[asterisk-users] rfc3680, reginfo+xml
Hi, RFC3680 defines a SIP event package for registration. This event package which can be used through NOTIFY-SUBSCRIBE methods, seems very useful for free sitting or presence applications. This package is supported in various SIP phones (at least Thomson ST2030) : when turned on, this feature adds a new login/logout menu among other things. It can also be used to send Welcome notices to mobile users : whenever a mobile user comes in, a SIP MESSAGE is sent by a software application which has previously subscribed to be notified of any registration event related to this mobile user. It appears Asterisk supports SIP NOTIFY-SUBSCRIBE methods. But I couldn't find any trace of this specific Registration Event package support (but I won't swear I searched the right way). How can I make sure this feature is supported or not ? More precisely, this Registration Event package support relies on these headers : SIP SUBSCRIBE reg Event SIP SUBSCRIBE application/reginfo+xml Accept SIP NOTIFY reg Event SIP NOTIFY application/reginfo+xml Content How shall I check ? Regards ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to re-read values from database in Trixbox
Hello guys, I'm using Trixbox and I have a PHP application that updates a value in the MySQL asterisk database as an interface to have a dynamic customizable IVR. After execute the UPDATE SQL query, the php application is supossed to reload asterisk or restart amportal in order to get the change working, but nor asterisk -rx reload nor amportal restart got the change working. So, the question is how can I re-read the new value from the database to be effective in asterisk? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to re-read values from database in Trixbox
You are updating the MySQL config, which is not propagated to the Asterisk config files. Only after you regenerate the configuratios, you can reload asterisk. Dirty hack: need_reload flag must be set to true. Real solution: retrieve_conf + asterisk reload On Wednesday 22 August 2007 10:22, Edgar Guadamuz wrote: Hello guys, I'm using Trixbox and I have a PHP application that updates a value in the MySQL asterisk database as an interface to have a dynamic customizable IVR. After execute the UPDATE SQL query, the php application is supossed to reload asterisk or restart amportal in order to get the change working, but nor asterisk -rx reload nor amportal restart got the change working. So, the question is how can I re-read the new value from the database to be effective in asterisk? ___ --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] Saftware RAID1 or Hardware RAID1 with Asterisk
Steve Totaro wrote: I guess I am just lucky to have 24 hour manned data centers with staff that walk around looking for flashing LEDs. I am sure there is some error thrown in /var/log/messages about a failure that could be used to trigger a notification quite trivially. Both smartd and mdadm can be configured to send emails. Regards, Richard ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Which interface?
Hi: If any body use meetmemanager or conman or web-meetme please say how about is it.I'd appreciated any idea. Regards. - Be a better Heartthrob. Get better relationship answers from someone who knows. Yahoo! Answers - Check it out. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How do I configure asterisk?
Hi: Which one is better and easier for configure asterisk,directly or by GUI ? I'd appreciate any idea. Regards. - Building a website is a piece of cake. Yahoo! Small Business gives you all the tools to get online.___ --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] rfc3680, reginfo+xml
Olivier, This feature is not supported in Asterisk. I can tell this looking at the code. If you want to test this yourself, send Asterisk a SUBSCRIBE message with Event: reg header in it. You can either use an off-the-shelf UA that supports RFC 3680 to do this or you can use SIPp (an open-source SIP test tool) to do this. Since Asterisk does not support reg event-package, it'll respond back with a 489 (Bad Event) response. Raj On 8/22/07, Olivier [EMAIL PROTECTED] wrote: Hi, RFC3680 defines a SIP event package for registration. This event package which can be used through NOTIFY-SUBSCRIBE methods, seems very useful for free sitting or presence applications. This package is supported in various SIP phones (at least Thomson ST2030) : when turned on, this feature adds a new login/logout menu among other things. It can also be used to send Welcome notices to mobile users : whenever a mobile user comes in, a SIP MESSAGE is sent by a software application which has previously subscribed to be notified of any registration event related to this mobile user. It appears Asterisk supports SIP NOTIFY-SUBSCRIBE methods. But I couldn't find any trace of this specific Registration Event package support (but I won't swear I searched the right way). How can I make sure this feature is supported or not ? More precisely, this Registration Event package support relies on these headers : SIP SUBSCRIBE reg Event SIP SUBSCRIBE application/reginfo+xml Accept SIP NOTIFY reg Event SIP NOTIFY application/reginfo+xml Content How shall I check ? Regards ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco firmwares 3.6.3 vs 3.8.6
Hi All, A question for those with Cisco 7940/60 SIP phones. I used to load POS3-06-03-00 Firmware to the cisco phones. A month or so ago, I ran some tests and found that latest 3.8.6 firmware worked well, and solved an issue or two on the phones. I've a number of users who work outside of the LAN. Our phones use DNS names to talk to A*k, so in theory, just enabling NAT makes the phone work outside the LAN (home users, remote users, etc). However, when we loaded the 3.8.6 firmware to these phones, we've found the phones no longer work outside of the LAN. Using Etherreal, we've found that the communication between the Phone and A*k breaks (A*k never sees the Register packets, but the phone does seem to send them. I'll post more detail if its needed, but I wondered if anyone else has seen this ? The size of the IP packet for register is different (larger on the 3.8.6), but the important content of the Register message seems the same. I've ruled out ISP/firewall interference, as its happened on a number of users. Obviously there are fixes in 3.8.6, so I don't want to downgrade the phones again, but I can't see why they'd fail... Adrian Marsh ___ --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] compatibility of PRI Two B channel transfers TBTC/2BTC
On 8/21/07, Matthew Fredrickson [EMAIL PROTECTED] wrote: Matt Florell wrote: Hello, A client has asked for Two B channel Transfer capability (known as TBCT or 2BCT, similar to other features such as ECT, RTL and Q,SIG Path Replacement) in a new Asterisk system and so I researched the capability and came up with quite a few gaps in documentation. From what I've gathered, the official Digium statement is that is works with DMS100 only, and only in Asterisk 1.4.X : http://kb.digium.com/entry/26/140/ This definitely works. I wrote it and tested it myself. Although in a bugtracker posting with a patch from over two years ago, Matt Fredrickson from Digium says that it works with 5ESS under Asterisk 1.2.X: http://bugs.digium.com/view.php?id=3554 There's an implementation I scrubbed out a couple of years ago, but I think there was a bug in it that I was not able to fix. When push came to shove, and I needed a switch to debug it on (and when I had more time to work on it), nobody offered switch access so that I could debug it. So I don't think it is working right now. There are also bounties and claims of this feature working on NI2 protocol(although no patches posted) on the voip-info.org Wiki: http://www.voip-info.org/wiki/view/Asterisk+bounty+PRI+2B+channel+transfer+for+NI2+PRI+line http://www.voip-info.org/wiki/index.php?page=Asterisk%20bounty%20PRI%202B%20channel%20transfer Yeah, well, they're really old :-) Try getting a hold of the authors. I am trying to, I have sent a message to whitehawk82 on the digium forums and hopefully he will post back to me. If anyone knows who that actually is, I would like to get a hold of them, Please email me their contact details. Thanks for clearing all of this up Matt, Hopefully I'll be able to fix the notes out there to give a better picture of all of this once I'm done with this project. Thanks, MATT--- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DUNDi, So Easy A Caveman Could Do It!
Well done! It's top-news on AstPligg right now. http://oinko.net/astpligg/story.php?title=DUNDi_So_Easy_A_Caveman_Could_Do_It Thanks l. On Wed, 22 Aug 2007 03:51:51 +0200, JR Richardson [EMAIL PROTECTED] wrote: Here you go folks: ftp://ftp.ntcp.net/DUNDi_So_Easy.pdf If someone would be so kind as to upload to the wiki, it will be much appriciated. Thank you all who replied to my poll questions. As always, I hope this help. JR -- Loway Research - Home of QueueMetrics http://queuemetrics.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco firmwares 3.6.3 vs 3.8.6
FYI about cisco firmware: http://www.cisco.com/warp/public/707/cisco-sr-20070821-sip.shtml A. On Wed, 2007-08-22 at 12:26 +0100, Adrian Marsh wrote: Hi All, A question for those with Cisco 7940/60 SIP phones. I used to load POS3-06-03-00 Firmware to the cisco phones. A month or so ago, I ran some tests and found that latest 3.8.6 firmware worked well, and solved an issue or two on the phones. I've a number of users who work outside of the LAN. Our phones use DNS names to talk to A*k, so in theory, just enabling NAT makes the phone work outside the LAN (home users, remote users, etc). However, when we loaded the 3.8.6 firmware to these phones, we've found the phones no longer work outside of the LAN. Using Etherreal, we've found that the communication between the Phone and A*k breaks (A*k never sees the Register packets, but the phone does seem to send them. I'll post more detail if its needed, but I wondered if anyone else has seen this ? The size of the IP packet for register is different (larger on the 3.8.6), but the important content of the Register message seems the same. I've ruled out ISP/firewall interference, as its happened on a number of users. Obviously there are fixes in 3.8.6, so I don't want to downgrade the phones again, but I can't see why they'd fail... Adrian Marsh ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] rfc3680, reginfo+xml
Thanks for replying, Raj. Do you think such feature should, ideally, be implemented in Asterisk should it be implemented in a dedicated software (presence ?) ? It seems to me it should, though I'm not aware of many devices using this feature, beside SIP hardphones. Would it be difficult to extend current code to comply with this RFC, when rfc3265 mechanism is already in place ? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk with FAX problem
Dear all I have setup of asterisk 1.2.14 and this is working fine. first i want to explain you my setup of asterisk on network i have connect my asterisk with mediant 2000 gateway and PRI terminated on mediant. [fax_machin]--[audio_code_fxs]-[Asterisk]---[mediant_2000]---PRI my fax machine connected with audiocode 24 fxs extention and which is connected with asterisk and asterisk connected with mediant 2000 now i am not able to send FAX outside my company so is there any special configuration for T.38 protocal ?? can anyone explain me how do i go ahead with this setup to start FAX - Pinpoint customers who are looking for what you sell. ___ --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] Polycom and NAT
I have both of those command lines for my natted sip device. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Darryl Dunkin Sent: Wednesday, 22 August 2007 1:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom and NAT In your sip.conf, for the user: nat=yes To send keepalives for the UDP connection (depending on how flimsy the device handling NAT is): qualify=yes From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Klaverstyn, David C Sent: Tuesday, August 21, 2007 17:51 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Polycom and NAT Hi All, I have a Polycom 501 that is behind a NAT. When it registers to the Asterisk server it is using the IP address on the private network and not the public IP of the NAT address. Can someone tell me what I need to do so the phone registerers using an internet address rather than the remote network NAT address. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How do I configure asterisk?
On 8/22/07, fateme fatah [EMAIL PROTECTED] wrote: Hi: Which one is better and easier for configure asterisk,directly or by GUI ? I'd appreciate any idea. Regards. It's up to you to decide what's easier for you and your needs. For beginners GUI is ok, but if you need some fancy functionality, you will need to code config files for yourself. This question doesn't have definite answer - some people prefer GUI management of their servers, and thus choose MS IIS, and so on, but some prefer plain config files (and choose Linux). As a programmer i prefer config files (and that's more nerdy). Regards, Atis -- Atis Lezdins, IT Responsible of BEST Riga, [EMAIL PROTECTED] ICQ: 142239285 Skype: atis.lezdins Cell Phone: +371 28806004 [Tele2, Latvia] Work phone: +1 800 7502835 [Toll free, USA] ?BEST? - www.BEST.eu.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] Polycom behind NAT won't register to * server behind ALG
Henry L.Coleman wrote: I think what Alex was trying to say was that Polycom IP Phones are an example of immature product development. While they look very nice and have a nice display the product doesn't compete very well compared to other manufacturers. The two most obvious flaws are that they cannot be NAT'ed so they cannot be used as Off Premise eXtensions phones and the other being that they take so long to configure and re-boot. I have a golden rule with any phone that I plan on installing for a customerIf I can't get it working within 20 minutes then don't use it. I'm afraid Polycom breaks my golden rule. With such a lot of competition in this market they should have sorted this out two years ago. Reboots should not happen very often. This is a non-issue for most people. I've never seen a phone that could not work with NAT with Asterisk. Polycoms work just fine with NAT and Asterisk. The nice thing about Asterisk's NAT support is that the phone does not need to support NAT. ___ --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] 99 bottles of beer
I've been working on an X10 component already. It works, but I wish the CMA15 would work correctly in Linux (I know it's suppose to, but for whatever reason the one I have just doesnt.) It's just a little AGI script that I have working with Cepstral that throws http PUTs to the Windows box that has Apache-PHP and the command line app. Yeah, I know. But it wouldnt be this tedious if the CMA15 would appear correctly on my * box. (Oh, did I mention I made a LCARS Web GUI for this as well? :P) Steve Edwards wrote: On Tue, 21 Aug 2007, Russell Bryant wrote: Nice! While we're on the subject of silly but fun dialplan bits, check out my TV remote extension. When I moved a few months ago, there was a while when I couldn't find the wireless keyboard that I usually use as my TV remote to control MythTV. So, I built dialplan so I could use a wireless phone as my remote, instead. The dialplan reads digits from the phone and sends the correct commands to a MythTV network control interface for the frontend application. I posted my tested .conf version and the untested AEL version to the MythTV wiki. The AEL version would probably be prettier with macros, now that I think of it ... http://www.mythtv.org/wiki/index.php/Controlling_MythTV_from_any_phone_using_Asterisk And practical :) Almost every room in my house has a phone -- if I could teach my kids to put them back where they belong. This could easily be extended to recognize which phone was used so it could control the Myth FE in that room. Also, it could/should be extended to control x10 devices as well... To control the tv in this room, press 1. To control a tv in another room, press 2. To control the outside lights, press 3. To control the sprinklers, press 4, ... Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Saftware RAID1 or Hardware RAID1 with Asterisk
For RAID1, I am not sure. But for RAID 5, You should always use hardware RAID. If you use software RAID and your CPU spikes for too long, you can corrupt your disks. I have seen this several times. -- -- Steven http://www.glimasoutheast.org Vidura Senadeera [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Dear All, I would like to get community's feedback with regard to RAID1 ( Software or Hardware) implementations with asterisk. This is my setup Motherboard with SATA RAID1 support CENT OS 4.4 Asterisk 1.2.19 Libpri/zaptel latest release 2.8 Ghz Intel processor 2 80 GB SATA Hard disks 256 MB RAM digium PRI/E1 card Following are the concerns I am having I'm planing to put this asterisk server in production enviorment which is having E1 connection to the asterisk server, approximately 20 con-current calls, Music on hold, voice mail boxes. 1. If I use Software RAID, what would be the impact to my deployment? ( problems that I have to face with regard to the call flow ) 2. If I use Hardware based RAID 1, what would be the impact to the system? 3. According to your practical experiance what is the ideal solution among both options? I will be highly appreciate your feedback on this regard. -- Thanks Regards, Vidura Senadeera, -- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Saftware RAID1 or Hardware RAID1 with Asterisk
On Wed, 22 Aug 2007, Steven wrote: For RAID1, I am not sure. But for RAID 5, You should always use hardware RAID. If you use software RAID and your CPU spikes for too long, you can corrupt your disks. I have seen this several times. Please report this to the linux-raid mailling list, especially if you can repeat it. But I have to say, I've been using linux-raid for 8 years on some seriously overused/abused servers and *never* had disk corruption in all those years that I couldn't track down to a hardware problem. Gordon ___ --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] TDM400P FXO click sounds
Have you tried installing this board in another PC to test your FXOs ? What motherboard are you using ? Are your FXOs boards original digium or they are chineses versions ? Luis A P Barbosa 2007/8/15, [EMAIL PROTECTED] [EMAIL PROTECTED]: Hello, I have a TDM400P with 4 FXO ports, currently using three. When sending or receiving calls on this card, there is a nearly constant popping/clicking sound, it is related to the echo cancellation?. I adjusted my gains properly, but to no avail. I even found that setting echotraining=no in zapata.conf didn't change the scenario at all. I've plugged analog handsets into the same jacks, and the line is crystal-clear. Below is my zapata.conf, if you guys have any ideas how I might resolve this, I'd appreciate it. I have installed from sources Asterisk 1.2.22 and zaptel-1.2.19 on a debian etch x86_64. [channels] language=es context=ent-4229 ;rxwink=300 usecallerid=yes hidecallerid=no ; Whether or not to enable call waiting on internal extensions ; With this set to 'yes', busy extensions will hear the call-waiting ; tone, and can use hook-flash to switch between callers. The Dial() ; app will not return the BUSY result for extensions. ; callwaiting=yes threewaycalling=yes transfer=yes ;canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no echotraining=yes echotraining=128 relaxdtmf=yes rxgain=3.0 txgain=3.0 callgroup=1 pickupgroup=1 immediate=no ;busydetect=yes ;busycount=4 callprogress=no ;busypattern=500,500 ;answeronpolarityswitch=yes ;hanguponpolarityswitch=yes ;callprogress=yes faxdetect=incoming faxdetect=outgoing signalling=fxs_ks group=1 channel=1 signalling=fxs_ks group=2 channel=2; singalling=fxs_ks group=3 channel=3; ;singalling=fxs_ks ;group=1 ;channel=4 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Home Automation (was: Re: 99 bottles of beer)
On Wed, 2007-08-22 at 08:50 -0500, [EMAIL PROTECTED] wrote: Date: Tue, 21 Aug 2007 21:01:50 -0400 From: David Cook [EMAIL PROTECTED] Subject: Re: [asterisk-users] 99 bottles of beer To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii On 8/21/07, Steve Edwards [EMAIL PROTECTED] wrote: To control the tv in this room, press 1. To control a tv in another room, press 2. To control the outside lights, press 3. To control the sprinklers, press 4, ... Before this thread I already had a Firecracker on the server, a fair assortment of lights and the sprinklers are on an X10Pro Irrigation Controller. Damn, now I'm gonna be up all night. Isn't this kind of Asterisk interface to home automation what the xPL package in Trixbox is supposed to offer? Is there a source for clear, concise, *tested* guides and instructions for Asterisk/xPL home automation somewhere other than just a needle in the http://www.google.com/search?q=xpl+%22home+automation%22+asterisk haystack? Or maybe there's a better interface than xPL. -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [OT] IAX2 WiFi phone?
Does such a beastie exist? I've tried a couple of UT Starcom WiFi SIP phones (the F1000g and F3000 respectively), and found them both to be seriously lacking - regular crashes (especially the F3000), poor battery life, and poor reception in particular. However, whilst SIP phones are great, I'd really like an IAX2 phone if there is one, as I can make that work natively though the firewall, connected directly to a remote Asterisk server (remote = the other end of a broadband link). Cheers, Ade. No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.484 / Virus Database: 269.12.1/965 - Release Date: 21/08/2007 16:02 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [OT] IAX2 WiFi phone?
/me goes to work. There are none that I know of. There are only a couple of IAX(2) hard phones, and none of them, that I know of, are manufactured in the US anyways, and have problems. (Of course, what is manufactured in the US these days) That would be a great device, would love to see it come about. -bk - Original Message - From: Ade Vickers [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, August 22, 2007 9:27:09 AM (GMT-0600) America/Chicago Subject: [asterisk-users] [OT] IAX2 WiFi phone? Does such a beastie exist? I've tried a couple of UT Starcom WiFi SIP phones (the F1000g and F3000 respectively), and found them both to be seriously lacking - regular crashes (especially the F3000), poor battery life, and poor reception in particular. However, whilst SIP phones are great, I'd really like an IAX2 phone if there is one, as I can make that work natively though the firewall, connected directly to a remote Asterisk server (remote = the other end of a broadband link). Cheers, Ade. No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.484 / Virus Database: 269.12.1/965 - Release Date: 21/08/2007 16:02 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Agent status on Polycom phone?
Has anyone designed a method to allow callback agents (Asterisk 1.2) to log in on a Polycom SoundPoint IP phone and have the phone visually indicate the agents logged in status? ___ --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] Que on A2Billing
Hello All, Stable release of A2Billing has solved most of my problems and so far everything is OK... Right now the only problem I am facing with my SIP clients are: - - Three-way Calling Three-way calling works fine, but when SIP client hangs up the call, the other two channels are still active and talking. - Call Forwarding I created the context for *72, *73, *90, *91, *52, and *53. SIP client can enable and disable but it never works because a2billing.php will time out and hang up the SIP channel. - Voice Mail I created the context for voice mail, but the calls will never go to voice mail because a2billing.php after 60 sec will hang up the channel. No doubt A2Billing is a great software, but the above features are also essential for home SIP users... Anyone can show or share their setup if they have implemented the above features with A2Billing Software. Cheers, Nitesh Al Bochter wrote: In a2billing just change the 9 to what you need it is right in the conf file. Best regards, Al Bochter Bochter Services -- Need to call me use our web phone at the link below http://www.bochterservices.com/voip/iaxphone.php?cn=250 -- Can you WIN gold today? Click on the link and see. http://www.bochterservices.com/?t=USbill_email -- Need cash we buy silver and gold -- Nitesh Divecha wrote: Thanks everyone for the input... In real world we can not ask the customers to dial 9, if they want to call another SIP user... and trust me its confusing for a customer also... meaning when to dial 9 and when to not... We have a custom proprietary system which does this part very well... Before it sends the call on a Trunk it will check the DID, if it exists within the local system. If it does then it will just use IP to IP call, else send the call to Trunk... I think its possible to do this by creating some basic dial plans... Same like creating local extensions. Cheers, Nitesh John Novack wrote: Given that Asterisk is modeled on, in the telephone industry, an obsolete PBX design, without many of the modern day hybrid features, and only recently has any effort been made to provide buttons and lights for lines ( Is that yet working in 1.4??) one would have to do some very careful number parsing to not use a trunk digit. If every phone in the system had buttons and lights representing external connections and internal connections on other button(s) ( intercom ) this wouldn't be an issue. Most legacy systems have been able to do this for the last 20 years or so. John Novack Nitesh Divecha wrote: Thanks man, Is there any other way without dialing 9... it will be kinda pain for a customer to dial 9 every time and plus they need to know also... Is there any intelligent way to identify? if its a local SIP then don't route to Trunk else route to Trunk. Cheers, Nitesh Guillermo Salas M. wrote: On Tue, 2007-06-19 at 09:36 -0400, Nitesh Divecha wrote: Thanks man... So far everything worked as expected... How can I make internal calls stay within the PBX. For example, when one SIP-Friend tries to call another SIP-Friend without sending the call out on Trunk and receive it back. Same like dialing from one extension number to another extension. My SIP-Friends are using US DID numbers and I would like to keep the local calls within the network. Right now when I try to call other SIP-Friend, I get a message saying The number you have dialer is currently not available... while the SIP-Friend is registered. Try dialing the number 9 before the sip/iax2 friend number. Regards, Cheers, Nitesh ___ --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 Inbound (clean). Database: 000750-2, 06/19/2007 - 6/19/2007 4:22:13 PM
Re: [asterisk-users] phone as control interface (was 99 bottles of beer)
On Tue, 21 Aug 2007, Steve Prior wrote: Steve Edwards wrote: To control the tv in this room, press 1. To control a tv in another room, press 2. To control the outside lights, press 3. To control the sprinklers, press 4, ... A while back I was thinking along the lines of using a phone as a home automation interface, though I was thinking of it in combination with a voice recognitition system such as Lumenvox. It occured to me that when you want to turn the lights on, you don't really want to pick up a phone, dial a special extension, and then start using menus. Personally, I hate voice recognition systems. Voice prompts are great, but don't take away my keypad. Maybe I'm too far out on the edge of the bell curve, but I CAN remember what Alison was prattling about long enough to be told which key to press. Also, keys are much faster, especially once memorized as will happen quickly for something as frequently used as the TV. Think of a task like muting the TV: ) Off-hook ) Dialtone ) Press ** (change to remote mode) ) To control the... ) Press 1 ) To change the vol... ) Press 1 ) To mut... ) Press 0 I'll be finished before you can say HAL, Computer, Zen or whatever. Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TDM400P Not hanging up fast enough
Hi List, I have a client who has a TDM400P with 4 FXO. He has a problem them when some one calls, then hangs up it takes a good 10-15 seconds or more of the card to realize that the line was hung up on. The phones keep reigning After a bit it hangs up on the line. Also there has been some hanging. (After a user on the PBX side hangs up the card does not release the line). I am using asterisk 1.2.21.1 with Zaptel 1.2.18. Thanks. Dovid___ --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] Agent status on Polycom phone?
Answered my own question - use buddy watch on the Polycom and create a hint priority extension for the agent channel... From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Damon Estep Sent: Wednesday, August 22, 2007 8:42 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Agent status on Polycom phone? Has anyone designed a method to allow callback agents (Asterisk 1.2) to log in on a Polycom SoundPoint IP phone and have the phone visually indicate the agents logged in status? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [OT] IAX2 WiFi phone?
Brandon Kruse wrote: /me goes to work. There are none that I know of. There are only a couple of IAX(2) hard phones, and none of them, that I know of, are manufactured in the US anyways, and have problems. (Of course, what is manufactured in the US these days) That would be a great device, would love to see it come about. That was pretty much what I'd concluded from my googling :( Of course, for me, I'd like to see a device available in the UK/Europe the US is fine, but your power supply is a bit incompatible with ours, and the shipping cost is a nightmare ;) If I could convert my Grandstream GXP-2000's to IAX2, i'd be as happy as a pig in, er, you get the picture; I could then drop the one Asterisk per site setup I'm currently stuck with - although I suppose the advantage of that particular setup is all internal calls are truly internal... Anyway - if anyone hears of an IAX2 WiFi phone in the works, please do drop a line in here Cheers, Ade. No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.484 / Virus Database: 269.12.1/965 - Release Date: 21/08/2007 16:02 ___ --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] TDM400P Not hanging up fast enough
Hi Dovid, I had the same problem with the same configuration 3 lines worked fine but the fourth couldn't detect disconnect supervision (it was a Foriegn Exchange Line). There are two things you should try before anything else. 1. Reverse the Tip and Ring connections on the CO Lines and test again. 2. Make sure your lines are set up for Kewl Start. There are lots of people who have had these problems so I would check the archives on the TAUG Good Luck (you may need it) -- Henry L. Coleman. Dovid B Hi List, I have a client who has a TDM400P with 4 FXO. He has a problem them when some one calls, then hangs up it takes a good 10-15 seconds or more of the card to realize that the line was hung up on. The phones keep reigning After a bit it hangs up on the line. Also there has been some hanging. (After a user on the PBX side hangs up the card does not release the line). I am using asterisk 1.2.21.1 with Zaptel 1.2.18. Thanks. Dovid___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Chan_mobile and Asterisk SVN-branch-1.4-r80183 compile error
Hi all, I receive this error while compiling chan_mobile: gcc -g -c -fPIC -fPIC -o chan_mobile.o chan_mobile.c chan_mobile.c: In function `mbl_load_config': chan_mobile.c:1745: erreur: trop d'arguments pour la fonction « ast_config_load » make[1]: *** [chan_mobile.o] Erreur 1 make[1]: Leaving directory `/usr/src/asterisk-addons' Does anyone know what's the problem? -- Daniel ___ --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] Chan_mobile and Asterisk SVN-branch-1.4-r80183 compile error
Administrator TOOTAI wrote: Hi all, I receive this error while compiling chan_mobile: gcc -g -c -fPIC -fPIC -o chan_mobile.o chan_mobile.c chan_mobile.c: In function `mbl_load_config': chan_mobile.c:1745: erreur: trop d'arguments pour la fonction « ast_config_load » make[1]: *** [chan_mobile.o] Erreur 1 make[1]: Leaving directory `/usr/src/asterisk-addons' Does anyone know what's the problem? You're trying to use a module written for trunk on 1.4. -- Jason Parker Digium ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to re-read values from database in Trixbox
Thanks Diego :) It works now. For the rest of the people, the commands needed were /var/lib/asterisk/bin/retrieve_conf and then asterisk -rx reload ___ --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] phone as control interface (was 99 bottles of beer)
Steve Edwards wrote: Personally, I hate voice recognition systems. Voice prompts are great, but don't take away my keypad. I never proposed to take away your keypad, I just wanted to add the voice option as well. What I do want to get rid of is the step below where you press ** to get into remote mode. What I think would be smoother is to have the extensions organized in such a way that the first button press (or voice utterance) is enough to determine whether the session is a phone call or an automation request. Then as soon as the phone is picked up you'd get into the automation context and the voice menu you gave would start right away. Then if it turned out that first button meant that the user really wanted to make a call, then Asterisk would shift into the normal call dialplan, but reprocess that already pressed key as part of the phone number to dial. I think that we can provide something more intelligent when the phone is first picked up than your basic dialtone and not require extra button presses to get into the right mode. Your desire is for speed and so is mine. Steve Maybe I'm too far out on the edge of the bell curve, but I CAN remember what Alison was prattling about long enough to be told which key to press. Also, keys are much faster, especially once memorized as will happen quickly for something as frequently used as the TV. Think of a task like muting the TV: ) Off-hook ) Dialtone ) Press ** (change to remote mode) ) To control the... ) Press 1 ) To change the vol... ) Press 1 ) To mut... ) Press 0 ___ --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] Chan_mobile and Asterisk SVN-branch-1.4-r80183 compile error
Jason Parker a écrit : Administrator TOOTAI wrote: Hi all, I receive this error while compiling chan_mobile: gcc -g -c -fPIC -fPIC -o chan_mobile.o chan_mobile.c chan_mobile.c: In function `mbl_load_config': chan_mobile.c:1745: erreur: trop d'arguments pour la fonction « ast_config_load » make[1]: *** [chan_mobile.o] Erreur 1 make[1]: Leaving directory `/usr/src/asterisk-addons' Does anyone know what's the problem? You're trying to use a module written for trunk on 1.4. Ah, ok, sorry. Was thinking it's for 1.4 trunk ;-) -- Daniel ___ --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] Saftware RAID1 or Hardware RAID1 with Asterisk
I've gotten burned by software raid so I'll probably be sticking with hardware in the future. If your drive dies for some reason it could affect the SATA bus and cause the system to crash. That's what happened to me. It wouldn't come back up on the second Raid 1 drive until I removed the bad one. With hardware raid that would NEVER happen. They are designed to isolate the Sata interface of a drive gone bad and just keep on running. Are there Hardware Raid cards that don't come with software for monitoring and emailing warnings? I know the 3ware cards come with linux software that can run as a web interface or command line so the whole monitoring argument is mute IMHO. -Original Message- From: Arnaud Ligot [mailto:[EMAIL PROTECTED] Sent: Tuesday, August 21, 2007 12:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Saftware RAID1 or Hardware RAID1 with Asterisk My servers run in a datacenter, 50km away from my office... if a led flash, if the speaker beep... I think I'll not see/hear it ... My servers are monitored using nagios which has a plugin for software raid... so if one array goes down, I receive a mail/sms/call/... futher more, everything is on the same panel: raid, http servers, free disk space, ... I think it is better than any led flashing into the DC :-D A. On Tue, 2007-08-21 at 10:30 -0400, Steve Totaro wrote: I thought that was what the flashing LEDs on the front of the server's HDs were for (besides showing activity). Some I have seen also have an LED near the power button to indicate HD problems. I guess if you are building your own boxen and not using enterprise grade servers, this is not the case. Thanks, Steve Totaro C F wrote: While hardware RAID tend to be more reliable, it is not always possible to properly monitor hardware raid in a linux system, unless you write your own code. Consider this: ~# cat /proc/mdstat Personalities : [raid1] md0 : active raid1 sdb2[2](F) sda2[1] 76139968 blocks [2/1] [_U] unused devices: none The above is from an active system that one hdd failed. It would take way longer to find such a thing on a hardware raid. Unless it came with a program that emails me notification on such a failure. On 8/20/07, Vidura Senadeera [EMAIL PROTECTED] wrote: Dear All, I would like to get community's feedback with regard to RAID1 ( Software or Hardware) implementations with asterisk. This is my setup Motherboard with SATA RAID1 support CENT OS 4.4 Asterisk 1.2.19 Libpri/zaptel latest release 2.8 Ghz Intel processor 2 80 GB SATA Hard disks 256 MB RAM digium PRI/E1 card Following are the concerns I am having I'm planing to put this asterisk server in production enviorment which is having E1 connection to the asterisk server, approximately 20 con-current calls, Music on hold, voice mail boxes. 1. If I use Software RAID, what would be the impact to my deployment? ( problems that I have to face with regard to the call flow ) 2. If I use Hardware based RAID 1, what would be the impact to the system? 3. According to your practical experiance what is the ideal solution among both options? I will be highly appreciate your feedback on this regard. -- Thanks Regards, Vidura Senadeera, ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] phone as control interface (was 99 bottles of beer)
Quoting Steve Prior [EMAIL PROTECTED]: personally my favourite still is phone in intercom mode listening at all times, if you have something to say, say it. otherwise pickup and dial for control or to talk or whatever. nothing preventing you from ignoring one of the options if you don't like it, or have a phone that supports it. Steve Edwards wrote: Personally, I hate voice recognition systems. Voice prompts are great, but don't take away my keypad. I never proposed to take away your keypad, I just wanted to add the voice option as well. What I do want to get rid of is the step below where you press ** to get into remote mode. What I think would be smoother is to have the extensions organized in such a way that the first button press (or voice utterance) is enough to determine whether the session is a phone call or an automation request. Then as soon as the phone is picked up you'd get into the automation context and the voice menu you gave would start right away. Then if it turned out that first button meant that the user really wanted to make a call, then Asterisk would shift into the normal call dialplan, but reprocess that already pressed key as part of the phone number to dial. I think that we can provide something more intelligent when the phone is first picked up than your basic dialtone and not require extra button presses to get into the right mode. Your desire is for speed and so is mine. Steve Maybe I'm too far out on the edge of the bell curve, but I CAN remember what Alison was prattling about long enough to be told which key to press. Also, keys are much faster, especially once memorized as will happen quickly for something as frequently used as the TV. Think of a task like muting the TV: ) Off-hook ) Dialtone ) Press ** (change to remote mode) ) To control the... ) Press 1 ) To change the vol... ) Press 1 ) To mut... ) Press 0 ___ --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 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 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] Chan_mobile and Asterisk SVN-branch-1.4-r80183 compile error
Administrator TOOTAI wrote: Jason Parker a écrit : Administrator TOOTAI wrote: Hi all, I receive this error while compiling chan_mobile: gcc -g -c -fPIC -fPIC -o chan_mobile.o chan_mobile.c chan_mobile.c: In function `mbl_load_config': chan_mobile.c:1745: erreur: trop d'arguments pour la fonction « ast_config_load » make[1]: *** [chan_mobile.o] Erreur 1 make[1]: Leaving directory `/usr/src/asterisk-addons' Does anyone know what's the problem? You're trying to use a module written for trunk on 1.4. Ah, ok, sorry. Was thinking it's for 1.4 trunk ;-) There is no such thing as 1.4 trunk. -- Jason Parker Digium ___ --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] Chan_mobile and Asterisk SVN-branch-1.4-r80183 compile error
Jason Parker wrote: Administrator TOOTAI wrote: Hi all, I receive this error while compiling chan_mobile: gcc -g -c -fPIC -fPIC -o chan_mobile.o chan_mobile.c chan_mobile.c: In function `mbl_load_config': chan_mobile.c:1745: erreur: trop d'arguments pour la fonction « ast_config_load » make[1]: *** [chan_mobile.o] Erreur 1 make[1]: Leaving directory `/usr/src/asterisk-addons' Does anyone know what's the problem? You're trying to use a module written for trunk on 1.4. Try checking out r421 of asterisk-addons, and replacing ast_debug(1, with ast_log(LOG_DEBUG, in all instances in chan_mobile.c. (Still only compile chan_mobile.c. This appears to work with 421, but not 423. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Multiple servers using realtime
I am in the process of setting up several * servers using realtime and connecting to mysql. I am trying to figure out if I should just use one database and one set of tables for all of the user data. Or if I should have separate databases for each * box. The boxes are independent of each other in that customerA only connects to box A. They will never fail over to box B or anything like that. I want to use realtime for queues,voicemail, sippeers and extensions. The only issue that I have come up with so far is that a common voicemail table would cause each box to try and send out mwi indicators since it appears each * box pulls all of the voicemail boxes from the DB every 10 seconds. Any thoughts on whether I should go with one DB, or separate per box DB's? There is one mysql box, I am not referring to mysql on each box, I am referring to whether I should use separate DB's within the one mysql box for each * box. Thanks. Peder ___ --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] Multiple servers using realtime
Seems to me that as long as all the contacts / reachability info / URIs are distinct for each user, there is not a problem with using one big database, and that it certainly presents less of a maintenance headache. It also provides easier migration path to future options you may want to explore that *do* take advantage of its shared aspect. On Wed, 22 Aug 2007, Peder @ NetworkOblivion wrote: I am in the process of setting up several * servers using realtime and connecting to mysql. I am trying to figure out if I should just use one database and one set of tables for all of the user data. Or if I should have separate databases for each * box. The boxes are independent of each other in that customerA only connects to box A. They will never fail over to box B or anything like that. I want to use realtime for queues,voicemail, sippeers and extensions. The only issue that I have come up with so far is that a common voicemail table would cause each box to try and send out mwi indicators since it appears each * box pulls all of the voicemail boxes from the DB every 10 seconds. Any thoughts on whether I should go with one DB, or separate per box DB's? There is one mysql box, I am not referring to mysql on each box, I am referring to whether I should use separate DB's within the one mysql box for each * box. Thanks. Peder ___ --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 -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --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] Chan_mobile and Asterisk SVN-branch-1.4-r80183 compile error
Jason Parker a écrit : Administrator TOOTAI wrote: Jason Parker a écrit : Administrator TOOTAI wrote: Hi all, I receive this error while compiling chan_mobile: gcc -g -c -fPIC -fPIC -o chan_mobile.o chan_mobile.c chan_mobile.c: In function `mbl_load_config': chan_mobile.c:1745: erreur: trop d'arguments pour la fonction « ast_config_load » make[1]: *** [chan_mobile.o] Erreur 1 make[1]: Leaving directory `/usr/src/asterisk-addons' Does anyone know what's the problem? You're trying to use a module written for trunk on 1.4. Ah, ok, sorry. Was thinking it's for 1.4 trunk ;-) There is no such thing as 1.4 trunk. Thought that svn co http://svn.digium.com/svn/asterisk/branches/1.4 asterisk-1.4 was considered as trunk for stable branch. -- Daniel ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Zaptel 1.2.20.1 and 1.4.5.1 released
The Asterisk.org development team has announced the release of Zaptel versions 1.2.20.1 and 1.4.5.1. These releases are to correct an error in the install target in the Makefile of the 1.4.5 and 1.2.20 Zaptel releases, as well as a handful of other issues. See the respective Changelogs for more details. Both releases are available as a tarball as well as a patch against the previous release. They are available for download from downloads.digium.com. Thank you for your support! ___ --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] phone as control interface (was 99 bottles of beer)
Jon Pounder wrote: Quoting Steve Prior [EMAIL PROTECTED]: personally my favourite still is phone in intercom mode listening at all times, if you have something to say, say it. otherwise pickup and dial for control or to talk or whatever. nothing preventing you from ignoring one of the options if you don't like it, or have a phone that supports it. Computer: close bulkheads on Deck 40! Deck 40 does not exist. Uh oh. -Stephen- ___ --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] Saftware RAID1 or Hardware RAID1 with Asterisk
Gordon Henderson wrote: You do (sometimes) need the hardware RAID controller to be supported by Linux and this is a weak area. Some controllers just look like a standard drive, so they are transparent to the system, but then you need to use either the BIOS utilities to set it up in the first place, or (typically) a Windows utility, although some controllers are now being supported by Linux with user-land tools to manage and check the arrays. Most proper (ie, not fakeraid) RAID controllers support Linux now. They are practically unsellable if they do not. -Stephen- ___ --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] Saftware RAID1 or Hardware RAID1 with Asterisk
Zane C.B. wrote: 1: Software RAID on Linux is way less than impressive. Plus last a I checked Linux can't handle mirroring a entire disk. Last I looked at it around a year ago you were limited to only mirroring partitions, which is a joke from a administrative standpoint. How is this any different in FreeBSD? Could you explain to me how else you are going to mirror an entire disk in software when your boot partition is on the disk? -Stephen- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] VoIP encryption with SIP and IAX
Dear all, I have an Asterisk server with SIP and IAX softphones clients, and I need to encrypt the voip calls among them: *For SIP clients I use Twinkle which implements the ZRTP/SRTP encryption mechanism client-2-client; I read it's the better security mechanism nowadays created by Phill Zimmerman who created PGP. *For IAX clients I used Kiax but I don't know exactly if there is any encryption mechanism for this protocol. Two short questions: 1) Do you think ZRTP/SRTP is the best option to encrypt SIP voip calls ??? 2) What is the best way to encrypt IAX voip calls ??? Really thanks Alejandro ___ --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-biz] Skype Outage Leaves Millions Speechless
Matthew Rubenstein wrote: Imagine if the world's largest online marketplace operated the world's largest alternative (and one of the largest in general) telco and an unregulated global online banking monopoly. And the telco suddenly went down, unexplained, for hours or days. That sounds like a serious threat to global economy and security, right? If the global economy is depending on a free, unguaranteed third-party VoIP service for critical communications, it deserves to go down in flames. I don't use Skype for anything important. It's nothing more than a nice to have. A tempest in a teapot. Embarrassing for Skype and eBay? Sure! A sign of Armageddon? Hardly. If anything, this is another warning against relying on Microsoft Windows. -Stephen- ___ --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] Zaptel 1.2.20.1 and 1.4.5.1 released
On Wed, Aug 22, 2007 at 01:19:14PM -0500, Asterisk Development Team wrote: The Asterisk.org development team has announced the release of Zaptel versions 1.2.20.1 and 1.4.5.1. These releases are to correct an error in the install target in the Makefile of the 1.4.5 and 1.2.20 Zaptel releases, as well as a handful of other issues. See the respective Changelogs for more details. Both releases are available as a tarball as well as a patch against the previous release. They are available for download from downloads.digium.com. Don't seem to be on www.asterisk.org (1.2.19 and 1.4.4) Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Queue Agents from Dialplan
Is there any way to get the channel of the first agent called in a queue? Say I have a queue with 5 agents setup in roundrobin. I want the voicemail to go to the first person that was called. Say a call comes in and rings 1,2,3, then I want it to go to vm for 1. Say the next call rings 4,5,1, I want it to go to vm for 4. I am looking for a way to get that info into the dialplan so that I can send the calls to the appropriate voicemail. Thanks. Peder ___ --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] Multiple servers using realtime
I use a centralized database (with replication) for several servers, and it works very well. I keep all the mysql traffic on a separate network from the SIP traffic. It makes it easy to add capacity. If you are doing all the mySQL on one box anyway, I don?t see any adavantage to using separate databases. -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nombre de Peder @ NetworkOblivion Enviado el: miercoles, 22 de agosto de 2007 19:06 Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: [asterisk-users] Multiple servers using realtime I am in the process of setting up several * servers using realtime and connecting to mysql. I am trying to figure out if I should just use one database and one set of tables for all of the user data. Or if I should have separate databases for each * box. The boxes are independent of each other in that customerA only connects to box A. They will never fail over to box B or anything like that. I want to use realtime for queues,voicemail, sippeers and extensions. The only issue that I have come up with so far is that a common voicemail table would cause each box to try and send out mwi indicators since it appears each * box pulls all of the voicemail boxes from the DB every 10 seconds. Any thoughts on whether I should go with one DB, or separate per box DB's? There is one mysql box, I am not referring to mysql on each box, I am referring to whether I should use separate DB's within the one mysql box for each * box. Thanks. Peder ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Users Conference - [EMAIL PROTECTED]:30 PM EDT: Founders of Voicepulse
For this week's conference, the two founders of Voicepulse, Ravi Sakaria and Ketan Patel, will be joining us. For those of you who are not aware, Voicepulse is an asterisk friendly VOIP provider that has won awards for service and innovation. We will also have Trixbox news, updates, as well as discount codes. Lastly, we are working feverishly to bring you more information regarding legal issues surrounding VOIP in the coming weeks. So please join us for this week's conference: http://www.AsteriskUsersConference.org You can find out more about Voicepulse at: http://www.voicepulse.com Voicepulse's asterisk section located at: http://connect.voicepulse.com Info about the founders of Voicepulse: http://www.voicepulse.com/corporate/Management.aspx ___ --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] Dialogic support
Sorry for not indenting, I am stuck using OWA for the moment. If your customer seriously wants to pursue that option, please let me know what they have. Model numbers, used/new, how many, and any other details. I can probably get them a much better price than Ebay or something. Thank, Steve My customer has tones of DM3 cards (DM/V600, DM/N1200, and D600-2E1), they want to see if they can use them in Asterisk. My advise to them is to sell those cards and buy Sangoma E1 cards, and still have money left. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Armin Schindler Sent: Tuesday, August 21, 2007 3:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: [EMAIL PROTECTED] Subject: Re: [asterisk-users] Dialogic support On Tue, 21 Aug 2007, Wai Wu wrote: Can someone share pointers to Asterisk's Dialogic support? Which boards are supported, driver status, and etc. Which type of boards are you interested in? I don't know about other cards, but the DIVA Server ISDN cards are well supported. Armin ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users winmail.dat___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [OT] IAX2 WiFi phone?
We bought a few IAX ata's a while ago (virbiage) and they worked quite wellone of those with a standard cordless phone would be an idea... PaulH On Wed, 2007-08-22 at 15:27 +0100, Ade Vickers wrote: Does such a beastie exist? I've tried a couple of UT Starcom WiFi SIP phones (the F1000g and F3000 respectively), and found them both to be seriously lacking - regular crashes (especially the F3000), poor battery life, and poor reception in particular. However, whilst SIP phones are great, I'd really like an IAX2 phone if there is one, as I can make that work natively though the firewall, connected directly to a remote Asterisk server (remote = the other end of a broadband link). Cheers, Ade. No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.484 / Virus Database: 269.12.1/965 - Release Date: 21/08/2007 16:02 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to get callee extension in applicationmap(features.conf)
hello,
I use trixbox.I had define a feature code testfeature:
[applicationmap]
#include features_applicationmap_additional.conf
testfeature = *3,callee,Macro,vote
[featuremap]
blindxfer = ## ; Blind Transfer
disconnect = **; Disconnect Call
automon = *1 ; One Touch Record
atxfer = *2; Attended Xfer
testfeature = *3
here is my macro-vote:
[macro-vote]
exten = s,1,Noop('Macro-vote')
exten = s,2,BackGround(custom/0703)
;exten = s,3,
exten = s,3,goto(voting,s,1)
[voting]
;exten = s,1,Background(custom/0703)
exten = s,1,Noop('Now Let's voting!')
exten = 1,1,Set(Rate=100)
exten = 1,2,SayDigits(${Rate})
exten = 2,1,Set(Rate=80)
exten = 2,2,SayDigits(${Rate})
exten = 3,1,Set(Rate=60)
exten = 3,2,SayDigits(${Rate})
exten = 4,1,Set(Rate=0)
exten = 4,2,SayDigits(${Rate})
exten = t,1,goto(s,1)
exten = i,1,Playback(invalid)
exten = i,2,goto(s,1)
That is two problem:
1.When callee press *3,asterisk start to execute macro vote,execute
BackGround(custom/0703),when playing,the caller press 1,asterisk goto
VM context,it can't goto voting context.
I had another try:callee press *3,asterisk start to execute macro
vote,execute BackGround(custom/0703), let asterisk play to end,callee
press *3 once again,at this time asterisk goto voting cntext
correct.How can i correct this problem?
2.How to get callee's extension in my vote or voting Context?
Best Regard
yfeng lee
--
---
说我所做，做我所说，做我所想
___
--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-users Digest, Vol 37, Issue 88
with the * box on a 192.168.1.x the polycoms were able to communicate however sustained a lot of one way audio problems. Moving thim onto the same network is the only thing we have been able to reliable do. According to Polycom Support this is what they are intended for and no definitive answer as to whether they would support Stun or another method in the future. At least as of 6 months ago. Matt Although I do appreciate your response, I didn't intend to paint this as a NAT issue in my original post. I have successfully deployed Polycom phones behind NAT many times in the past when the * box was on a public IP without a NAT or ALG present. This leads me to focus on the ALG as part of issue in this case (not that the ALG in and of itself is the issue, but the combination of Polycom and the ALG since other brands of phones work properly). The link that I referred to in my original post referenced an issue with the MD5 hash being different on either end due to differences in the URI, causing a registration authentication problem (as I understand it). I was just asking for assistance understanding what the link was recommended as a fix. Thanks! No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.484 / Virus Database: 269.12.1/963 - Release Date: 8/20/2007 5:44 PM -- Message: 8 Date: Wed, 22 Aug 2007 08:53:28 +0200 From: Olivier [EMAIL PROTECTED] Subject: [asterisk-users] rfc3680, reginfo+xml 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 Hi, RFC3680 defines a SIP event package for registration. This event package which can be used through NOTIFY-SUBSCRIBE methods, seems very useful for free sitting or presence applications. This package is supported in various SIP phones (at least Thomson ST2030) : when turned on, this feature adds a new login/logout menu among other things. It can also be used to send Welcome notices to mobile users : whenever a mobile user comes in, a SIP MESSAGE is sent by a software application which has previously subscribed to be notified of any registration event related to this mobile user. It appears Asterisk supports SIP NOTIFY-SUBSCRIBE methods. But I couldn't find any trace of this specific Registration Event package support (but I won't swear I searched the right way). How can I make sure this feature is supported or not ? More precisely, this Registration Event package support relies on these headers : SIP SUBSCRIBE reg Event SIP SUBSCRIBE application/reginfo+xml Accept SIP NOTIFY reg Event SIP NOTIFY application/reginfo+xml Content How shall I check ? Regards -- next part -- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070822/347dae 77/attachment-0001.htm -- Message: 9 Date: Wed, 22 Aug 2007 01:22:13 -0600 From: Edgar Guadamuz [EMAIL PROTECTED] Subject: [asterisk-users] How to re-read values from database in Trixbox 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 Hello guys, I'm using Trixbox and I have a PHP application that updates a value in the MySQL asterisk database as an interface to have a dynamic customizable IVR. After execute the UPDATE SQL query, the php application is supossed to reload asterisk or restart amportal in order to get the change working, but nor asterisk -rx reload nor amportal restart got the change working. So, the question is how can I re-read the new value from the database to be effective in asterisk? -- Message: 10 Date: Wed, 22 Aug 2007 10:42:52 +0300 From: Diego Iastrubni [EMAIL PROTECTED] Subject: Re: [asterisk-users] How to re-read values from database in Trixbox To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 You are updating the MySQL config, which is not propagated to the Asterisk config files. Only after you regenerate the configuratios, you can reload asterisk. Dirty hack: need_reload flag must be set to true. Real solution: retrieve_conf + asterisk reload On Wednesday 22 August 2007 10:22, Edgar Guadamuz wrote: Hello guys, I'm using Trixbox and I have a PHP application that updates a value in the MySQL asterisk database as an interface to have a dynamic customizable IVR. After execute the UPDATE SQL query, the php application is supossed to reload asterisk or restart amportal in order to get the change working, but nor asterisk -rx reload nor amportal restart got the change working. So, the question is how can I re-read the new value from the database to be effective in asterisk? -- Message: 11 Date: Wed
