Re: [asterisk-users] SCCP Questions
Hello, Thanks you for the replies ill take a look at the driver you sent over. Im going to run some test and see what happens, hopefully the driver in 1.8 is soild and nothing needs to be messed with, but we will see :) On Thu, Jun 14, 2012 at 5:06 AM, Tim Nelson tnel...@rockbochs.com wrote: Greetings Ron- Just wanted to give you a heads up about an alternative SCCP channel driver available for Asterisk. Please see here: http://freecode.com/projects/chan-sccp-b I have no experience with it (nor SCCP in general) but just wanted to give you an option in the event the included SCCP driver does not give you satisfactory results. --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SCCP Questions
Is the chan-sccp-b project the same one that got put in SVN of 1.8 branch? I have not been able to find anything definitive that says so, I really need 1.8 branch so trying to see which is the best way to go. Thanks On Thu, Jun 14, 2012 at 9:34 AM, Ron McCarthy ronmc...@gmail.com wrote: Hello, Thanks you for the replies ill take a look at the driver you sent over. Im going to run some test and see what happens, hopefully the driver in 1.8 is soild and nothing needs to be messed with, but we will see :) On Thu, Jun 14, 2012 at 5:06 AM, Tim Nelson tnel...@rockbochs.com wrote: Greetings Ron- Just wanted to give you a heads up about an alternative SCCP channel driver available for Asterisk. Please see here: http://freecode.com/projects/chan-sccp-b I have no experience with it (nor SCCP in general) but just wanted to give you an option in the event the included SCCP driver does not give you satisfactory results. --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SCCP Questions
Hi List, Has anyone been running SCCP with a larger number of phones? Im looking to deploy like 75+ phones and I want to keep SCCP so I don't have to upgrade them and for the SLA, some phones also have no SIP software for them so im forced to keep SCCP. Does anyone have any experience with this? From what ive read the SCCP support works and works well, im just worried about trying to run this many phones and if im missing any sort of issues that could come up. Thanks! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Avaya 9640 Convert to SIP (slightly off topic)
Hi List, Ive got a bunch of Avaya IP9640 that we want to convert to SIP and then hook up to Asterisk so we can dump this overpriced Avaya system. Ive got ahold of the SIP firmware, but I cannot find anything on how to convert the phone itself to SIP, when I go into setup mode it wants a command which im guessing is the program code like the rest of the Avaya systems works. Has anyone been able to convert this to SIP and if so, any suggestions for me? Any help would be great, Ive Googled and have found a bunch of dead ends. Thanks! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Multiple Digium cards with one NFAS trunkgroup
Hi list, Ive got a server with 6 ports on it (4+2 port card) we have a DS3 delivering all voice DS1's to us. Carrier has a trunkgroup for the first 8 span (we only have the first 6 plugged in right now). Everything works fine until we fail the primary D channel (D's are on 24,48) the secondary then picks up and outbound calls do not work, if we reboot Asterisk the D on 48 comes up and it works fine, but we have to reboot, a reload chan_zap.so or no interaction causes all outgoing calls to fail and no inbound audio. Any ideal on what would cause this? I tried the latest Zaptel and then a older version to, same issue. Here is what the config files look like, any help would be great on this, ive Googled and can't seem to find what else to try! zaptel.conf loadzone=us defaultzone=us span=1,1,0,esf,b8zs bchan=1-23 dchan=24 span=2,1,0,esf,b8zs bchan=25-47 dchan=48 span=3,1,0,esf,b8zs bchan=49-72 #dchan=72 span=4,1,0,esf,b8zs bchan=73-96 #dchan=96 span=5,1,0,esf,b8zs bchan=97-120 #dchan=120 span=6,1,0,esf,b8zs bchan=121-144 #dchan=144 zapata.conf [trunkgroups] trunkgroup = 1,24,48 spanmap = 1,1,0 spanmap = 2,1,1 spanmap = 3,1,2 spanmap = 4,1,3 spanmap = 5,1,6 spanmap = 6,1,7 [channels] switchtype=4ess signalling=pri_cpe context=trunkinbound group = 0 channel = 1-23 channel = 25-47 channel = 49-144 ;,25-47,49-144 ;,25-47,48-288 ;channel = 1-23,25-47,49-144 ;,49-71,73-95,97-119,121-143 ___ -- 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 Digium cards with one NFAS trunkgroup
Ill try another mx of libPRI, it has to be something goofy. Thanks for the input guys. On Mon, Dec 28, 2009 at 11:00 AM, Kevin P. Fleming kpflem...@digium.comwrote: Steve Totaro wrote: I would call Digium but last I knew, NFAS only worked across one card. NFAS is implemented in libpri and Asterisk (chan_zap or chan_dahdi), which means it has no concept of 'cards' at all. Cards are handled at the Zaptel/DAHDI layer, and are presented as spans full of channels to the userspace application... so there really can't be any such restriction. NFAS should work fine across multiple cards. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.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 ___ -- 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 listen in on a SIP channel?
Hi list, I see their is ExtenSpy(), I want to monitor calls (in and out I hope) from another phone, all the channels are SIP. ChanSpy() looks like you cannot give it a context and I want to be able to only monitor certain calls. Any Ideals on how to do this? Thanks! ___ -- 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 listen in on a SIP channel?
Works like a champ. I have to use the b option as well otherwise it just goes into a endless beep, sounds good though! Thanks for the help! On Wed, Jan 7, 2009 at 4:02 PM, Mark Michelson mmichel...@digium.comwrote: Ron McCarthy wrote: Hi list, I see their is ExtenSpy(), I want to monitor calls (in and out I hope) from another phone, all the channels are SIP. ChanSpy() looks like you cannot give it a context and I want to be able to only monitor certain calls. Any Ideals on how to do this? Thanks! You can use ChanSpy for this, using its grouping feature. When a call is made to or from a phone which you would like to listen to, set the SPYGROUP variable to some number. Then when you call the Chanspy application, supply it with the g option and use that number as an argument. You can get more details on this by issuing the command core show application Chanspy in the Asterisk CLI. Specifically look at the g option. Hope this helps. Mark Michelson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Using manager originate and Dial() once inside dialplan
Hi List, We are trying to make a click 2 call button, we have a PHP script that passes the 1st phone number of the 1st leg to a manager script, that then dials the 1st call, then the 2nd call gets placed inside of Asterisk using a normal dial command. Problem is, we get no status codes, we cannot see if their was a hangup, a answer anything, and also once the callers hangs up, it's killed and cannot execute more commands any dial plan. Any advice on what to do? Is their another way to start a call but from the CLI or something, we just want to dial part 1, if they pick up/press 1 then it dials part 2, then they bridge the call, pretty easy I would think Any help would be great! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Adtran TA-750 channels go onhook
Hi List, Ive got some of these boxes hooked up to a Digium card running EM wink. Sometimes these channels go onhook for no reason, or when a person hangs the phone up they stay on hook. Are their settings on the channel bank or the card itself I am missing? If anyone has any help or answers on this weird problem please let me know, any help would be great! Thanks ___ -- 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] app_valetparking.c anyone using it on 1.4?
Ahh ok, this makes a little more sense now! I guess it is a different way of looking of it. Thanks On Feb 6, 2008 5:28 AM, Steve Langstaff [EMAIL PROTECTED] wrote: -Original Message- From: Lacy Moore Sent: 06 February 2008 10:54 On Feb 6, 2008 3:46 AM, Steve Langstaff [EMAIL PROTECTED] wrote: ValetParking doesn't announce anything because the whole point of ValetParking is to be able to explicitly park a call at a known spot. I was under the impression that the Valet part of ValetParking meant that you *don't* explicitly park a call at a known spot - the valet takes your call, finds a free spot for it and then tells you where it has been parked. Is that what the builtin function ParkAndAnnounce does? Of course, parking voice may be nothing like parking vehicles :) SInce when has anything dealing with computer systems made sense? :-) Oh, I feel like such a fool! ___ -- 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] Parking Valet
Did you ever get this to work? I can get it to park calls, but it will not announce what parking space all in the same cycle, I can't see how this is useful is you have to transfer to park it, then dial another extension just to see where it was parked! Anyone got this working? On Jul 18, 2007 6:58 PM, Russell Bryant [EMAIL PROTECTED] wrote: Kevin Kiely wrote: app_valetparking listed here http://www.freeswitch.org/asterisk_stuff/ Indicates support for Asterisk 1.4. The documentation listed suggests an install like so: cd /usr/src/asterisk cp contrib/scripts/astxs /usr/bin/ cd apps wget http://www.bkw.org/app_valetparking.c cd .. astxs -install apps/app_valetparking.c However astxs doesn't seem to be present in asterisk 1.4 Does anyone have this working with 1.4? and any suggestions on how to install? astxs won't work with the build system in 1.4, so it's not there anymore. However, if you drop the file into the apps directory, Asterisk will automatically build and install it for you when you run make and make install. -- Russell Bryant Software Engineer Digium, Inc. ___ --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] app_valetparking.c anyone using it on 1.4?
Hi List, I have this running, but after I park a call it will not announce where it is at, it's like you have to call another application just to say where it is parked at. I have tried a second priority option for the same extension with that ValetParkList but it seems once ValetParkCall has been ended it will not process anymore priorities in this extension. Any ideals or help would be great! ___ -- 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] Shared line appearance phones?
Any examples yet Russell? Thanks! On Dec 3, 2007 6:43 PM, shadowym [EMAIL PROTECTED] wrote: That would be VERY much appreciated Russell, There seems to be a lack of info and the accompanying confusion/misinformation about this. -Original Message- From: Russell Bryant [mailto:[EMAIL PROTECTED] Sent: Friday, November 30, 2007 4:11 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Shared line appearance phones? Mark Wiater wrote: I fought with this in 1.4.5 with polycom phones. I was hoping to share a DID from a PRI on several Polycom IP430's. Might you be willing to share some specific configurations for such a situation? There are some basic examples in doc/sla.pdf in the 1.4 tree. However, I have on my to-do list to spend a week with an SLA test environment and coming up with an extensive set of examples of the different ways it can be used. I will post something to this list when that is available. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ --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] Call Parking with multiple lots
Hi List, I need to have one PBX but have multiple call parking for many different context. Basically for hosted VoIP, anyway this can be achineved? We really want to use the Snom's or something like that with a light on the phone so we can what caller is in each parking space/line. I have not seen anyway to do this, any ideals anyone? Thanks! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Parking with multiple lots
How many contexts have you had this running on? And for the ring back, you cant have it park and then on the same call return the info, has to hangup then ring back? Thanks! On Jan 23, 2008 4:48 PM, Darryl Dunkin [EMAIL PROTECTED] wrote: Look at app_valetparking, available here: http://www.freeswitch.org/asterisk_stuff/ I do not know about phone notification (I just use ringback/overhead paging), but it handles multiple contexts just fine. -- *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *Ron McCarthy *Sent:* Wednesday, January 23, 2008 15:39 *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Call Parking with multiple lots Hi List, I need to have one PBX but have multiple call parking for many different context. Basically for hosted VoIP, anyway this can be achineved? We really want to use the Snom's or something like that with a light on the phone so we can what caller is in each parking space/line. I have not seen anyway to do this, any ideals anyone? Thanks! ___ -- 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] Call Parking with multiple lots
I agree it is old, some people won't adopt. We run into this with clients who are to use to legacy key systems. I have found no other real way around this when you need this feature, some way for another person in a office to pick up a call. Its a hassle, wish some people would change! On 1/23/08, Darryl Dunkin [EMAIL PROTECTED] wrote: I've had two live, it's a pretty archaic feature that emulates older PBXs so it isn't a popular feature at all. Just check the source on your options: -= Info about application 'ValetParkCall' =- [Synopsis] Valet Park Call [Description] ValetParkCall(exten|lotname|timeout[|return_ext][|return_pri][ |return_context]) Park Call at exten in lotname until someone calls ValetUnparkCall on the same exten + lotname set exten to 'auto' to auto-choose the slot. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ron McCarthy Sent: Wednesday, January 23, 2008 16:04 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call Parking with multiple lots How many contexts have you had this running on? And for the ring back, you cant have it park and then on the same call return the info, has to hangup then ring back? Thanks! On Jan 23, 2008 4:48 PM, Darryl Dunkin [EMAIL PROTECTED] wrote: Look at app_valetparking, available here: http://www.freeswitch.org/asterisk_stuff/ I do not know about phone notification (I just use ringback/overhead paging), but it handles multiple contexts just fine. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ron McCarthy Sent: Wednesday, January 23, 2008 15:39 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Call Parking with multiple lots Hi List, I need to have one PBX but have multiple call parking for many different context. Basically for hosted VoIP, anyway this can be achineved? We really want to use the Snom's or something like that with a light on the phone so we can what caller is in each parking space/line. I have not seen anyway to do this, any ideals anyone? Thanks! ___ -- 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] Adtran 750 and EM Wink
Hi List, I have some T1 750s; hookup via a T1 to a TE420P card. We have them all set as EM wink trunks. We get a dial tone and can call numbers, but when i call another number the ringing I hear on my phone is very weird, its not the standard ring tone. What will cause this? Is this a * setting or something in the Adtran I am fixing, I just want good ole ringing to be heard! Any help would be great! Thanks ___ -- 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] Shared line appearance phones?
Hi List, What phones support shared line appearance? I would like a phone where we can place calls on a line and have them picked up at another phone, but we don't want to use call parking. I want to use this in a multi tenant environment so I would need multiple lots. Any ideals for me? Thanks! ___ --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: Best firmware for Linksys Router thatis SIP AWARE
The VoIP load includer SER on the router, only difference I am aware of. On Nov 28, 2007 8:44 AM, Dovid B [EMAIL PROTECTED] wrote: - Original Message - From: David Boyd [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, November 26, 2007 4:29 PM Subject: Re: [asterisk-users] OT: Best firmware for Linksys Router thatis SIP AWARE On Mon, 2007-11-26 at 08:08 -0600, Erik Anderson wrote: On Nov 26, 2007 7:51 AM, Dovid B [EMAIL PROTECTED] wrote: Hi, I am currently playing with DD-WRT and I like it. I am looking for something that is more SIP Aware. Anyone know one those that are out there ? Dovid - what exactly are you hoping this sip aware firmware will do that dd-wrt doesn't? I've been using dd-wrt in combination with various SIP ITSPs for several years and have had no problems - just add the necessary port forwards and a few traffic shaping rules and it works just fine. I do know that they (the dd-wrt people) have a voip edition of dd-wrt available. I'm not sure what additional functionality it has over the standard version, though. -erik Erik, I struggle with the traffic shaping rules, would you be willing to provide additional details as to what you have done in past? Any additional information would be greatly appreciated. Thanks, Dave So do I. I set SIP to high how ever the calls are still bad. I guess I need to read up a bit more on the firmware and how to set it up correctly. ___ --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] Shared line appearance phones?
Well we need a light on the phone to blink when a call is on hold, but we want to pick it up from any phone, so its a BLF key/light tied to it. Maybe you can intergrate that with ques, I guess I need to look into that more, just have never heard of that being done! Thanks for the suggestion, Ill take a look. On Nov 28, 2007 8:37 AM, Ricardo Carvalho [EMAIL PROTECTED] wrote: I don't know if I understood you right, but can't that be solved with call queues? http://www.voip-info.org/wiki/index.php?page=Asterisk+call+queues http://www.voip-info.org/wiki/view/Asterisk+RealTime+Queue http://www.voip-info.org/wiki/view/Asterisk+config+queues.conf Regards, Ricardo Carvalho ___ --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] Shared line appearance phones?
Asterisk 1.4 im guessing? I did not know the Snom's worked with that, Ill have to check it out then! Thanks! Brad On Nov 28, 2007 9:28 AM, Russell Bryant [EMAIL PROTECTED] wrote: Ron McCarthy wrote: What phones support shared line appearance? I would like a phone where we can place calls on a line and have them picked up at another phone, but we don't want to use call parking. I want to use this in a multi tenant environment so I would need multiple lots. Any ideals for me? I have successfully used the SLA applications in Asterisk with Polycom, Aastra, and Snom phones. I like working with the Polycom phones the most, but depending on how many buttons you want and how much you're willing to pay, the others may suit your needs better. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ --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] SIP INVITE failing and AgentCallBackLogin()
Hi List, Ive got a few * boxes connecting together, one box is doing AgentCallBackLogin() and then the 2nd box is holding some phones at a remote site. I have users login to the main box and * shows the user is logged into a extension that resides on the other box, problem is, when I go to make a call to a agent, I get May 16 05:59:08 NOTICE[13897]: chan_sip.c:9750 handle_response_invite: Failed to authenticate on INVITE to 'Ex 301 sip:[EMAIL PROTECTED] ;tag=as4e18cbb4' I have a peer setup in the box doing the AgentCallBackLogin() with insecure=very, ive also tried insecure=invite as well, no luck!! Asterisk 1.2.13 I am using on both boxes. Can anyone provide any help on this? I think is rellly weird invites are failing when im telling * to ignore them basically!!! Phones are Snom 360's as well. Thanks! Ron ___ --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] Call Parking is slow with park orbit on Snom 3xx / 360
Hi List, I have a client who is using park heavily, but once we hit the cal button (in this a hotkey tied to park orbit on the Snom's), we have a 3 second delay before we here the digit the call is parked on. Is their anyway around this at all? Does anyone know if we have these same delays if using the DTMF digits? Any suggestions would be great! Thanks! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Snom 360 Caller ID in missed / recieved calls
Hi List, We have noticed on our Snom 360s that under missed/recieved calls the number is cut off, so you cannot see the entire phone number. Does anyone have a work around or is this a bug Snom is working on? Cheers! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AgentCallBackLogin Help!
Hi List! Im using (or trying to) use AgentCallBackLogin() to have * find roaming users, here is a diagram. Server A (Hq) Server B(Branch Site) Server C (Branch Site) All my que users are on Server A, I have Server B/C dial a extension to call AgentCallBackLogin() on Server A, the user logs in and * shows the user logged in at the branch site, all is well! You cann the Agent, it rings the phone on Server B/C, but then follows the next process for that extension on Server B/C, instead I need it to go back to Server A extension flow, (call the agent, if no answer send to AGENTS voicemail, not the phones voicemail). Is there anyway I can get * to send the call back to the main server and continue on with call flow if the agent doesnt pickup at one of the branch sites? Any help on this would be great, ive tried everything I can think of, all wish no such luck!! Also, if someone has done this successfully im willing to pay for help as well :) Thanks!! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Zap Channel Deadlocks
Hey List, Asterisk 1.2.13 with Sangoma Card and beta 14 drivers. I am having problems with deadlock channels and having to kill asterisk, and then restart it, cannot make calls in or outbound. This has happend about 4 times now, and the system was running fine for a few months fine. Any suggestions or comments would be greaet, and im in a world of hurt here! Thanks Brad ___ --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] Zap Channel Deadlocks
I gues ill look and see what version they are on, its a production system, so that always scares me!!! But, good ideal!! :) On 3/8/07, shadowym [EMAIL PROTECTED] wrote: Ummm. How about upgrading to production released drivers? -Original Message- From: Ron McCarthy [mailto:[EMAIL PROTECTED] Sent: Thursday, March 08, 2007 1:44 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Zap Channel Deadlocks Hey List, Asterisk 1.2.13 with Sangoma Card and beta 14 drivers. I am having problems with deadlock channels and having to kill asterisk, and then restart it, cannot make calls in or outbound. This has happend about 4 times now, and the system was running fine for a few months fine. Any suggestions or comments would be greaet, and im in a world of hurt here! Thanks Brad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple parking lot
In order to use this patch, i have to download the complete version of SVN asterisk? I highly doubt this will work with the metermaid patch that allows the call park buttons to work with Snoms. Last time I let anyone share a PBX!! Any comments on this would be great! Thanks Brad On 1/26/07, Olle E Johansson [EMAIL PROTECTED] wrote: 25 jan 2007 kl. 08.26 skrev Darryl Dunkin: There is an SVN branch with this feature: http://svn.digium.com/view/asterisk/team/oej/multiparking/ I had hope this would be a feature added to Asterisk 1.4, but fail to see it on the changelog. It wasn't approved due to some architecture issues. I'll see if I get time to fix them for next release. /O ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Multiple parking lot
Hi list, Does anyone know any ways to have mutiple parking lots? I've got a pbx that 2 customers share, both need their own, and then have lights on the phone flash when they park the call (snom phones). Any ideals I'm not thinking of?!? Any help would be great! Thanks Ron ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Snom has dialtone after putting a person on hold
Hi List, I cant seem to find the setting that changes this! You put a person on hold, they are on hold like normal, but after a few seconds the phone will then start having dialtone coming from the speakerphone, really weird!! Anyone know how to fix this? I see where it could be nice, but in this case, we just want them on hold is all, no dialtone! Any help would be great! Thanks! Ron ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Snom Record / Voice Recorder Button
Hi, This does nothing for me at all! :( I dont have to map the record key to dial *1 or anything? Also, have you figured a way just to make the record button work when not on phone, like a memo button prehaps? Maybe set one speed key as a memo type button to send them a voicemail, and use record button to record calls perhaps. Let me know if you think of anything!! Thanks! Brad On 1/12/07, Ale [EMAIL PROTECTED] wrote: Hi, Ron McCarthy wrote: Hi List, Has anyone got the record button to work on the Snom's? Im looking to have it send a email with a attachemnt of what the user records I hope. It looks like you just point the button to [EMAIL PROTECTED] and just have that extension record it. Any clue on how to do this, just use Record() or Voicemail(s1234). Any suggesstions, any luck with this? You can choose various way to record call with snom, the easy one is: 1)leave the default configuration on snom phone, Function Key - Record, set it as key Event number F_REC 2)insert the following line on features.conf [featuremap] automon = *1 ; One Touch Record 3)update you Dial command on extension.conf add wW modifier, example Dial(SIP/100,30,jrwWt) You can find other infos at: http://www.voip-info.org/wiki/view/Asterisk+config+features.conf Thanks! Ciao Ciao, Ale ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Snom Record / Voice Recorder Button
Intresting, there is no way to have *1 not place the tone while on the phone is it? On 1/12/07, Steve Davies [EMAIL PROTECTED] wrote: I think that the support for the SNOM button (which uses a SIP message to request the recording) is part of the bristuff patch, otherwise only *1 will work. http://www.junghanns.net/downloads/ Cheers, Steve On 1/12/07, Ron McCarthy [EMAIL PROTECTED] wrote: Hi, This does nothing for me at all! :( I dont have to map the record key to dial *1 or anything? Also, have you figured a way just to make the record button work when not on phone, like a memo button prehaps? Maybe set one speed key as a memo type button to send them a voicemail, and use record button to record calls perhaps. Let me know if you think of anything!! Thanks! Brad On 1/12/07, Ale [EMAIL PROTECTED] wrote: Hi, Ron McCarthy wrote: Hi List, Has anyone got the record button to work on the Snom's? Im looking to have it send a email with a attachemnt of what the user records I hope. It looks like you just point the button to [EMAIL PROTECTED] and just have that extension record it. Any clue on how to do this, just use Record() or Voicemail(s1234). Any suggesstions, any luck with this? You can choose various way to record call with snom, the easy one is: 1)leave the default configuration on snom phone, Function Key - Record, set it as key Event number F_REC 2)insert the following line on features.conf [featuremap] automon = *1 ; One Touch Record 3)update you Dial command on extension.conf add wW modifier, example Dial(SIP/100,30,jrwWt) You can find other infos at: http://www.voip-info.org/wiki/view/Asterisk+config+features.conf ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Snom Record / Voice Recorder Button
Hi List, Has anyone got the record button to work on the Snom's? Im looking to have it send a email with a attachemnt of what the user records I hope. It looks like you just point the button to [EMAIL PROTECTED] and just have that extension record it. Any clue on how to do this, just use Record() or Voicemail(s1234). Any suggesstions, any luck with this? Thanks! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Setting outgoing caller id on a zap channel for one sip extension only
Thanks guys for all the help. For this setup I just did a GoToIf(), I will look into multiple context though, looks like thats whats needed for having alot of different outbound caller ids! Thanks again! On 12/6/06, C F [EMAIL PROTECTED] wrote: Asterisk supports whats called context, using a context just for that phone you can set a different callerid, then use a default context for all the other phones. On 12/6/06, Ron McCarthy [EMAIL PROTECTED] wrote: Hi List, Ive got one extension/login that when they call out from that it needs to show a different name/number, and then the rest of the phone will have a default one. Whats the best way to do this? I know it can be done, just cant figure out how! Ive looked around and seem to see no docs on it. Any help or examples would be great on this! Thanks! Ron ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Setting outgoing caller id on a zap channel for one sip extension only
Hi List, Ive got one extension/login that when they call out from that it needs to show a different name/number, and then the rest of the phone will have a default one. Whats the best way to do this? I know it can be done, just cant figure out how! Ive looked around and seem to see no docs on it. Any help or examples would be great on this! Thanks! Ron ___ --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] Setting outgoing caller id on a zap channel for one sip extension only
Hi Rob,
Well see that would work great if I knew the numbers they would be calling,
but all I know is the source number/phone, i have no clue who they will be
calling. Any ideals now? I wish it was that easy!
Thanks!
On 12/6/06, Rob Schall [EMAIL PROTECTED] wrote:
Ron,
I believe you would just want to edit your extensions.conf file so that
the extension you want separate has its own rule set.
exten = 4567,1,Set(CALLERID(all)=000-000-)
exten = 4567,n,Dial(SIP/4567)
all other calls would just fit in like:
exten = _4.,1,Set(CALLERID(all)=111-000-)
exten = 4567,n,Dial(SIP/${EXTEN})
Hope that gives a bit of insight or puts you in the right direction.
Rob
Ron McCarthy wrote:
Hi List,
Ive got one extension/login that when they call out from that it needs
to show a different name/number, and then the rest of the phone will
have a default one. Whats the best way to do this? I know it can be
done, just cant figure out how! Ive looked around and seem to see no
docs on it. Any help or examples would be great on this!
Thanks!
Ron
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Re: [asterisk-users] Setting outgoing caller id on a zap channel for one sip extension only
Yeah,
Bascailly lets say extension 2 places a outbound call, it needs to show
that persons private DID name and number, and anyone else gets the global
callerid name/number. I guess you do this via a if statement, im trying but
having a hell of a time getting it to work!
On 12/6/06, Rob Schall [EMAIL PROTECTED] wrote:
Ron,
By source #, i assume you mean you have something like a SIP phone on
the network with the extension like 4455, and you want that to have a
different caller id when you make outgoing calls, then the rest of the
phones on your network (the rest would show a global company number).
Based on where you put the exten, it works as either an incoming or
outgoing handle. For example, I have it setup on our network, that if
you dial out and connect to our local area code, that the callerid is
one number. If you call long distance, you would see our 1800 number.
Is this the setup you are looking for?
Rob
Ron McCarthy wrote:
Hi Rob,
Well see that would work great if I knew the numbers they would be
calling, but all I know is the source number/phone, i have no clue who
they will be calling. Any ideals now? I wish it was that easy!
Thanks!
On 12/6/06, *Rob Schall* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Ron,
I believe you would just want to edit your extensions.conf file so
that
the extension you want separate has its own rule set.
exten = 4567,1,Set(CALLERID(all)=000-000-)
exten = 4567,n,Dial(SIP/4567)
all other calls would just fit in like:
exten = _4.,1,Set(CALLERID(all)=111-000-)
exten = 4567,n,Dial(SIP/${EXTEN})
Hope that gives a bit of insight or puts you in the right direction.
Rob
Ron McCarthy wrote:
Hi List,
Ive got one extension/login that when they call out from that it
needs
to show a different name/number, and then the rest of the phone
will
have a default one. Whats the best way to do this? I know it can
be
done, just cant figure out how! Ive looked around and seem to see
no
docs on it. Any help or examples would be great on this!
Thanks!
Ron
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Re: [asterisk-users] RE: Snom 360 Multiple calls on hold help
Hi Guys, So the new firmware seems to work great, except. if you hit transfer and then dont hit a key, or dial a extension within literally 2 seconds, the two calls on hold bridge. As you can imagine, chaos!!! Is this a firmware problem, or a setting im missing? Thanks! On 11/27/06, Jamie Heckford [EMAIL PROTECTED] wrote: Thanks again for this new beta release, I couldnt of asked for a quicker response time, my hat is truly off to Snom for actually caring about the customer! I'll 2nd that, we use mainly Snom's now and its mostly down to the fact they provide excellent customer service and support. And they also make very good handsets :) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RE: Snom 360 Multiple calls on hold help
Upgraded the client this morning, lets hope this works for good :) Can you tell us what the best way to suggest bug features or additions is? For example, Arizona is not a option under time zones, we dont have DST and never change time, would be nice if that was added! Also, a digital clock, alot of americans are lazy and I guess still cant figure out how to read a analog clock :( Thanks again for this new beta release, I couldnt of asked for a quicker response time, my hat is truly off to Snom for actually caring about the customer! Thanks again! On 11/23/06, Sven Fischer [EMAIL PROTECTED] wrote: Hi, try our latest beta version 6.5.2 which can be found here: http://www.snom.com/wiki/index.php/Snom360/Firmware/Beta_Versions http://www.snom.com/wiki/index.php/Snom320/Firmware/Beta_Versions http://www.snom.com/wiki/index.php/Snom300/Firmware/Beta_Versions Release Notes: http://www.snom.com/wiki/index.php/Snom360/Firmware/Release_Notes#6.5.2_beta http://www.snom.com/wiki/index.php/Snom320/Firmware/Release_Notes#6.5.2_beta http://www.snom.com/wiki/index.php/Snom300/Firmware/Release_Notes#6.5.2_beta Regards, Sven On Wednesday 22 November 2006 17:56, Ron McCarthy wrote: Yeah, doing more testing shows that the speed keys are broken, but dialing it works!!! Ugg!!! can you let me know if you get a new firmware? Im going to try and downgrade... Thanks! On 11/22/06, Alban [EMAIL PROTECTED] wrote: Yes, already. Waiting now for a new firmware... Regards, Alban Le Mercredi 22 Novembre 2006 13:50, Steve Davies a écrit: On 11/22/06, Alban [EMAIL PROTECTED] wrote: I'm having the same problem, pressing a speed dial/extension when 2 calls are on the phone connect the 2 calls together. Typing the number instead of using speed dial works. With older firmware, 6.2.1 or 6.3, it was working... But then other problem with pickup, deadlocking the phone (or slowing it down). Certainly due to the dp bug (fixed in 6.5.1). Regards, Alban. Has this been reported to snom by anyone? They are generally pretty good at fixing this type of issue and providing beta firmware. Regards, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RE: Snom 360 Multiple calls on hold help
Yeah, doing more testing shows that the speed keys are broken, but dialing it works!!! Ugg!!! can you let me know if you get a new firmware? Im going to try and downgrade... Thanks! On 11/22/06, Alban [EMAIL PROTECTED] wrote: Yes, already. Waiting now for a new firmware... Regards, Alban Le Mercredi 22 Novembre 2006 13:50, Steve Davies a écrit: On 11/22/06, Alban [EMAIL PROTECTED] wrote: I'm having the same problem, pressing a speed dial/extension when 2 calls are on the phone connect the 2 calls together. Typing the number instead of using speed dial works. With older firmware, 6.2.1 or 6.3, it was working... But then other problem with pickup, deadlocking the phone (or slowing it down). Certainly due to the dp bug (fixed in 6.5.1). Regards, Alban. Has this been reported to snom by anyone? They are generally pretty good at fixing this type of issue and providing beta firmware. Regards, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Is this a PRI problem, *, or the phone???
Hey List, Im sure you have all seen my frantic posts today, was going to see if someone else can shed some light on this. I am running * 1.2.13, Zaptel 1.2.13, and beta 12 of the Sangoma drivers with a a101u card. When using Snom 360s,320s (going to test on a Polycom soon as well). I put two calls on hold, pick one call up and go to transfer them out to a extension or voicemail and the two Zap channels bridge, the calls disappear from the phone, and those two callers are now bridged! As you can see, very, very very weird and not good at all. At first we thought this was a Snom problem, but everyone has said that all my configs look good. Could this be some config on the PRI, as far as being the zapata config? Something like when the calls are on hold the system is getting confused? I have one span setup (channels 1-23), and all in the same pickup/call group (group 1). This is correct isnt it? Or should this be setup differently? Anyone have any ideals on this? Ive tried everything, could it be my config in extensions.conf ? If it does with the Polycom as well, im truly confused on what this could be!!! If anyone has any suggestions or comments, please let me know!!! Thanks Ron ___ --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] Termination Rates
Im going to get a trial account, .014 to US is not bad at all!Only downside is that g729 is only codec they allow :(On 9/17/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: I saw this termination company, www.BuyMin.comthe rates looks good. Has anyone any experience with this company? I use Gafachi, very reliable but expensive. ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom programmable buttons
Yeah i was messing around last night and saw that! Now if I can only get the other caller to not hear the DTMF digits id be set! I didnt know you could remap the keys to DTMF digits, but since I can do that this will work perfect for the most part! Thanks for the info!On 9/17/06, Noah Miller [EMAIL PROTECTED] wrote: Hi Ron - Is there a way to program one of the buttons on the 501 (Like the services button) to do on the fly call recording? So in the middle of the phonecall you can record the call without have to do a transfer type of setup. Ive looked at the manual but cant seem how to do that, I only see how you can remap a key function.You can use the asterisk automon feature along with Polycom keyremapping.In features.conf, just set the DTMF code you'd like to useautomon, and then in the Polycom sip.cfg file, remap your Services keyto this DTMF code.The recordings will be placed in /var/spool/asterisk/automon- Noah___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom programmable buttons
Hello list!Is there a way to program one of the buttons on the 501 (Like the services button) to do on the fly call recording? So in the middle of the phonecall you can record the call without have to do a transfer type of setup. Ive looked at the manual but cant seem how to do that, I only see how you can remap a key function. Any help would be great!TIARon ___ --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] Idiot questions
FXO is coming from the PSTN, FXS is what devices connect to (like a analog phone).If you are using VOIP phone then you dont need the FXS modules, just FXO.On 8/24/06, joea, j4computers [EMAIL PROTECTED] wrote: As a complete newcomer to Asterisk, Digium and PBX, I have several questions.But I'll start with this.To setup a simple system with only a couple of POTS lines, I gather I will need a TDM400 board with FXO and/or FXS modules. So, a TDM400 card will support up to two analog (POTS) lines?joea___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Multiple site multi server setup
Hi List, I want to do a layout like this: Corporate-Asterisk01 Site A-Asterisk02Site B-Asterisk03 I will have phones register to each server at each location, and also want to store the users voicemail there. Now here is my question. Can I setup the phones (Polycom I was hoping) to register to the server its connected to, and the main server, so in the event the server it is normally connected is down it could still make interoffice calls and make calls via the PSTN from the main server. Would this be done with RegOp() function or something else? Also then all the servers would have to be connected via NFS due to the fact of the voicemails wanting to be stored on another machine while its primary is down, or is this not even possible? Any help would be great! Thanks Ron ___ --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] Clustering
Regarding OSPF, so your saying you have multiple * boxes setup with same exact config and then just have OSPF fail everthing over to the new server if it cant get to it? That makes sense, just never of even thought of doing it that way. Heck, if you want to get real complex just run BGP and you could then setup priorties for each server and all kinds of cool stuff. Are you then using regexten on all servers so when a * tries to make a call it can find where to go, or are you using something else? Thanks! RonOn 3/12/06, Douglas Garstang [EMAIL PROTECTED] wrote: It doesn't. It's transparent to the user agent.-Original Message-From: Wai Wu [mailto:[EMAIL PROTECTED]]Sent: Sunday, March 12, 2006 9:40 AMTo: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] ClusteringHow does OSPF tell the remote end (assuming he does not know your setup) start sending RTP packets to the other interface?-Original Message-From: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED]]On Behalf Of Douglas GarstangSent: Sunday, March 12, 2006 1:41 AMTo: Asterisk Users Mailing List - Non-Commercial Discussion; AsteriskUsers Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] Clustering No, only if a network interface in the server fails. We have two network interfaces per system (actually we have four, but two are on a private network with a MySQL server). If one of the network interfaces fails, OSPF will switch the default route over to the other interface pretty quick smart. There's probably a little luck involved here too.-Original Message-From: Gabriel Afana [mailto:[EMAIL PROTECTED]]Sent: Sat 3/11/2006 10:07 PM To: Asterisk Users Mailing List - Non-Commercial DiscussionCc:Subject: Re: [Asterisk-Users] ClusteringSo you are actually able to maintain a call in progress even if the server its connected to fails (by routing to another)?- Gabe- Original Message -From: David Coulson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.comSent: Saturday, March 11, 2006 7:15 PM Subject: Re: [Asterisk-Users] Clustering From what I can find online, OSPF seems to be a technology ormethod, not necessarily a program.What are you using to perform OSPF? OSPF is a routing protocol. Quagga (quagga.net) is a good open source implementation of OSPF for Unix. David ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Clustering
Hi Gabe, Well I was going to use the SBC to have all phone point to the SBC, and then the SBC takes care of what servers it needs to register with, and then keep a state of what server the RTP stream and the phone need to connect to. Basically like a load balancer would. This is what I understood from Juniper's site. Have you seen anything on this? Thanks! RonOn 3/11/06, Gabriel Afana [EMAIL PROTECTED] wrote: Hi Ron, I've been following your thread. I noticed you mentioned about a Juniper Session Border Controller. I checked online and read about it, but was unsure exactly how it could intergrate with Asterisk. How would you have planned to use that device? I am interested because one of my upstream providers mentioned I should be using an SBC. - Gabe - Original Message - From: Ron McCarthy To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Friday, March 10, 2006 11:22 AM Subject: [Asterisk-Users] Clustering Hello All,Ive been doing more and more research on trying to setup a cluster/load balancer for Asterisk. All the Asterisk boxes would be using a config that is the same between them all (via a DB), but we want one location to point the phones to, and from there that machine/device will send it to a Asterisk server so the call can be processed. I know you cant balance the whole call, ie: once the call is started the RTP stream has to go to the same server, but a new call could go to a different server if perhaps the 1st server was unreachable.Has anyone tried this, or got this to work? Ive been looking at using a Juniper Session Border Controller, but not sure if thats gonna do the trick, and then we also have SER..Any comments would be great!ThanksRon ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Clustering
Why even bother with OSPF, why not just run 802.1ad or a LAG on the two NIC's? But anyways, so Ser just shows the phones registered with all the servers that you have? I guess im getting confused, cause when you go to ring the phone what Asterisk box does it use, just the 1st one there is (since the phone is registered on all of them...) Thanks! RonOn 3/12/06, Douglas Garstang [EMAIL PROTECTED] wrote: Uhm, No. We have multiple Asterisk boxes. OSPF only fails over between interfaces in a single Asterisk system. We're not using regexten (cuz there's no frikkin docs for it!!!). We're using OpenSER's send() command to forward registrations from a phone to all Asterisk systems. -Original Message-From: Ron McCarthy [mailto:[EMAIL PROTECTED]]Sent: Sunday, March 12, 2006 1:29 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] ClusteringRegarding OSPF, so your saying you have multiple * boxes setup with same exact config and then just have OSPF fail everthing over to the new server if it cant get to it? That makes sense, just never of even thought of doing it that way. Heck, if you want to get real complex just run BGP and you could then setup priorties for each server and all kinds of cool stuff.Are you then using regexten on all servers so when a * tries to make a call it can find where to go, or are you using something else?Thanks!Ron On 3/12/06, Douglas Garstang [EMAIL PROTECTED] wrote: It doesn't. It's transparent to the user agent.-Original Message-From: Wai Wu [mailto:[EMAIL PROTECTED]]Sent: Sunday, March 12, 2006 9:40 AMTo: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] ClusteringHow does OSPF tell the remote end (assuming he does not know your setup) start sending RTP packets to the other interface?-Original Message-From: [EMAIL PROTECTED][mailto: [EMAIL PROTECTED]]On Behalf Of Douglas GarstangSent: Sunday, March 12, 2006 1:41 AMTo: Asterisk Users Mailing List - Non-Commercial Discussion; AsteriskUsers Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] ClusteringNo, only if a network interface in the server fails. We have two network interfaces per system (actually we have four, but two are on a private network with a MySQL server). If one of the network interfaces fails, OSPF will switch the default route over to the other interface pretty quick smart. There's probably a little luck involved here too.-Original Message-From: Gabriel Afana [mailto:[EMAIL PROTECTED]]Sent: Sat 3/11/2006 10:07 PM To: Asterisk Users Mailing List - Non-Commercial DiscussionCc:Subject: Re: [Asterisk-Users] ClusteringSo you are actually able to maintain a call in progress even if the server its connected to fails (by routing to another)?- Gabe- Original Message -From: David Coulson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Sent: Saturday, March 11, 2006 7:15 PM Subject: Re: [Asterisk-Users] Clustering From what I can find online, OSPF seems to be a technology ormethod, not necessarily a program.What are you using to perform OSPF? OSPF is a routing protocol. Quagga (quagga.net) is a good open source implementation of OSPF for Unix. David ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http
Re: [Asterisk-Users] Clustering
Hi Gabe, Well im guessing your ServerIron wont work because its not near smart enough to know how SIP works, let alone, 99.9% of load balancers I have seen use private IP's on server side, the and Load Balncer then has the public IP's assigned to them. Right there, this creates a problem in itself. But im assiumg the Foundry isnt smart enough to keep track of multiple phones from the same IP, and all the RTP sessions associated with it, since like you said, several hundred port numbers are being used. The Juniper box seems to rewrite the actual SIP header on the outbound transversal to the Internet, this solving the NAT return path problem, and then it keeps track in a state table as to what ports go to what server, etc, etc. But I think there is no way this could failvoer in the middle of the car, since it would somehow have to change the RTP stream to another port, but also the phone would have to get to get registered on that server as well, which its not, which is why Douglas is using SER to have it register on several different machines, so when the failover occurs the phone is registered and the RTP stream just needs to pick up. Im trying to see exactly how he is doing this, since this is the exact thing I need, and then Ill just run OSPF on my core router (not sure if that will work yet). I woudl perfer to do this all in hardware vs software since a Cisco/Juniper box is musch less prone to failure then a server with software, but I guess more research will tell what ill be using in the end :) Once I get this going, I want to post a entire howto on the wiki. Thanks! RonOn 3/12/06, Gabriel Afana [EMAIL PROTECTED] wrote: Hi Ron, If the SBC would have served mearly as a load balancer...I already have one and it didn't work too well. I have a Foundry ServerIron XL load balancer and I've tried using it with Asterisk. It has had positive and negative results. Positive: It *would* load balance between asterisk servers for whatever port I set (I was using 5060 for SIP). However, I didn't mess with the RTP because its got so many ports (and you can't add ranges for virtual server ports, you have to enter exact ports - at least I think) and because I have no idea how that would work if SIP signaling goes to one server and RTP goes to another??? (probably not!) I would create a virtual IP on the load balancers and have all the phones register to this IP. When checking status of the ports on each server, it showed 5060 for all servers was unused (0 current connections). When I would make a call, it would show the 5060 port on one of the * servers in use (1 current connections) and it worked finethis is where the problem started. Negative: When I would hang up the phone, it would still show 1 current connection to that server's 5060 port. Every call I would make from then on would *still* go to that same server. It seems the ports are sticky or set with a keepalive. Of course I can define these options on the ServerIron, but even with sticky disabled and keepalive disabled, the port would appear active (like keepalive was enabled) and every call would go to the same server (like sticky was enabled). Even if I would shutdown asterisk on that server, it would still show an active user on that port and when I would make the call, the call would not go through. The LB was not failing the port. I think maybe if I keep playing with it...? Any suggestions? If I can get my ServerIron working, I will do a complete write up on it...but it works only partially. This is why I was so interested in the Juniver SBCif it would be able to act a proxy, do all the load balancing and instantly failover if a server fails; basically a VoIP Load Balancer. But I guess thats not what it does. Does a VoIP load balancer hardware exist or is the only solution right now software proxies like SER? - Gabe - Original Message - From: Ron McCarthy To: Gabriel Afana ; Asterisk Users Mailing List -Non-Commercial Discussion Sent: Sunday, March 12, 2006 1:16 PM Subject: Re: [Asterisk-Users] Clustering Hi Gabe,Well I was going to use the SBC to have all phone point to the SBC, and then the SBC takes care of what servers it needs to register with, and then keep a state of what server the RTP stream and the phone need to connect to. Basically like a load balancer would. This is what I understood from Juniper's site. Have you seen anything on this?Thanks!Ron On 3/11/06, Gabriel Afana [EMAIL PROTECTED] wrote: Hi Ron, I've been following your thread. I noticed you mentioned about a Juniper Session Border Controller. I checked online and read about it, but was unsure exactly how it could intergrate with Asterisk. How would you have planned to use that device? I am interested because one of my upstream providers mentioned I should be using an SBC. - Gabe
Re: [Asterisk-Users] Clustering
* go to that same server. It seems the ports are sticky or set with a keepalive. Of course I can define these options on the ServerIron, but even with sticky disabled and keepalive disabled, the port would appear active (like keepalive was enabled) and every call would go to the same server (like sticky was enabled). Even if I would shutdown asterisk on that server, it would still show an active user on that port and when I would make the call, the call would not go through. The LB was not failing the port. I think maybe if I keep playing with it...? Any suggestions? If I can get my ServerIron working, I will do a complete write up on it...but it works only partially. This is why I was so interested in the Juniver SBCif it would be able to act a proxy, do all the load balancing and instantly failover if a server fails; basically a VoIP Load Balancer. But I guess thats not what it does. Does a VoIP load balancer hardware exist or is the only solution right now software proxies like SER? - Gabe - Original Message - From: Ron McCarthy To: Gabriel Afana ; Asterisk Users Mailing List -Non-Commercial Discussion Sent: Sunday, March 12, 2006 1:16 PM Subject: Re: [Asterisk-Users] Clustering Hi Gabe,Well I was going to use the SBC to have all phone point to the SBC, and then the SBC takes care of what servers it needs to register with, and then keep a state of what server the RTP stream and the phone need to connect to. Basically like a load balancer would. This is what I understood from Juniper's site. Have you seen anything on this?Thanks!Ron On 3/11/06, Gabriel Afana [EMAIL PROTECTED] wrote: Hi Ron, I've been following your thread. I noticed you mentioned about a Juniper Session Border Controller. I checked online and read about it, but was unsure exactly how it could intergrate with Asterisk. How would you have planned to use that device? I am interested because one of my upstream providers mentioned I should be using an SBC. - Gabe - Original Message - From: Ron McCarthy To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Friday, March 10, 2006 11:22 AM Subject: [Asterisk-Users] Clustering Hello All,Ive been doing more and more research on trying to setup a cluster/load balancer for Asterisk. All the Asterisk boxes would be using a config that is the same between them all (via a DB), but we want one location to point the phones to, and from there that machine/device will send it to a Asterisk server so the call can be processed. I know you cant balance the whole call, ie: once the call is started the RTP stream has to go to the same server, but a new call could go to a different server if perhaps the 1st server was unreachable.Has anyone tried this, or got this to work? Ive been looking at using a Juniper Session Border Controller, but not sure if thats gonna do the trick, and then we also have SER..Any comments would be great!ThanksRon ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Clustering
I had no ideal this thread would get this big! Im going to look more into the regcontext. I planned on using DUNDi since im going to have lots of * servers that will be tied over a ATM backbone, so DUNDi can then do on-net calling much easier and never have to hit the internet, can just ride our private backbone. If anyone has more examples, please post, im gonna to start with some mock configs this weekend I hope, and start testing. There has be to a way to achieve N+1 redudancy, its just a matter of getting it all to play nicely :) I will be glad to post all steps needed to this to a wiki, so we can share this setup. Thanks! RonOn 3/10/06, Douglas Garstang [EMAIL PROTECTED] wrote: Hi JR. I'm dying to know... where'd you find your DUNDi documentation? Has something new appeared since I looked at it 2-3 months ago? The O'Reilly book's DUNDi section was impossible to follow, and the examples in the Asterisk DUNDi config files are no better. You do a search online and get almost no results (still wondering when Digium is going to realease some docs for what they call their protocol). I spent a few weeks working on it, tearing my hair out, and gave up. So did my boss.Doug.-Original Message-From: JR Richardson [mailto:[EMAIL PROTECTED]]Sent: Fri 3/10/2006 8:55 PMTo: [EMAIL PROTECTED]; asterisk-users@lists.digium.comCc: [EMAIL PROTECTED] Subject: re: [Asterisk-Users] Clustering--Message: 6Date: Fri, 10 Mar 2006 12:22:12 -0700From: Ron McCarthy [EMAIL PROTECTED]Subject: [Asterisk-Users] ClusteringTo: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.comMessage-ID:[EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1Hello All,Ive been doing more and more research on trying to setup a cluster/loadbalancer for Asterisk. All the Asterisk boxes would be using a config that is the same between them all (via a DB), but we want one location to pointthe phones to, and from there that machine/device will send it to a Asteriskserver so the call can be processed. I know you cant balance the whole call, ie: once the call is started the RTP stream has to go to the same server,but a new call could go to a different server if perhaps the 1st server wasunreachable.Has anyone tried this, or got this to work? Ive been looking at using a Juniper Session Border Controller, but not sure if thats gonna do the trick,and then we also have SER..Any comments would be great!ThanksRon-- next part -- Ron,I'm doing something similar with clustering.I haven't gotten the total design down yet but so far I have 1 * server holding all the routes to several child/registration * servers where my iax and sip clients actually register to.I'm using the 1 * server running dundi to peer with all the registration servers and all the registration servers only peer with the 1 * server, I call this the Registration Presence Server or RPS.When a child/registration server does a lookup request to the RPS, the RPS does a lookup on all the other registration servers and knows who is registered where and relays that info back to the original requesting registration server.Use a dundi ttl=1 in the RPS and ttl=2 in each registration server to avoid routing loops.I'm still in the testing phase but it's going well, but I'm running into some cache timeout issues when a client drops off and re-registers to a different registration server, have to flush dundi to pickup the new location.There is a cache timeout parameter I have yet to play with.I don't have the load balancing session border controller function down yet, but that is on the list of things to do.Hope this helps.JRJR RichardsonEngineering for the Masses___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Clustering
Hello All, Ive been doing more and more research on trying to setup a cluster/load balancer for Asterisk. All the Asterisk boxes would be using a config that is the same between them all (via a DB), but we want one location to point the phones to, and from there that machine/device will send it to a Asterisk server so the call can be processed. I know you cant balance the whole call, ie: once the call is started the RTP stream has to go to the same server, but a new call could go to a different server if perhaps the 1st server was unreachable. Has anyone tried this, or got this to work? Ive been looking at using a Juniper Session Border Controller, but not sure if thats gonna do the trick, and then we also have SER.. Any comments would be great! Thanks Ron ___ --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] Is everyone getting mails except me?
I havent got any mails since 2:42 this morning..usually i get at least the normal 10-15 a hour, if someone gets this can they reply? Thanks! Ron ___ --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] Lowering Server Load
Also, SATA on a onboard SATA card will eat more CPU then a SCSI system. Are you running software RAID by chance with your SATA? SCSI or SCSI Raid will not each CPU near as much since the HBA does all the work and does tie up the CPU with all its I/O's. We have successfulyl recorded 5+ calls at a time via dual xeon 3.0 with 10K SCSI drives in RAID-5 with no issuses running about 30 PRI channels and anywhere from 50-75 SIP channels, all with g729 encoding. Hope this helps! RonOn 3/2/06, Anton Krall [EMAIL PROTECTED] wrote: Yep, I tried it and indeed, it lowers cpu usage, so I switched from wav togsm format and Im thinking about doing the ramdisk solution for recording...Sounds like a good move?|-Original Message- |From: [EMAIL PROTECTED]|[mailto:[EMAIL PROTECTED]] On Behalf Of |Matt Riddell [NZ]|Sent: Thursday, March 02, 2006 2:04 AM|To: Asterisk Users Mailing List - Non-Commercial Discussion|Subject: Re: [Asterisk-Users] Lowering Server Load||Can you try not recording for a bit and see if that helps? ||--|Cheers,||Matt Riddell|___||http://www.sineapps.com/news.php (Daily Asterisk News - html)|http://freevoip.gedameurope.com (Free Asterisk Voip Community) |http://www.sineapps.com/rssfeed.php (Daily Asterisk News -|rss) ___|--Bandwidth and Colocation provided by Easynews.com --| |Asterisk-Users mailing list|To UNSUBSCRIBE or update options visit:| http://lists.digium.com/mailman/listinfo/asterisk-users|| ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Topology
Hello again, Well all out * boxes will be on a channelized DS3 running back to the main * server, so I guess we could actaully have it be extra smart and have it route directly to the other * box (not the main one) if it needs to make a inter-branch call it can. Ill have to look around into DUNDI somemore then, I figure it would be playing a part into this! Thanks for the help! RonOn 2/25/06, yusuf [EMAIL PROTECTED] wrote: Ron McCarthy wrote: Hi List, Im planning on setting up asterisk for a large scale enviorment, with multiple sites. We will be doing quite a bit of inner office calling at each site, and want to place a smaller scale * box at each site with no PRI's, and have that connect to our main * servers at our data center that will have the PRI connections. Can this be done? I havent seen to much of this on the mailing list, im guessing each server would talk to the main * server via a IAX trunk or a SIP peer. Also one other key point would then be to keep the voicemail for each office on its local * server instead of having it go to the data center. My main concern is the dialplan, I guess if the peer is not local it would then go out the IAX or SIP gateway to the main * server and then check in its dial plan/routing table there, correct? Any help/suggesstion on this would be great! Thanks RonHi Ron,Yes, all of this can be done without too much hassle.I am assuming ateach local site you will have IP/soft phones connected to that asterisk box, so each local astreisk box will have it own local users. Yes, thanThe voicemail will sit on each local asterisk, becuase the user isdefined there.Your concern with the dialplan is, i assume, is what numbers are local numbers, inter branch numbers, and what numbers must go the the main *server.If you use DUNDI, this will sort out the interbranch numbers,since each * box will advertise its local numbers to other asterisk servers.One question, can each of the local @ servers 'see' each other, or canthey only 'see' the main server, because this will change your dialplan.yusuf___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Topology
Hi List, Im planning on setting up asterisk for a large scale enviorment, with multiple sites. We will be doing quite a bit of inner office calling at each site, and want to place a smaller scale * box at each site with no PRI's, and have that connect to our main * servers at our data center that will have the PRI connections. Can this be done? I havent seen to much of this on the mailing list, im guessing each server would talk to the main * server via a IAX trunk or a SIP peer. Also one other key point would then be to keep the voicemail for each office on its local * server instead of having it go to the data center. My main concern is the dialplan, I guess if the peer is not local it would then go out the IAX or SIP gateway to the main * server and then check in its dial plan/routing table there, correct? Any help/suggesstion on this would be great! Thanks Ron ___ --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
