Re: [asterisk-users] Dealing with progress codes
On Thu, Oct 30, 2008 at 1:40 PM, Juan RodrÃguez [EMAIL PROTECTED] wrote: With a script connecting to a DB server and looking for the prefix, is a good solution. This way you don't need to force the user to dial the the leading 1 (or not to do it), you just look on the DB server and if it does not matches a local prefix then you dial with the leading 1. What method do you use to determine if an area code or NPA-NXX is local to your area? In my area, with the presence of things like EACS extended calling or other methods, I haven't found any reliable method for this. In case EACS is something specific to my area, its a line-based feature that extends a number's calling scope, but seems to be different from metro calling and not specifically defined anywhere but Telcordia documents, I suppose. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and MiniITX setups
Here are my numbers, with CentOS 4.4 processor : 0 vendor_id : CentaurHauls cpu family : 6 model : 9 model name : VIA Nehemiah stepping: 10 cpu MHz : 533.573 cache size : 64 KB fdiv_bug: no hlt_bug : no f00f_bug: no coma_bug: no fpu : yes fpu_exception : yes cpuid level : 1 wp : yes flags : fpu vme de pse tsc msr cx8 mtrr pge cmov pat mmx fxsr sse rng rng_en ace ace_en bogomips: 1067.68 g723 gsm ulaw alaw g726 adpcm slin lpc10 g729 speex ilbc g723 - - - - - - - - - - - gsm - - 5 512 5 419 - -74 ulaw -16 - 1 9 2 116 - -71 alaw -16 1 - 9 2 116 - -71 g726 -22 8 8 - 8 722 - -77 adpcm -16 2 2 9 - 116 - -71 slin -15 1 1 8 1 -15 - -70 lpc10 -281414211413 - - -83 g729 - - - - - - - - - - - speex - - - - - - - - - - - ilbc -27131320131227 - - - On 12/29/06, C F [EMAIL PROTECTED] wrote: Gordon, how did you get such good numbers? Here is my setup: [EMAIL PROTECTED]:~# cat /proc/cpuinfo processor : 0 vendor_id : CentaurHauls cpu family : 6 model : 10 model name : VIA Esther processor 1200MHz stepping: 9 cpu MHz : 399.054 cache size : 128 KB fdiv_bug: no hlt_bug : no f00f_bug: no coma_bug: no fpu : yes fpu_exception : yes cpuid level : 1 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge cmov pat clflush acpi mmx fxsr sse sse2 tm pni est tm2 rng rng_en ace ace_en bogomips: 799.99 g723 gsm ulaw alaw g726 adpcm slin lpc10 g729 speex ilbc g723 - - - - - - - - - - - gsm - -111134131052 - - 202 ulaw -35 - 125 4 143 - - 193 alaw -35 1 -25 4 143 - - 193 g726 -562323 -252264 - - 214 adpcm -36 3 326 - 244 - - 194 slin -34 1 124 3 -42 - - 192 lpc10 -683535583734 - - - 226 g729 - - - - - - - - - - - speex - - - - - - - - - - - ilbc -69363659383577 - - - What distro are you running? On 12/29/06, Gordon Henderson [EMAIL PROTECTED] wrote: On Fri, 29 Dec 2006, Mark Greene wrote: How well do you think asterisk could run on a miniITX board like the ones linked below with the call volume of say a small doctors office or something? http://www.mini-box.com/s.nl/sc.8/category.15/.f I have a lot of installations using this board: http://www.icp-epia.co.uk/index.php?act=viewProdproductId=90 The key thing is to compile asterisk for a i586. This is vitally important, as the VIA processor on those boards is lacking some MMX instructions that asterisk uses. 20 calls aren't an issue here. Transcoding is. You *really* don't want to be using speex or ilbc here! Here is the output of a show translation recalc 30 on one of these: g723 gsm ulaw alaw g726 adpcm slin lpc10 g729 speex ilbc g723 - - - - - - - - - - - gsm - - 4 412 4 320 - -72 ulaw - 7 - 110 2 118 - -70 alaw - 7 1 -10 2 118 - -70 g726 -14 9 9 - 9 825 - -77 adpcm - 7 2 210 - 118 - -70 slin - 6 1 1 9 1 -17 - -69 lpc10 -171212201211 - - -80 g729 - - - - - - - - - - - speex - - - - - - - - - - - ilbc -18131321131229 - - - 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 ___ --Bandwidth and Colocation provided by
Re: [asterisk-users] Re: [OT] Wifi SIP phones - LinkSys WIP330
On 12/28/06, Bryan M. Johns [EMAIL PROTECTED] wrote: I recommend the hitachi wifi phones for use with asterisk. Bryan M. Johns Partner Shelton Johns Technology Group Office: (678) 248-2637 X: 1500 Direct: (678) 229-1809 http://www.sheltonjohns.com **Sent from my mobile phone** -Original Message- From: Steven [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: 12/28/2006 4:30 PM Subject: [asterisk-users] Re: [OT] Wifi SIP phones - LinkSys WIP330 I bought a WIP300 to test and it was aweful. It would either not register a keypress or register it twice. It would also freeze up few minutes at a time. It looks like the WIP330 has a new keypad, so maybe that problem is gone. The WIP300 worked with asterisk, but I can not recall the quality at this point. -- -- Steven http://www.glimasoutheast.org Wayne [EMAIL PROTECTED] wrote in message news: [EMAIL PROTECTED] Hi List, Hope everyone is recovering from the festive season :) (ok we still have new years i guess!) Anyways, I was wondering if anyone has had any successful dealings with WiFi phones and operation with '*' at all? I've been keeping my eye on the LinkSys WIP330 ( http://preview.tinyurl.com/nccxn ) and wondered your collective thoughts? Would I be correct in thinking that (as long as the relevant ports were open on the firewall) it would be possible to still be an extension to * if you could access the internet from, say, a wifi hot spot that was not a part of the lan? Thanks Wayne . Funny you should mention this. I just pulled a WIP300 out of a box about 5 minutes ago to test it. First impression: The speaker sucks. All calls sound like there's an ill-tuned radio in the background, with some kind of squealing always present. Also a fair amount of static. The AP is about 5 feet away, so I don't think its the connectivity. I'm not giving up on this phone yet though. Will report back with more if this topic still lives when I'm finished. ___ --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] How to kill a meet me room at midnight
Or you could AGI a PHP script that runs before each caller enters the conference room that sets the AbsoluteTimeout on the channel to midnight that night (calculating the proper seconds, of course). The conference would of course die at midnight after all its participants left. Nathan On 11/27/06, Noah Miller [EMAIL PROTECTED] wrote: You could write an extension which executes meetme kick, for all the channels, but I am not sure how to execute such a thing at a given time. Create a call file, and schedule it to run with cron. The following page on the wiki shows something similar: http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out You can adapt it to suit your needs. - Noah on Saturday 11/25/2006 Eric Bishop([EMAIL PROTECTED]) wrote Not quite what I'm looking for. I ant to hang up all channels (zap or sip) in meetme room 5 On 11/23/06, Michiel van Baak [EMAIL PROTECTED] wrote: On 19:18, Thu 23 Nov 06, Eric Bishop wrote: Other than rebooting the server or restarting Asterisk from cron does anyone know how to kill a meetme room at midnight. Or perhaps other creative ways people deal with callers who don't hang up. You can use soft hangup chan -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called 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 Not quite what I'm looking for. I ant to hang up all channels (zap or sip) in meetme room 5brbrdivspan class=gmail_quoteOn 11/23/06, b class=gmail_sendernameMichiel van Baak/b lt;a href=mailto: [EMAIL PROTECTED] [EMAIL PROTECTED]/agt; wrote:/spanblockquote class=gmail_quote style=border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;On 19:18, Thu 23 Nov 06, Eric Bishop wrote:brgt; Other than rebooting the server or restarting Asterisk from cron does anyone brgt; know how to kill a meetme room at midnight. Or perhaps other creative waysbrgt; people deal with callers who don't hang up.brbrYou can use soft hangup lt;changt;brbr--brMichiel van Baakbra href=mailto:[EMAIL PROTECTED] [EMAIL PROTECTED]/abra href=http://michiel.vanbaak.eu; http://michiel.vanbaak.eu/abrGnuPG key: a href= http://pgp.mit.edu:11371/pks/lookup?op=getamp;search=0x71C946BD; http://pgp.mit.edu:11371/pks/lookup?op=getamp;search=0x71C946BD /abrbrquot;Why is it drug addicts and computer afficionados are both called users?quot;brbr___br--Bandwidth and Colocation provided by a href=http://Easynews.com; Easynews.com/a --brbrasterisk-users mailing listbrTo UNSUBSCRIBE or update options visit:brnbsp;nbsp; a href= http://lists.digium.com/mailman/listinfo/asterisk-users; http://lists.digium.com/mailman/listinfo/asterisk-users /abr/blockquote/divbr ___ --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 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [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 ___ --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] Transfers - No ringback or moh
On 7/24/06, Douglas Garstang [EMAIL PROTECTED] wrote: Douglas Garstang wrote: -Original Message- When the extension on your desk is ringing, after you have pressed transfer key a second time(soft or hard key), does the original caller still hear music on hold, or ringback or nothing? Following your example, pressing transfer once, entering the extension (Caller C's) does not yield a second transfer option until C answers. Pressing the button anyway, does not get a response from the phone. When (B) selects the initial transfer, I have Cancel, Name, Blind. (A) hears hold music. During the transfer and Before (C) answers, the phone options are Cancel, Split. Once C answers, I have, Hold, Cancel, Transfer, More We have SIP version 1.6.3. Polycom must have changed something... Doug. Not exactly. There's an option in the Polycom config to disallow unattended attended transfers. In other words, you do not have the option to press Transfer while you are getting in a RINGING progress state. Sounds Like one of you has that option enabled, the other has it disabled. I don't know exactly what the option is, but I've seen it before. Nathan ___ --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] Grandstream Budgetone and Mac mini?
Mac Ethernet ports are auto-switching. Don't need a cross-cable :) On 4/18/06, Mark Phillips [EMAIL PROTECTED] wrote: Just for shits and giggles, have you tried using a cross over cable? I'mnot saying it's gonna work because everything I read says you're doing the right thing but it's worth a try.Mark, G7LTT/KC2ENIRandolph, NJhttp://www.g7ltt.comDmitry Ivanov wrote: Hallo! Anyone tried connect PC port of BT-102 to Mac mini? I have four BT-102. Looks like none of them works with Mac mini G4... ___ --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 listTo 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: Random music not so 'random'
On 4/3/06, Roger McCoy [EMAIL PROTECTED] wrote: Scratch that!Apparently asterisk just needed to be restarted.. wonder why a reload didn't work? Reloads don't update music on hold settings. moh reload seems to work just fine for me. Nathan ___ --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] VoiceMail realtime not working in asterisk-1.2.6
On 4/4/06, asterisk user [EMAIL PROTECTED] wrote: hi all,I can not get voicemail working in realtime withasterisk-1.2.6. extconfig.conf is correct voicemail = odbc,asterisk,voicemail_usersi am getting the fallowing errorExecuting Answer(SIP/xx.xx.xx.xxx-0a02e1c0, ) innew stack -- Executing Set(SIP/xx.xx.xxx-0a02e1c0, foo=102) in new stack -- Executing NoOp(SIP/xx.xx.xx.xxx-0a02e1c0,102) in new stack -- Executing GotoIf(SIP/xx.xx.xx.xxx-0a02e1c0,0?5:7) in new stack -- Goto (default,102,7) -- ExecutingVoiceMail(SIP/xx.xx.xx.xxx-0a02e1c0, 102) in newstackApr5 11:57:34 WARNING[14612]: app_voicemail.c:2385leave_voicemail: No entry in voicemail config file for '102'.i also tried with adding searchcontexts=yesinvoicemail.conf but i got segmentationfault. What does odbc show give for output? I'm running 1.2.6 on a box and its working just fine, pulling mailbox information from Realtime. Nathan ___ --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] cdr_odbc appears to have fields missing
On 3/29/06, Brian Roy [EMAIL PROTECTED] wrote: I'm currently using Asterisk running version 1.2.5 and trying to use cdr_odbc to connect to a Microsoft SQL database. I have everything running, but the insert statement being sent to database doesn't appear to have the start, answer, end information in it. Below is the insert statement that MS Profiler shows being sent. As you can see those fields are missing. INSERT INTO cdr (calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode) VALUES When I look at the code, in this case calldate is actually the cdr-start value. I'm working on a patch to record answer and end as well. Nathan ___ --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] MusicOnHold with mpg123
Alright, I've come across a really strange issue and I've been banging my head trying to figure it out. I have 3 machines. 1 Dell Dimension 4100, Pentium 3. 1 Dell 400SC, Pentium 4. 1 Dell 1600SC, Xeon. I run mpg123 0.59r on each machine. Using RH9 with a 2.4.20-8 kernel, each machine plays MoH flawlessly. As RH9 gets older and older, however, the need to upgrade arose. So I upgraded each machine to CentOS 4.3, with various 2.6 kernels ranging from 2.6.22 to 2.6.34 (yes, with the spinlock error). With these machines in this current state, the old Dell Dimension plays MoH flawlessly (the provided fpm MoH with the distro), and the two newer dells (400SC and 1600SC) all have heavy static, crackling, and other undesirable noises introduced into what I can only guess is the decoding of the MP3 files. Neither of these newer machines can transcode between audio file formats without introducing this same static using sox as well. I tried moving to native MoH, using the format_mp3 module, but found the lack of volume control to be problematic for us. Since we've been dealing with either no MoH, or moh that has undesirable qualities (too loud, some static/crackling, etc) If anyone has some ideas regarding this, I'd be happy to hear them. Then maybe Jared and Leif won't have to put up with my exasperated, repeated attempts at fixing this :) ___ --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] Feedback from VON expo! Info on * HA and Polycom phone!!
On 3/16/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Q: What the deal with the limit on the number of people you can monitor for presence? A: There is no limit in the phone. This is an Asterisk limitation. BS. Polycom even stated it's a polycom limitation which will be fixed in the next firmware release (polycom is increasing the limit from 7 to 48 iirc). Apparently, its actually a limit in the Asterisk Business Edition, and to fully comply with their interop policies with Digium (one of their partners), Polycom has also made this a limit in the phone's hardware. That's what I've heard from quite a few places, before this thread popped up, anyway. ___ --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] how to show called name on calling polycom display
On 3/15/06, Giorgio Incantalupo [EMAIL PROTECTED] wrote: Hi, we have an asterisk 1.2.1 box and 2 polycom SIP phones. We'd like to show the called name on the calling polycom display instead of his /her extensions as I do with the caller name on the called polycom. Is it possible? If yes, how? TIA Giorgio Incantalupo I believe something like this is being worked on in the bugtracker at bugs.digium.com. I don't remember how far along the project is though. ___ --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] how to show called name on calling polycomdisplay
On 3/15/06, Alexander Lopez [EMAIL PROTECTED] wrote: This is a function of the Phone itself. Asterisk has nothing to do with it as it does not know anything about the call until after the SIP device 'sends' it. To my knowledge it is not posible. I don't even think a SIP standard is available for this. I guess the developers that have worked on implementing this should be told that, then. http://bugs.digium.com/view.php?id=6643 This 'feature' along with changing CallerID Display after a call has been answered is something missing from the RFC. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gabriel Afana Sent: Wednesday, March 15, 2006 12:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] how to show called name on calling polycomdisplay I was looking for this exactly as well Any ideas? - Gabe - Original Message - From: Giorgio Incantalupo [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, March 15, 2006 12:52 AM Subject: [Asterisk-Users] how to show called name on calling polycom display Hi, we have an asterisk 1.2.1 box and 2 polycom SIP phones. We'd like to show the called name on the calling polycom display instead of his /her extensions as I do with the caller name on the called polycom. Is it possible? If yes, how? 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] Cisco 7960 SIP - Displaying Time
On 3/13/06, Chris Stenton [EMAIL PROTECTED] wrote: I have had no issues with 8.2 so far! Chris Except the Caller ID issue reported in another thread? This issue has been fixed in SIP firmware 7.5 Omar A. Sabek Yes, and I read that SIP 7.5 firmware have some other issues. They recommend using 7.4 firmware. I'm not sure how good in new 8.2 firmware. Tomislav ___ --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] Cisco 7960 SIP - Displaying Time
On 3/9/06, Greg Oliver [EMAIL PROTECTED] wrote: On Thu, 2006-03-09 at 10:36 -0500, Nabeel Jafferali wrote: Is there a way to display the time of the 7960 running firmware 7.4? Im unable to find any information. Add the following to SIPDefault.cnf or SIPMAC.cnf: sntp_server: time.nrc.ca sntp_mode: unicast time_zone: ESTOn my 7960 with 7.4 firmware, the time automagically disappears for someunknown reason. The phone still functions, but the time goes awayuntil I reboot it.Not a big deal to me, so I have not investigated it further.-Greg I use anycast. Seems like I read something about directbroadcast not working in recent SIP versions. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Follow Me
On 2/23/06, Darrick Hartman [EMAIL PROTECTED] wrote: Dinesh Nair wrote: On 02/22/06 11:08 C F said the following: http://bugs.digium.com/view.php?id=5574 That is a patch that will do just that. while an app is nice, followme could have been done thru some nifty dialplan work as well.True, but why not accept the app?It sure makes the dial plan alot easier to read.Why not make things easy when possible, rather thanforcing a newer user to have to jump through all sorts of hoops toimplement a simple feature like this.Darrick In addition, using the dialplan to do this can make a mess of your CDRs, whereas an application can take better control of that situation. Nathan ___ --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 IP 601 Buddy Watch problems
On 2/23/06, Olle E Johansson [EMAIL PROTECTED] wrote: Isaac Xiao wrote: We have the same issue happened to all Asterisk versions of 1.2.X (I tried all). In CLI, it shows "-- Incoming call: Got SIP response 500 Internal Server Error back from 192.168.2.104". Once you see this msg, the buddy watch won't work any more until you reboot the phone. I also upgrade polycom phone from version 1.6.2 to 1.6.4.0064, but no luck.As always with SIP errors, it is very hard to say anything without a SIPtransaction log./O It may have something to do with the watch limit on the Polycom firmware. I have one phone that does this as well, nearly all the time. I believe its the only one that exceeds the 6-7 buddy watch limit as well. I'll try to get a sip debug to see whats going on. ___ --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] Voicemail Changes
On 10/21/05, Sean Cook [EMAIL PROTECTED] wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Sherwood McGowan Sent: Friday, October 21, 2005 3:45 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Voicemail Changes You know, I was thinking I was going crazy... I've had that issue, and another issue where whenever someone called one of our customers, the customer would get 4 to 5 blank voicemails Still haven't tracked that one down. Sherwood A while back I found that with the current voicemail system, the mwi would show that their was voicemail before the recording of a message was complete. If the user went in to the voicemail system, they would get an empty message and the message would be lost. Has this been fixed in the beta? Or is it on schedule for 1.2? Regards, Sean I am sure it is related... there needs to be a maildir like storage for voicemail so that these race conditions do not occur. Voicemail on recording, needs to be written to a tmp directory then moved into the appropriate area upon completion. Sean I just ran into this today, on 1.2.3 with Polycom IP 501 phones. Message was from a potential customer looking for a PBX too... imagine that embarrassment :) Anyone know how to get this resolved? Thanks, Nathan ___ --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 IP301: Pass-through ethernet port unusable?
On 1/31/06, Damon Estep [EMAIL PROTECTED] wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Jerry Glomph Black Sent: Monday, January 30, 2006 11:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Polycom IP301: Pass-through ethernet port unusable? Have just done a deployment of 45 of these puppies. They are doing their main job quite well, but of course there are minor kinks. A not-so-minor one is that if one attempts to plug a PC into the 2nd RJ-45 jack, as soon as you send any reasonable amount of traffic (even casual web surfing) the phone seizes. We had to run a bunch of cables in a big rush to users' PCs, having (erroneously) believed that the passthru RJ45 would be a usable port! Has anyone out there experienced this? No issues on the IP501 with 2.6.2 bootrom and 1.5.3 SIP. Ethernet port works fine for the PC. Works fine here, with an IP501, 2.6.2 bootrom and 1.6.3 SIP. Was using ethernet port for a while, even downloading large files through it without any hiccups. ___ --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] Shared Line Appearance
On 1/27/06, Kevin P. Fleming [EMAIL PROTECTED] wrote: Sean Cook wrote: Is there an implementation for shared line support in asterisk? I know that with hint I can watch line status... I just want to be able to pick up on an extension when ringing or jumping in on a call by punching the line. You are confusing shared line appearance with shared extension appearance. It is possible today to watch an extension's status and use the key on the phone to either call that extension (if it is not in use) or pick up a ringing call at that extension. With some creative dialplan programming it may also be possible to force any call that extension is involved in into a MeetMe and then join it... thereby joining the call. This is all 'shared extension appearance' stuff. Pick up a ringing call at that extension? I can see how you would do the rest of the things you mentioned, but how would you pick up a ringing call going to that extension? Shared Line Appearance is much more complex to implement, but we are very seriously considering doing it in the near future, since there is so much demand. Keep in mind that you will _never_ be able to fully simulate a key system using Asterisk unless you seriously dumb down the Asterisk features that don't make sense for a key system... but we can at least get this part functional. Stay tuned :-) That would be great to see. ___ --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] Problems with MusicOnHold
Greetings, I have two machines. One is a P3 Dell Dimension 4100, the other is a PowerEdge SC420. Both are running Asterisk 1.0.7, the PowerEdge has a TE405P card in it, the Dimension has a Digium X100P present (although not modprobed). Each machine has mpg123 0.59r loaded, and is using the exact same set of MP3s for music on hold (both the distributed ones and some of our own). Neither box is sharing any interrupts. I use the same 7960G to test the Music on Hold. On the Dimension 4100, MusicOnHold works flawlessly. No static, no glitches, nothing. On the PowerEdge SC420, MusicOnHold has a lot of static, pops, crackles, and almost everything you can imagine. I can't think of anything else that is applicable. Basically, the machines seem pretty much identical to me. I expected MoH to work the same as well, but it isn't. If anyone has any ideas, please let me know. Thanks, Nathan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Trouble with Realtime
On Fri, 11 Mar 2005 16:38:38 -0600, Nathan Bowyer [EMAIL PROTECTED] wrote: On Fri, 11 Mar 2005 15:04:03 -0600, Matthew Boehm [EMAIL PROTECTED] wrote: You can't have this: [from-sip] switch = Realtime/[EMAIL PROTECTED] The context in your extensions.conf must be different from your Realtime context. Okay, I'll try it, but that doesn't explain why voicemail doesn't work. The extension to access the voicemail is static in extensions.conf. -Matthew Nathan Bowyer wrote: Greetings, I'm having some trouble with the realtime engines. When asterisk loads, everything looks fine, there don't seem to be any problems via notices or anything. Furthermore, cdr_odbc is working, and actively logging my failed call attempts to db through ODBC using the same DSN. unixODBC and the mysql drivers are installed from source. Here are the relevant parts of the config: Extconfig.conf (Under the [settings] section) sipusers = odbc,voip,sip_users sippeers = odbc,voip,sip_users voicemail = odbc,voip,voicemail_users extensions = odbc,voip,extensions_table res_odbc.conf [asterisk] dsn = MySQL-asterisk username = voip password = temp123 pre-connect = yes Under extensions.conf, in the [from-sip] context: switch = Realtime/[EMAIL PROTECTED] Running isql MySQL-asterisk voip pass connects to the DB, and queries return the proper data. I have the following tables in the mysql databases: +--+ Tables_in_voip | +--+ cdr | extensions_table | sip_users| voicemail_users | +--+ In voicemail_users I have an entry for 100101, and in extensions_table I have an extension 520, priority 1 to playback tt-monkeys. Asterisk fails to acknowlege the existence of either. sip_users is blank, and cdr holds the (working) CDR information. In /usr/local/etc/odbc.ini I have: [MySQL-asterisk]] Description = MySQL ODBC Driver Testing Driver = MySQL #Socket = /var/run/mysqld/mysqld.sock Server = 10.10.15.30 User= voip Password= temp123 Database= voip Option = 3 #Port = and odbcinst.ini: [MySQL] Description = MySQL ODBC MyODBC Driver Driver = /usr/lib/libmyodbc3.so FileUsage = 1 UsageCount = 2 If I've missed some relevant part of the configuration, let me know, but I think I got all of it. I'm pretty mistified at the moment, after a few hours of working on it. Oh yes. I also tried a realtime update mailbox 100101 password 1357 from the * CLI, but it errored out. It suggested to check the debug log, but the debug log shows absolutely nothing about Realtime. I've loaded and unloaded app_realtime.so, to no effect. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Trouble with Realtime
Greetings, I'm having some trouble with the realtime engines. When asterisk loads, everything looks fine, there don't seem to be any problems via notices or anything. Furthermore, cdr_odbc is working, and actively logging my failed call attempts to db through ODBC using the same DSN. unixODBC and the mysql drivers are installed from source. Here are the relevant parts of the config: Extconfig.conf (Under the [settings] section) sipusers = odbc,voip,sip_users sippeers = odbc,voip,sip_users voicemail = odbc,voip,voicemail_users extensions = odbc,voip,extensions_table res_odbc.conf [asterisk] dsn = MySQL-asterisk username = voip password = temp123 pre-connect = yes Under extensions.conf, in the [from-sip] context: switch = Realtime/[EMAIL PROTECTED] Running isql MySQL-asterisk voip pass connects to the DB, and queries return the proper data. I have the following tables in the mysql databases: +--+ | Tables_in_voip | +--+ | cdr | | extensions_table | | sip_users| | voicemail_users | +--+ In voicemail_users I have an entry for 100101, and in extensions_table I have an extension 520, priority 1 to playback tt-monkeys. Asterisk fails to acknowlege the existence of either. sip_users is blank, and cdr holds the (working) CDR information. In /usr/local/etc/odbc.ini I have: [MySQL-asterisk]] Description = MySQL ODBC Driver Testing Driver = MySQL #Socket = /var/run/mysqld/mysqld.sock Server = 10.10.15.30 User= voip Password= temp123 Database= voip Option = 3 #Port = and odbcinst.ini: [MySQL] Description = MySQL ODBC MyODBC Driver Driver = /usr/lib/libmyodbc3.so FileUsage = 1 UsageCount = 2 If I've missed some relevant part of the configuration, let me know, but I think I got all of it. I'm pretty mistified at the moment, after a few hours of working on it. Thanks, Nathan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Trouble with Realtime
On Fri, 11 Mar 2005 15:04:03 -0600, Matthew Boehm [EMAIL PROTECTED] wrote: You can't have this: [from-sip] switch = Realtime/[EMAIL PROTECTED] The context in your extensions.conf must be different from your Realtime context. Okay, I'll try it, but that doesn't explain why voicemail doesn't work. The extension to access the voicemail is static in extensions.conf. -Matthew Nathan Bowyer wrote: Greetings, I'm having some trouble with the realtime engines. When asterisk loads, everything looks fine, there don't seem to be any problems via notices or anything. Furthermore, cdr_odbc is working, and actively logging my failed call attempts to db through ODBC using the same DSN. unixODBC and the mysql drivers are installed from source. Here are the relevant parts of the config: Extconfig.conf (Under the [settings] section) sipusers = odbc,voip,sip_users sippeers = odbc,voip,sip_users voicemail = odbc,voip,voicemail_users extensions = odbc,voip,extensions_table res_odbc.conf [asterisk] dsn = MySQL-asterisk username = voip password = temp123 pre-connect = yes Under extensions.conf, in the [from-sip] context: switch = Realtime/[EMAIL PROTECTED] Running isql MySQL-asterisk voip pass connects to the DB, and queries return the proper data. I have the following tables in the mysql databases: +--+ Tables_in_voip | +--+ cdr | extensions_table | sip_users| voicemail_users | +--+ In voicemail_users I have an entry for 100101, and in extensions_table I have an extension 520, priority 1 to playback tt-monkeys. Asterisk fails to acknowlege the existence of either. sip_users is blank, and cdr holds the (working) CDR information. In /usr/local/etc/odbc.ini I have: [MySQL-asterisk]] Description = MySQL ODBC Driver Testing Driver = MySQL #Socket = /var/run/mysqld/mysqld.sock Server = 10.10.15.30 User= voip Password= temp123 Database= voip Option = 3 #Port = and odbcinst.ini: [MySQL] Description = MySQL ODBC MyODBC Driver Driver = /usr/lib/libmyodbc3.so FileUsage = 1 UsageCount = 2 If I've missed some relevant part of the configuration, let me know, but I think I got all of it. I'm pretty mistified at the moment, after a few hours of working on it. Thanks, Nathan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Advanced Agents - Need a nice web interface
On Wed, 19 Jan 2005 12:56:59 -0500, Paul Rodan [EMAIL PROTECTED] wrote: [snip] 3. Create historical report to pull agent activity. Should display login/logout activity. Be able to pull information by rep and timeframe. This could probably be done with the CDRs and queue_log. 4. Create hold calls/bypass statuses for agent login. This status should allow the rep to pause all incoming calls to their login for reasons such as: 1-Break, 2-Lunch, 3-Meeting, 4-Project, 5-Other. This status should not log the agent out of the phone, but only temporarily take them out of the queue to receive the next available call until they end the hold/bypass status and make themselves available for incoming calls. There was a patch in the bug tracker (bugs.digium.com) a week or so ago about pausing agents. It would temporarily stop calls coming to their station, but not log them out, as I recall. I'm thinking no, but I figured I'd ask anyways before telling my bosses they're out of their minds. Even if there's an existing interface out there that can provide 1 or 2 of these things, it'd be a nice start. Most of it I'd have to work with a developer to get created, and I'm thinking option 4 is impossible, but 1 2 and 3 is possible with time. Help? Everything is possible with time :) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Compiling zaptel 1.0.2 on Fedora Core
On Mon, 29 Nov 2004 17:08:23 -0800, Brian Wright [EMAIL PROTECTED] wrote: I'm trying to get zaptel 1.0.2 compiled on FC2 or FC3 and I'm getting compile time errors. Systems include: FC2: Linux 2.6.9-1.3_FC2 #1 Mon Nov 15 14:46:43 EST 2004 i686 i686 i386 GNU/Linux FC3: Linux 2.6.9-1.681_FC3 #1 Thu Nov 18 15:13:22 EST 2004 x86_64 x86_64 x86_64 GNU/Linux /usr/home/bwright/zaptel-1.0.2 make linux26 ... make -C /usr/src/linux-2.6 SUBDIRS=/usr/home/bwright/zaptel-1.0.2 modules make[1]: Entering directory `/usr/src/linux-2.6.9' Makefile:461: .config: No such file or directory CC [M] /usr/home/bwright/zaptel-1.0.2/zaptel.o In file included from /usr/home/bwright/zaptel-1.0.2/zconfig.h:9, from /usr/home/bwright/zaptel-1.0.2/zaptel.c:40: include/linux/config.h:4:28: linux/autoconf.h: No such file or directory In file included from /usr/home/bwright/zaptel-1.0.2/zaptel.c:40: /usr/home/bwright/zaptel-1.0.2/zconfig.h:10:27: linux/version.h: No such file or directory [snip] What appears to be happening is that the #include statement is trying to include linux/*.h from both /usr/src/linux-2.6/include/linux and /usr/include/linux. Apparently, this conflicting include directory namespace doesn't work when compiling kernel drivers on Fedora Core. So, it appears to be ignoring /usr/include/linux entirely. The compile, thus, fails finding autoconf.h, version.h and possibly other header files located in /usr/include/linux which shouldn't be in /usr/src/linux-2.6/include/linux. The other thing I've noticed is the following: more /usr/include/linux/autoconf.h #error Invalid kernel header included in userspace more /usr/include/linux/config.h #ifndef _LINUX_CONFIG_H #define _LINUX_CONFIG_H #ifdef __KERNEL__ #error Incorrectly using glibc headers for a kernel module #endif #endif Clearly, if creating a kernel driver on FC2 or FC3, they don't really want you including these header files anyway and will error out anyway if you do manage to include them. So, I'd have to say that the zaptel drivers are going to need a bit of work to compile properly under FC2 or FC3 anyway. This might also explain why the compiler is ignoring /usr/include/linux. So, is there any way around this problem? I've already had my hand at editing the Makefile, but I can't seem to make it do anything different than this. If anyone has any ideas, please let me know. I'd like to get this driver compiled. Did you make the symlink from /usr/src/linux-2.6 to /lib/modules/`uname -r`/build/? As stated in README.Linux26, it says that creating a kernel module no longer requires the full kernel source to compile against. Do that and it should work. Its worked for me on both FC2 and FC3. Nathan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Phone Selection
On Sun, 28 Nov 2004 16:32:10 -0600, Rich Adamson [EMAIL PROTECTED] wrote: I'm looking at the Sayson 480i or the Cisco CP-7960. Which one would you suggest and why please ? I briefly tested the 480i a couple of weeks ago. Had a problem in that it would not use the tftp server address contained in the dhcp response, so had to define everything from the keypad to make it work. The quality of the audio was good, the speakerphone function worked, and all other very basic phone functions that I tried (not an expensive test at all) worked as expected. There seemed to be a lot of this function will be implemented in a later software release kind of thing going on. I did not write down the s/w version that it was running, but I do remember there were two additional releases available after the one I had. I would not deploy this phone in large quantities at this time as they would be a support nightmare. For small quantities, not a bad phone at all. That's about all I can tell you on it. I use a 7960 for day to day business use and like it very well. It feels like a phone, works like a phone, excellent speakerphone, and continues to function well. Probably a little over priced these days. I'll stay with it for now. You should probably dig through the wiki as I'm sure there is more detail there on lots of different phones. Rich Someone once told me that he would never consider using a SIP phone unless it had been through several software releases / revisions. In my experience, this kind of thinking seems to work well. For example, this 480i is relatively new to the market, having only been out less than a year. Bugs are probably still being worked out, features still to be implemented, and with those features, more bugs. I'd recommend the Cisco 7940/7960 series phones. Another phone I've been impressed with so far, although haven't tested extensively is the Polycom Soundpoint IP500. It seems to be a solid phone, with a feature set that gives Cisco a run for its money. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DTMF and Access Codes
Hello, I have a problem which I've found quite strange, to say the least. I have a client who uses long distance access codes from their LD provider. The codes are 4-digits, nothing extraordinary there. The problem is, if you dial the digits quickly, without pauses inbetween them, the LD company does not recognize those digits. If you dial the code slowly, everything works. Phones I'm using are Cisco 7960G phones, * is connected by PRI to the PSTN. 7960Gs are on SIP v6.3 If I set the phones to use inband DTMF, and Asterisk to use rfc2833, the LD codes work no matter how fast or slow I key them in. It Just Works. If I set both Asterisk and the Phone to either inband or rfc2833, fast digit dialing breaks the LD codes. Anyone ever see anything like this before, or know of any way to fix it? Thanks, Nathan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Strategy in App_queue
On Mon, 08 Nov 2004 15:05:33 +0800, el Flynn [EMAIL PROTECTED] wrote: Nathan Bowyer wrote: Doesn't seem to work for me that way. Anyone else got any ideas? When I look at the code, it looks like copying what roundrobin does, then simply removing the pos whenever you complete a call (or one abandons) would reset the queue back to its original state. I can't seem to accomplish this, though. What about assigning penalties to the agents? The agent to call first would have the lowest penalty, increasing as you add agents to the list. Flynn While it does put them in the correct order this way, it seems to have a hard time progressing in penalties. Phone one will ring many many times, and if no one answers it will simply keep ringing. I suppose I could play with the metrics and penalties, making the second ring place the second phone as the lowest metric (the phone to be called). I'll have to check that out. Is anyone interested in something like this, or is this a change I should just keep to myself? :) Thanks, Nathan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Strategy in App_queue
Doesn't seem to work for me that way. Anyone else got any ideas? When I look at the code, it looks like copying what roundrobin does, then simply removing the pos whenever you complete a call (or one abandons) would reset the queue back to its original state. I can't seem to accomplish this, though. Nathan On Wed, 27 Oct 2004 21:48:37 -0400, Robert Jackson [EMAIL PROTECTED] wrote: If you have a group of agents as the only member of the queue like so: member = Agent/@1 And specify the agents in agents.conf in the order you want like so: agent = 1,1234,Test1 agent = 2,1234,Test2 agent = 3,1234,Test3 The agents will be called in 1,2,3 order regardless of the strategy that you specify. This has been my experience. I am not sure if it was designed this way on purpose, but it seems to work this way for me nonetheless. Good luck, Robert Jackson -Original Message- From: Nathan Bowyer [mailto:[EMAIL PROTECTED] Sent: Wednesday, October 27, 2004 9:40 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] New Strategy in App_queue Hello, I've been looking at and working on a new queue strategy for about a week now, off and on. However, being that I'm not really a C programmer (yet, anyway) I have not made much progress. The concept is rather simple, probably the easiest of all the queue strategies. I simply want to it to ring the interfaces / agents in the order they are listed in queues.conf, and start over at the beginning when a new call comes in. If you were doing this with Dial() and in the dial plan, it would look something like this: exten = s,1,Dial(SIP/phone1,35) exten = s,2,Dial(SIP/phone2,30) exten = s,3,Dial(SIP/phone3,35) There is a very real reason for wanting to do this with app_queue rather than Dial, but its rather outside the scope of this message, I think. In any case, if any of you have interest in seeing something like this happen, or have any pointers or tips that would aid in this endeavour, please let me know. Thanks, Nathan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/as terisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] New Strategy in App_queue
Hello, I've been looking at and working on a new queue strategy for about a week now, off and on. However, being that I'm not really a C programmer (yet, anyway) I have not made much progress. The concept is rather simple, probably the easiest of all the queue strategies. I simply want to it to ring the interfaces / agents in the order they are listed in queues.conf, and start over at the beginning when a new call comes in. If you were doing this with Dial() and in the dial plan, it would look something like this: exten = s,1,Dial(SIP/phone1,35) exten = s,2,Dial(SIP/phone2,30) exten = s,3,Dial(SIP/phone3,35) There is a very real reason for wanting to do this with app_queue rather than Dial, but its rather outside the scope of this message, I think. In any case, if any of you have interest in seeing something like this happen, or have any pointers or tips that would aid in this endeavour, please let me know. Thanks, Nathan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
