Re: [asterisk-users] VOIP Provider wooes
Gregory Malsack wrote on 1/4/08 4:48 PM: Does anyone know of a good VOIP dialtone provider in the northern Chicago area. My client has tried Broadvoice and Mix and is having problems with latency in the middle of the traceroute between him and the provider. I use Broadvoice and haven't had any problems with them, and I'm in Southwest Michigan... oddly enough, I use broadvoice's New York node rather than their Chicago node, despite being a closer traceroute to Chicago from here (I go through Chicago on the way to anywhere on the net from here), because I had problems with their Chicago node dropping out. But I get pretty good connections with minimal latency from New York, despite sending the packets right past the Chicago one. :) Strange, but it works for me. -- Dave Miller http://www.justdave.net/ System Administrator, Mozilla Corporation http://www.mozilla.com/ Project Leader, Bugzilla Bug Tracking System http://www.bugzilla.org/ ___ --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] e911
Mike Hammett wrote on 11/20/07 1:27 PM: One of my providers has a different SIP account for each number. I have all of my users in one outbound context (caller ID passes fine). How do I ensure that the callers get routed down their correct SIP account with my provider for e911 purposes without each having their own context? I think the easiest answer is going to be to go ahead and put each in their own context. Note that you can include contexts from each other... so say they're all in [downstream-phones] right now (for example)... you can do something like this: [phones-in-account1] include = downstream-phones exten = 911,s,Goto(DialViaAccount1) [phones-in-account2] include = downstream-phones exten = 911,s,Goto(DialViaAccount2) etc. -- Dave Miller http://www.justdave.net/ System Administrator, Mozilla Corporation http://www.mozilla.com/ Project Leader, Bugzilla Bug Tracking System http://www.bugzilla.org/ ___ --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-addons compilation error: dereferencing pointer to incomplete type
Jeremy Malcolm wrote on 7/13/07 10:06 AM: I am having trouble getting asterisk-addons 1.4.2 to compile (after a successful configure). Asterisk itself (and AsteriskGUI) compile fine. I get: [snipped] Sounds like you're missing the -devel package for MySQL would be my first guess. -- Dave Miller http://www.justdave.net/ System Administrator, Mozilla Corporation http://www.mozilla.com/ Project Leader, Bugzilla Bug Tracking System http://www.bugzilla.org/ ___ --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] Phantom Calls
Lee Jenkins wrote on 6/19/07 9:56 AM: Vadim Berezniker wrote: Enable verbose logging for the asterisk log Set verbose level to 4 Review the log file for anything that looks like a phantom call. There should be enough information to get some idea of why this is happening. Thanks, I have done this yesterday by setting up putty to log to a file, but the customer employees have inadvertently shut it down on a couple of a occasions :) Hopefully it will be running when this happens again so I can try to track down the problem. You should be able to tell it to log to a file in addition to the console in logger.conf. Something like: full = notice,warning,error,verbose Then it should show up in /var/log/asterisk/full and you wouldn't need to keep a session open to the console to see it, just go back and look at the file later. -- Dave Miller http://www.justdave.net/ System Administrator, Mozilla Corporation http://www.mozilla.com/ Project Leader, Bugzilla Bug Tracking System http://www.bugzilla.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
Re: [asterisk-users] reset Polycom phones remotely
Forum wrote on 5/26/07 5:32 PM: I have provisioned a bunch of Polycom 301 phones to get the config files from my ftp server. Out of the 4 phones 2 get the config file however the other 2 cannot contact the boot server. I have reboot the phones a number of times remotely (the client is 400 km away) through vnc and logging onto the web config internally. No matter what I change on the web config page it is not saved. I feel I need to reset or reformat the phones - if so how can I do this remotely? Can anyone think of a reason why these 2 phones cannot contact the boot server when the other 2 can? Have you checked their boot server type, and does it match what you have available? If FTP is all you have set up on the boot server and those two phones are set to use TFTP then you would have this issue. -- Dave Miller http://www.justdave.net/ System Administrator, Mozilla Corporation http://www.mozilla.com/ Project Leader, Bugzilla Bug Tracking System http://www.bugzilla.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
Re: [asterisk-users] reset Polycom phones remotely
Got any rogue DHCP options configured that are set for that MAC address or range? I've noticed when you put in options that the Polycom looks for and have them configured syntactically incorrect, the Polycoms will refuse the entire transaction instead of just the option that was screwed up (timezones with an illegal value in my case). Forum wrote on 5/27/07 12:07 PM: It's definitely ftp. I have given the phone a static ip. When I set it to dhcp it just hangs and cannot get an IP. I can ping the phone and see the web config page so it is on the network. Any more suggestions. Steve -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Miller Sent: Sunday, 27 May 2007 5:56 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] reset Polycom phones remotely Forum wrote on 5/26/07 5:32 PM: I have provisioned a bunch of Polycom 301 phones to get the config files from my ftp server. Out of the 4 phones 2 get the config file however the other 2 cannot contact the boot server. I have reboot the phones a number of times remotely (the client is 400 km away) through vnc and logging onto the web config internally. No matter what I change on the web config page it is not saved. I feel I need to reset or reformat the phones - if so how can I do this remotely? Can anyone think of a reason why these 2 phones cannot contact the boot server when the other 2 can? Have you checked their boot server type, and does it match what you have available? If FTP is all you have set up on the boot server and those two phones are set to use TFTP then you would have this issue. -- Dave Miller http://www.justdave.net/ System Administrator, Mozilla Corporation http://www.mozilla.com/ Project Leader, Bugzilla Bug Tracking System http://www.bugzilla.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
Re: [asterisk-users] Call someone to instantly join conference using MeetMe
Arpit Mehta wrote on 5/19/07 10:18 PM: I was just wondering how would the application be where the Asterisk calls a number and that number joins the conference as soon as the call connects. There would be only one conference already defined in meetme.conf and there is one person already joined the conference. Currently MeetMe requires a person dialing into it and the joining the conference. How could this be done using MeetMe or any other conference application? Any suggestions/hints/links are welcome. Set up an extension that dials directly into the conference in question, then use that extension via the Local channel as the source of a call to the number you want to dial, triggered via the Management API or a call file. [meetme-dialin] exten = 1234,1,Answer() exten = 1234,n,MeetMe(4321) Pipe the following into the Manager API with an extra blank line at the end: Action: Originate Channel: Local/[EMAIL PROTECTED] Context: from-inside (or whatever context is appropriate) Exten: (the number you want to call) Priority: 1 I'm going from memory, so you may have to play with it a little bit but that's the basic idea. -- Dave Miller http://www.justdave.net/ System Administrator, Mozilla Corporation http://www.mozilla.com/ Project Leader, Bugzilla Bug Tracking System http://www.bugzilla.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
Re: [asterisk-users] asterisk telemarketer torture sound files
Adam Jacob Muller wrote on 5/5/07 1:06 PM: Hi, I have some annoying telemarketer calling me on a recurring basis, but I'd like to discourage them a bit and have some fun. I found this: http://www.voip-info.org/wiki/view/Asterisk+AEL+Telemarketer+Torture but the link to download the sound files is dead (wyoming.e-tools.com is NXDOMAIN). Anyone have a copy of these? I believe they're included in Asterisk's extra sounds package now. Look for the sounds with a tt- prefix on the filenames. -- Dave Miller http://www.justdave.net/ System Administrator, Mozilla Corporation http://www.mozilla.com/ Project Leader, Bugzilla Bug Tracking System http://www.bugzilla.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
Re: [asterisk-users] asterisk telemarketer torture sound files
Adam Jacob Muller wrote on 5/5/07 1:38 PM: On May 5, 2007, at 1:15 PM, Dave Miller wrote: Adam Jacob Muller wrote on 5/5/07 1:06 PM: Hi, I have some annoying telemarketer calling me on a recurring basis, but I'd like to discourage them a bit and have some fun. I found this: http://www.voip-info.org/wiki/view/Asterisk+AEL+Telemarketer+Torture but the link to download the sound files is dead (wyoming.e-tools.com is NXDOMAIN). Anyone have a copy of these? I believe they're included in Asterisk's extra sounds package now. Look for the sounds with a tt- prefix on the filenames. Unfortunately, this doesn't seem to be the case :/ -=[~/asterisk-extra-sounds-en-gsm-current]=- -=[Sat May 05]=- -=[13:32:42]=- [EMAIL PROTECTED] ls -l tt-* ls: tt-*: No such file or directory checked through the files as well, i don't see them here or in the core sounds, though there are a few tt-* files in the core package tt-allbusy.gsm tt-monkeys.gsm tt-monkeysintro.gsm tt-somethingwrong.gsm tt-weasels.gsm Ah, those are the ones I was thinking of. For some reason I didn't think those would be in core :) -- Dave Miller http://www.justdave.net/ System Administrator, Mozilla Corporation http://www.mozilla.com/ Project Leader, Bugzilla Bug Tracking System http://www.bugzilla.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
Re: [asterisk-users] Calls in ulaw, not gsm as desired
G.711µ is another name for ulaw, G.711A is another name for alaw. Rob Schall wrote on 5/2/07 5:30 PM: I see this. However, ulaw works fine. I believe alaw works as well. So I don't think their site is either correct, or they didn't mean it like that. Rob Jerry Jones wrote: A simple glance at their website will tell you this about the 501 G.711 μ/A and G.729A (Annex B) configuration On May 2, 2007, at 12:22 PM, Jaswinder Singh wrote: Try ilbc if the phone supports (free) or g729 ( better but your asterisk will need license if you want to transcode calls from g729 to other codecs or want to record calls ) . Also check your phones config if its support multiple codecs . . On 02/05/07, Rob Schall [EMAIL PROTECTED] wrote: So I reloaded things and had just gsm set for 2 of my polycom 501 phones. However, the logs say No codec found, which leads me to believe that polycom 501 phones can't use gsm. Does anyone have something like this working? If not gsm, is there a better option with these phones over a low bandwidth situation? ___ --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 -- Dave Miller http://www.justdave.net/ System Administrator, Mozilla Corporation http://www.mozilla.com/ Project Leader, Bugzilla Bug Tracking System http://www.bugzilla.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
Re: [asterisk-users] TDM400P and TE405P
Nitesh Divecha wrote on 5/1/07 10:28 AM: Is it possible to have both Digium cards installed on one Server (TDM400P and TE405P)? I have one site which requires both connection POT and T1/E1. How can I configure both cards? Should work just fine. The Zaptel drivers will pick up both. Just be forewarned, the T1/E1 channels will all get numbered before the POTS channels, no matter what order they're on the bus, so 1-24 will be your T1 and 25-28 the POTS, for example. (I think E1 goes to 32?) -- Dave Miller http://www.justdave.net/ System Administrator, Mozilla Corporation http://www.mozilla.com/ Project Leader, Bugzilla Bug Tracking System http://www.bugzilla.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
Re: [asterisk-users] Music on Hold issue with asterisk 1.4.2
Steve Finkelstein wrote on 4/28/07 12:21 AM: my musiconhold.conf: [default] mode=quietmp3 directory=/var/lib/asterisk/mohmp3 and finally in my extensions.conf: asterisk-1.4.2 # grep 100 /etc/asterisk/extensions.conf exten = 100,1,MusicOnHold(30) exten = 100,2,Hangup When I dial 100 however, I receive the following: [Apr 28 00:17:33] WARNING[30975]: res_musiconhold.c:947 local_ast_moh_start: No class: 30 The parameter to MusicOnHold is the class of music to play. You have no class named 30 just like the error says. :) You do have a class named default in the config snippet you pasted, so MusicOnHold(default) should work. -- Dave Miller http://www.justdave.net/ System Administrator, Mozilla Corporation http://www.mozilla.com/ Project Leader, Bugzilla Bug Tracking System http://www.bugzilla.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
Re: [asterisk-users] Random Asterisk deaths
Wayne Jensen wrote on 4/24/07 7:24 PM: Every once in a while for no apparent reason, Asterisk has been dying on me, dropping all calls in progress. There's nothing in the log file or on the Asterisk console that indicates the reason. Some days it doesn't happen at all. Other days it happens two or three times. The problem began on Friday, but the last time anything was changed on that box was at least a week before that. Any suggestions on what to do/where to look to find out what's going on and fix the problem? There was a security update for Asterisk released yesterday which addresses a denial-of-service class vulnerability in which malformed SIP packets could cause asterisk to crash. Our server also randomly died once on Monday morning with no apparent cause. It's possible someone was exploiting this. I upgraded ours last night. -- Dave Miller http://www.justdave.net/ System Administrator, Mozilla Corporation http://www.mozilla.com/ Project Leader, Bugzilla Bug Tracking System http://www.bugzilla.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] MeetMe + IAX2 + Asterisk 1.2.18 = calls dropped
We upgraded our asterisk server to 1.2.18 last night to pick up the security update. Since then, any calls coming in on IAX2 links get dropped if they try to enter a MeetMe conference room. The log shows this: Apr 26 08:33:16 NOTICE[27362]: chan_iax2.c:3167 iax2_read: I should never be called! Hanging up. I've temporarily worked around it by switching our inbound provider to use SIP instead of IAX, but that's not an ideal solution. -- Dave Miller http://www.justdave.net/ System Administrator, Mozilla Corporation http://www.mozilla.com/ Project Leader, Bugzilla Bug Tracking System http://www.bugzilla.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
Re: [asterisk-users] MeetMe + IAX2 + Asterisk 1.2.18 = calls dropped
Dave Miller wrote on 4/26/07 11:46 AM: We upgraded our asterisk server to 1.2.18 last night to pick up the security update. Since then, any calls coming in on IAX2 links get dropped if they try to enter a MeetMe conference room. The log shows this: Apr 26 08:33:16 NOTICE[27362]: chan_iax2.c:3167 iax2_read: I should never be called! Hanging up. I've temporarily worked around it by switching our inbound provider to use SIP instead of IAX, but that's not an ideal solution. Quick turnaround on the bug tracker, bug is resolved fixed already :) http://bugs.digium.com/view.php?id=9600 guess that'll be fixed in the next release. -- Dave Miller http://www.justdave.net/ System Administrator, Mozilla Corporation http://www.mozilla.com/ Project Leader, Bugzilla Bug Tracking System http://www.bugzilla.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
Re: [asterisk-users] Two devices registrating same extension
Frederico Madeira wrote on 4/26/07 12:18 PM: Hi guys, Is it possible to asterisk manage multiple devices registration with same extension ? When asterisk receive a call for that extension, it send call to all devices registered with that extension, and rtp go to first one that answer the call. Sure. They need separate userids in your sip/iax/zap/whatever.conf so asterisk knows that they're two separate devices. Then in your dialplan for that extension just tell it to ring both (with an between them). Dial(SIP/device1SIP/device2,15) -- Dave Miller http://www.justdave.net/ System Administrator, Mozilla Corporation http://www.mozilla.com/ Project Leader, Bugzilla Bug Tracking System http://www.bugzilla.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
Re: [asterisk-users] Re: MeetMe + IAX2 + Asterisk 1.2.18 = calls dropped
Tony Mountifield wrote on 4/26/07 12:26 PM: In article [EMAIL PROTECTED], Dave Miller [EMAIL PROTECTED] wrote: We upgraded our asterisk server to 1.2.18 last night to pick up the security update. Since then, any calls coming in on IAX2 links get dropped if they try to enter a MeetMe conference room. The log shows this: Apr 26 08:33:16 NOTICE[27362]: chan_iax2.c:3167 iax2_read: I should never be called! Hanging up. I've temporarily worked around it by switching our inbound provider to use SIP instead of IAX, but that's not an ideal solution. What was the last version that successfully worked for you? 1.2.17. But the problem has been found and fixed (see my other post) -- Dave Miller http://www.justdave.net/ System Administrator, Mozilla Corporation http://www.mozilla.com/ Project Leader, Bugzilla Bug Tracking System http://www.bugzilla.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
Re: [asterisk-users] Two devices registrating same extension
Philipp Kempgen wrote on 4/26/07 12:53 PM: Frederico Madeira wrote: Is it possible to asterisk manage multiple devices registration with same extension ? When asterisk receive a call for that extension, it send call to all devices registered with that extension, and rtp go to first one that answer the call. Not really but I think what you are looking for is just something like exten = 10,1,Dial(SIP/15SIP/16SIP/17) Or you could put them in a queue with strategy=ringall. Might be worth pointing out that the sip device name is not an extension number. A lot of folks (particularly the web-based config editors) seem to do that by default, and it confuses people into thinking the device names are the extension number. You could just as easily make them SIP/george or SIP/harry. We started out with one of those web-based systems that named them all with the extension number, so when we started needing additional devices for the same extension (like softphones) we just started tacking suffixes on them. SIP/204 and SIP/204soft for example. -- Dave Miller http://www.justdave.net/ System Administrator, Mozilla Corporation http://www.mozilla.com/ Project Leader, Bugzilla Bug Tracking System http://www.bugzilla.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
Re: [asterisk-users] Polycom IP 501 is displaying wrong time
Crazy Boy wrote on 4/19/07 11:41 PM: Thank you for your response. As you said, I set it for -5. But, its displaying wrong time. I don't enter any SNTP Server. Is it must? How can I solve this problem? Can you tell me? Yeah, there's no way to set the clock except by using an NTP server, so you need to set one. -- Dave Miller http://www.justdave.net/ System Administrator, Mozilla Corporation http://www.mozilla.com/ Project Leader, Bugzilla Bug Tracking System http://www.bugzilla.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
Re: [asterisk-users] Polycom IP 501 is displaying wrong time
Chris Mason (Lists) wrote on 4/19/07 6:10 AM: If your phone is getting its parameters by DHCP from a linux server, add the NTP server option to that server: in /etc/dhcpd.conf option time-servers 192.168.0.3; If your phone is getting an NTP server setting by DHCP server, you can't override that from any setting. I came across this where a polycome 501 was connected to the internet directly and comcast was setting NTP to 10.10.x.x, which was ridiculous. Their tech support could never understand why this was a problem and would not address the problem despite repeated calls. Also of note is that the time zone can also be set via DHCP, and if it is, that can't be overridden in the phone, either. -- Dave Miller http://www.justdave.net/ System Administrator, Mozilla Corporation http://www.mozilla.com/ Project Leader, Bugzilla Bug Tracking System http://www.bugzilla.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
Re: [asterisk-users] Meetme cant handle more than 5 users in a call?? hmmmm
Dean Collins wrote on 3/28/07 9:27 AM: Meetme cant handle more than 5 users in a call?? H Heh, that's a laugh. We regularly get 40 or more callers in a conference room in MeetMe with no problems. In fact, the call quality is better than some of those 800# conference services we used to use before we had Asterisk. :) The story is likely what hardware you have it running on. If you expect your phone system to be an enterprise-class PBX, it needs to run on enterprise-class hardware, not some leftover 486 box from the back closet. -- Dave Miller http://www.justdave.net/ System Administrator, Mozilla Corporation http://www.mozilla.com/ Project Leader, Bugzilla Bug Tracking System http://www.bugzilla.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
Re: [asterisk-users] Polycom Questions
Dave Fullerton wrote on 3/6/07 9:33 AM: Polycom's 2.1.0 firmware has the new DST settings as the default. This is what they use for the SNTP element: SNTP tcpIpApp.sntp.resyncPeriod=86400 tcpIpApp.sntp.address= tcpIpApp.sntp.address.overrideDHCP=0 tcpIpApp.sntp.gmtOffset= tcpIpApp.sntp.gmtOffset.overrideDHCP=0 tcpIpApp.sntp.daylightSavings.enable=1 tcpIpApp.sntp.daylightSavings.fixedDayEnable=0 tcpIpApp.sntp.daylightSavings.start.month=3 tcpIpApp.sntp.daylightSavings.start.date=8 tcpIpApp.sntp.daylightSavings.start.time=2 tcpIpApp.sntp.daylightSavings.start.dayOfWeek=1 tcpIpApp.sntp.daylightSavings.start.dayOfWeek.lastInMonth=0 tcpIpApp.sntp.daylightSavings.stop.month=11 tcpIpApp.sntp.daylightSavings.stop.date=1 tcpIpApp.sntp.daylightSavings.stop.time=2 tcpIpApp.sntp.daylightSavings.stop.dayOfWeek=1 tcpIpApp.sntp.daylightSavings.stop.dayOfWeek.lastInMonth=0/ The one thing I'm not sure about is the tcpIpApp.sntp.daylightSavings.start.date=8 line. According to the 2.1.0 admin guide that means the second week of the month but none of the guides before that mention this as a valid option. Thanks! One question I have... with this applied (and even with the original config I had before changing it to this), the start.dayOfWeek setting shows up as Monday on the web interface on the phone. Is the web interface goofed up, or should that be Sunday? -- Dave Miller http://www.justdave.net/ System Administrator, Mozilla Corporation http://www.mozilla.com/ Project Leader, Bugzilla Bug Tracking System http://www.bugzilla.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
