be installed in different machines with different usage rates. If you really want to avoid this the discs would have to be selected out of different batches. It should never be forgotten that originally RAID stood for Redundant Array of Inexpensive Disks, I never could understand how a RAID could be made up using SCSI disks seeing that they are certainly not inexpensive. The other common misconception is tape backups there is an old story in computing that the main problem with tapes was that the bits fell off, how many people actually test the restore capability before they actually need it? -- Dave Cotton [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] busy/hangup/answer detection in PRI E1 channels
You can use the hangupcause variable which us the pri cause code
supplied when a call is ended over a PRI line. For example this is the
maco we use to dial a number over PRI.
[macro-pridial]
exten = s,1,GotoIf($[${ARG1:0:2} != 00]?noint)
exten = s,n,Set(DENYINT=${DB(denyinternational/${CALLERIDNUM})})
exten = s,n,GotoIf($[ ${DENYINT} = yes ]?congestion)
exten = s,n(noint),Set(BLOCKCID=${DB(blockcid/${CALLERIDNUM})})
exten = s,n,GotoIf($[ ${BLOCKCID} = yes ]?prohib:cont)
exten = s,n(prohib),SetCallerPres(prohib)
exten = s,n(cont),Dial(ZAP/g1/${ARG1},60,Tr)
exten = s,n,Set(CDR(userfield)=${HANGUPCAUSE}.${DIALSTATUS})
exten = s,n,GotoIf($[ ${DIALSTATUS} = BUSY ]?busy)
exten = s,n,GotoIf($[ ${DIALSTATUS} = CONGESTION ]?congestion)
exten = s,n,GotoIf($[ ${HANGUPCAUSE} = 28 ]?unrecognised)
exten = s,n,GotoIf($[ ${HANGUPCAUSE} = 1 ]?discon)
exten = s,n,GotoIf($[ ${DIALSTATUS} = CHANUNAVAIL ]?congestion)
exten = s,n,Hangup
exten = s,n(busy),Busy
exten = s,n(congestion),GotoIf($[ ${HANGUPCAUSE} = 34 ]?error)
exten = s,n,Congestion
exten = s,n(error),Answer
exten = s,n,SendText(${HANGUPCAUSE}: ERROR: No channels available)
exten = s,n,Wait(1)
exten = s,n,Playback(all-outgoing-lines-unavailable)
exten = s,n,Wait(10)
exten = s,n,Hangup
exten = s,n(unrecognised),Answer
exten = s,n,SendText(${HANGUPCAUSE}: Unrecognised No.)
exten = s,n,Wait(1)
exten = s,n,Playback(that-is-not-rec-phn-num)
exten = s,n,Wait(10)
exten = s,n,Hangup
exten = s,n(discon),Answer
exten = s,n,SendText(${HANGUPCAUSE}:Out Of Service)
exten = s,n,Wait(1)
exten = s,n,Playback(discon-or-out-of-service)
exten = s,n,Wait(10)
exten = s,n,Hangup
On Thu, 2007-03-15 at 10:09, Vidura Senadeera wrote:
Hi,
Please discribe me how we define busy/hang/answer detection with PRI
E1 channels.
Since busydetect, callprogress, busycount giving falts hangup and call
drops what is the solution on PRI channels?
--
Thanks Regards,
Vidura B. Senadeera.
__
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: While the VoIP-Info.org site is down...
DC == Dave Cotton [EMAIL PROTECTED] writes: DC I never could understand how a RAID could be made up using SCSI DC disks seeing that they are certainly not inexpensive. Small Computer Systems Interface. SCSI was vastly cheaper and (perceived as, at least) less reliable than the proper mainframe disks at the time. /Benny ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] busy/hangup/answer detection in PRI E1 channels
Vidura Senadeera wrote: Hi, Please discribe me how we define busy/hang/answer detection with PRI E1 channels. Since busydetect, callprogress, busycount giving falts hangup and call drops what is the solution on PRI channels? PRI channels have call supervision and Asterisk will see the hangup/answers just fine. The busydetect, callprogress, busycount should be removed from your setup. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] voip-info.org status update
If voip-info.us would allow an rsync of the database, I would gladly host a mirror. Since they won't, I have setup the domain listed below. If the community is worried enough/upset enough, please consider putting information at voip-wiki.us. I have no problem with people rsyncing the database off of the central mirror (for consistency sake) or even some other idea to keep the data synced. We have over 80 machines in our datacenter, and none of them have ever had a cataclyzmic failure. I'm not sure what the program with voip-info is. Just a note.. the address is voip-wiki.us, not voip-info.us :) On 3/15/07, OCOSA List Acct. [EMAIL PROTECTED] wrote: Hi All, Personally all of you who are complaining you need to stop becoming part of the problem and become part of the solution. Everyone makes mistakes and if you all depend on James' site so much then you need to donate some time or contact him about getting a mirror. The so called new site at voip-info.us can be mirrored to the .org one. Let's stop all the *%^%$#% cause it's not coming up right nowwe are all in this together and we all have one common goal to use voip and provide a service to our customersLets all come back to earth and get back on target and help this great site get back online. I will offer a mirror site once up no problem may even offer a dedicated server. Actually we can offer a site in (couple of hours) provided James has all the information... Otis Surratt Jr. / [EMAIL PROTECTED] OCOSA Communications, LLC PH: 918.585.9882 x 205 Fax: 918.585.5857 Visit Tulsa's Best Internet Data Center: http://www.ocosa.com/hosting/colo/index.asp Stephen Bosch wrote: shadowym wrote: Hard to expect the business community to take Asterisk seriously when this sort of stuff happens IMHO. I can't understand how 3 of 4 hard drives could just suddenly fail simultaneously. There must be more too it. No UPS? Someone spilled their coffee into it? Something! Either way, it's amateur hour! If I can't be confident enough in an important source of information like this then I can't be confident enough to provide an Asterisk solution to businesses. That's the way I see it. Yea, it's a wiki but it's the best source of info out there. Well, it's always bothered me that the most authoritative and current source of configuration information is an iffy wiki operated by someone not connected with Digium at all. The documentation needs to be better, or we need a better wiki :) The trouble is that Asterisk changes so rapidly that any static document is going to be obsolete before it's finished, so the wiki model makes good sense; but it has to be structured better, at least a little bit the way Wikipedia is operated. -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: What happend to voip-info?
Gordon Henderson wrote: The site is pingable, so I'd suggest it's either crashed in some awkward way and just needs resetting, but you never know... Voip-info.org is down due to a hardware failure. Will be back soon. Thanks for using voip-info.org! [EMAIL PROTECTED] -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] DECT to SIP gateway experiences
Hi, I have a Siemens Gigaset DECT base connected to a Sipura SPA3000. The Message Waiting indicator on the handset works fine in this configuration. (I've used both the S100 and SL100 phones / bases, the operation is identical.) The illuminating Message button on the handset can also be configured to dial *97 so it's a single press if you have voicemail waiting. One 'gotcha' with the SPA3000 (and probably other similar devices) is that by default it internally recognises *nn codes and these all need clearing to allow Asterisk to handle everything. You also may need the latest V3 firmware (even on V2 hardware) especially if you want to use the FXO port as an asterisk trunk. I assume the other Sipura / Linksys devices will have similar features. I've got a SPA2100 dual FXS unit, but I've not got that in use at the moment. I think MWI worked on that but I can't be 100% sure. For the last week I've also been using a Nokia E65 as a WiFi/SIP handset. I'm still experimenting with the overall system setup, but so far this seems to work fine - one handset for both fixed line mobile calls. Robert. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Daniel Pittman Sent: 15 March 2007 03:32 To: asterisk-users@lists.digium.com Subject: [asterisk-users] DECT to SIP gateway experiences G'day. I hope this isn't off-topic for the list. I am looking at an Asterisk setup that includes cordless phones. The three choices I can see, at this stage, are: * wifi phones * an ATA and a cordless analog phone * a DECT to SIP basestation The various wifi phone options don't grab us as suitable -- they are costly, have poor battery life and even the best have pretty mixed reviews. They just don't, at least in Australia, compare well to the non-wifi options. I know a lot of people have success with an ATA and a standard analog cordless phone. We figure that is the fallback, but: It seems to me that a direct SIP to DECT gateway could have significant advantages in terms of supporting the MWI (voicemail) indicator on the DECT phone directly -- there just isn't any way I could trigger it on any of the analog sets I have at the moment. Unfortunately I can't local any information on this; the documentation for the Zyxel DECT gateways and Siemens Gigaset DECT bases don't say *anything* about their supporting MWI hardware from a SIP server. The other killer feature that a DECT base could theoretically offer is some sort of soft menu system -- ADSI, XML, or whatever. That would make for extremely nice integration with the CallerID database in Asterisk, voicemail, etc. So, can anyone comment on support for MWI in the SIP DECT gateways? How about soft menu support? Regards, Daniel -- Digital Infrastructure Solutions -- making IT simple, stable and secure Phone: 0401 155 707email: [EMAIL PROTECTED] http://digital-infrastructure.com.au/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DNIS/DNID
Hi, i had the same problem recently for sip. my scenario was that i connected 2 asterisk servers and dialed from one asterisk to another. and for sending the DNID i used the following comand: exten= 1,1,Dial(SIP/[EMAIL PROTECTED]) here riz is the channel name. hope this works with zap also On 3/15/07, Mark Quitoriano [EMAIL PROTECTED] wrote: Hi i have an asterisk pbx with E1 port connected to another PBX. Im trying to send the DNID/DNIS to the PBX here's my dialplan exten = 888111,1,Dial(ZAP/g2) exten = 888111,n,Hangup() The PBX just get the number 2 as it's DNIS when i change it to ZAP/1 or ZAP/g1 the PBX get the number 1. What should i add to send the extension number as DNID/DNIS? Thanks! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Regards Rizwan Hisham Software Engineer ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DECT to SIP gateway experiences
Am Thu, 15 Mar 2007 14:32:28 +1100 schrieb Daniel Pittman [EMAIL PROTECTED]: It seems to me that a direct SIP to DECT gateway could have significant advantages in terms of supporting the MWI (voicemail) indicator on the DECT phone directly -- there just isn't any way I could trigger it on any of the analog sets I have at the moment. Unfortunately I can't local any information on this; the documentation for the Zyxel DECT gateways and Siemens Gigaset DECT bases don't say *anything* about their supporting MWI hardware from a SIP server. The MWI depends on the vendor who implements this feature. IIRC, MWI is not part of the DECT specifiations so you won't find any generic information regarding this topic. The other killer feature that a DECT base could theoretically offer is some sort of soft menu system -- ADSI, XML, or whatever. Theoretically! The soft menus that you see on DECT handsets are proprietary so they don't work with 3rd party DECT base stations. So, can anyone comment on support for MWI in the SIP DECT gateways? How about soft menu support? MWI works on the KIRK Wireless gateways we are using. There is support for soft menus which come from the DECT base but they cannot be sent by the SIP server. MWI only works on KIRK handsets; our attempts with Siemens and Philips handsets were unsuccessful. Best regards, Henning Holtschneider -- LocaNet oHG - http://www.loca.net Lindemannstrasse 81, D-44137 Dortmund tel +49 231 91596-25, fax +49 231 91596-55 sip [EMAIL PROTECTED] Registergericht Amtsgericht Dortmund HRA 14208 Geschäftsführer Sven Haufe, Henning Holtschneider signature.asc Description: PGP signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: What happend to voip-info?
I can mirror too, if needed. I have lots of bandwidth, email me off list On 3/15/07, Tomislav Parcina [EMAIL PROTECTED] wrote: Gordon Henderson wrote: The site is pingable, so I'd suggest it's either crashed in some awkward way and just needs resetting, but you never know... Voip-info.org is down due to a hardware failure. Will be back soon. Thanks for using voip-info.org! [EMAIL PROTECTED] -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] voip-info.org status update
On Thursday 15 March 2007 12:32 am, shadowym wrote: Hard to expect the business community to take Asterisk seriously when this sort of stuff happens IMHO. I can't understand how 3 of 4 hard drives could just suddenly fail simultaneously. There must be more too it. No UPS? Someone spilled their coffee into it? Something! Obviously you didn't read Google's research paper on drive failures. And aside from that, you're also obviously pushing an agenda with these inciteful comments. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] voip-info.org status update
Andrew Kohlsmith wrote: On Thursday 15 March 2007 12:32 am, shadowym wrote: Hard to expect the business community to take Asterisk seriously when this sort of stuff happens IMHO. I can't understand how 3 of 4 hard drives could just suddenly fail simultaneously. There must be more too it. No UPS? Someone spilled their coffee into it? Something! Obviously you didn't read Google's research paper on drive failures. And aside from that, you're also obviously pushing an agenda with these inciteful comments. There are also hardware raid controller failures to deal with. To make it worse, allowing the wrong person to replace failed drives and controllers can be disastrous. As for backup, they reported that it will be back up and running. That would be hard to do without backup when 3 of 4 drives failed. Suppose you are running raid5 on 3 drives with a 4th as ready spare? Suppose that overheating is a factor? One of the drives fails and the system begins building the spare in the background. Then the spare fails. Then 1 of the 2 remaining drives fails. Some of us can spec out a fancy new server that comes with a 24/7 fast-response service contract. Then the complainers can donate the money to buy it and put it in a good data center. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] While the VoIP-Info.org site is down...
Stephen Bosch wrote: Patrick May wrote: On Wed, Mar 14, 2007 at 10:05:20PM -0400, Matt wrote: Yikes.. you'd think a server would be running RAID. At any rate.. Please feel free to visit http://www.voip-wiki.us I have set this up to be able to hold information for the Asterisk community. I will also gladly allow others to mirror it. It is sitting in a climate controlled data center in Central PA on a server with RAID. Additionally, it is at the end of 95Megabytes/second on a BGP redundant connection. Please feel free to use it, if the community feels it can be useful... additionally, I would love to setup some rsync mirrors with others so that we can have redundant backups of this very valuable information. The previous message to the list was they lost 3 of 4 drives in the array. I'm not sure of any RAID that can sustain 75% hardware loss and still function. As somebody else has already pointed out -- There must be more to it. Let's say three of four drives failed -- the odds of them failing at the same time are vanishingly slim; but if you're not paying attention, and you operate with a degraded volume, well... then you get what you deserve. RAID or no RAID, the site should have one or more mirrors. -Stephen- The odds of multiple drive failure are a lot higher than you think. Failing power supplies or power spikes are common to all drives and controller failure on a drive can throw noise back onto the SCSI bus causing corruption on other drives. Although the other affected drives are not physically damaged, your data has evaporated none the less. We have already had one multi-drive RAID failure on our main file server (only one drive was physically failed) and a single drive and power supply failure on our Asterisk box. RAID 1 and redundant power supplies saved the day. Spring and Fall are the special Hardware Failure Seasons! Seems to affect power supplies, hard drives and light bulbs in particular. regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] voip-info.org status update
Hard to expect the business community to take Asterisk seriously when this sort of stuff happens IMHO. I can't understand how 3 of 4 hard drives could just suddenly fail simultaneously. There must be more too it. No UPS? Someone spilled their coffee into it? Something! Sure, there always is. For example, from our own little cache of stories: Bad component in the power supply blows, momentarily spiking voltages throughout the server. Colo cooling failed and temps rose ten degrees, baking the drives a bit. Someone let slip with a cart and banged into the rack. Drives were spinning continuously for several years, and then power went out. Two of four don't spin back up. Anyone who's been in the industry for any length of time will have stories. Some of them even interesting. I remember a few years ago when the roof/wall of an ATT data center was destroyed during a storm. Either way, it's amateur hour! If I can't be confident enough in an important source of information like this then I can't be confident enough to provide an Asterisk solution to businesses. That's the way I see it. Yea, it's a wiki but it's the best source of info out there. If you're not smart enough to have a local snapshot of anything that is critical to what you're providing to customers, then, well, you're right, it *is* amateur hour. As for voip-info.org, I cannot comprehend why you would attack a very nice public service in this manner. Perhaps I am mistaken, but I thought that it was a general VOIP resource, not specific to Asterisk. While I have found it a very convenient interface to Asterisk information, you seem to be suggesting that it is the only source of information. It is not. We ought to all be thanking the fine folks at voip-info.org for their fantastic store of information. Hopefully, if there is any need for assistance to cover additional backup hosting, cash to cover the expense of new drives, or whatever they happen to need, they'll post here and let us all know. We're happy to make a no-strings-attached contribution of some sort, because the resource has been quite useful over the years. ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again. - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] voip-info.org status update
On 3/15/07, Andrew Kohlsmith [EMAIL PROTECTED] wrote: Obviously you didn't read Google's research paper on drive failures. This one? http://labs.google.com/papers/disk_failures.html -HJC ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] While the VoIP-Info.org site is down...
Just a heads up guys. I'm currently attempting to recover the website through spidering the Google cache. I'll let you know how it turns out. On 3/15/07, Drew Gibson [EMAIL PROTECTED] wrote: Stephen Bosch wrote: Patrick May wrote: On Wed, Mar 14, 2007 at 10:05:20PM -0400, Matt wrote: Yikes.. you'd think a server would be running RAID. At any rate.. Please feel free to visit http://www.voip-wiki.us I have set this up to be able to hold information for the Asterisk community. I will also gladly allow others to mirror it. It is sitting in a climate controlled data center in Central PA on a server with RAID. Additionally, it is at the end of 95Megabytes/second on a BGP redundant connection. Please feel free to use it, if the community feels it can be useful... additionally, I would love to setup some rsync mirrors with others so that we can have redundant backups of this very valuable information. The previous message to the list was they lost 3 of 4 drives in the array. I'm not sure of any RAID that can sustain 75% hardware loss and still function. As somebody else has already pointed out -- There must be more to it. Let's say three of four drives failed -- the odds of them failing at the same time are vanishingly slim; but if you're not paying attention, and you operate with a degraded volume, well... then you get what you deserve. RAID or no RAID, the site should have one or more mirrors. -Stephen- The odds of multiple drive failure are a lot higher than you think. Failing power supplies or power spikes are common to all drives and controller failure on a drive can throw noise back onto the SCSI bus causing corruption on other drives. Although the other affected drives are not physically damaged, your data has evaporated none the less. We have already had one multi-drive RAID failure on our main file server (only one drive was physically failed) and a single drive and power supply failure on our Asterisk box. RAID 1 and redundant power supplies saved the day. Spring and Fall are the special Hardware Failure Seasons! Seems to affect power supplies, hard drives and light bulbs in particular. regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mitchel Constantin Snap - A desktop user interface for Asterisk www.snapanumber.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Which SIP method/option to display a short text message ?
Hi, After further research, it seems SIP MESSAGE rfc3428) and SIP INFO (rfc2976) methods could be the more relevant for this feature. I'm still wondering whether SIP hardphones or Asterisk implement these methods in such a way you could make a welcome message, for example, appear on you contact phone screen. Cheers ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: DECT to SIP gateway experiences
HH == Henning Holtschneider [EMAIL PROTECTED] writes: HH MWI works on the KIRK Wireless gateways we are using. Kirk ip600/3? If so, how do you configure it? /Benny ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] voip-info.org status update
shadowym wrote: . I can't understand how 3 of 4 hard drives could just suddenly fail simultaneously. There must be more too it. No UPS? Someone spilled their coffee into it? Something! If you can't understand it, do some research before mouthing off (as everyone on this list is encouraged to do). Multi-drive failures are common, one drive or power supply fails and corrupts or damages other drives on the bus. Either way, it's amateur hour! You said it! regards, Drew ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] qozap: t3 timer expired for span ...
Hi all message: qozap: t4 timer expired for span 2 qozap: t4 timer expired for span 3 qozap: t3 timer expired for span 2 qozap t3 timer expired for span 3 wow -- what does this mean!? all of a sudden showing up on my server ... no change after reboot .. Junghanns QuadBRI card in place affecting outgoing faxing?! (between bridged TDM400 analog card and QuadBRI) Not a clue why this is .. incoming/outgoing voice calls work, incoming faxes even work but when outgoing fax is dialed, says no one is availale to answer at this time The error has not ever been there before and as far as I know, no isdn wiring has been changed or anything ideas, appreciated! -- Chris Earle System Solutions Specialist -- This message has been scanned for viruses and dangerous content and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] snom led not working with asterisk 1.4.1
Hi, I'm testing Asterisk 1.4.1 with Snom phones but leds are not working to show which devices are busy/not connected. The same phone worked with Asterisk 1.2.9.1. I would appreciate anyone who knows how to setup Asterisk 1.4.1 to behave as 1.2.9.1. TIA Giorgio Incantalupo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: SIP unicode support ?
Benny Amorsen wrote: KD == Klaus Darilion [EMAIL PROTECTED] writes: KD Hi! Is there unicode support in Asterisk for SIP? E.g. How can I KD have a displayname with special characters? KD E.g. if I want to have the Umlaut ä in the display name: KD callerid=Jeff Gräser 11 Is your sip.conf UTF-8-encoded? Will it work with UTF-8 encoded sip.conf? -- Klaus Darilion nic.at ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] voip-info.org status update
On Thu, Mar 15, 2007 at 08:08:57AM -0600, Joe Greco said: Anyone who's been in the industry for any length of time will have stories. Some of them even interesting. I remember a few years ago when the roof/wall of an ATT data center was destroyed during a storm. Yep. Ashburn VA datacenter. Tornado hit it. Water was pouring in on someone's servers, and surprisingly they didn't go down! ATT did bring them down due to safety concerns. Our servers, in that datacenter, were unaffected and had zero downtime. Most of the damage was to unoccupied portions of the datacenter. I've also had multiple drives fail simultaineously on a 0+1 Raid. It totally sucks when it happens. One online spare was not enough and didn't have time to rebuild before the second and third drives failed. We did have backups and were able to restore everything within 4 hours, but we still lost some data between the last backup the night before and when the drives failed. These were not cheapo IDE drives either, they were server grade scsi (HP branded Seagates.) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] qozap: t3 timer expired for span ...
On Thu, Mar 15, 2007 at 10:30:23AM -0500, Chris Earle (CBL) wrote: Hi all message: qozap: t4 timer expired for span 2 qozap: t4 timer expired for span 3 qozap: t3 timer expired for span 2 qozap t3 timer expired for span 3 Which version is it of bristuff? wow -- what does this mean!? all of a sudden showing up on my server ... no change after reboot .. Junghanns QuadBRI card in place Anything connected to it? Where exactly? affecting outgoing faxing?! (between bridged TDM400 analog card and QuadBRI) Not a clue why this is .. incoming/outgoing voice calls work, incoming faxes even work but when outgoing fax is dialed, says no one is availale to answer at this time The error has not ever been there before and as far as I know, no isdn wiring has been changed or anything ideas, appreciated! -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] snom led not working with asterisk 1.4.1
On Thu, 2007-03-15 at 15:33 +0100, Giorgio Incantalupo wrote: Hi, I'm testing Asterisk 1.4.1 with Snom phones but leds are not working to show which devices are busy/not connected. The same phone worked with Asterisk 1.2.9.1. I would appreciate anyone who knows how to setup Asterisk 1.4.1 to behave as 1.2.9.1. Giorgio-- That's a pretty generic question! But that aside, there's been a substantive change in the configs for SIP phones, that could easily affect your device state monitoring. So, suggestion: read the example sip config file in the src/configs dir, pay close attention to stuff like call-limit, the limitonpeers stuff, etc, and then make sure you update all your phone entries in sip.conf. Restart asterisk, or reload sip, and hopefully your lights will work. In general, EVERYONE, here's some advise: When you upgrade from version 1.x to 1.(x+2), always review ALL your config files against the new config file examples. Things change! Hopefully, for the better! murf -- Steve Murphy Software Developer Digium ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] voip-info.org status update
OCOSA List Acct. wrote: Hi All, Personally all of you who are complaining you need to stop becoming part of the problem and become part of the solution. Everyone makes mistakes and if you all depend on James' site so much then you need to donate some time or contact him about getting a mirror. The point has been made amply -- he refused requests to mirror. That is ultimately why people are angry. You want to talk community? Then walk the walk. And for all the other people who say that a free service is an excuse for amateur execution, it's not, and there are ample examples. Community means just that -- you involve the community. You don't sit on the egg because you're worried about losing Google ad revenue, or the extra cost of bandwidth that comes with mirroring. If it's a thankless enterprise for James, then he should ask for help! It's been offered. And finally -- no, the wiki is not the only source of information, but it happens to have become the only *comprehensive* source. Being unmirrored, that's not a good circumstance. Here's an undeniable fact -- look how much list traffic this outage has generated! People have been using and depending on it. Some people have pointed out things that could be improved; offers of help have been ignored or worse, rejected. That doesn't sound like community to me. Community is everybody working together. One person sitting on information, no matter the personal sacrifice, does not constitute community. -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] While the VoIP-Info.org site is down...
mitcheloc wrote: Just a heads up guys. I'm currently attempting to recover the website through spidering the Google cache. I'll let you know how it turns out. Great stuff! I'll be keen to hear how it goes. -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] voip-info.org status update
Matt you are right it is voip-wiki.us I looked at my browser tab. LOL sorry...but my POV still stands... good day. Otis Surratt Jr. / [EMAIL PROTECTED] OCOSA Communications, LLC PH: 918.585.9882 x 205 Fax: 918.585.5857 Visit Tulsa's Best Internet Data Center: http://www.ocosa.com/hosting/colo/index.asp Matt wrote: If voip-info.us http://voip-info.us would allow an rsync of the database, I would gladly host a mirror. Since they won't, I have setup the domain listed below. If the community is worried enough/upset enough, please consider putting information at voip-wiki.us http://voip-wiki.us. I have no problem with people rsyncing the database off of the central mirror (for consistency sake) or even some other idea to keep the data synced. We have over 80 machines in our datacenter, and none of them have ever had a cataclyzmic failure. I'm not sure what the program with voip-info is. Just a note.. the address is voip-wiki.us http://voip-wiki.us, not voip-info.us http://voip-info.us :) On 3/15/07, *OCOSA List Acct. * [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi All, Personally all of you who are complaining you need to stop becoming part of the problem and become part of the solution. Everyone makes mistakes and if you all depend on James' site so much then you need to donate some time or contact him about getting a mirror. The so called new site at voip-info.us http://voip-info.us can be mirrored to the .org one. Let's stop all the *%^%$#% cause it's not coming up right nowwe are all in this together and we all have one common goal to use voip and provide a service to our customersLets all come back to earth and get back on target and help this great site get back online. I will offer a mirror site once up no problem may even offer a dedicated server. Actually we can offer a site in (couple of hours) provided James has all the information... Otis Surratt Jr. / [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] OCOSA Communications, LLC PH: 918.585.9882 x 205 Fax: 918.585.5857 Visit Tulsa's Best Internet Data Center: http://www.ocosa.com/hosting/colo/index.asp http://www.ocosa.com/hosting/colo/index.asp Stephen Bosch wrote: shadowym wrote: Hard to expect the business community to take Asterisk seriously when this sort of stuff happens IMHO. I can't understand how 3 of 4 hard drives could just suddenly fail simultaneously. There must be more too it. No UPS? Someone spilled their coffee into it? Something! Either way, it's amateur hour! If I can't be confident enough in an important source of information like this then I can't be confident enough to provide an Asterisk solution to businesses. That's the way I see it. Yea, it's a wiki but it's the best source of info out there. Well, it's always bothered me that the most authoritative and current source of configuration information is an iffy wiki operated by someone not connected with Digium at all. The documentation needs to be better, or we need a better wiki :) The trouble is that Asterisk changes so rapidly that any static document is going to be obsolete before it's finished, so the wiki model makes good sense; but it has to be structured better, at least a little bit the way Wikipedia is operated. -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
OT: Re: [asterisk-users] voip-info.org status update
wrote: *snipped If I can't be confident enough in an important source of information like this then I can't be confident enough to provide an Asterisk solution to businesses. That's the way I see it. Yea, it's a wiki but it's the best source of info out there. *snipped sorry to see you go! that is unless you were being *overly dramatic*. G ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: SIP unicode support ?
Klaus Darilion wrote: Benny Amorsen wrote: KD == Klaus Darilion [EMAIL PROTECTED] writes: KD Hi! Is there unicode support in Asterisk for SIP? E.g. How can I KD have a displayname with special characters? KD E.g. if I want to have the Umlaut ä in the display name: KD callerid=Jeff Gräser 11 Is your sip.conf UTF-8-encoded? Will it work with UTF-8 encoded sip.conf? By converting sip.conf into UTF-8 I manged to display german Umlaute äöü on a SNOM 360 softphone. But I had no luck with háček like ǎ. But I think this is a limitation of the SNOM phone. Obviously Asterisk treat the UTF-8 encoding just as characters and send them without knowing that is is unicode. Is it safe to use UTF-8 encoded configuration files? regards klaus -- Klaus Darilion nic.at ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FW: Seamless Multi Office Asterisk Deployment
Brandon, What it sounds like you are looking at as far as having the phones register to the system and then have users login to a phone should be possible, I have not tried. I would suspect that you could build a dial plan menu to prompt the caller for their credentials and then take the phone's identification and add an entry to a database for that phone and the caller id, extension, voice mail box etc.. You could then use the realtime engine to query the table for the information when an extension is dialed. Doing the login through a sort of IVR would make it hardware independent. One note on the QOS, You might be right about it being okay with only 3 calls at a time, but I would offer this example. We us a MPLS network, in which most site have 384k and our main sites have 1.5mb. When I first placed a test call over the link using gsm from a 384 site to the main site, they had no issues, but I had terrible problems with audio being dropped or delayed and playing over top itself. I implemented QOS because the connection is not only for voice, so I could at least give priority to my audio and set aside bandwidth for it. On your time frame, it is hard to say, your users and existing hardware and training all factor into it. Having done asterisk systems before, I have deployed small sites, like 2-5 people in very short time frames, typically a few days building the system off site and testing then a week or less on site dealing with wiring, setup/testing and training. On a site of your size I would almost consider spending a couple weeks on site. You mentioned trixbox, I started with [EMAIL PROTECTED] myself and must say it was a great thrill to place a call between two softphones after a hour or so. But what I eventually realized was that if I had to troubleshoot the dialplan I was going to be lost in macros and AGI. I started out writing my configs and then using svn repositories for each site and copying in a base config for each new site. Worked great for a while and I knew the dialplan inside out. I'm now moving more and more to realtime and database storage for config files and dialplan sections which make managing multiple sites config files much easier. Also the use of Dundi in [EMAIL PROTECTED] was not through the GUI, and I understood it much better once I began using the files. The bottom line: in my experience [EMAIL PROTECTED]/Trixbox/freepbx are great ways to get your feet wet and are a proof of concept and even great for a basic system, but for what you are wanting to do and what I did. Asterisk is the only way to go. If your worried about not knowing enough, goto a bootcamp or some other training. If you want ease administration for several IT people, then you could look at some of the web interfaces that connect and edit the conf files. Hope that helps On 3/13/07, Brandon Comouche [EMAIL PROTECTED] wrote: For startes I will keep it on the list and we can discuss some major concepts, and I will possibly make some contact off list later for the nitty-gritty :) In-reply to Steve: I did have a look at the bicomsystems product and it does appear to do everything I am looking for. However, I have looked in to vendor systems and have decided to go with an Asterisk system. Hench asking for assistance on the Asterisk mailing list ;) On the discussion at hand: At this time I am not going to worry about the QoS with my T1 network lines, I have been wondering what the quality will be like. I do not plan to have more than maybe three calls on a line at peak times. But I know that there will be more in the future. I am working with a total employee base of around 30, and the remote offices have two to four employees at a time, not a huge traffic demand. What I am most curious about at this time is the methods used to move from server to server. *Ideally* I would like to sit down at a phone, enter my extension/password and have that phone ring as my extension. Essentially, I would like a log in system on the phone. This presents me with two issues: I have to make my phones allow simple logon as a SIP device, and I need to get my credentials to move between Asterisk servers. What methods have others used, or where should I look for more information? At this point I have two Polycom phones (430 and 501) for testing, they seem to be talked about as very flexible. If they will not allow me to add a user friendly login prompt, maybe I need to find alternatives though. But this is the Asterisk list and I don't want to go too far off topic, so the main concern is how I would synchronize my information between asterisk servers. One final topic on this message I would like to cover is time frame. I am thinking maybe around 6 months to have at least a partial functioning system up and tested. By partial I mean deployable with a basic infrastructure feature set. I don't know if this is too little time or too much time. My co-workers are excited about what Asterisk has to offer. Any other thoughts on
RE: [asterisk-users] voip-info.org status update
I was expecting a response like this. First of all. I do NOT rely on any one source of information seeing as how so much of it is outdate and/or just plain wrong. I always try get at least 2 or 3 sources of info. Hey, don't blame the messenger. EVERYTHING comes with a manual right? Why do you suppose that is? Has nothing to do with how much someone knows or doesn't know about the product. In the case of Asterisk, yes it is changing rapidly and hardcopies quickly get outdated so a wiki kind of makes sense. As far as trying to be part of the solution instead of the problem. I have contributed a LOT of info to that wiki. I feel I have earned the right to critisize it's availability or lack thereof. If you want mirrors there appears to be no shortage of people willing to provide this if you would ONLY ask! -Original Message- From: Bill Hackensack [mailto:[EMAIL PROTECTED] Sent: Wednesday, March 14, 2007 11:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] voip-info.org status update On 3/15/07, OCOSA List Acct. [EMAIL PROTECTED] wrote: Hi All, and if you all depend on James' site so much then you need to donate some time or contact him about getting a mirror. The so called new site Google didn't go down, and if you had bothered searching the archives of this list you would have known this has all been discussed before. Whoever the powers that be that run the wiki did not want help before. People had begged to be able to mirror the site. This could have all been avoided. Maybe this time they will be a little more interested in getting some mirrors. And as for whoever said this reflects on Asterisk, if you have to depend on the wiki to help your clients maybe you need to step back and see how this reflects on you. If you're charging customers by the hour for something, you need to know the stuff and not have to spend most of that time searching for answers. Yeah, I know, most consultants these days have no clue about what they sell to clients. We see that every day on this list alone. Step up and be different. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] voip-info.org status update
A percentage of all my profits go back to the community. What about you? -Original Message- From: Gordon Henderson [mailto:[EMAIL PROTECTED] Sent: Thursday, March 15, 2007 1:42 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] voip-info.org status update On Wed, 14 Mar 2007, shadowym wrote: Hard to expect the business community to take Asterisk seriously when this sort of stuff happens IMHO. I think you hit the nail on the head with one word: community. Asterisk is free, community supported, and the voip-info site has been provided for free - with the support of the community. The site would appear to be financially supported by a small number of quite unobtrusive google ads, and therein lies the problem... Hosting isn't free. If you can't/won't pay for hosting, then you have to support it by advertising. I can sell you web space/servers/co-lo facilities with full disk/server/location redundancy, backups and so on, but would you be willing to pay for it? Probably not. So you takes your chances with a popular hosting company, put in a small number of google ads to pay for a basic hosting package and go with it. After-all, there are millions of websites hosted on millions of servers throughout the world - it's a highly competitive business - there are offers of hosting for £1 a month or even less, but do you think it's a sustainable model? I don't. Well, maybe it is when you have 1000s of clients with 10s of 1000s of websites (spread over 100s of servers!) but with scale comes more issues. I can't understand how 3 of 4 hard drives could just suddenly fail simultaneously. There must be more too it. No UPS? Someone spilled their coffee into it? Something! That does strike me as odd, but I've seen it myself with a bad batch of disks. (IBM DeathStar, Hitachi, etc.) You usually get warnings, but if you're employing monkeys paying them peanuts, then they usually just treat them as fire forget once installed in the rack and plumbed into their automated selling/billing system. Either way, it's amateur hour! It's the way 99% of all co-lo facilities work. Buy big, sell cheap with little or no SLA - hope that the hardware/premises/internet is reliable enough, employ monkeys, pay peanuts. If you want quality, then be prepared to pay for it, and £1 a month does not give you quality IMO, and in my experience as someone who runs a small co-lo facility, people will not pay for quality hosting. A quality server costs me £650, more if the client insists on a Dull. Sure, I can put together something with pair of disks for under £300, but I know (from experience!) it won't last the 4+ years I want it to last, nor deliver the preformance my clients (who are willing to pay for such a service) demand. I'm not blaming James here because that's the way it is! I bet he's spent 100s of hours (unpaid) setting it up, running it and maintaining it, and resorted to google ads. purely to fund it. I don't envy him at all. If I can't be confident enough in an important source of information like this then I can't be confident enough to provide an Asterisk solution to businesses. That's the way I see it. Yea, it's a wiki but it's the best source of info out there. So how much are you willing to pay to support such a service? Gordon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] While the VoIP-Info.org site is down...
75% failure at EXACTLY the same time? Come on! We all know better than that. Probably lost one drive at a time over weeks or months with no automated warnings! Amateur hour! -Original Message- From: Patrick May [mailto:[EMAIL PROTECTED] Sent: Wednesday, March 14, 2007 8:10 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] While the VoIP-Info.org site is down... On Wed, Mar 14, 2007 at 10:05:20PM -0400, Matt wrote: Yikes.. you'd think a server would be running RAID. At any rate.. Please feel free to visit http://www.voip-wiki.us I have set this up to be able to hold information for the Asterisk community. I will also gladly allow others to mirror it. It is sitting in a climate controlled data center in Central PA on a server with RAID. Additionally, it is at the end of 95Megabytes/second on a BGP redundant connection. Please feel free to use it, if the community feels it can be useful... additionally, I would love to setup some rsync mirrors with others so that we can have redundant backups of this very valuable information. The previous message to the list was they lost 3 of 4 drives in the array. I'm not sure of any RAID that can sustain 75% hardware loss and still function. Patrick ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: OT: Re: [asterisk-users] voip-info.org status update
lol yeh all will miss you :D . . its like stopping to use internet if google is down sometime . On 15/03/07, Richard Lyman [EMAIL PROTECTED] wrote: wrote: *snipped If I can't be confident enough in an important source of information like this then I can't be confident enough to provide an Asterisk solution to businesses. That's the way I see it. Yea, it's a wiki but it's the best source of info out there. *snipped sorry to see you go! that is unless you were being *overly dramatic*. G ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voip-Wiki Site Information
Matt wrote: Community, I have put up www.voip-wiki.us http://www.voip-wiki.us My apologies to our fellow Asteristians outside the us... this was the only easy domain available. What's wrong with voip-info.org ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk n-way call problem
Hi,
i am using the n-way-call dialplan solution found on voip-info. i have
added its entry in applicationmap of features.conf file. the problem
is..its not working. to activate the n-way call i dial *0 but nothing
happens. i have played around with dtmf and codec settings but no success.
the extensions and sip configuration is below if you want to have a look. I
dont have any clue why its not working.
###extensions.conf###
[local]
exten = _XX,1,Set(DYNAMIC_FEATURES=nway-start)
exten = _XX,2,SIPDtmfMode(inband)
exten= 10,3,Dial(SIP/saad,,tT)
exten= 10,n,Hangup
exten= 11,3,Dial(SIP/riz,,tT)
exten= 11,n,Hangup
exten= 12,3,Dial(SIP/rehmat,,tT)
exten= 12,n,Hangup
[dynamic-nway]
exten = _XXX,1,Answer
exten = _XXX,n,Set(CONFNO=${EXTEN})
exten = _XXX,n,Set(MEETME_EXIT_CONTEXT=dynamic-nway-invite)
exten = _XXX,n,Set(DYNAMIC_FEATURES=)
exten = _XXX,n,MeetMe(${CONFNO},pdMX)
exten = _XXX,n,Hangup
[dynamic-nway-invite]
exten = 0,1,Read(DEST,dial,,i)
exten = 0,n,Set(DYNAMIC_FEATURES=nway-inv#nway-noinv)
exten = 0,n,Dial(Local/[EMAIL PROTECTED],,g)
exten = 0,n,Set(DYNAMIC_FEATURES=)
exten = 0,n,Goto(dynamic-nway,${CONFNO},1)
exten = i,1,Goto(dynamic-nway,${CONFNO},1)
[dynamic-nway-dest]
exten = _XXX,1,Dial(SIP/${EXTEN})
[macro-nway-start]
exten = s,1,Set(CONFNO=${FindFreeConf()})
;exten = s,n,ChannelRedirect(${BRIDGEPEER},dynamic-nway,${CONFNO},1)
exten = s,n,ChannelRedirect(${BRIDGEPEER},dynamic-nway,${CONFNO},1)
exten = s,n,Read(DEST,dial,,i)
exten = s,n,Set(DYNAMIC_FEATURES=nway-inv#nway-noinv)
exten = s,n,Dial(Local/[EMAIL PROTECTED],,g)
exten = s,n,Set(DYNAMIC_FEATURES=)
exten = s,n,Goto(dynamic-nway,${CONFNO},1)
[macro-nway-ok]
exten = s,1,ChannelRedirect(${BRIDGEPEER},dynamic-nway,${CONFNO},1)
[macro-nway-notok]
exten = s,1,SoftHangup(${BRIDGEPEER})
#sip.conf###
[saad]
userid=saad
secret=1234
host=dynamic
type=friend
context=local
qualify=4000
insecure=invite,port
dtmfmode = inband
disallow = all
allow=ulaw
[riz]
userid=riz
secret=1234
host=dynamic
type=friend
context=local
qualify=4000
dtmfmode = inband
disallow = all
allow=ulaw
[rehmat]
userid=rehmat
secret=1234
host=dynamic
type=friend
context=local
qualify=4000
insecure=invite,port
dtmfmode = inband
disallow = all
allow=ulaw
#features.conf###
[applicationmap]
nway-start = *0,self,caller,Macro,nway-start
nway-inv = **,self,caller,Macro,nway-ok
nway-noinv = *#,self,caller,Macro,nway-notok
;nway-start = *0,caller,Macro,nway-start
;nway-inv = **,caller,Macro,nway-ok
;nway-noinv = *#,caller,Macro,nway-notok
--
Regards
Rizwan Hisham
Software Engineer
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] voip-info.org status update
I'm curious what you think that agenda might be? If it is to push the perception of Asterisk as a solid alternative to Traditional PBX's into the mainstream then I am guilty as charged! -Original Message- From: Andrew Kohlsmith [mailto:[EMAIL PROTECTED] Sent: Thursday, March 15, 2007 6:41 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] voip-info.org status update On Thursday 15 March 2007 12:32 am, shadowym wrote: Hard to expect the business community to take Asterisk seriously when this sort of stuff happens IMHO. I can't understand how 3 of 4 hard drives could just suddenly fail simultaneously. There must be more too it. No UPS? Someone spilled their coffee into it? Something! Obviously you didn't read Google's research paper on drive failures. And aside from that, you're also obviously pushing an agenda with these inciteful comments. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Linksys not Ringing
I do not have any answer int he dialplan. what I mean is that when I call any other SIP phone is does the answer in the CLI. Even if I put and answer() in the dialplan still no ringing Jason Luki wrote: shouldn't there be an answer in there somewhere?... like... No... you can (and probably should) Dial() an extension before answering the incoming call. Do a sip debug and see if the Sipura is getting the INVITE message (and responding with an ACK), and if it sends back a RINGING message. Something strange is going here, and my bet is on some kind of NAT screw-up. --Luki ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] voip-info.org status update
A percentage of all my profits go back to the community. What about you? I think we've been contributing various resources to various online Internet communities for about two decades, more if you go back into the BBS era. We're still dedicating more than a quarter of a gigabit of bandwidth to the free exchange of Usenet news, something we've been doing since the '80's. Challenging people on this list about what they've contributed to the community over the years is going to be a losing proposition. I guarantee it. Don't do it, you make yourself look silly. ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again. - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] voip-info.org status update
Nobody said anything about power supply problems did they? Besides, this has NOTHING to do with one machine and what may or may not have happened to it. It has EVERYTHING to do with the availability of the information however that may be acomplished. Half that info on the wiki is out of date or just plain wrong anyways so maybe someone will use this as an opportunity of alleviate that. The few times I tried to delete or change info people got upset so I just won't bother anymore. If you want to blame the messenger feel free. At least it's getting attention now so hopefully things will improve. _ From: Drew Gibson [mailto:[EMAIL PROTECTED] Sent: Thursday, March 15, 2007 7:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] voip-info.org status update shadowym wrote: I can't understand how 3 of 4 hard drives could just suddenly fail simultaneously. There must be more too it. No UPS? Someone spilled their coffee into it? Something! If you can't understand it, do some research before mouthing off (as everyone on this list is encouraged to do). Multi-drive failures are common, one drive or power supply fails and corrupts or damages other drives on the bus. Either way, it's amateur hour! You said it! regards, Drew ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] voip-info.org status update
shadowym wrote: A percentage of all my profits go back to the community. What about you? -Original Message- From: Gordon Henderson [mailto:[EMAIL PROTECTED] Sent: Thursday, March 15, 2007 1:42 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] voip-info.org status update On Wed, 14 Mar 2007, shadowym wrote: Hard to expect the business community to take Asterisk seriously when this sort of stuff happens IMHO. I think you hit the nail on the head with one word: community. Asterisk is free, community supported, and the voip-info site has been provided for free - with the support of the community. The site would appear to be financially supported by a small number of quite unobtrusive google ads, and therein lies the problem... Hosting isn't free. If you can't/won't pay for hosting, then you have to support it by advertising. I can sell you web space/servers/co-lo facilities with full disk/server/location redundancy, backups and so on, but would you be willing to pay for it? Probably not. So you takes your chances with a popular hosting company, put in a small number of google ads to pay for a basic hosting package and go with it. After-all, there are millions of websites hosted on millions of servers throughout the world - it's a highly competitive business - there are offers of hosting for £1 a month or even less, but do you think it's a sustainable model? I don't. Well, maybe it is when you have 1000s of clients with 10s of 1000s of websites (spread over 100s of servers!) but with scale comes more issues. I can't understand how 3 of 4 hard drives could just suddenly fail simultaneously. There must be more too it. No UPS? Someone spilled their coffee into it? Something! That does strike me as odd, but I've seen it myself with a bad batch of disks. (IBM DeathStar, Hitachi, etc.) You usually get warnings, but if you're employing monkeys paying them peanuts, then they usually just treat them as fire forget once installed in the rack and plumbed into their automated selling/billing system. Either way, it's amateur hour! It's the way 99% of all co-lo facilities work. Buy big, sell cheap with little or no SLA - hope that the hardware/premises/internet is reliable enough, employ monkeys, pay peanuts. If you want quality, then be prepared to pay for it, and £1 a month does not give you quality IMO, and in my experience as someone who runs a small co-lo facility, people will not pay for quality hosting. A quality server costs me £650, more if the client insists on a Dull. Sure, I can put together something with pair of disks for under £300, but I know (from experience!) it won't last the 4+ years I want it to last, nor deliver the preformance my clients (who are willing to pay for such a service) demand. I'm not blaming James here because that's the way it is! I bet he's spent 100s of hours (unpaid) setting it up, running it and maintaining it, and resorted to google ads. purely to fund it. I don't envy him at all. If I can't be confident enough in an important source of information like this then I can't be confident enough to provide an Asterisk solution to businesses. That's the way I see it. Yea, it's a wiki but it's the best source of info out there. So how much are you willing to pay to support such a service? Gordon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Guys don't you think this thread has gone on long enough. We all support this community!!! My suggestion would be the same as many others have stated, another place for additional information is great. I would not want to totally write off the guys at voip-info. I would suggest that we create a new wiki, make it solely for Asterisk topics, as not to offend or replace voip-info. Build mirrors to multiple sites and multiple domain names. This would give this community a second resource with redundancy which is what I think ALL of us are looking for. I have taken the pleasure, of registering the domain name ASTERISKONLINE.ORG. I will donate a dedicated server with bandwidth to the cause. I am looking for additional people to help populate the wiki with useful information and to help maintain the site. I would suggest that ee have maybe 4 or 5 mirrors to start off and a core group of admins to help maintain the site. I am willing to work with anyone else that is about providing a solution to our current issue. If you guys want to REALLY work toward a solution, here's the chance. For the individuals that are interested in helping e-mail me. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:
RE: [asterisk-users] voip-info.org status update
I completely agree with the cheap hosting commments - my company competes against it all the time. Things go bad with the host in one way or another, sites move, and the cycle repeats. Is that how someone reputable wants to run a business moving their site around every couple months when things break ? Or do people want a company that is reliable and actually strives to deliver a decent SLA ? As for the hosting of voip-info, I don't see anything wrong with the model of providing something useful for free but sprinkling a few ads through it to help pay for the costs. Yes its annoying when something you rely on is not available, but what right has anyone got to complain that is not paying to have it available ? Maybe a better situation would be to partner with at least one more person or group that has hosting capacity, and split revenue in some manner to offset the costs, and have it hosted at at least 2 locations to guard against disaster, but with a wiki its not all that simple since its updating all the time, and straight mirrors won't work. Something to look into, but it would take even more volunteer hours to setup. A service my company offers (I am not trying to plug myself, but simply offering an alternative to the way it is now) is called livebackup. Hosting is all setup for a mirror of the complete setup which is copied over at some interval. Should problems arise with the primary site that can't be fixed quickly, dns is simply changed to point to the backup site and it operates as if it was the primary. This is meant to cover these sorts of situations where a disaster is not quickly recoverable, but running two sites in parallel has other issues which make it too complicated for anything without a super high budget. Quoting shadowym [EMAIL PROTECTED]: A percentage of all my profits go back to the community. What about you? -Original Message- From: Gordon Henderson [mailto:[EMAIL PROTECTED] Sent: Thursday, March 15, 2007 1:42 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] voip-info.org status update On Wed, 14 Mar 2007, shadowym wrote: Hard to expect the business community to take Asterisk seriously when this sort of stuff happens IMHO. I think you hit the nail on the head with one word: community. Asterisk is free, community supported, and the voip-info site has been provided for free - with the support of the community. The site would appear to be financially supported by a small number of quite unobtrusive google ads, and therein lies the problem... Hosting isn't free. If you can't/won't pay for hosting, then you have to support it by advertising. I can sell you web space/servers/co-lo facilities with full disk/server/location redundancy, backups and so on, but would you be willing to pay for it? Probably not. So you takes your chances with a popular hosting company, put in a small number of google ads to pay for a basic hosting package and go with it. After-all, there are millions of websites hosted on millions of servers throughout the world - it's a highly competitive business - there are offers of hosting for £1 a month or even less, but do you think it's a sustainable model? I don't. Well, maybe it is when you have 1000s of clients with 10s of 1000s of websites (spread over 100s of servers!) but with scale comes more issues. I can't understand how 3 of 4 hard drives could just suddenly fail simultaneously. There must be more too it. No UPS? Someone spilled their coffee into it? Something! That does strike me as odd, but I've seen it myself with a bad batch of disks. (IBM DeathStar, Hitachi, etc.) You usually get warnings, but if you're employing monkeys paying them peanuts, then they usually just treat them as fire forget once installed in the rack and plumbed into their automated selling/billing system. Either way, it's amateur hour! It's the way 99% of all co-lo facilities work. Buy big, sell cheap with little or no SLA - hope that the hardware/premises/internet is reliable enough, employ monkeys, pay peanuts. If you want quality, then be prepared to pay for it, and £1 a month does not give you quality IMO, and in my experience as someone who runs a small co-lo facility, people will not pay for quality hosting. A quality server costs me £650, more if the client insists on a Dull. Sure, I can put together something with pair of disks for under £300, but I know (from experience!) it won't last the 4+ years I want it to last, nor deliver the preformance my clients (who are willing to pay for such a service) demand. I'm not blaming James here because that's the way it is! I bet he's spent 100s of hours (unpaid) setting it up, running it and maintaining it, and resorted to google ads. purely to fund it. I don't envy him at all. If I can't be confident enough in an important source of information like this then I can't be confident enough to provide an Asterisk solution to businesses. That's the way I
RE: [asterisk-users] While the VoIP-Info.org site is down...
Quoting shadowym [EMAIL PROTECTED]: 75% failure at EXACTLY the same time? Come on! We all know better than that. Probably lost one drive at a time over weeks or months with no automated warnings! Amateur hour! a power supply or backplane problem could easily physically damage the entire array. a bus or controller problem could also destroy the data on all drives making it equally useless. Unless you were there, cut the guy some slack since you don't know the details. -Original Message- From: Patrick May [mailto:[EMAIL PROTECTED] Sent: Wednesday, March 14, 2007 8:10 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] While the VoIP-Info.org site is down... On Wed, Mar 14, 2007 at 10:05:20PM -0400, Matt wrote: Yikes.. you'd think a server would be running RAID. At any rate.. Please feel free to visit http://www.voip-wiki.us I have set this up to be able to hold information for the Asterisk community. I will also gladly allow others to mirror it. It is sitting in a climate controlled data center in Central PA on a server with RAID. Additionally, it is at the end of 95Megabytes/second on a BGP redundant connection. Please feel free to use it, if the community feels it can be useful... additionally, I would love to setup some rsync mirrors with others so that we can have redundant backups of this very valuable information. The previous message to the list was they lost 3 of 4 drives in the array. I'm not sure of any RAID that can sustain 75% hardware loss and still function. Patrick ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] voip-info.org status update
If a wiki site about automobiles crashes, should I buy a horse? shadowym wrote: I'm curious what you think that agenda might be? If it is to push the perception of Asterisk as a solid alternative to Traditional PBX's into the mainstream then I am guilty as charged! -Original Message- From: Andrew Kohlsmith [mailto:[EMAIL PROTECTED] Sent: Thursday, March 15, 2007 6:41 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] voip-info.org status update On Thursday 15 March 2007 12:32 am, shadowym wrote: Hard to expect the business community to take Asterisk seriously when this sort of stuff happens IMHO. I can't understand how 3 of 4 hard drives could just suddenly fail simultaneously. There must be more too it. No UPS? Someone spilled their coffee into it? Something! Obviously you didn't read Google's research paper on drive failures. And aside from that, you're also obviously pushing an agenda with these inciteful comments. - ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] voip-info.org status update
What a bunch of whiny people! If you travel to the website now you'll see the following note: begin quote-- Voip-info.org is down due to a hardware failure. Will be back soon. Due to the kind offers of mirror services from many people, once the site is back online, there will be a number of read-only mirrors of the site available as alternate access. Thanks for using voip-info.org! [EMAIL PROTECTED] end quote-- -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Re: Which SIP method/option to display a shorttext message
From: Olivier [EMAIL PROTECTED] Date: Thu, 15 Mar 2007 15:21:15 +0100 Hi, After further research, it seems SIP MESSAGE rfc3428) and SIP INFO (rfc2976) methods could be the more relevant for this feature. I'm still wondering whether SIP hardphones or Asterisk implement these methods in such a way you could make a welcome message, for example, appear on you contact phone screen. Cheers There was a thread indicating that you can do that with SendText() with capable hard phones. Yuan Liu ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] DNIS/DNID
From: Mark Quitoriano [EMAIL PROTECTED] Date: Thu, 15 Mar 2007 11:59:30 +0800 Hi i have an asterisk pbx with E1 port connected to another PBX. Im trying to send the DNID/DNIS to the PBX here's my dialplan exten = 888111,1,Dial(ZAP/g2) I thought you'd get an error message about the syntax above? If the PBX is configured to take DNIS as DTMF string, D() flag could be used. Yuan Liu exten = 888111,n,Hangup() The PBX just get the number 2 as it's DNIS when i change it to ZAP/1 or ZAP/g1 the PBX get the number 1. What should i add to send the extension number as DNID/DNIS? Thanks! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voip-Wiki Site Information
#1 - It's down #2 - The owner is prohibiting anyone from mirroring it. On 3/15/07, Trevor Peirce [EMAIL PROTECTED] wrote: Matt wrote: Community, I have put up www.voip-wiki.us http://www.voip-wiki.us My apologies to our fellow Asteristians outside the us... this was the only easy domain available. What's wrong with voip-info.org ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Replacement Wiki - options (Formerly 'status of voip-info')
I would suggest that we create a new wiki, make it solely for Asterisk topics, as not to offend or replace voip-info. Build mirrors to multiple sites and multiple domain names. This would give this community a second resource with redundancy which is what I think ALL of us are looking for. I have taken the pleasure, of registering the domain name ASTERISKONLINE.ORG. I would like to know what the community feels about an Asterisk-only wiki. I can see pros and cons of Asterisk-only vs. Asterisk/FreeSwitch/Yate/OpenPBX/etc. My gut says keep it open for everything OSS/VoIP. (I have no logical reason for feeling that way - it's just a gut feeling.) I will donate a dedicated server with bandwidth to the cause. I am looking for additional people to help populate the wiki with useful information and to help maintain the site. I would suggest that ee have maybe 4 or 5 mirrors to start off and a core group of admins to help maintain the site. Thanks for putting your money where your mouth is! This is the kind of action the community needs. I am willing to work with anyone else that is about providing a solution to our current issue. If you guys want to REALLY work toward a solution, here's the chance. For the individuals that are interested in helping e-mail me. I hope you get some respondents. In the meantime it might be good to check out the fledgling wiki here: http://www.voip-wiki.us It uses MediaWiki which has a nice, clean interface and seems pretty easy to use. -MC ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voip-Wiki Site Information
On 3/15/07, Matt [EMAIL PROTECTED] wrote: #1 - It's down #2 - The owner is prohibiting anyone from mirroring it. Have you checked the message on voip-info.org recently? http://voip-info.org/ Voip-info.org is down due to a hardware failure. Will be back soon. Due to the kind offers of mirror services from many people, once the site is back online, there will be a number of read-only mirrors of the site available as alternate access. Thanks for using voip-info.org! [EMAIL PROTECTED] -Erik ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Freepbx Incoming call's configuration
Hi every body, I've set up a Trixbox Server with TE110P,all things seem to work fine(Thank You Malling lists irc Forums), but i need your help, i ve 30 numbre from 60 to 89, i need to specify for each sip extension a Zap number for example to call the sales service the caller must call 555-4570 and automaticly the caller will be redirected to the 202 ( sales service ) so nobody else can use this number ..70 im using freepbx, so can someone please help me :) Kind Regards ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voip-Wiki Site Information
Excelent! Then once it comes up voip-wiki.us will be glad to provide a read-only mirror. On 3/15/07, Erik Anderson [EMAIL PROTECTED] wrote: On 3/15/07, Matt [EMAIL PROTECTED] wrote: #1 - It's down #2 - The owner is prohibiting anyone from mirroring it. Have you checked the message on voip-info.org recently? http://voip-info.org/ Voip-info.org is down due to a hardware failure. Will be back soon. Due to the kind offers of mirror services from many people, once the site is back online, there will be a number of read-only mirrors of the site available as alternate access. Thanks for using voip-info.org! [EMAIL PROTECTED] -Erik ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DNIS/DNID
Mark Quitoriano wrote:
Hi i have an asterisk pbx with E1 port connected to another PBX. Im
trying to send the DNID/DNIS to the PBX here's my dialplan
exten = 888111,1,Dial(ZAP/g2)
exten = 888111,n,Hangup()
The PBX just get the number 2 as it's DNIS when i change it to ZAP/1
or ZAP/g1 the PBX get the number 1. What should i add to send the
extension number as DNID/DNIS?
exten = 888111,1,Dial(ZAP/g2/${EXTEN})
Right now you're trying to dial the number g2, instead of using group 2.
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Can I use an Intertel IPPhone Plus 7704500 with Asterisk somehow
I may be able to get my hands on a few of these units as we are phasing them out at a company. I could not find much in the way of connecting these to non Intertel systems. Anyone have an idea or success with this? Thanks! Bill ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Freepbx Incoming call's configuration
Hi Younss, You just need to setup Inbound Routes in FreePBX. The inbound routes allow you to route calls based upon caller ID or DID. Since you want to route based upon the number your caller dialed, you want to route based on DID. For your example: 1. Create a new inbound route. 2. In the DID field, enter the number you wish to route (555-4570). Keep in mind that this must match what your provider sends. Some providers send +1554570, some sent just 4570, and some send something in between. Check with your provider for their format. 3. At the bottom, select where you'd like that number to be routed to (I believe you need to select Core: Extension 202). Save the route, apply the settings (via clicking on the red bar), and that's it! Alex On 3/15/07, younss azzayani [EMAIL PROTECTED] wrote: Hi every body, I've set up a Trixbox Server with TE110P,all things seem to work fine(Thank You Malling lists irc Forums), but i need your help, i ve 30 numbre from 60 to 89, i need to specify for each sip extension a Zap number for example to call the sales service the caller must call 555-4570 and automaticly the caller will be redirected to the 202 ( sales service ) so nobody else can use this number ..70 im using freepbx, so can someone please help me :) Kind Regards ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Call center manager for Asterisk (Release 0.3)
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of nik600 Sent: Wednesday, March 14, 2007 12:36 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Call center manager for Asterisk (Release 0.3) Hi i just want to let you know that is available a new release of ccmanager. I've added the possibility to import queue_log information in a mysql database and to generate reports using this information. The software is in a beta state and provides this functionality: - users management - call generation (making a GET or POST request on a certain URL) - queue management (LOGIN / LOGOUT / QUEUE STATUS) - pickup a call from a queue even if the user isn't logged in the queue - outbound call in customizable context - queue stats import from queue_log - queue reports creation (using an open xml format) Please note, i think that the xml definition of a report is very important, if many people share each other their reports there is the possibility to build a reports-repository, so the final user can use many reports and, if the user know what he is doing, he can customize the reports. I am looking for people to improve this project, any help would be appreciated. - developers (php / mysql / postgres / ajax ) - tester - graphics (div css) Here there are some screenshots https://sourceforge.net/dbimage.php?id=115442 https://sourceforge.net/dbimage.php?id=115440 https://sourceforge.net/dbimage.php?id=114381 And here there is the sourceforge project. https://sourceforge.net/projects/ccmanager Thanks, nik Nik, This looks REALLY COOL! Just an FYI in case you didn't know, there is also a callcenter asterisk mailing list that you could post this to. I am not sure how many users are subscribed but it is most certainly more of your target audience. At any rate, I can't wait to get this setup and take it for a test drive. I will provide and feedback and help that I can. Is your plan to make this a commercial product sometime down the road? Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] voip-info.org status update
Of course you should buy a horse. But then there are the questions like. Do I get one like the Budweiser ones? Or just a mule (they can be helpful). What about color? Maybe a spotted one? Will my horse be able to talk to other horses using SIP? Or will it only be able to use IAX? Man, so many decisions if we have to go that way. Paul wrote: If a wiki site about automobiles crashes, should I buy a horse? shadowym wrote: I'm curious what you think that agenda might be? If it is to push the perception of Asterisk as a solid alternative to Traditional PBX's into the mainstream then I am guilty as charged! -Original Message- From: Andrew Kohlsmith [mailto:[EMAIL PROTECTED] Sent: Thursday, March 15, 2007 6:41 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] voip-info.org status update On Thursday 15 March 2007 12:32 am, shadowym wrote: Hard to expect the business community to take Asterisk seriously when this sort of stuff happens IMHO. I can't understand how 3 of 4 hard drives could just suddenly fail simultaneously. There must be more too it. No UPS? Someone spilled their coffee into it? Something! Obviously you didn't read Google's research paper on drive failures. And aside from that, you're also obviously pushing an agenda with these inciteful comments. - ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Which SIP method/option to display a shorttext message
Hi, Thanks for the pointer. I will check previous threads (as I've not found yet any sendText compliant hardphone). Cheers ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] voip-info.org status update
You'll have to check the horse-wiki and pray it never goes down. Alternatively, you could get a Cisco horse. While it may cost more, at least you'll have a number you can call for tech support should your horse throw a shoe. The downside being, of course, if you want to modify your horse (e.g. - adding a rear spoiler, tinting its blinders, or adding a saddle with a piece of spinny plastic that makes it look like you're actually walking *backwards*) you'll have to use proprietary parts only purchasable from stables.cisco.com. :( Jay Rob Schall wrote: Of course you should buy a horse. But then there are the questions like. Do I get one like the Budweiser ones? Or just a mule (they can be helpful). What about color? Maybe a spotted one? Will my horse be able to talk to other horses using SIP? Or will it only be able to use IAX? Man, so many decisions if we have to go that way. Paul wrote: If a wiki site about automobiles crashes, should I buy a horse? shadowym wrote: I'm curious what you think that agenda might be? If it is to push the perception of Asterisk as a solid alternative to Traditional PBX's into the mainstream then I am guilty as charged! -Original Message- From: Andrew Kohlsmith [mailto:[EMAIL PROTECTED] Sent: Thursday, March 15, 2007 6:41 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] voip-info.org status update On Thursday 15 March 2007 12:32 am, shadowym wrote: Hard to expect the business community to take Asterisk seriously when this sort of stuff happens IMHO. I can't understand how 3 of 4 hard drives could just suddenly fail simultaneously. There must be more too it. No UPS? Someone spilled their coffee into it? Something! Obviously you didn't read Google's research paper on drive failures. And aside from that, you're also obviously pushing an agenda with these inciteful comments. - ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dropped calls in Asterisk - A general question
Hey all, I have a question for those administrating/building out systems with over 30 users on them. How often do you experience the dropped call phenomena. Would you care to share your experiences including what versions of * you were using, what kind of connectivity was present (T1, Fractional T, Intergrated T, DSL, Cable). Echo? Solutions? (e.g. we bought an X_Brand Echo Canceller). Also, which phones most found favorable with Asterisk on a full functional level. Not Polycoms because they're so neat! Or: Cisco rocks!. Something more to the tune of X_Brand phones worked well with Asterisk 1.2.xx for 70 users on a Data T. We had an X_Brand switch which did/didn't do PoE running Asterisk on a SuperX_Brand server with X amount of memory. Any response is appreciated as long as its something productive. No My SuperX_Brand system has a new logo and a shiny silver box that the vendor states `surpasses unforseen functionality due to hyperbolic hooplah blah blah`. Short, sweet effective. Thanks. -- J. Oquendo http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x1383A743 sil . infiltrated @ net http://www.infiltrated.net The happiness of society is the end of government. John Adams smime.p7s Description: S/MIME Cryptographic Signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What happend to voip-info?
On Wed, 14 Mar 2007, Stephen Bosch wrote: Gordon Henderson wrote: On Wed, 14 Mar 2007, Jonathan k. Creasy wrote: I would be willing to mirror it also?. At the risk of sounding like an AOLer, Me Too ... (UK based mirror?) The site is pingable, so I'd suggest it's either crashed in some awkward way and just needs resetting, but you never know... If it never comes up, someone is going to have to write a real manual for Asterisk. I actually learned most of what I needed initially out of 2 O'Reilly books: Asterisk: The future of telephony, and Switching To VoIP... I didn't find out about the WiKi until much later. It's been good though and I'd even started to put a little bit up myself. Lets hope for the best! Gordon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] shutdown
somebody can help me with this message I don´t understand *CLI stop now Beginning asterisk shutdown Executing last minute cleanups == Destroying musiconhold processes Yuck! Error in buffer handling...: Connection reset by peer Yuck! Error in buffer handling...: Connection reset by peer Asterisk cleanly ending (0). thanks _ José -- MENSAJE ENVIADO CON WMAIL 1.01 UNIVERSIDAD DEL CAUCA ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call center manager for Asterisk (Release 0.3)
Nik, This looks REALLY COOL! thanks Just an FYI in case you didn't know, there is also a callcenter asterisk mailing list that you could post this to. I am not sure how many users are subscribed but it is most certainly more of your target audience. thanks, i'll subscribe on it. At any rate, I can't wait to get this setup and take it for a test drive. I will provide and feedback and help that I can. ok, thanks if you want i've started a mailing list for this project https://lists.sourceforge.net/lists/listinfo/ccmanager-users Is your plan to make this a commercial product sometime down the road? I've not taken any decision but i think that the project will be released under GPL, and maybe i can provide commercial support on installation / customization / maintainace Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- /*/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/nikstresser ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Dropped calls in Asterisk - A general question
I've got 415 phones, mostly Cisco 7960's. The only time I see dropped calls is when either end hangs up, or I restart asterisk. Using all T1 PRI. HW mainly: Dell 1750 w/2GB, Digium TE410 or TE412P's. Raid1 w/PERC. I use Dell 1950's for the VM servers, but anything with a Digium card is a Dell 1750. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of J. Oquendo Sent: Thursday, March 15, 2007 2:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Dropped calls in Asterisk - A general question Hey all, I have a question for those administrating/building out systems with over 30 users on them. How often do you experience the dropped call phenomena. Would you care to share your experiences including what versions of * you were using, what kind of connectivity was present (T1, Fractional T, Intergrated T, DSL, Cable). Echo? Solutions? (e.g. we bought an X_Brand Echo Canceller). Also, which phones most found favorable with Asterisk on a full functional level. Not Polycoms because they're so neat! Or: Cisco rocks!. Something more to the tune of X_Brand phones worked well with Asterisk 1.2.xx for 70 users on a Data T. We had an X_Brand switch which did/didn't do PoE running Asterisk on a SuperX_Brand server with X amount of memory. Any response is appreciated as long as its something productive. No My SuperX_Brand system has a new logo and a shiny silver box that the vendor states `surpasses unforseen functionality due to hyperbolic hooplah blah blah`. Short, sweet effective. Thanks. -- J. Oquendo http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x1383A743 sil . infiltrated @ net http://www.infiltrated.net The happiness of society is the end of government. John Adams ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call center manager for Asterisk (Release 0.3)
On 3/14/07, Steve Totaro [EMAIL PROTECTED] wrote: Just an FYI in case you didn't know, there is also a callcenter asterisk mailing list that you could post this to. I am not sure how many users are subscribed but it is most certainly more of your target audience. Where do you subscribe to the mailing list? Thanks! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call center manager for Asterisk (Release 0.3)
i haven't found any call center asterisk mailing list, but i've found this: http://lists.digium.com/mailman/listinfo/asterisk-biz On 3/15/07, Lacy Moore - Aspendora [EMAIL PROTECTED] wrote: On 3/14/07, Steve Totaro [EMAIL PROTECTED] wrote: Just an FYI in case you didn't know, there is also a callcenter asterisk mailing list that you could post this to. I am not sure how many users are subscribed but it is most certainly more of your target audience. Where do you subscribe to the mailing list? Thanks! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- /*/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/nikstresser ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] A200 card problem
Hi - I just got an A200 card with 1 FXO and 1 FXS module. Sadly, I can't make it work- currently, asterisk will not startup because of a bad module. Below are some log files/config files. If anyone has any suggestions, I'd appreciate it. I used Trixbox 2.0 and followed instructions on (http:// sangoma.editme.com/wanpipe-linux-asterisk-atHome) - no problems running through or compiling. I also downloaded trixbox 2.0 with sangoma drivers included directly from sangoma (http:// wiki.sangoma.com/Trixbox-1xx). I know how this list feels about trixbox, but still, the card/configs are the same, no? Any advice is appreciated. thanks Todd +++ /var/log/asterisk/full Mar 15 16:12:37 VERBOSE[31964] logger.c: [chan_mgcp.so]Mar 15 16:12:37 VERBOSE[31964] logger.c: [chan_mgcp.so] = (Media Gateway Control Protocol (MGCP)) Mar 15 16:12:37 VERBOSE[31964] logger.c: [chan_local.so]Mar 15 16:12:37 VERBOSE[31964] logger.c: [chan_local.so] = (Local Proxy Channel) Mar 15 16:12:37 VERBOSE[31964] logger.c: [chan_phone.so]Mar 15 16:12:37 VERBOSE[31964] logger.c: [chan_phone.so] = (Linux Telephony API Support) Mar 15 16:12:37 VERBOSE[31964] logger.c: [chan_zap.so]Mar 15 16:12:37 VERBOSE[31964] logger.c: [chan_zap.so] = (Zapata Telephony w/PRI) Mar 15 16:12:37 WARNING[31964] chan_zap.c: Unable to specify channel 1: No such device or address Mar 15 16:12:37 ERROR[31964] chan_zap.c: Unable to open channel 1: No such device or address here = 0, tmp-channel = 1, channel = 1 Mar 15 16:12:37 ERROR[31964] chan_zap.c: Unable to register channel '1' Mar 15 16:12:37 WARNING[31964] loader.c: chan_zap.so: load_module failed, returning -1 Mar 15 16:12:37 WARNING[31964] loader.c: Loading module chan_zap.so failed! [EMAIL PROTECTED] ~]# +++ END /var/log/asterisk/full +++ /var/log/messages [EMAIL PROTECTED] ~]# tail -20 /var/log/messages Mar 15 16:24:08 asterisk1 safe_asterisk: [pbx_config.so] = (Text Extension Configuration) Mar 15 16:24:08 asterisk1 safe_asterisk: [pbx_functions.so] = (Builtin dialplan functions) Mar 15 16:24:08 asterisk1 safe_asterisk: [pbx_ael.so] = (Asterisk Extension Language Compiler) Mar 15 16:24:08 asterisk1 safe_asterisk: [pbx_dundi.so] = (Distributed Universal Number Discovery (DUNDi)) Mar 15 16:24:08 asterisk1 safe_asterisk: [pbx_loopback.so] = (Loopback Switch) Mar 15 16:24:08 asterisk1 safe_asterisk: [pbx_spool.so] = (Outgoing Spool Support) Mar 15 16:24:08 asterisk1 safe_asterisk: [chan_iax2.so] = (Inter Asterisk eXchange (Ver 2)) Mar 15 16:24:08 asterisk1 safe_asterisk: [chan_agent.so] = (Agent Proxy Channel) Mar 15 16:24:08 asterisk1 safe_asterisk: [chan_skinny.so] = (Skinny Client Control Protocol (Skinny)) Mar 15 16:24:08 asterisk1 safe_asterisk: [chan_h323.so] = (Objective Systems H323 Channel) Mar 15 16:24:08 asterisk1 safe_asterisk: [chan_sip.so] = (Session Initiation Protocol (SIP)) Mar 15 16:24:08 asterisk1 safe_asterisk: [chan_features.so] = (Feature Proxy Channel) Mar 15 16:24:08 asterisk1 safe_asterisk: [skipping chan_oss.so] Mar 15 16:24:08 asterisk1 safe_asterisk: [chan_mgcp.so] = (Media Gateway Control Protocol (MGCP)) Mar 15 16:24:08 asterisk1 safe_asterisk: [chan_local.so] = (Local Proxy Channel) Mar 15 16:24:08 asterisk1 safe_asterisk: [chan_phone.so] = (Linux Telephony API Support) Mar 15 16:24:08 asterisk1 safe_asterisk: [chan_zap.so] = (Zapata Telephony w/PRI) Mar 15 16:24:08 asterisk1 safe_asterisk: Asterisk died with code 1. Mar 15 16:24:08 asterisk1 safe_asterisk: Automatically restarting Asterisk. Mar 15 16:24:08 asterisk1 asterisk: safe_asterisk startup succeeded [EMAIL PROTECTED] ~]# +++ END /var/log/messages ++/etc/zaptel.conf+++ [EMAIL PROTECTED] ~]# more /etc/zaptel.conf # Autogenerated by /usr/local/sbin/sangoma/setup-sangoma -- do not hand edit # Zaptel Channels Configurations (zaptel.conf) # loadzone=us defaultzone=us # Sangoma A200 [slot:9 bus:1 span: 1] fxsks=1 fxsks=2 fxoks=3 fxoks=4 [EMAIL PROTECTED] ~]# ++END /etc/zaptel.conf+++ ++/etc/asterisk/zapata.conf+++ [EMAIL PROTECTED] asterisk]# more zapata.conf ; ; Zapata telephony interface ; ; Configuration file [trunkgroups] [channels] language=en context=from-zaptel signalling=fxs_ks rxwink=300 ; Atlas seems to use long (250ms) winks ; ; Whether or not to do distinctive ring detection on FXO lines ; ;usedistinctiveringdetection=yes usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no echotraining=800 rxgain=0.0 txgain=0.0 group=0 callgroup=1 pickupgroup=1 immediate=no ;faxdetect=both faxdetect=incoming ;faxdetect=outgoing ;faxdetect=no
[asterisk-users] Re: Replacement Wiki - options (Formerly 'status of voip-info')
Michael Collins wrote: I would suggest that we create a new wiki, make it solely for Asterisk topics, as not to offend or replace voip-info. Build mirrors to multiple sites and multiple domain names. This would give this community a second resource with redundancy which is what I think ALL of us are looking for. I have taken the pleasure, of registering the domain name ASTERISKONLINE.ORG. I would like to know what the community feels about an Asterisk-only wiki. I can see pros and cons of Asterisk-only vs. Asterisk/FreeSwitch/Yate/OpenPBX/etc. My gut says keep it open for everything OSS/VoIP. (I have no logical reason for feeling that way - it's just a gut feeling.) I will donate a dedicated server with bandwidth to the cause. I am looking for additional people to help populate the wiki with useful information and to help maintain the site. I would suggest that ee have maybe 4 or 5 mirrors to start off and a core group of admins to help maintain the site. Thanks for putting your money where your mouth is! This is the kind of action the community needs. I am willing to work with anyone else that is about providing a solution to our current issue. If you guys want to REALLY work toward a solution, here's the chance. For the individuals that are interested in helping e-mail me. I hope you get some respondents. In the meantime it might be good to check out the fledgling wiki here: http://www.voip-wiki.us It uses MediaWiki which has a nice, clean interface and seems pretty easy to use. -MC I'm okay with OSS/VoIP. Just need confirmation that we all want to do this. I don't want to allocate a server to the cause and it just sit idle. I'm willing to work with the guys with www.voip-wiki.us. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Incoming Caller ID
Does anyone know if I can get Incoming caller id name and number on a sagnoma PRI. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming Caller ID
Rob Vinson wrote: Does anyone know if I can get Incoming caller id name and number on a sagnoma PRI. Yes. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming Caller ID
That should be provided by your telco, if your referring to a PRI on a Sangoma T-1 card. On 3/15/07, Rob Vinson [EMAIL PROTECTED] wrote: Does anyone know if I can get Incoming caller id name and number on a sagnoma PRI. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sip_nat.conf - Asterisk with two Ethernet Interfaces
Will this do the intended thing? This is in sip_nat.conf which is included in sip.conf: externip=192.168.0.200 localnet=192.168.0.200/255.255.255.0 externip=64.168.237.110 localnet=192.168.1.2/255.255.255.0 I have Asterisk running on a box with two Ethernet interfaces and bound to both. One interface, 192.168.1.2 services clients outside the firewall who are led to believe that Asterisk is running at 64.168.237.110. The other interface, 192.168.0.200 services clients inside the firewall who are led to believe that Asterisk is running at 192.168.0.200. I am worried that Asterisk might not do what I am intending to do. What I want is for Asterisk to send SIP/SDP Connection Info as 64.168.237.110 to those clients that contact it on 192.168.1.2, and to send SIP/SDP Connection Info as 64.168.237.110 to those that contact it on 192.168.0.200. I am worried that it might use the last externip=... declaration for both interfaces and ignore the other one. Could somebody help me with determining the treatment of the multiple externip=... declarations? Or point me to how I could verify? I don't want to wait until the evening to find out from outside the firewall that this doesn't work. (It is working fine from within the firewall where I contact asterisk at 192.168.0.200) Thank you! Best regards, Anjul. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] A200 card problem
Sangoma gives excellent support Suggest you try there first They probably will want SSH access to the box. Send them an e-mail John Novack Todd H wrote: Hi - I just got an A200 card with 1 FXO and 1 FXS module. Sadly, I can't make it work- currently, asterisk will not startup because of a bad module. Below are some log files/config files. If anyone has any suggestions, I'd appreciate it. I used Trixbox 2.0 and followed instructions on (http://sangoma.editme.com/wanpipe-linux-asterisk-atHome) - no problems running through or compiling. I also downloaded trixbox 2.0 with sangoma drivers included directly from sangoma (http://wiki.sangoma.com/Trixbox-1xx). I know how this list feels about trixbox, but still, the card/configs are the same, no? Any advice is appreciated. thanks Todd +++ /var/log/asterisk/full Mar 15 16:12:37 VERBOSE[31964] logger.c: [chan_mgcp.so]Mar 15 16:12:37 VERBOSE[31964] logger.c: [chan_mgcp.so] = (Media Gateway Control Protocol (MGCP)) Mar 15 16:12:37 VERBOSE[31964] logger.c: [chan_local.so]Mar 15 16:12:37 VERBOSE[31964] logger.c: [chan_local.so] = (Local Proxy Channel) Mar 15 16:12:37 VERBOSE[31964] logger.c: [chan_phone.so]Mar 15 16:12:37 VERBOSE[31964] logger.c: [chan_phone.so] = (Linux Telephony API Support) Mar 15 16:12:37 VERBOSE[31964] logger.c: [chan_zap.so]Mar 15 16:12:37 VERBOSE[31964] logger.c: [chan_zap.so] = (Zapata Telephony w/PRI) Mar 15 16:12:37 WARNING[31964] chan_zap.c: Unable to specify channel 1: No such device or address Mar 15 16:12:37 ERROR[31964] chan_zap.c: Unable to open channel 1: No such device or address here = 0, tmp-channel = 1, channel = 1 Mar 15 16:12:37 ERROR[31964] chan_zap.c: Unable to register channel '1' Mar 15 16:12:37 WARNING[31964] loader.c: chan_zap.so: load_module failed, returning -1 Mar 15 16:12:37 WARNING[31964] loader.c: Loading module chan_zap.so failed! [EMAIL PROTECTED] ~]# +++ END /var/log/asterisk/full +++ /var/log/messages [EMAIL PROTECTED] ~]# tail -20 /var/log/messages Mar 15 16:24:08 asterisk1 safe_asterisk: [pbx_config.so] = (Text Extension Configuration) Mar 15 16:24:08 asterisk1 safe_asterisk: [pbx_functions.so] = (Builtin dialplan functions) Mar 15 16:24:08 asterisk1 safe_asterisk: [pbx_ael.so] = (Asterisk Extension Language Compiler) Mar 15 16:24:08 asterisk1 safe_asterisk: [pbx_dundi.so] = (Distributed Universal Number Discovery (DUNDi)) Mar 15 16:24:08 asterisk1 safe_asterisk: [pbx_loopback.so] = (Loopback Switch) Mar 15 16:24:08 asterisk1 safe_asterisk: [pbx_spool.so] = (Outgoing Spool Support) Mar 15 16:24:08 asterisk1 safe_asterisk: [chan_iax2.so] = (Inter Asterisk eXchange (Ver 2)) Mar 15 16:24:08 asterisk1 safe_asterisk: [chan_agent.so] = (Agent Proxy Channel) Mar 15 16:24:08 asterisk1 safe_asterisk: [chan_skinny.so] = (Skinny Client Control Protocol (Skinny)) Mar 15 16:24:08 asterisk1 safe_asterisk: [chan_h323.so] = (Objective Systems H323 Channel) Mar 15 16:24:08 asterisk1 safe_asterisk: [chan_sip.so] = (Session Initiation Protocol (SIP)) Mar 15 16:24:08 asterisk1 safe_asterisk: [chan_features.so] = (Feature Proxy Channel) Mar 15 16:24:08 asterisk1 safe_asterisk: [skipping chan_oss.so] Mar 15 16:24:08 asterisk1 safe_asterisk: [chan_mgcp.so] = (Media Gateway Control Protocol (MGCP)) Mar 15 16:24:08 asterisk1 safe_asterisk: [chan_local.so] = (Local Proxy Channel) Mar 15 16:24:08 asterisk1 safe_asterisk: [chan_phone.so] = (Linux Telephony API Support) Mar 15 16:24:08 asterisk1 safe_asterisk: [chan_zap.so] = (Zapata Telephony w/PRI) Mar 15 16:24:08 asterisk1 safe_asterisk: Asterisk died with code 1. Mar 15 16:24:08 asterisk1 safe_asterisk: Automatically restarting Asterisk. Mar 15 16:24:08 asterisk1 asterisk: safe_asterisk startup succeeded [EMAIL PROTECTED] ~]# +++ END /var/log/messages ++/etc/zaptel.conf+++ [EMAIL PROTECTED] ~]# more /etc/zaptel.conf # Autogenerated by /usr/local/sbin/sangoma/setup-sangoma -- do not hand edit # Zaptel Channels Configurations (zaptel.conf) # loadzone=us defaultzone=us # Sangoma A200 [slot:9 bus:1 span: 1] fxsks=1 fxsks=2 fxoks=3 fxoks=4 [EMAIL PROTECTED] ~]# ++END /etc/zaptel.conf+++ ++/etc/asterisk/zapata.conf+++ [EMAIL PROTECTED] asterisk]# more zapata.conf ; ; Zapata telephony interface ; ; Configuration file [trunkgroups] [channels] language=en context=from-zaptel signalling=fxs_ks rxwink=300 ; Atlas seems to use long (250ms) winks ; ; Whether or not to do distinctive ring detection on FXO lines ; ;usedistinctiveringdetection=yes usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no echotraining=800 rxgain=0.0 txgain=0.0
Re: [asterisk-users] Single sign on PC + phone?
Patrick wrote: Thanks for the info Trevor. Was your proof of concept also with Windows PCs or *nix PCs? I haven't played with realtime yet so I might be in for a bit of a learning curve. This was just on Linux user stations with a simple bash script that send a request to a web server. The web server did the rest based on the PC's IP address and user's username. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming Caller ID
Rob Vinson wrote: Does anyone know if I can get Incoming caller id name and number on a sagnoma PRI The bigger question is if your telco is sending it to you. asterisk generally takes care of everything automatically, provided it's available and you've configured your PRI properly. Number comes instantaneously with your call, while name shows up a brief moment later. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie Question
On 3/8/07, Chris Nighswonger [EMAIL PROTECTED] wrote: Thanks for the responses. iptables on the * box has no rules and all tables default to 'accept.' I have not got to the point of placing calls out across the internet yet. The issue here is no audio back from the * box when running through the demo routine. I'll try to set it up to make a call outside tomorrow. Ok. I have not been able to setup the box to call outside, however, watching the packet traffic I see plenty of data flowing from the xlite client to the * server, but never any packets from the server to the client. (That is, during the course of the call.) The server and client talk just fine when establishing the connection, just no audio data from the server to the client. Any thoughts? From everything I've read, the initial setup should be much easier than mine has gone so far... :( Chris ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie Question
On 3/15/07, Chris Nighswonger [EMAIL PROTECTED] wrote: Ok. I have not been able to setup the box to call outside, however, watching the packet traffic I see plenty of data flowing from the xlite client to the * server, but never any packets from the server to the client. (That is, during the course of the call.) The server and client talk just fine when establishing the connection, just no audio data from the server to the client. Any thoughts? Setup the demo IVR on your Atrisk box and call that from your xlite softphone. The entire call will be on your local network so you'll be able to see if the problem is local or not. -HJC ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Replacement Wiki - options (Formerly 'status of voip-info')
I would take an alternative stance and say that an Asterisk only solution is needed. This is a wildly growing product with nearly limitless possibilities. Trying to cram too much on a site just causes confusion. KISS (no I am not calling anybody in particular stupid.) On 3/15/07, Davis Sylvester III [EMAIL PROTECTED] wrote: Michael Collins wrote: I would suggest that we create a new wiki, make it solely for Asterisk topics, as not to offend or replace voip-info. Build mirrors to multiple sites and multiple domain names. This would give this community a second resource with redundancy which is what I think ALL of us are looking for. I have taken the pleasure, of registering the domain name ASTERISKONLINE.ORG. I would like to know what the community feels about an Asterisk-only wiki. I can see pros and cons of Asterisk-only vs. Asterisk/FreeSwitch/Yate/OpenPBX/etc. My gut says keep it open for everything OSS/VoIP. (I have no logical reason for feeling that way - it's just a gut feeling.) I will donate a dedicated server with bandwidth to the cause. I am looking for additional people to help populate the wiki with useful information and to help maintain the site. I would suggest that ee have maybe 4 or 5 mirrors to start off and a core group of admins to help maintain the site. Thanks for putting your money where your mouth is! This is the kind of action the community needs. I am willing to work with anyone else that is about providing a solution to our current issue. If you guys want to REALLY work toward a solution, here's the chance. For the individuals that are interested in helping e-mail me. I hope you get some respondents. In the meantime it might be good to check out the fledgling wiki here: http://www.voip-wiki.us It uses MediaWiki which has a nice, clean interface and seems pretty easy to use. -MC I'm okay with OSS/VoIP. Just need confirmation that we all want to do this. I don't want to allocate a server to the cause and it just sit idle. I'm willing to work with the guys with www.voip-wiki.us. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: zapata with Tiger3XX compilation error
Ok so I read the Linux 2.6 related README and finally compiled propperly, I thought but at the end I notice that lscpi does report the cards, but I cant modprobe wcfxo nor zaptel and I do have wcfxo.ko in the /lib/modules/2.6.8/extra/ directory, so what gives? This is a Debian Sarge, thanks! # # make clean starts here # make[1]: Entering directory `/usr/src/zaptel-1.4.0/menuselect' rm -f menuselect *.o make[2]: Entering directory `/usr/src/zaptel-1.4.0/menuselect/mxml' /bin/rm -f mxmldoc.o testmxml.o mxml-attr.o mxml-entity.o mxml-file.o mxml-index.o mxml-node.o mxml-search.o mxml-set.o mxml-private.o mxml-string.o libmxml.a mxmldoc doc/mxml.3 doc/mxmldoc.1 testmxml mxml.xml /bin/rm -f mxmldoc-static libmxml.a /bin/rm -f *.bck *.bak /bin/rm -f config.cache config.log config.status /bin/rm -f -r autom4te*.cache make[2]: Leaving directory `/usr/src/zaptel-1.4.0/menuselect/mxml' make[1]: Leaving directory `/usr/src/zaptel-1.4.0/menuselect' rm -f torisatool makefw tor2fw.h radfw.h rm -f fxotune fxstest sethdlc-new ztcfg ztdiag ztmonitor ztspeed zttest zttool rm -f *.o ztcfg tzdriver sethdlc sethdlc-new rm -f libtonezone.so libtonezone.a *.lo make -C /usr/src/linux SUBDIRS=/usr/src/zaptel-1.4.0 clean make[1]: Entering directory `/usr/src/kernel-source-2.6.8' CLEAN /usr/src/zaptel-1.4.0/wct4xxp CLEAN /usr/src/zaptel-1.4.0/.tmp_versions make[1]: Leaving directory `/usr/src/kernel-source-2.6.8' rm -f xpp/*.ko xpp/*.mod.c xpp/.*o.cmd rm -f xpp/*.o xpp/*.mod.o rm -rf .tmp_versions rm -f gendigits tones.h rm -f libtonezone* rm -f tor2ee rm -f fxotune rm -f core rm -f ztcfg-shared fxstest rm -rf misdn* rm -rf mISDNuser* # # ./configure starts here # checking for gcc... gcc checking for C compiler default output file name... a.out checking whether the C compiler works... yes checking whether we are cross compiling... no checking for suffix of executables... checking for suffix of object files... o checking whether we are using the GNU C compiler... yes checking whether gcc accepts -g... yes checking for gcc option to accept ISO C89... none needed checking how to run the C preprocessor... gcc -E checking for a BSD-compatible install... /usr/bin/install -c checking whether ln -s works... yes checking for GNU make... make checking for grep... /bin/grep checking for sh... /bin/sh checking for ln... /bin/ln checking for grep that handles long lines and -e... (cached) /bin/grep checking for egrep... /bin/grep -E checking for ANSI C header files... yes checking for sys/types.h... yes checking for sys/stat.h... yes checking for stdlib.h... yes checking for string.h... yes checking for memory.h... yes checking for strings.h... yes checking for inttypes.h... yes checking for stdint.h... yes checking for unistd.h... yes checking for initscr in -lcurses... yes checking curses.h usability... yes checking curses.h presence... yes checking for curses.h... yes checking for initscr in -lncurses... yes checking for curses.h... (cached) yes checking for newtBell in -lnewt... yes checking newt.h usability... yes checking newt.h presence... yes checking for newt.h... yes checking for usb_init in -lusb... no configure: creating ./config.status config.status: creating build_tools/menuselect-deps config.status: creating makeopts configure: *** Zaptel build successfully configured *** # # make linux26 starts here # make[1]: Entering directory `/usr/src/zaptel-1.4.0/menuselect' make[2]: Entering directory `/usr/src/zaptel-1.4.0/menuselect' make[3]: Entering directory `/usr/src/zaptel-1.4.0/menuselect/mxml' gcc -O -Wall -c mxml-attr.c gcc -O -Wall -c mxml-entity.c gcc -O -Wall -c mxml-file.c gcc -O -Wall -c mxml-index.c gcc -O -Wall -c mxml-node.c gcc -O -Wall -c mxml-search.c gcc -O -Wall -c mxml-set.c gcc -O -Wall -c mxml-private.c gcc -O -Wall -c mxml-string.c /bin/rm -f libmxml.a /usr/bin/ar crvs libmxml.a mxml-attr.o mxml-entity.o mxml-file.o mxml-index.o mxml-node.o mxml-search.o mxml-set.o mxml-private.o mxml-string.o a - mxml-attr.o a - mxml-entity.o a - mxml-file.o a - mxml-index.o a - mxml-node.o a - mxml-search.o a - mxml-set.o a - mxml-private.o a - mxml-string.o ranlib libmxml.a make[3]: Leaving directory `/usr/src/zaptel-1.4.0/menuselect/mxml' gcc -Wall -o menuselect.o -g -c -D_GNU_SOURCE menuselect.c gcc -Wall -o menuselect_curses.o -g -c -D_GNU_SOURCE menuselect_curses.c gcc -Wall -o strcompat.o -g -c -D_GNU_SOURCE strcompat.c gcc -g -Wall -o menuselect menuselect.o menuselect_curses.o strcompat.o mxml/libmxml.a -lncurses make[2]: Leaving directory `/usr/src/zaptel-1.4.0/menuselect' make[1]: Leaving directory `/usr/src/zaptel-1.4.0/menuselect' gcc gendigits.c -lm -o gendigits ./gendigits tones.h gcc -o makefw makefw.c ./makefw tormenta2.rbt tor2fw tor2fw.h Loaded 69900 bytes from file ./makefw pciradio.rbt radfw radfw.h Loaded 42096 bytes from file make -C /usr/src/linux SUBDIRS=/usr/src/zaptel-1.4.0 modules make[1]: Entering directory `/usr/src/kernel-source-2.6.8' CC [M]
Re: [asterisk-users] A200 card problem
Exactly, Sangoma support is THE BEST! :) Best Regards Josué 2007/3/15, John Novack [EMAIL PROTECTED]: Sangoma gives excellent support Suggest you try there first They probably will want SSH access to the box. Send them an e-mail John Novack Todd H wrote: Hi - I just got an A200 card with 1 FXO and 1 FXS module. Sadly, I can't make it work- currently, asterisk will not startup because of a bad module. Below are some log files/config files. If anyone has any suggestions, I'd appreciate it. I used Trixbox 2.0 and followed instructions on (http://sangoma.editme.com/wanpipe-linux-asterisk-atHome) - no problems running through or compiling. I also downloaded trixbox 2.0 with sangoma drivers included directly from sangoma (http://wiki.sangoma.com/Trixbox-1xx). I know how this list feels about trixbox, but still, the card/configs are the same, no? Any advice is appreciated. thanks Todd +++ /var/log/asterisk/full Mar 15 16:12:37 VERBOSE[31964] logger.c: [chan_mgcp.so]Mar 15 16:12:37 VERBOSE[31964] logger.c: [chan_mgcp.so] = (Media Gateway Control Protocol (MGCP)) Mar 15 16:12:37 VERBOSE[31964] logger.c: [chan_local.so]Mar 15 16:12:37 VERBOSE[31964] logger.c: [chan_local.so] = (Local Proxy Channel) Mar 15 16:12:37 VERBOSE[31964] logger.c: [chan_phone.so]Mar 15 16:12:37 VERBOSE[31964] logger.c: [chan_phone.so] = (Linux Telephony API Support) Mar 15 16:12:37 VERBOSE[31964] logger.c: [chan_zap.so]Mar 15 16:12:37 VERBOSE[31964] logger.c: [chan_zap.so] = (Zapata Telephony w/PRI) Mar 15 16:12:37 WARNING[31964] chan_zap.c: Unable to specify channel 1: No such device or address Mar 15 16:12:37 ERROR[31964] chan_zap.c: Unable to open channel 1: No such device or address here = 0, tmp-channel = 1, channel = 1 Mar 15 16:12:37 ERROR[31964] chan_zap.c: Unable to register channel '1' Mar 15 16:12:37 WARNING[31964] loader.c: chan_zap.so: load_module failed, returning -1 Mar 15 16:12:37 WARNING[31964] loader.c: Loading module chan_zap.so failed! [EMAIL PROTECTED] ~]# +++ END /var/log/asterisk/full +++ /var/log/messages [EMAIL PROTECTED] ~]# tail -20 /var/log/messages Mar 15 16:24:08 asterisk1 safe_asterisk: [pbx_config.so] = (Text Extension Configuration) Mar 15 16:24:08 asterisk1 safe_asterisk: [pbx_functions.so] = (Builtin dialplan functions) Mar 15 16:24:08 asterisk1 safe_asterisk: [pbx_ael.so] = (Asterisk Extension Language Compiler) Mar 15 16:24:08 asterisk1 safe_asterisk: [pbx_dundi.so] = (Distributed Universal Number Discovery (DUNDi)) Mar 15 16:24:08 asterisk1 safe_asterisk: [pbx_loopback.so] = (Loopback Switch) Mar 15 16:24:08 asterisk1 safe_asterisk: [pbx_spool.so] = (Outgoing Spool Support) Mar 15 16:24:08 asterisk1 safe_asterisk: [chan_iax2.so] = (Inter Asterisk eXchange (Ver 2)) Mar 15 16:24:08 asterisk1 safe_asterisk: [chan_agent.so] = (Agent Proxy Channel) Mar 15 16:24:08 asterisk1 safe_asterisk: [chan_skinny.so] = (Skinny Client Control Protocol (Skinny)) Mar 15 16:24:08 asterisk1 safe_asterisk: [chan_h323.so] = (Objective Systems H323 Channel) Mar 15 16:24:08 asterisk1 safe_asterisk: [chan_sip.so] = (Session Initiation Protocol (SIP)) Mar 15 16:24:08 asterisk1 safe_asterisk: [chan_features.so] = (Feature Proxy Channel) Mar 15 16:24:08 asterisk1 safe_asterisk: [skipping chan_oss.so] Mar 15 16:24:08 asterisk1 safe_asterisk: [chan_mgcp.so] = (Media Gateway Control Protocol (MGCP)) Mar 15 16:24:08 asterisk1 safe_asterisk: [chan_local.so] = (Local Proxy Channel) Mar 15 16:24:08 asterisk1 safe_asterisk: [chan_phone.so] = (Linux Telephony API Support) Mar 15 16:24:08 asterisk1 safe_asterisk: [chan_zap.so] = (Zapata Telephony w/PRI) Mar 15 16:24:08 asterisk1 safe_asterisk: Asterisk died with code 1. Mar 15 16:24:08 asterisk1 safe_asterisk: Automatically restarting Asterisk. Mar 15 16:24:08 asterisk1 asterisk: safe_asterisk startup succeeded [EMAIL PROTECTED] ~]# +++ END /var/log/messages ++/etc/zaptel.conf+++ [EMAIL PROTECTED] ~]# more /etc/zaptel.conf # Autogenerated by /usr/local/sbin/sangoma/setup-sangoma -- do not hand edit # Zaptel Channels Configurations (zaptel.conf) # loadzone=us defaultzone=us # Sangoma A200 [slot:9 bus:1 span: 1] fxsks=1 fxsks=2 fxoks=3 fxoks=4 [EMAIL PROTECTED] ~]# ++END /etc/zaptel.conf+++ ++/etc/asterisk/zapata.conf+++ [EMAIL PROTECTED] asterisk]# more zapata.conf ; ; Zapata telephony interface ; ; Configuration file [trunkgroups] [channels] language=en context=from-zaptel signalling=fxs_ks rxwink=300 ; Atlas seems to use long (250ms) winks ; ; Whether or not to do distinctive ring detection on FXO lines ; ;usedistinctiveringdetection=yes usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes
Re[2]: [asterisk-users] A200 card problem
I couldn't agree more. -Original Message- From: Josué Conti [EMAIL PROTECTED] To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Cc: Sent: Thu, 15 Mar 2007 20:31:53 -0300 Delivered: Thu, 15 Mar 2007 20:20:22 Subject:[asterisk-users] A200 card problem Exactly, Sangoma support is THE BEST! :) Best Regards Josué 2007/3/15, John Novack [EMAIL PROTECTED]: Sangoma gives excellent support Suggest you try there first They probably will want SSH access to the box. Send them an e-mail John Novack Todd H wrote: Hi - I just got an A200 card with 1 FXO and 1 FXS module. Sadly, I can't make it work- currently, asterisk will not startup because of a bad module. Below are some log files/config files. If anyone has any suggestions, I'd appreciate it. I used Trixbox 2.0 and followed instructions on (http://sangoma.editme.com/wanpipe-linux-asterisk-atHome) - no problems running through or compiling. I also downloaded trixbox 2.0 with sangoma drivers included directly from sangoma (http://wiki.sangoma.com/Trixbox-1xx). I know how this list feels about trixbox, but still, the card/configs are the same, no? Any advice is appreciated. thanks Todd +++ /var/log/asterisk/full Mar 15 16:12:37 VERBOSE[31964] logger.c: [chan_mgcp.so]Mar 15 16:12:37 VERBOSE[31964] logger.c: [chan_mgcp.so] = (Media Gateway Control Protocol (MGCP)) Mar 15 16:12:37 VERBOSE[31964] logger.c: [chan_local.so]Mar 15 16:12:37 VERBOSE[31964] logger.c: [chan_local.so] = (Local Proxy Channel) Mar 15 16:12:37 VERBOSE[31964] logger.c: [chan_phone.so]Mar 15 16:12:37 VERBOSE[31964] logger.c: [chan_phone.so] = (Linux Telephony API Support) Mar 15 16:12:37 VERBOSE[31964] logger.c: [chan_zap.so]Mar 15 16:12:37 VERBOSE[31964] logger.c: [chan_zap.so] = (Zapata Telephony w/PRI) Mar 15 16:12:37 WARNING[31964] chan_zap.c: Unable to specify channel 1: No such device or address Mar 15 16:12:37 ERROR[31964] chan_zap.c: Unable to open channel 1: No such device or address here = 0, tmp-channel = 1, channel = 1 Mar 15 16:12:37 ERROR[31964] chan_zap.c: Unable to register channel '1' Mar 15 16:12:37 WARNING[31964] loader.c: chan_zap.so: load_module failed, returning -1 Mar 15 16:12:37 WARNING[31964] loader.c: Loading module chan_zap.so failed! [EMAIL PROTECTED] ~]# +++ END /var/log/asterisk/full +++ /var/log/messages [EMAIL PROTECTED] ~]# tail -20 /var/log/messages Mar 15 16:24:08 asterisk1 safe_asterisk: [pbx_config.so] = (Text Extension Configuration) Mar 15 16:24:08 asterisk1 safe_asterisk: [pbx_functions.so] = (Builtin dialplan functions) Mar 15 16:24:08 asterisk1 safe_asterisk: [pbx_ael.so] = (Asterisk Extension Language Compiler) Mar 15 16:24:08 asterisk1 safe_asterisk: [pbx_dundi.so] = (Distributed Universal Number Discovery (DUNDi)) Mar 15 16:24:08 asterisk1 safe_asterisk: [pbx_loopback.so] = (Loopback Switch) Mar 15 16:24:08 asterisk1 safe_asterisk: [pbx_spool.so] = (Outgoing Spool Support) Mar 15 16:24:08 asterisk1 safe_asterisk: [chan_iax2.so] = (Inter Asterisk eXchange (Ver 2)) Mar 15 16:24:08 asterisk1 safe_asterisk: [chan_agent.so] = (Agent Proxy Channel) Mar 15 16:24:08 asterisk1 safe_asterisk: [chan_skinny.so] = (Skinny Client Control Protocol (Skinny)) Mar 15 16:24:08 asterisk1 safe_asterisk: [chan_h323.so] = (Objective Systems H323 Channel) Mar 15 16:24:08 asterisk1 safe_asterisk: [chan_sip.so] = (Session Initiation Protocol (SIP)) Mar 15 16:24:08 asterisk1 safe_asterisk: [chan_features.so] = (Feature Proxy Channel) Mar 15 16:24:08 asterisk1 safe_asterisk: [skipping chan_oss.so] Mar 15 16:24:08 asterisk1 safe_asterisk: [chan_mgcp.so] = (Media Gateway Control Protocol (MGCP)) Mar 15 16:24:08 asterisk1 safe_asterisk: [chan_local.so] = (Local Proxy Channel) Mar 15 16:24:08 asterisk1 safe_asterisk: [chan_phone.so] = (Linux Telephony API Support) Mar 15 16:24:08 asterisk1 safe_asterisk: [chan_zap.so] = (Zapata Telephony w/PRI) Mar 15 16:24:08 asterisk1 safe_asterisk: Asterisk died with code 1. Mar 15 16:24:08 asterisk1 safe_asterisk: Automatically restarting Asterisk. Mar 15 16:24:08 asterisk1 asterisk: safe_asterisk startup succeeded [EMAIL PROTECTED] ~]# +++ END /var/log/messages ++/etc/zaptel.conf+++ [EMAIL PROTECTED] ~]# more /etc/zaptel.conf # Autogenerated by /usr/local/sbin/sangoma/setup-sangoma -- do not hand edit # Zaptel Channels Configurations (zaptel.conf) # loadzone=us defaultzone=us # Sangoma A200 [slot:9 bus:1 span: 1] fxsks=1 fxsks=2 fxoks=3 fxoks=4 [EMAIL PROTECTED] ~]# ++END /etc/zaptel.conf+++
[asterisk-users] voip-info.org is back!
Looks like the site is back up. Don't all hit it at once, it might go down again ;-) Sean ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] voip-info.org is back!
That's awesome, we were nearly done with the spider too! On 3/15/07, Sean Bright [EMAIL PROTECTED] wrote: Looks like the site is back up. Don't all hit it at once, it might go down again ;-) Sean ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mitchel Constantin Snap - A desktop user interface for Asterisk www.snapanumber.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom call parking feature and Asterisk call parking
On 3/14/07, Mojo with Horan Company, LLC [EMAIL PROTECTED] wrote:
The third field, in my case Local/4${BRIDGEPEER:5:[EMAIL PROTECTED]
is the channel to announce the parked call slot to. In my case,
extensions beginning with 1xx are the phones themselves, and extensions
4xx are the same phones but will make them auto-answer (like paging).
You might have a better way to do this because this is a little cumbersome.
The auto answer part of this is just too cool! You don't have to put
the phone on the hook and parking, you don't have to worry about
answering or fumbling around and hanging up on the callback, or
anything. If you have the handset in your hands, it calls back, plays
the number in your ear and it's done. If you're on speakerphone, it
calls back and plays the number and it's done. Almost as good as
parking using chan_sccp with the Cisco (it also displays the parking
spot on the park... hmmm... that could possibly be done...).
On my phones, when I press park, I have to press a number (any number
works) and then press park again. Is this the case for everyone else,
or am I missing something? Not a big deal, just seems odd.
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: sip_nat.conf - Asterisk with two Ethernet Interfaces
I figured it out by examining the log files. The following works but not as hypothesized above. 1. the second externip overrides the first, so externip can only be specified once 2. the first localnet and the second localnet are BOTH understood and used. 3. asterisk tests the destination IP against BOTH localnet specifications, and if the destination IP does NOT match, THEN it substitutes its SOURCE IP with externip. 4. if the source IP is not substituted then the correct local source IP is used depending upon which interface the request came from. so, all's good, at least for my scenario. quote who=Anjul Srivastava Will this do the intended thing? This is in sip_nat.conf which is included in sip.conf: externip=192.168.0.200 localnet=192.168.0.200/255.255.255.0 externip=64.168.237.110 localnet=192.168.1.2/255.255.255.0 I have Asterisk running on a box with two Ethernet interfaces and bound to both. One interface, 192.168.1.2 services clients outside the firewall who are led to believe that Asterisk is running at 64.168.237.110. The other interface, 192.168.0.200 services clients inside the firewall who are led to believe that Asterisk is running at 192.168.0.200. I am worried that Asterisk might not do what I am intending to do. What I want is for Asterisk to send SIP/SDP Connection Info as 64.168.237.110 to those clients that contact it on 192.168.1.2, and to send SIP/SDP Connection Info as 64.168.237.110 to those that contact it on 192.168.0.200. I am worried that it might use the last externip=... declaration for both interfaces and ignore the other one. Could somebody help me with determining the treatment of the multiple externip=... declarations? Or point me to how I could verify? I don't want to wait until the evening to find out from outside the firewall that this doesn't work. (It is working fine from within the firewall where I contact asterisk at 192.168.0.200) Thank you! Best regards, Anjul. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] voip-info.org is back!
Sean Bright wrote: Looks like the site is back up. Don't all hit it at once, it might go down again ;-) ...and now... mirrormirrormirrormirrormirrormirrormirrormirrormirrormirrormirror -stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Help! Echo problem even at T1 PRI?
Hello, We have an asterisk setup at our client's site using a TE205P. The line to telco is a 23 channels T1 PRI, however the line has random echo problems (about 5-10% of the calls)! Can anybody tell me if echo cancellation is really needed even at a T1 PRI to the telco? Because people keep saying when they deploy voip solution in Hong Kong using T1 PRI, there is no need of echo cancellation. (even the local Digium distributor) Asterisk is 1.2.13, zaptel is 1.2.10. I choosed the MARK2 canceller in the zaptel. The setting in zaptel is default: echocancel=yes echocancelwhenbridged=yes echotraining=yes rxgain=0 txgain=0 When I tried to adjust the echocancel value to a higher taps (e.g. 256) or adjust the rxgain/txgain value, I can even hear echo much easier. Anyone can confirm with me we should rather turn echocancelwhenbridged=no, or even echocancel=no when we only use a T1 PRI at the zaptel? Or can have other suggestions to try solving this problem? Because of some people's information that T1 PRI does not need any echo cancellation, add a hardware echo cancellation module is not an option here. I think I need some sort of information from another 3rd party before my boss will even agree to just try something out. Really thank you!! Vincent ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: zapata with Tiger3XX compilation error
On Thu, Mar 15, 2007 at 03:38:20PM -0700, pedro noticioso wrote: Ok so I read the Linux 2.6 related README and finally compiled propperly, I thought but at the end I notice that lscpi does report the cards, but I cant modprobe wcfxo nor zaptel and I do have wcfxo.ko in the /lib/modules/2.6.8/extra/ directory, so what gives? This is a Debian Sarge, thanks! I believe the docs have misled you. I bet /usr/src/linux is a link to some kernel source that happens to have a kernel almost configured correctly. What is the output of: modinfo wcfxo Anything? apt-get install linux-headers-`uname -r` and repeat the build. Alternatively, echo deb http://updates.xorcom.com/rapid sarge main /etc/apt/sources.list apt-get update apt-get install zaptel zaptel-modules-`uname -r` genzaptelconf # optional: #echo '#include zapata-channels.conf' /etc/asterisk/zapata.conf #/etc/init.d/asterisk start If you want asterisk 1.4: that should be a simple matter of adding two symlinks, I believe. I'll have to dig in further. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] voip-info.org status update
We had 2 of 3 SCSI drives fail in a RAID a couple of weeks ago - its hard to explain that to a customer! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of cb Sent: Thursday, March 15, 2007 12:00 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] voip-info.org status update On Mar 15, 2007, at 12:32 AM, shadowym wrote: Hard to expect the business community to take Asterisk seriously when this sort of stuff happens IMHO. I can't understand how 3 of 4 hard drives could just suddenly fail simultaneously. There must be more too it. It is drifting off topic, but if all the drives in the array where bought from the same batch, and it was a bad batch, they could all fail at about the same time. I've seen it happen in non-raid drives, I had a batch of drives all bought at the same time, that all went bad within about a week of each other. Each was in a different PC so they had slightly different up times and usage. I could see if those drives had been in a RAID array and were being stressed equally, they may have all failed within hours or even minutes of each other. -chris www.mythtech.net ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: busy/hangup/answer detection in PRI E1 channels (Vidura Senadeera)
Hi Gareth Blades Doug,
Thanks so much for for the feedback. I have searched on lot of documents
but couldn't able to find clear answer regarding it.
I hope you guys replies are very much help all in aterisk community.
Thanks Regards,
Vidura Senadeera,
Network Engineer,
Debug Solutions
Sri Lanka .
Tel - +94114520036
Mobile - +9466596
Web - www.debug.lk
--
Message: 14
Date: Thu, 15 Mar 2007 15:39:07 +0530
From: Vidura Senadeera [EMAIL PROTECTED]
Subject: [asterisk-users] busy/hangup/answer detection in PRI E1
channels
To: asterisk-users@lists.digium.com
Message-ID:
[EMAIL PROTECTED]
Content-Type: text/plain; charset=iso-8859-1
Hi,
Please discribe me how we define busy/hang/answer detection with PRI E1
channels.
Since busydetect, callprogress, busycount giving falts hangup and call
drops
what is the solution on PRI channels?
--
Thanks Regards,
Vidura B. Senadeera.
-- next part --
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20070315/e9cae81c/attachment-0001.htm
--
Message: 16
Date: Thu, 15 Mar 2007 10:35:16 +
From: Gareth Blades [EMAIL PROTECTED]
Subject: Re: [asterisk-users] busy/hangup/answer detection in PRI E1
channels
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID:
[EMAIL PROTECTED]
Content-Type: text/plain
You can use the hangupcause variable which us the pri cause code
supplied when a call is ended over a PRI line. For example this is the
maco we use to dial a number over PRI.
[macro-pridial]
exten = s,1,GotoIf($[${ARG1:0:2} != 00]?noint)
exten = s,n,Set(DENYINT=${DB(denyinternational/${CALLERIDNUM})})
exten = s,n,GotoIf($[ ${DENYINT} = yes ]?congestion)
exten = s,n(noint),Set(BLOCKCID=${DB(blockcid/${CALLERIDNUM})})
exten = s,n,GotoIf($[ ${BLOCKCID} = yes ]?prohib:cont)
exten = s,n(prohib),SetCallerPres(prohib)
exten = s,n(cont),Dial(ZAP/g1/${ARG1},60,Tr)
exten = s,n,Set(CDR(userfield)=${HANGUPCAUSE}.${DIALSTATUS})
exten = s,n,GotoIf($[ ${DIALSTATUS} = BUSY ]?busy)
exten = s,n,GotoIf($[ ${DIALSTATUS} = CONGESTION ]?congestion)
exten = s,n,GotoIf($[ ${HANGUPCAUSE} = 28 ]?unrecognised)
exten = s,n,GotoIf($[ ${HANGUPCAUSE} = 1 ]?discon)
exten = s,n,GotoIf($[ ${DIALSTATUS} = CHANUNAVAIL ]?congestion)
exten = s,n,Hangup
exten = s,n(busy),Busy
exten = s,n(congestion),GotoIf($[ ${HANGUPCAUSE} = 34 ]?error)
exten = s,n,Congestion
exten = s,n(error),Answer
exten = s,n,SendText(${HANGUPCAUSE}: ERROR: No channels available)
exten = s,n,Wait(1)
exten = s,n,Playback(all-outgoing-lines-unavailable)
exten = s,n,Wait(10)
exten = s,n,Hangup
exten = s,n(unrecognised),Answer
exten = s,n,SendText(${HANGUPCAUSE}: Unrecognised No.)
exten = s,n,Wait(1)
exten = s,n,Playback(that-is-not-rec-phn-num)
exten = s,n,Wait(10)
exten = s,n,Hangup
exten = s,n(discon),Answer
exten = s,n,SendText(${HANGUPCAUSE}:Out Of Service)
exten = s,n,Wait(1)
exten = s,n,Playback(discon-or-out-of-service)
exten = s,n,Wait(10)
exten = s,n,Hangup
On Thu, 2007-03-15 at 10:09, Vidura Senadeera wrote:
Hi,
Please discribe me how we define busy/hang/answer detection with PRI
E1 channels.
Since busydetect, callprogress, busycount giving falts hangup and call
drops what is the solution on PRI channels?
--
Thanks Regards,
Vidura B. Senadeera.
Message: 18
Date: Thu, 15 Mar 2007 07:06:30 -0400
From: Doug Lytle [EMAIL PROTECTED]
Subject: Re: [asterisk-users] busy/hangup/answer detection in PRI E1
channels
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Vidura Senadeera wrote:
Hi,
Please discribe me how we define busy/hang/answer detection with PRI
E1 channels.
Since busydetect, callprogress, busycount giving falts hangup and call
drops what is the solution on PRI channels?
PRI channels have call supervision and Asterisk will see the
hangup/answers just fine. The busydetect, callprogress, busycount
should be removed from your setup.
Doug
--
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel version for asterisk 1.2.16
On 3/14/07, Kevin P. Fleming [EMAIL PROTECTED] wrote: There is no need for any 'map'; any Asterisk 1.2.x release should be usable with any Zaptel 1.2.x release, but of course we'd suggest using the latest releases of both. There are no API changes or feature additions (generally) in release branches, so frequently you can update _only_ Asterisk if you are happy with the version of Zaptel you have installed and running. Thanks for that. That's actually good news for those of us that have local patches to apply to apply to zaptel, to make wacky phones ring at a different frequency, for example. Kevin, I would recommend adding the above to the download page so we can see it. Regards, Randy R ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help! Echo problem even at T1 PRI?
Hi Vincent - Can anybody tell me if echo cancellation is really needed even at a T1 PRI to the telco? Because people keep saying when they deploy voip solution in Hong Kong using T1 PRI, there is no need of echo cancellation. (even the local Digium distributor) I have to do echo cancellation on a PRI for one of my customers, even though the Telco office is only two blocks away. Asterisk is 1.2.13, zaptel is 1.2.10. I choosed the MARK2 canceller in the zaptel. There's also the aggressive option for MARK2, you might try that. Or you could try the MG2 echo can. The setting in zaptel is default: echocancel=yes echocancelwhenbridged=yes echotraining=yes The best settings for us: echocancel=yes echocancelwhenbridged=no echotraining=no You'll probably have to do some experimenting, as your best values may be different. Good Luck! - Noah ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
