Re: [asterisk-users] SLA Appearance between 2 Cisco 7960's (SIP)
Speaking as someone who did an installation with quite a few Cisco phones running sip, they do not support hinting at all with the sip firmware. The latest version of the Cisco phones (79x1) are primarily sip phones, however with the CCM specific extensions, they tend to be fairly difficult to get to work with asterisk. I wouldn't recommend banging your head on the wall trying, took a while for the bump on my head to go away. :) Anyone trying to get this to work would be better off getting a phone that does support hinting, and has plenty of documentation on how to do it (Polycom, Aastra, SNOM, etc). Jason Howk wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I run the phone with sip firmware so I can confirm it does. ;) Actually the G means global and replaces the actual text on the buttons with icons instead. The gigabit interfaces come on the later - -GE models. My question was more directed to if anyone has gotten SIP hints to work on the older 7960s at all. Looks like I might just have to give the new snom 370 a try... - --J. On Apr 25, 2007, at 7:59 PM, Brad Sumrall wrote: I am very confident the 7960G has a sip load. I know for sure the regular 7960 does and the G just means gigabit interface. The 7970 was the only one that didn't because of all the color interface/touch screen, and Cisco was still pushing call manager big time, so skinny was the only load available. If you log into cisco.com, they have it under software. Sometimes people post it on the internet. Asterisk is supposed to be more skinny friendly these days. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jason Howk Sent: Wednesday, April 25, 2007 7:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SLA Appearance between 2 Cisco 7960's (SIP) From reading the SLA docs, SIP hints are use to get the lights on the phone to show the correct state. I was under the impression that the SIP firmware on the 7960's didn't support the SIP hints properly (or at all), which means that SLA won't work properly on a 7960. If anyone has gotten this to work, I'd like to hear about it. --Jason. John C. Wolosuk Jr. wrote: Has anyone had any success with getting SLA going between 2 SIP phones? (Particularly a set of Cisco 79xx's) The SLA document that comes with the asterisk source is about as clear as mud. Does anyone have a working sip.conf, sla.conf, and extensions.conf that I can use for reference? The part I'm most confused about is how to build the lines in sip.conf and how the phones should behave. It seems apparent that the phones should not register with asterisk, otherwise all the phones will try to register to be THE phone for a given extension. should these lines be built like a trunk/peer? if I could be an example of how lines for SLA should look in sip.conf, that would be helpful. Also I'm somewhat annoyed that I have to compile zaptel drivers that I don't use in order to compile the app_meetme.so module so I can have the SLA functions available to the dialplan... Any feedback is greatly appreciated! ___ --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 -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (Darwin) iD8DBQFGMD8nyocPzc/H1dsRAtlVAJ4gLjMENCyW2wDFMhxMRO6eIX76yQCdESBt G83ykWxG1EWcxLNqZfyp5ME= =RVLT -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Aaron Daniel Community Relations Specialist [EMAIL PROTECTED] (256) 428-6010 ___ --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] SLA Appearance between 2 Cisco 7960's (SIP)
Russell Bryant wrote: John C. Wolosuk Jr. wrote: Has anyone had any success with getting SLA going between 2 SIP phones? (Particularly a set of Cisco 79xx's) The SLA document that comes with the asterisk source is about as clear as mud. Mud, huh? I guess I should work on that at some point, then ... You say two phones. What do you intend to use on the trunk side? I assume you want a SIP trunk. Does anyone have a working sip.conf, sla.conf, and extensions.conf that I can use for reference? sip.conf: This is configured just like any other SIP device. In your scenario of two SIP phones and one SIP trunk, sip.conf would contain three entries. For example: [station1] type=friend secret=station1 host=dynamic context=sla_stations dtmfmode=rfc2833 disallow=all allow=ulaw [station2] type=friend secret=station2 host=dynamic context=sla_stations dtmfmode=rfc2833 disallow=all allow=ulaw [providerA] type=friend secret=something host=providerA.com context=line1 dtmfmode=rfc2833 disallow=all allow=ulaw sla.conf: (From sla.pdf, page 7) Here you create a definition for a single line and two stations. [line1] type=trunk device=Local/[EMAIL PROTECTED] [station](!) type=station trunk=line1 [station1](station) device=SIP/station1 [station2](station) device=SIP/station2 extensions.conf: [line1] ; This is used for incoming calls from SIP/providerA because providerA ; has context=line1 in sip.conf. Incoming calls immediately go into the ; SLATrunk application. Then, the appropriate stations will start ; ringing. exten = s,1,SLATrunk(line1) [line1_outbound] ; This context is used by the SLA code. line1 in sla.conf was ; configured to use a device called Local/[EMAIL PROTECTED] ; That means that when someone presses the line button for line1, ; it will get connected to Disa. Disa will provide dialtone and ; allow the caller to dial any other extensions that live in this ; context. In this case, there is only one available pattern. When ; it gets dialed, the call goes out to SIP/providerA. exten = disa,1,Disa(no-password|line1_outbound) exten = _1NXXNXX,1,Dial(SIP/[EMAIL PROTECTED]) [sla_stations] ; These extensions are called by the stations . ; This extension should be called when the the phone for ; SIP/station1 is taken off hook without pressing a line button. exten = station1,1,SLAStation(station1) ; This extension should be called when the user presses the ; line1 key on the phone. exten = station1_line1,1,SLAStation(station1_line1) ; The line1 key on the phone for station1 should be configured ; to subscribe to the state of the extension station1_line1. ; This will allow Asterisk to control the light to make it turn ; on, off, or blink, as appropriate. exten = station1_line1,hint,SLA:station1_line1 exten = station2,1,SLAStation(station2) exten = station2_line1,hint,SLA:station2_line1 exten = station2_line1,1,SLAStation(station2_line1) The part I'm most confused about is how to build the lines in sip.conf and how the phones should behave. It seems apparent that the phones should not register with asterisk, otherwise all the phones will try to register to be THE phone for a given extension. should these lines be built like a trunk/peer? if I could be an example of how lines for SLA should look in sip.conf, that would be helpful. Actually, the phones *do* register to Asterisk. But, the line appearance buttons themselves are not registrations to Asterisk. They are simply subscribers to the state of extensions. You set these up just like you would for any other hint in Asterisk. Just an FYI, Cisco phones running SIP do *not* do shared line appearances, on *any* system. -- Aaron Daniel Community Relations Specialist [EMAIL PROTECTED] (256) 428-6010 ___ --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] New Community Blogs
Hello all, I'd like to introduce you to a new feature that we're opening up for all users on the AsteriskNOW.org website. Anyone with an account can now post a blog on the front page. This feature will give you the opportunity to post stories about what you've done with Asterisk and AsteriskNOW for everyone to see. After you log into AsteriskNOW.org, you will see a box on the right, containing a link that says create content. Click there, then blog entry on the main page. Fill out the boxes, and click submit, and then you're done. In the process of adding this feature, we have updated the Communications Rules located at http://www.asterisk.org/community/rules, so check those out before posting. Thanks, Aaron Daniel Community Relations Specialist [EMAIL PROTECTED] (256) 428-6010 ___ --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] AsteriskNOW Migration
That is incorrect. AsteriskNOW (actually, the AsteriskGUI) edits files in place, leaving any old information in them. This allows you to fully customize your users and dialplan without interfering with the GUI's operation. Aaron Daniel Community Relations Specialist [EMAIL PROTECTED] (256) 428-6010 - Original Message - From: Tzafrir Cohen [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, February 12, 2007 2:51:21 PM GMT-0600 US/Central Subject: Re: [asterisk-users] AsteriskNOW Migration On Mon, Feb 12, 2007 at 02:06:31PM -0500, Michelle Dupuis wrote: I would suggest you grab the menu from the .conf file and paste it into the new setup. (After even a little asterisk experience, they should be able to get away from the gui). Note that confiugration of AsteriskNow rewrites extensions.conf, and thus an #include of an external file will not work as planned. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] 2 devices using same sip account
We use this function regularly (you should see my phone's dialstring...). If one phone responds that it's unavailable, the rest of the phones will still ring through. In the event that none of the other phones are answered, the extension is considered unanswered, so depending on how you program your dialplan, the call will go to the unavailable voicemail. If you watch the CLI in this situation, you'll see Asterisk try all the devices in the group at the same time, and it'll just bypass any devices that are unavailable. Also, the problem with multiple phones registering with Asterisk at the same name is that Asterisk only stores the information about the device once, and is overwritten with each subsequent register. If you have a softphone and a hardphone both registered, whichever one has a faster re-register rate will win out over the slower one. The only way around this is through the call groups, as several people have stated. Aaron On Tue, 2006-12-19 at 10:55 -0800, Carla Schroder wrote: Hmm, I don't know what happens when one of the lines is busy and none of the lines get answered. It's easy enough to test. If it doesn't go to voicemail, then perhaps this is what you want: http://www.voip-info.org/wiki/view/Asterisk+tips+findme On Tuesday 19 December 2006 9:58 am, René Enskat wrote: how isit possible to get the VM there when one line is busy? regards rene On Tue, 19 Dec 2006 09:48:01 -0800 Carla Schroder [EMAIL PROTECTED] wrote: Your phones only register once, when they first start up. Seems to me that having multiple phones on the same account is asking for trouble- why not set up multiple accounts in the usual way, and create a ring group for all the phones you want to use? Like this example that rings two phones at the same time: exten = 100,1,Dial(SIP/101SIP/102,30,t) exten = 100,2,VoiceMail([EMAIL PROTECTED]) There are all kinds of fancy variations on this theme, but the idea is the same: one user with many phones, one extension, one voicemail box. -- Aaron Daniel Senior Voice Analyst Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --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] MOH Between Asterisk Servers
Set the musiconhold class on the original server if you want the MOH to match up correctly. Both servers notice that the call is on hold... makes sense to me. On Fri, 2006-12-15 at 14:56 -0700, Douglas Garstang wrote: Scenario: A call is sent from one Asterisk system to another with IAX. The remote Asterisk system runs the Queue application, which then starts to play a different music on hold class then the standard 'default'. The console on this system displays: -- Executing Queue(IAX2/xxx.yyy.142.203:4569-4, demo_QMain|t|||60) in new stack -- Started music on hold, class 'demo_MainOffice', on IAX2/xxx.yyy.142.203:4569-4 -- Called SIP/2943367 -- Called SIP/2943368 -- SIP/2943367-1bb8 is ringing -- SIP/2943368-537f is ringing However, on the first Asterisk system, we see this on the console: -- Called dundiapps:[EMAIL PROTECTED]/demo_EMain -- Call accepted by xxx.yyy.142.204 (format g729) -- Format for call is g729 -- Started music on hold, class 'default', on IAX2/xxx.yyy.142.203:4569-5 The music on hold class in use is not being conveyed back to the original Asterisk system. Please don't tell me this is a limitation. That would be very very bad. Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Aaron Daniel Senior Voice Analyst Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --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] anyone used vitelity?
We're using vitelity, not in large scale call center type numbers, but any long distance numbers we dial go out their system. They've been working great, but if you expect support for an asterisk system, don't bother calling them. The furthest they'll go is telling you that there are configs on the web and if you're not using a regular IP phone, they can't help you. We did have a hiccup with them yesterday, but other than that, calls are clear and seem to succeed well. On Wed, 2006-12-13 at 08:54 -0600, Curt Shaffer wrote: Just emailing the list to see if anyone out there has used Vitelity? If so what has been your experience with service, support etc? Thanks Curt ___ --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 -- Aaron Daniel Senior Voice Analyst Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --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] Pickup application
Does anyone have the pickup application working? I'm attempting to get it so that a particular extension programmed into a phone can be picked up by another phone with that extension programmed with a speed dial with a 'p' in front... basically, if you dial p100 and extension 100 is ringing, it'll pick up that extension, otherwise it dials the number. The problem I'm having is in the fact that my phones register with mac addresses instead of extensions, so I'm unsure as to what to put in the pickup app args. I've tried mac, extension, sip device name, etc... no luck. Anyone have any ideas? -- Aaron Daniel Senior Voice Analyst Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --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] PCI, PCI-X and PCI-e -- Server / Interface Card Selection
The Digium TE410P base card does indeed work in PCI-X slots. We're using two of the TE412P's in a PCI-X server with no problems :) On Sun, 2006-12-10 at 00:10 -0500, Time Bandit wrote: I can't risk spending a few thousand just to reach the conclusion that Digium's PRI or BRI cards do not work with a particular system's PCI-X slots/bus... Or, worse, staying with a dead card / system board in my hands ! :-( Anyone ? I don't know about Digium cards, but I just installed a Sangoma A101 card into an IBM server in a PCI-X slot and it is working perfectly. You should ask Digium hth ___ --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 -- Aaron Daniel Senior Voice Analyst Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --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] CLI History
On Mon, 2006-12-11 at 11:17 -0800, Carla Schroder wrote: On Monday 11 December 2006 9:31 am, Douglas Garstang wrote: What's wrong with the Asterisk CLI history? When I exit the CLI, and re-enter, the last command in the history always defaults to 'stop now'. This is very bad, and it's caused accidental shutdowns more than once. Connected to Asterisk 1.2.9.1 currently running on hera (pid = 17399) Verbosity is at least 3 hera*CLI A No such command 'A' (type 'help' for help) hera*CLI B No such command 'B' (type 'help' for help) hera*CLI C No such command 'C' (type 'help' for help) hera*CLI D No such command 'D' (type 'help' for help) hera*CLI E No such command 'E' (type 'help' for help) hera*CLI [10:[EMAIL PROTECTED](pbx3):~]# asterisk -trv Asterisk 1.2.9.1, Copyright (C) 1999 - 2006 Digium, Inc. and others. Created by Mark Spencer [EMAIL PROTECTED] Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'show license' for details. = Connected to Asterisk 1.2.9.1 currently running on hera (pid = 17399) Verbosity is at least 3 hera*CLI stop now -- I pressed the UP arrow upon re-entering the console! I'm a bit confused by your example. What are A,B,C, etc? To exit the Asterisk console, I type 'exit'. Asterisk continues to run, as it should. To re-enter the console I use asterisk -rvvv. He was demonstrating how the CLI history shows stop now as the last command (which um... it's a history? you're last command is gonna be the um... last command you ran... i.e. stop now). Douglas, why're you even running stop now on a live production server. If you're not, quit complaining and watch what you type before hitting enter. -- Aaron Daniel Senior Voice Analyst Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --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] MWI across multiple servers
We've got a setup similar to that. Depending on how you want to set it up, just use externnotify and a script that touches the msgnum.txt file in the user's vm directory on the other boxes. We're using ssh, you may choose to use a different method. It's an immediate MWI notification, and seems to work well. If you're interested, let me know, I'll shoot the scripts over to you. On Wed, 2006-12-06 at 09:20 -0700, Porier, Jeremy M. wrote: We are about to deploy six Asterisk servers across the state with SIP phones at each site registering to their local server. However, we are centralizing voicemail at our main campus to enable the transfer of voicemails between users regardless of site. It also simplifies our backup procedures for voicemail. Any tips for distributing MWI messages amongst those separate servers that phones are registering to? I suppose I could script something on the voicemail server to put a file in the inbox on the distributed servers but perhaps there is something more elegant I'm unaware of? If not, has anyone scripted this before and willing to share? Would be much appreciated. Thanks, Jeremy ___ --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 -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --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] MWI across multiple servers
I've posted the instructions and scripts on my blog for everyone to grab. This way I'm not sending random files to random people :) http://asterisk.mdaniel.net/?p=14 Let me know if I need to change anything. On Wed, 2006-12-06 at 10:12 -0700, David Thomas wrote: Aaron, Could you please send me the scripts as well. Thanks! 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 -- Aaron Daniel Senior Voice Analyst Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --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] Digium TE407P vs. Sangoma A104d
We have two of the TE407P's active in one of our gateways, and it's been an awesome card to deal with. I'd definitely have to say, once we installed those, the server's been extremely stable, the T1's have had no echo problems, it just works. I'd recommend the Digium cards. Disclaimer: I've never used the Sangoma cards. On Mon, 2006-12-04 at 12:21 -0800, Michael Collins wrote: Has anyone had experience with one or both of these cards? I’m in a position where I might need to recommend one over the other. I’ve read everything that I can find online, so now I’d like to hear of personal experiences. Everything I read on both cards is “5 stars! Awesome! It Rocks!” They both seem to have similar capabilities, similar pricing, etc. Could those of you who have seen these in action please give us some feedback? I’m interested in anything that might help me decide, be it warranty info, vendor responsiveness, etc. Thanks! -MC ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Aaron Daniel Senior Voice Analyst Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --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: Digium through Octasic
Same here. We've been running the TE412P/TE407P family since they came out, replacing our TE406P/TE411Ps. It's been a difference of night and day, and since day one, the cards have been awesome. I'd strongly suggest using the TE412P :) Aaron On Mon, 2006-11-27 at 09:01 -0500, Steven wrote: We used to have a TE411P with the old echo canceller and still had occasional echo. Last week, I replaced it with a TE412P with the Octastic EC and have had no echo reported since then. -- -- Steven http://www.glimasoutheast.org Heidi Mendoza [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] We're looking at using 4 or 8 port T1 cards with echo cancellation and are evaluating brands to go with. We know that Sangoma has excellent solutions especially when it comes to echo. But we still have to hear about actual performance of a Digium card using the same Octasic DSP echo canceller. Would appreciate hearing something on this. __ Sponsored Link Mortgage rates near 39yr lows. $420,000 Mortgage for $1,399/mo - Calculate new house payment __ ___ --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 -- Aaron Daniel Senior Voice Analyst Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --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] Load balance Asterisk servers?
Incorrect :) IAX2 most definitely does support regcontext. Also, I think what he means is the phone specific information must be exactly the same from system to system or the failover won't be as seamless as you expect. A lot of phones support some sort of SRV records, so in the event of a failure, the phones will automatically find the next available server. The other option there is to set up an HA environment so the failover is even transparent to the phones, they just start talking to the new IP address immediately. Another thought, in any failover situation, if you have any sort of automated failover, you must make sure phones that need specific features fail to the same server (i.e. hinting and such) as those features don't work cross server. Aaron On Tue, 2006-11-14 at 08:16 -0700, David Thomas wrote: On 11/14/06, Stelios Koroneos [EMAIL PROTECTED] wrote: JR Richardson gave a very nice presentation at Astricon on how to do that with DUNDI As I understand it JR Richardson's DUNDi solution does not support IAX. It uses regcontex which I believe is only available with SIP. (please correct me if I'm wrong) Also JR notes that... Associated SIP Users, business customers, require same registration and failover to the same servers so unless this is for a residential setup, it may not be of much use to you. Nevertheless it is great documentation, and may get you further than you are now. regards, 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 -- Aaron Daniel Senior Voice Analyst Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --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] Load balance Asterisk servers?
On Tue, 2006-11-14 at 12:00 -0700, David Thomas wrote: On 11/14/06, Aaron Daniel [EMAIL PROTECTED] wrote: Incorrect :) IAX2 most definitely does support regcontext. Also, I think what he means is the phone specific information must be exactly the same from system to system or the failover won't be as seamless as you expect. A lot of phones support some sort of SRV records, so in the event of a failure, the phones will automatically find the next available server. The other option there is to set up an HA environment so the failover is even transparent to the phones, they just start talking to the new IP address immediately. Another thought, in any failover situation, if you have any sort of automated failover, you must make sure phones that need specific features fail to the same server (i.e. hinting and such) as those features don't work cross server. Aaron On Tue, 2006-11-14 at 08:16 -0700, David Thomas wrote: On 11/14/06, Stelios Koroneos [EMAIL PROTECTED] wrote: JR Richardson gave a very nice presentation at Astricon on how to do that with DUNDI As I understand it JR Richardson's DUNDi solution does not support IAX. It uses regcontex which I believe is only available with SIP. (please correct me if I'm wrong) Also JR notes that... Associated SIP Users, business customers, require same registration and failover to the same servers so unless this is for a residential setup, it may not be of much use to you. Nevertheless it is great documentation, and may get you further than you are now. regards, 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 -- Aaron Daniel Senior Voice Analyst Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --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 There I go spreading mis-information again. :) Thanks Aaron for clearing up the IAX2 regcontext question. That is good to know. In the DUNDi scenario, are there any other features that would be affected or become unavailable in the event of a failure? It seems like normal call processing would continue once the client re-registered to a different box, but I haven't tried it yet. I assume one could use DNS-Round-Robin to load balance registrations between the boxes in the cluster, then pull the failed box out of DNS to prevent registration attempts while the box is dead. Is there a better way to do this ??? 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 The best practice for the failover situation will depend on the phone you're using, and what kind of failover you're looking for. Most *real* phones support SRV records which will contain the list of servers that your phone should talk to. If the phone sees one server down, it automatically jumps to the next available one. You are correct however, that call processing would still work as long as you have the servers configured correctly :) Also, if you configure the phone correctly, you should theoretically be able to not lose any calls in case of failure as well, however you would lose call details records for that call. Douglas Garstang will tell you round-robin won't work, I haven't tested it. Something about random packets going to round-robined server addresses :) The other thing to consider is actual failover ip addresses, where one computer automatically assumes another computer's ip address in the event of failure. The phones would automatically start talking to the new system immediately. -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --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] DUNDi Asterisk Cluster
On Tue, 2006-11-14 at 13:09 -0700, David Thomas wrote: We use only IP connections to our asterisk boxes. Given this our origination/termination providers usually send/receive traffic to/from our network on a single IP or limited number of IPs. In a DUNDi Asterisk Cluster, would each of the boxes need to be able to connect to our origination/termination providers directly, or would we need to setup a common gateway box to forward calls to/from our providers? How is this type of routing best handled in an all IP environment? Thanks, 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 I would set up a separate set of boxes that can directly connect to the providers. That gives you separation of services, and if one of those boxes goes down, you'll have enough saturation to not have to worry about it. Having the main boxes do the connections to the providers adds a level of complexity when you need to bring a new system online. Just my 2 cents :) -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --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] DUNDi precache
Doug, This may help you out a little. It's a whitepaper that JR wrote on how to get a DUNDi cluster working with two redundant primary servers that handle all the DUNDi legwork. Read through it, you might get some information you can use out of it. http://txaug.net/storage/users/3/3/images/17/Using%20DUNDi%20with%20a%20Cluster%20of%20Asterisk%20Servers.pdf On Thu, 2006-11-09 at 08:36 -0700, Douglas Garstang wrote: Does anyone have any information on how to use DUNDi precaching? Mark Spencer made a post 2 years ago where he hinted it may be possible to configure DUNDi such that you could centralise your DUNDi registration info by using precaching, instead of having each DUNDi peer meshed with every other one... http://lists.digium.com/pipermail/dundi/2004-October/000189.html However, it seems that no documentation exists for this in the known universe. Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --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] DUNDi precache
Why would you want to do that? Defeats the purpose of *having* the DUNDi protocol. Why not just program the extensions in at regular intervals or something? On Thu, 2006-11-09 at 10:16 -0700, Douglas Garstang wrote: Aaron. Thanks. JR sent me that article before it was published. He's not precaching registrations. He's doing something different. In his configuration, when a registration server gets a request for the location of a phone, it queries the DUNDi Lookup server, which in turn queries the other registration servers on it's behalf. It doesn't actually cache the registrations itself. According to what Mark Spencer wrote, it should be possible for this DUNDi Lookup server to hold, or store (ie cache) -all- phone registration info so that it doesn't have to query the other registration servers. Doug. -Original Message- From: Aaron Daniel [mailto:[EMAIL PROTECTED] Sent: Thursday, November 09, 2006 9:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DUNDi precache Doug, This may help you out a little. It's a whitepaper that JR wrote on how to get a DUNDi cluster working with two redundant primary servers that handle all the DUNDi legwork. Read through it, you might get some information you can use out of it. http://txaug.net/storage/users/3/3/images/17/Using%20DUNDi%20w ith%20a%20Cluster%20of%20Asterisk%20Servers.pdf On Thu, 2006-11-09 at 08:36 -0700, Douglas Garstang wrote: Does anyone have any information on how to use DUNDi precaching? Mark Spencer made a post 2 years ago where he hinted it may be possible to configure DUNDi such that you could centralise your DUNDi registration info by using precaching, instead of having each DUNDi peer meshed with every other one... http://lists.digium.com/pipermail/dundi/2004-October/000189.html However, it seems that no documentation exists for this in the known universe. Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --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] DUNDi precache
I still don't see a reason to use it. If you want immediate information about phones even in the event of a catastrophic failure, bypass the cache altogether (that's what we do) and have it do a lookup every time. Also, set your lookup time to an acceptable value in the event that the primary DUNDi servers don't find the phone. I think ours is extremely low since we've only got a small number of servers. From the stats that I got from JR's talk on DUNDi at Astricon, seems to me the overhead of doing the DUNDi lookups are almost nil, so personally I think caching is pointless in a highly available environment. On Thu, 2006-11-09 at 12:19 -0700, Douglas Garstang wrote: I also just realised a distinct advantage of the precache model. Lets say you have a central DUNDi cache server. He has in his cache the knowledge that appearance 2944093 is registered to pbx1 for the next hour. If pbx1 where to crash, then for the next hour, calls to 2944093 would fail. However, in the precache model, when the phone registers to an Asterisk box, the Asterisk box immediately precaches the information to the central DUNDi server, who maintains this information until it's updated. If pbx1 where to crash, and the phone failed over to pbx2, pbx2 would then send updated registration information to the DUNDi precache server, and thus calls would not fail for an hour. Douglas. -Original Message- From: Aaron Daniel [mailto:[EMAIL PROTECTED] Sent: Thursday, November 09, 2006 11:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] DUNDi precache Why would you want to do that? Defeats the purpose of *having* the DUNDi protocol. Why not just program the extensions in at regular intervals or something? On Thu, 2006-11-09 at 10:16 -0700, Douglas Garstang wrote: Aaron. Thanks. JR sent me that article before it was published. He's not precaching registrations. He's doing something different. In his configuration, when a registration server gets a request for the location of a phone, it queries the DUNDi Lookup server, which in turn queries the other registration servers on it's behalf. It doesn't actually cache the registrations itself. According to what Mark Spencer wrote, it should be possible for this DUNDi Lookup server to hold, or store (ie cache) -all- phone registration info so that it doesn't have to query the other registration servers. Doug. -Original Message- From: Aaron Daniel [mailto:[EMAIL PROTECTED] Sent: Thursday, November 09, 2006 9:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DUNDi precache Doug, This may help you out a little. It's a whitepaper that JR wrote on how to get a DUNDi cluster working with two redundant primary servers that handle all the DUNDi legwork. Read through it, you might get some information you can use out of it. http://txaug.net/storage/users/3/3/images/17/Using%20DUNDi%20w ith%20a%20Cluster%20of%20Asterisk%20Servers.pdf On Thu, 2006-11-09 at 08:36 -0700, Douglas Garstang wrote: Does anyone have any information on how to use DUNDi precaching? Mark Spencer made a post 2 years ago where he hinted it may be possible to configure DUNDi such that you could centralise your DUNDi registration info by using precaching, instead of having each DUNDi peer meshed with every other one... http://lists.digium.com/pipermail/dundi/2004-October/000189.html However, it seems that no documentation exists for this in the known universe. Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --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 -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --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] AEL2 in 1.2
I know I compiled AEL2 into 1.2 before, considering I just copied my source from one server to another, yet I can't seem to figure out why I'm getting this error. Anyone have any ideas? make[1]: Entering directory `/usr/local/src/asterisk-svn/asterisk/pbx' flex argdesc.l argdesc.l, line 19: unrecognized %option: reentrant argdesc.l, line 20: unrecognized %option: bison-bridge argdesc.l, line 21: unrecognized %option: bison-locations make[1]: *** [argdesc_lex.c] Error 1 make[1]: Leaving directory `/usr/local/src/asterisk-svn/asterisk/pbx' make: *** [subdirs] Error 1 -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --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] AEL2 in 1.2
Nevermind, seems scp'ing the source directory over didn't do me any good. On Fri, 2006-11-03 at 09:25 -0600, Aaron Daniel wrote: I know I compiled AEL2 into 1.2 before, considering I just copied my source from one server to another, yet I can't seem to figure out why I'm getting this error. Anyone have any ideas? make[1]: Entering directory `/usr/local/src/asterisk-svn/asterisk/pbx' flex argdesc.l argdesc.l, line 19: unrecognized %option: reentrant argdesc.l, line 20: unrecognized %option: bison-bridge argdesc.l, line 21: unrecognized %option: bison-locations make[1]: *** [argdesc_lex.c] Error 1 make[1]: Leaving directory `/usr/local/src/asterisk-svn/asterisk/pbx' make: *** [subdirs] Error 1 -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --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] Realtime, DUNDi and regexten
We've been using DUNDi, Realtime, and regexten extensively for months now, and it's been working great since we got it running. We don't use the NoOp's for anything other than extension discovery since creating the NoOp's in already configured contexts isn't very stable. Aaron On Thu, 2006-11-02 at 00:29 -0500, Andrew Joakimsen wrote: I can't even get regexten to work with config files On 11/1/06, Douglas Garstang [EMAIL PROTECTED] wrote: It seems that when you use Realtime static and possibly realtime realtime for sip users, that Asterisk fails to create the regexten context for DUNDi. Someone else had the same problem back in July. Doesn't look like they ever had a resolution. http://lists.digium.com/pipermail/asterisk-users/2006-July/160105.html Someone else posted what appears to be the same issue to the bug tracker 3 years ago: http://bugs.digium.com/view.php?id=3053 Mark Spencer closed the bug, saying it was a configuration issue, and to use 'includes'. Not sure what he means by that. Has anyone got this to work? Thanks, Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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 -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --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: AW: [asterisk-users] Snom or Cisco Phones?
That and Cisco won't give you the time of day if you don't use their stuff ;) We have about 1600 of the Cisco's on campus, and unless you run them on the call manager, you're not gonna have nearly as many features as any other phone that's designed with SIP in mind. That said, if you need a phone with dialtone, a pretty screen, and limited xml services, then I will say that the cisco's are extremely easy to provision once you figure out the upgrade paths. (Oh, and we're running 7940's and 7960's... if you're looking at the 7912's, etc, good luck, they're a _complete_ pain to work with) Aaron On Tue, 2006-10-31 at 20:41 +0100, Christian Stredicke wrote: I think one of the differences is: We do pay attention to Asterisk and this mailing list ;-) CS -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Joao Pereira Gesendet: Dienstag, 31. Oktober 2006 13:47 An: asterisk-users@lists.digium.com Betreff: [asterisk-users] Snom or Cisco Phones? Hello I need to buy IP Phones to work with Asterisk, and I'm in doubt between Snom and Cisco Phones. Can you gurus, please, give me your impression of these 2 brands? I need to focus more in SIP and Asterisk compatibility and less in pricing (yes, I know the Cisco are more expensive). Are there any features that Snom has, that Cisco doesnt? And are these features important? Thanks Joao Pereira ___ --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 -- Aaron Daniel Senior Voice Analyst Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --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 line phones with different contexts
Hey all,
Has anyone had any issues with phones having multiple lines that are in
different contexts? We've got a couple phones that we're testing
intercom functionality for, and I'm noticing that for some strange
reason, no matter what line we use, the phones tend to be completely in
one context or another, not segregated like I would expect.
Our contexts look like this:
context intercom {
_ = {
Answer;
check-cid();
Set(CALLERID(num)=${CALLERID(num)} (INT));
SIPAddHeader(Alert-Info: Ring Answer);
createds(${EXTEN});
Dial(SIP/${ds}|20);
Hangup;
};
};
context long-distance {
includes {
local;
};
_9011 = dialout(${EXTEN});
_91NXXNXX = dialout(${EXTEN});
};
The phones are configured as such:
[0004F2100526_1]
canreinvite=no
context=long-distance
host=dynamic
nat=no
qualify=6
secret=secret
type=peer
regexten=44198
[0004F2100526_2]
canreinvite=no
context=intercom
host=dynamic
nat=no
qualify=6
secret=secret
type=peer
regexten=44198
A sip debug from one of the intercoms:
-- SIP read from 10.20.136.130:5060:
INVITE sip:[EMAIL PROTECTED]:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.20.136.130;branch=z9hG4bK2a5fd1d91B78BACE
From: Aaron Daniel
sip:[EMAIL PROTECTED];tag=DDF0722-FFF8D457
To: sip:[EMAIL PROTECTED];user=phone
CSeq: 1 INVITE
Call-ID: [EMAIL PROTECTED]
Contact: sip:[EMAIL PROTECTED]
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE,
NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_430-UA/2.0.1.0291
Supported: 100rel,replaces
Allow-Events: talk,hold,conference
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 251
v=0
o=- 1161637564 1161637564 IN IP4 10.20.136.130
s=Polycom IP Phone
c=IN IP4 10.20.136.130
t=0 0
a=sendrecv
m=audio 2240 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
--- (14 headers 11 lines)---
Using INVITE request as basis request -
[EMAIL PROTECTED]
Sending to 10.20.136.130 : 5060 (non-NAT)
Found peer '0004F2100526_1'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 10.20.136.130:2240
Found description format PCMU
Found description format PCMA
Found description format G729
Found description format telephone-event
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x10c
(ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Looking for 4000 in long-distance (domain tcm1.shsu.edu)
Reliably Transmitting (no NAT) to 10.20.136.130:5060:
SIP/2.0 484 Address Incomplete
Via: SIP/2.0/UDP
10.20.136.130;branch=z9hG4bK2a5fd1d91B78BACE;received=10.20.136.130
From: Aaron Daniel
sip:[EMAIL PROTECTED];tag=DDF0722-FFF8D457
To: sip:[EMAIL PROTECTED];user=phone;tag=as04c17ab8
Call-ID: [EMAIL PROTECTED]
CSeq: 1 INVITE
User-Agent: SCM1 - Sip Call Manager 1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:[EMAIL PROTECTED]
Content-Length: 0
---
tcm1*CLI
-- SIP read from 10.20.136.130:5060:
ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.20.136.130;branch=z9hG4bK2a5fd1d91B78BACE
From: Aaron Daniel
sip:[EMAIL PROTECTED];tag=DDF0722-FFF8D457
To: sip:[EMAIL PROTECTED];user=phone;tag=as04c17ab8
CSeq: 1 ACK
Call-ID: [EMAIL PROTECTED]
Contact: sip:[EMAIL PROTECTED]
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE,
NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_430-UA/2.0.1.0291
Max-Forwards: 70
Content-Length: 0
--- (11 headers 0 lines)---
Destroying call '[EMAIL PROTECTED]'
Finally, a sip show peer on the intercom line proving asterisk knows
it's in the right context:
tcm1*CLI sip show peer 0004F2100526_2
tcm1*CLI
* Name : 0004F2100526_2
Secret : Set
MD5Secret: Not set
Context : intercom
Subscr.Cont. : Not set
Language :
AMA flags: Unknown
CallingPres : Presentation Allowed, Not Screened
Callgroup:
Pickupgroup :
Mailbox : [EMAIL PROTECTED]
VM Extension : asterisk
LastMsgsSent : 0
Call limit : 0
Dynamic : Yes
Callerid :
Expire : 2252
Insecure : port,invite
Nat : RFC3581
ACL : No
CanReinvite : No
PromiscRedir : No
User=Phone : No
Trust RPID : No
Send RPID: Yes
DTMFmode : rfc2833
LastMsg : 0
ToHost :
Addr-IP : 10.20.136.130 Port 5060
Defaddr-IP : 0.0.0.0 Port 5060
Def. Username: 0004F2100526_2
SIP Options : (none)
Codecs : 0x8000e (gsm|ulaw|alaw|h263)
Codec Order : (none)
Status : OK (14 ms)
Useragent: PolycomSoundPointIP-SPIP_430-UA/2.0.1.0291
Reg. Contact : sip:[EMAIL PROTECTED]
ANY help would be greatly
Re: [asterisk-users] Digium vs. Sangoma
I agree with Lacy. This is nothing more than a troll. The asterisk-users list is a forum for asterisk users to advance their knowledge of asterisk, not constantly rag on asterisk and it's developers. Some of us are actually interested in learning more and helping others understand new concepts. For those of you that think this list is your own personal bitching ground, grow up and have a professional discussion with the person/company you have your beef with. Keep in mind that many new users come here first for advice on how to do stuff, and if they continue to get this stuff in their mailbox, they're going to think EVERYONE in the asterisk community has nothing better to do than whine about who they think is better. Aaron On Mon, 2006-10-23 at 22:19 -0500, Lacy Moore - Aspendora wrote: I'm still wondering how this relates to the asterisk-users list. Take it elsewhere. Just like on IRC, take it elsewhere. Don't waste my time. On 10/23/06, TV Guy [EMAIL PROTECTED] wrote: For the record: The Digium people follower's think their shit doesn't stink. I would really like to see their VC cash in the dumper with Sangoma kicking their collective ass. Sangoma is a proper run company, not a bunch of gay little tarts showing off. Made in America - Deep South and All. On 10/23/06, Vulture [EMAIL PROTECTED] wrote: 1. Astricon is a conference and exhibition of Asterisk not Digium. 2. bkw__ has the right to free speech and his opinon of how childish or not Mark Spencer and Digum is or is not. 3. Corydon-w is dening bkw__'s right to free speech. 4. Mark Spencer and other Digium employees have no right to ask someone to leave because of a shirt that is within the limits of the dress code of the resturant. 5. CtRiX has the right to free speech and his opinon of both company's products. 6. Corydon-w is dening CtRiX's right to free speech. I would expect a public apology from Mark Spencer/Digium for asking bkw__ to leave the resturant. I would expect a public apology from Corydon-w to both bkw__ and CtRiX for kicking them off the irc server. I would expect both bkw__ and CtRiX's irc access to be returned immediatly. If Corydon-w has moderator access to the irc channel - then a warning to the user is a must. A user cannot be simply kicked because he is saying this or that. If that topic was not appropriate, then the warning should have silenced the issue. Now we have a good chunk of the asterisk comminuty aware of the events that took place. This makes everyone involved look worse. If I do not see an apology from Mark Spencer and Digium on this list, then I will not be purchasing any more Digium products nor will I publicly contribute any of my conceptual idea's or source code on how to make asterisk better. This is an open source project, and as such anyone can take and modify or integrate their software and/or hardware with it. Anyone disagreeing with my view on this can take it up with me directly off-list. -Jon Unmetered Pipe wrote: I don't mean to be a troll in any way shape or form. I was on IRC last night and I observed the following convo. below. What do you guys make of it ? [02:14] bkw__ Let me tell you how chidlish digium and Mark Spencer is. I walk into a restaurant with them all here at Astricon wearing my sangoma shirt and he asked me to leave. [02:15] Dovid u serious ? [02:15] *** mog ([EMAIL PROTECTED]) left () [02:15] bkw__ yes [02:15] Adam12 Wow, I didn't think Mark Spencer owned a restaurant! :) [02:15] Dovid btw how old is mark ? he looks like a kid [02:15] bkw__ 28 [02:15] Dovid damn [02:16] Dovid filthy rich boy [02:16] Dovid i knes he wasnt over 30 [02:16] CtRiX bkw_, trolling again :-) [02:16] Inez I must try with 1.4 [02:16] *** Corydon-w sets channel #asterisk mode +b *! [EMAIL PROTECTED] [02:16] *** bkw__ was kicked from #asterisk by Corydon-w (troll) [02:16] Dovid he asked u to leave cause u were wearin the competitors shirt ? u know u didnt have to leave. i
Re: [asterisk-users] Digium on Dell PowerEdge 1850
We're running 2 TE412P's in a Dell 1850 just fine, been running like this for well around 6 months to a year now without any problems. They're not exactly 212P's but I imagine it won't be much different. On Wed, 2006-10-18 at 10:54 +0200, Tomislav Parčina wrote: Does anybody have Digium TE212P interface card on Dell PowerEdge 1850? I'm planning to install * on that configuration so I'm looking for any positive/negative experience. Best regards, -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)270248 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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 -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cisco 7960 not registering after * restart
Heh, well, I actually just started a blog to keep track of various goings on, but I just started it so it's kinda scarce. I intend to update it in and out with various information I email to people so everyone can benefit from the questions and answers people use. I'd like to see other people register and start posting stuff there as well as it's got free registration and basically unlimited storage. Voip-info.org is great for learning how to do the basics, but I'd like to see more people join together and disseminate information about how they do things. Check it out: http://asterisk.mdaniel.net On Thu, 2006-10-12 at 10:16 -0400, Jay R. Ashworth wrote: On Wed, Oct 11, 2006 at 09:11:43AM -0500, Aaron Daniel wrote: That's a bug with the 7.5 firmware. I would suggest upgrading to the 8.4 version, we've been running it for a few weeks in a test environment and everyone's been pretty satisfied with the new firmware (read: nobody's complained). If the server goes out, they re-register after the timeout without problems. And that's *exactly* the sort of information that makes me wonder, Aaron: do you guys write a (publically accessible) blog on the goings on in your telecoms dept/asterisk project? I know you're a little closer to In The Real World than some folks, which might militate against... but you're a college, too. :-) And it seems that you're gonna know a whole lot of stuff. Just wondering... Cheers, -- jra -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --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] Increase VoiceMail Messages Recording Gain - AudioCalls are Ok
I've uploaded a patch to my host, it only does the volgain in int format (we use +7 which seems to work well). We've had no problems with it since we set it up back in February, and everyone seems to love it since nobody's blowing out their speakers anymore lol. The patch we use actually does a number of things. We convert from WAV to mp3 for better client support (i.e. my boss used his pda phone to listen to an mp3 voicemail), and we also change the From field of the email to come from the user leaving the voicemail instead of the server email. I think that's it, the file's posted here: http://asterisk.mdaniel.net/?p=5 Check it out, let me know if it works for ya'll. Aaron On Wed, 2006-10-11 at 10:49 +0100, Marco Mouta wrote: Hi Aaron! Could you please provid me your patch for 1.2? I didn't get you, it was a problem for you to get the messages into mp3 format? Did you have any problem until now with this patch on *1.2 ? My box is 1.2.5 and still very stable until now:) Hope you can help me, i can't figure out why no one though about this has a serious request on *1.2 , as this seems to happen always when you have asterisk behind a legacy pbx with zapata in telephony interface. On 10/11/06, Aaron Daniel [EMAIL PROTECTED] wrote: That doesn't always work :) There's two options... either port the volgain patch from 1.4 to 1.2 (If anyone wants a copy, we've been using it for months... however it also converts to mp3 so we'd have to strip that out)... or use 1.4 which includes the patch. Let me know if I should post a copy of the older code somewhere. The 1.4 patch is here: http://bugs.digium.com/view.php?id=6237 Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 On Tue, 2006-10-10 at 21:18 -0500, Shawn Kelley wrote: I had the same problem. Checking voicemail via the phone was perfectly normal but the email attachments were so quiet we had to turn the computer volume all the way up along with the speakers amps just to make the attachment understandable. Then just wait until someone forgets to turn the volume back down and a lovely windows message box pops up. Scares the (pick your word) out of everyone in the office! After much searching I found the solution: In the voicemail.conf file change the order in which the recording formats are specified. Asterisk will email the first format in the list. My original line: format=wav49|wav|gsm My new line: format=wav|wav49|gsm NOTE: My understanding is that the wav files are much larger attachments than the wav49 version. However, we haven't noticed much difference, still fairly small attachments. Definitely no problems on a LAN or Broadband connection. From: Marco Mouta [mailto:[EMAIL PROTECTED] Sent: Tuesday, October 10, 2006 2:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Increase VoiceMail Messages Recording Gain - AudioCalls are Ok Hi all I'm deploying a VoiceMailserver with Asterisk behind a legacy pbx, providing Voicemail to email services for Lecagy PBX extensions. On busy or unanswered calls, Legacy pbx will dial a specific DID (one per extension) to asterisk, and the call is handled by Voicemail application. I've several SIP extensions on this Asterisk box, and calls between Asterisk extensions and legacy PBX are just fine, at least no complaining from users, seems good to me:) The problem is: Right now, and i'm referring only to calls directly handled by VoiceMail application, the users get their audio files in email but the audio is very very low. I've thought about changing RX gain on PRI interface between legacy pbx and asterisk, but until now no complaining with audio calls. I'm afraid that changing this parameter to solve voicemail issues will get me in troubles with Voice Calls . Any advice, or previous similar experience? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users
Re: [asterisk-users] cisco 7960 not registering after * restart
That's a bug with the 7.5 firmware. I would suggest upgrading to the 8.4 version, we've been running it for a few weeks in a test environment and everyone's been pretty satisfied with the new firmware (read: nobody's complained). If the server goes out, they re-register after the timeout without problems. Aaron On Wed, 2006-10-11 at 15:35 +0200, Louis-David Mitterrand wrote: Hello, When I restart asterisk the cisco 7960/7940 phones (sip fw 7.5) fail to re-register themselves with asterisk, even though I put timer_register_expires: 60 in SIPDefault.cnf Is there a way to have these phones register themselves every 60 seconds? Alternatively, can asterisk be made to remember its dynamic sip hosts' registration after a restart? 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 -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] How big is *your* dialplan??
. On 10/11/06, Douglas Garstang [EMAIL PROTECTED] wrote: I see some awefully large dialplans here. Are people putting all this on one box or clustering it amongst a number of boxes? I think any business is going to be pretty annoyed if they suddenly lost access to 16,000+ extensions, and had to wait for a new box to be built and configured. -Original Message- From: George Pajari [mailto:[EMAIL PROTECTED] Sent: Tue 10/10/2006 10:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: Re: [asterisk-users] How big is *your* dialplan?? Single server, dual P3 866Mhz, 1.5Gb, TE407P, two PRIs to telco, one PRI to fax server, one PRI to T.38 gateway: 1791 extensions (4378 priorities) in 240 contexts -- George Pajari, netVOICE communications604 484 VOIP (484 8647 x102) Open Source VoIP/Telephony Specialists 1 877 NET VOIP (638 8647 x102) Hosted IP PBX Services for SOHO Small Businesses - www.ip-centrex.ca VoIP Service, Equipment, Systems, and Consulting - www.netvoice.ca ___ --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 ___ --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 -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How big is *your* dialplan??
Do you want single server stats, or cluster stats? Single server: -= 1004 extensions (1403 priorities) in 45 contexts. =- Aaron On Tue, 2006-10-10 at 14:16 -0600, Steve Murphy wrote: Hello! In my relentless quest for knowledge, I pose this question: who's got the biggest dialplans, and how big are these monsters? What's the biggest dialplan in use right now? If you feel you are a competitor, let me know how many contexts/extensions/priorities you are dealing with. Maybe the context with the most extensions, the extension with the most priorities would be interesting... For example: Digium's dialplan is roughly 50 contexts, 304 total extensions, 870 total priorities. My home system has 100 contexts, 400 total extensions, 935 total priorities. My biggest extension has 129 priorities... no inflation or useless cruft there, either... mostly. These would seem small compared to some dialplans out there, I'll bet. murf ___ --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 -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --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] Increase VoiceMail Messages Recording Gain - AudioCalls are Ok
That doesn't always work :) There's two options... either port the volgain patch from 1.4 to 1.2 (If anyone wants a copy, we've been using it for months... however it also converts to mp3 so we'd have to strip that out)... or use 1.4 which includes the patch. Let me know if I should post a copy of the older code somewhere. The 1.4 patch is here: http://bugs.digium.com/view.php?id=6237 Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 On Tue, 2006-10-10 at 21:18 -0500, Shawn Kelley wrote: I had the same problem. Checking voicemail via the phone was perfectly normal but the email attachments were so quiet we had to turn the computer volume all the way up along with the speakers amps just to make the attachment understandable. Then just wait until someone forgets to turn the volume back down and a lovely windows message box pops up. Scares the (pick your word) out of everyone in the office! After much searching I found the solution: In the voicemail.conf file change the order in which the recording formats are specified. Asterisk will email the first format in the list. My original line: format=wav49|wav|gsm My new line: format=wav|wav49|gsm NOTE: My understanding is that the wav files are much larger attachments than the wav49 version. However, we haven’t noticed much difference, still fairly small attachments. Definitely no problems on a LAN or Broadband connection. From: Marco Mouta [mailto:[EMAIL PROTECTED] Sent: Tuesday, October 10, 2006 2:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Increase VoiceMail Messages Recording Gain - AudioCalls are Ok Hi all I'm deploying a VoiceMailserver with Asterisk behind a legacy pbx, providing Voicemail to email services for Lecagy PBX extensions. On busy or unanswered calls, Legacy pbx will dial a specific DID (one per extension) to asterisk, and the call is handled by Voicemail application. I've several SIP extensions on this Asterisk box, and calls between Asterisk extensions and legacy PBX are just fine, at least no complaining from users, seems good to me:) The problem is: Right now, and i'm referring only to calls directly handled by VoiceMail application, the users get their audio files in email but the audio is very very low. I've thought about changing RX gain on PRI interface between legacy pbx and asterisk, but until now no complaining with audio calls. I'm afraid that changing this parameter to solve voicemail issues will get me in troubles with Voice Calls . Any advice, or previous similar experience? ___ --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 RT on Disk On Module Performance andDurability
that the SHORTEST expected life of a CF card in their test scenarios was over 70 years! How long is your power supply going to last? Even if the consumer level cards had 1/10 the life expectancy, that is still seven years. I expect to get at least that from my original AstLinux system. It's been two so far, I'll let you know how it is doing in another five years :). JFFS (and similar FSs) are not appropriate for CF cards or DOMs. They are meant to be used directly on flash memory and do their own wear leveling and in some cases, compression. All kinds of commercial devices use JFFS2. If you are using a CF or DOM with Linux, ext2 is the best FS to use. CF cards and DOMs use their own wear leveling, so none is required in the operating system or file system. CF cards and DOMs hide wear leveling from you and expose themselves as an ordinary IDE device. I echo Jeremy's conclusions. With a properly designed operating system, decent flash memory, and a reasonable usage pattern, I can tell you (with a great amount of certainty) that in most situations, CF cards will outlast just about any hard drive (even SCSI) when used 24/7. These days, it really is pretty tough to trash flash. However, if you are running a MySQL cluster or something with several, multi-gigabyte databases, no type of flash memory will last very long! :) To get back to answering your question, I HIGHLY recommend that you avoid MySQL and realtime on your box with a DOM. Nothing against either (MySQL or Realtime), but they will probably make your device more complicated than it needs to be while substantially shortening the life of your DOM. If you absolutely have to use MySQL, you might have better luck using a MySQL storage engine that uses fewer writes than InnoDB, but I am no expert on that. -- Kristian Kielhofner ___ --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 -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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 -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --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] password for vm users
You're digging a little further into it than any standard voicemail system would ever go. If you need more strenuous password functionality, I would suggest dropping users through a voicemail macro that does all the password functionality for you and then passed them to the actual voicemail app. That way it's more flexible, and you can allow the voicemail app to do what it's good at, voicemail ;) Aaron On Mon, 2006-10-09 at 14:07 -0700, stan ford wrote: how about password strength? or remembering and not allowing password? or password duration? Marco Mouta [EMAIL PROTECTED] wrote: just set your initial password to be equal to vm-account number, and Voicemail application will do that for you and will request users to setup a new password! On 10/9/06, stan ford [EMAIL PROTECTED] wrote: how does one force mandatory password change on login? and a period of time to pass before mandating a password change? im using trixbox so if you have that info. that would be good, if you have it for asterisk, im sure i could figure it out as well thx. __ Yahoo! Messenger with Voice. Make PC-to-Phone Calls to the US (and 30+ countries) for 2¢/min or less. ___ --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 -- Com os melhores cumprimentos, Marco Mouta ___ --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 __ Talk is cheap. Use Yahoo! Messenger to make PC-to-Phone calls. Great rates starting at 1¢/min. ___ --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 -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail MWI
For us the voicemail server doesn't have to know what phones are registered where. We have an externnotify script that drops x number of msgx.txt files into the respective voicemail folders after any call that goes through the voicemail server. x in this would be the number of messages. Since our servers are all exactly the same, we just blanket the script across all our call servers, but if you have specific servers where the phones are registered at, you can modify the SIP channel to update the db with server information and have the voicemail server just run the script across servers that that particular phone is currently registered with. The scripts we use are relatively lightweight, but can probably be turned into some sort of listening service to remove some of the ssh overhead required. Not a problem for us, but you might not like it much :) On Fri, 2006-10-06 at 11:49 -0600, Douglas Garstang wrote: I'd like to know if anyone has a suggested fix for this... You have a 'cluster' of Asterisk servers that use DUNDi etc for registration redundancy, finding other phones etc. You have a separate Asterisk box for voicemail. For voicemail deposit/retrieval you trunk the call over to the voicemail server. This all works fine. No issues there. What about MWI though? Your phones register with the cluster, not with the voicemail server, and therefore the voicemail server has no knowledge of where the phones are and therefore cannot send out SIP NOTIFY messages to phones. This is a general architectural problem with Asterisk. Has anyone solved it? Are the developers working on fixing problems like this for 1.6? Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --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] HTTP Connection Closed on 7960 SIP
This happens if you have a logo_url configured for your phone and the phone can't access it. I'm guessing you don't allow 80 through the firewall to the server that's serving the image. -- Aaron Daniel On Fri, October 6, 2006 20:13, Robert Goodyear wrote: Anyone know why I get HTTP Connection Closed on the display of a 7960 running a SIP image? Only seems to happen when registering against my Asterisk box from the WAN. I have 1:1 NAT happening on my firewall. Phones function perfectly otherwise. TFTP working fine across the firewall as well. Odd! Thanks in advance. -Rob. ___ --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] RPID
Thanks... I did some research and found that it's actually not what I was wanting (unless I missed something lol). I'm actually looking for a way to forward caller id information to the called party on a forwarded call. I may just need to dig deeper. On another note, I did find a patch in mantis that is considered experimental that does get it to where you can see the caller id of who you're calling based in the dialplan. Back to the drawing board though :) Aaron -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kristian Kielhofner Sent: Wednesday, September 27, 2006 4:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] RPID DANIEL, AARON MATTHEW wrote: Has anyone successfully gotten rpid working between two phones through asterisk? Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 Aaron, RPID is supported in Asterisk but many phones do not support it. Try adding the following to sip.conf: sendrpid=yes trustrpid=yes If it is going to work with your phones, it will just work. If not, chances are your phone does not support RPID. You can always look at a SIP debug to make sure it is getting sent. Even if your phones do not support RPID, From: usually works just fine :). -- Kristian Kielhofner ___ --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] University dumps CISCO VoIP for Asterisk
On Wed, 2006-09-20 at 08:26 -0500, Eric ManxPower Wieling wrote: joea, j4computers wrote: Ferguson, Michael[EMAIL PROTECTED] Wrote on: 9/20/2006 8:03 AM: G'Day List, Interesting article. Enjoy http://www.networkworld.com/news/2006/091206-von-sam-houston.html?t5 Mike The text states that asterisk cannot do secretarial functions, meaning one person, or admin, cannot answer multiple lines. This relates a bit to my recent post, asking about servicing multiple lines. Implication is that asterisk can do that, but I am now concerned that there is no uber function that can allow a single person answer any line, for reasons of convenience or design. Problem is, this was understood, rightly or wrongly, to exist, in preliminary inquiries (not here) and is a part of a potential clients desire. Can someone enlighten me? The problem with a BLF (Busy Lamp Field) is that it's hard to find a box with 6,000 buttons on it, as would be required by above university. Asterisk has several methods of picking up remote lines. Group Call Pickup, Directed Call Pickup, and the standard way Asterisk rings multiple extensions at the same time via in the Dial() Line, and BLF If you want the traditional Key System style of BLF, then you need a phone that supports it. The Polycom 601 Sidecar supports it in a limited way, and I've heard that SNOM supports it as well. What SPECIFICALLY are you trying to do that you are unable to do? You are correct, to an extent. We do have extensions that ring multiple phones on campus, however, BLF and SLA don't work in the current 1.2 branch. I know they're doing SLA work in 1.4, so we're hoping that the point is moot by the time it comes out. There are a number of patches that allow the polycoms and aastra's to do directed pickup on a line that's ringing combined with hinting to get the illusion of SLA, however, without extensive testing we haven't had a chance to implement the software. The biggest problem we have with the hinting functions is that you have to have the phones registered to the same server, and with two identical servers that could theoretically serve any phone we have, it's a management nightmare to guarantee that any given phone will be on the same server as any other given phone. On that note, for a small office, it would probably work great, it's just not feasible for us just yet, so we're looking into other options as well. :) -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --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] University dumps CISCO VoIP for Asterisk
On Wed, 2006-09-20 at 22:25 +0200, Olivier wrote: 2006/9/20, Aaron Daniel [EMAIL PROTECTED]: The biggest problem we have with the hinting functions is that you have to have the phones registered to the same server, and with two identical servers that could theoretically serve any phone we have, it's a management nightmare to guarantee that any given phone will be on the same server as any other given phone. On that note, for a small office, it would probably work great, it's just not feasible for us just yet, so we're looking into other options as well. :) -- Hi Aaron, I didn't know phones needed to be registered on the same server to benefit hinting functions (as we mainly install small offices). Do you think this comes from SIP and NOTIFY-SUBSCRIBE messages limitations or from current implementation ? Could this be worked around using SER or other software ? Regards The problem occurs in the subscribe message, I believe. The phone sends it's subscribe message to whatever server it's registered to, so only that server will know about it. I know D. Garstang has done a lot of painful work on multiple clustered asterisk servers attempting to do stuff like this with SER, but we haven't done anything to that extent yet. There are probably a number of code modifications we could make to replicate the information from server to server, but it seems like that would start to get somewhat unruly. -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --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] Fedora
Yes. We use fedora as a test system before moving it onto RHEL. On Mon, 2006-09-18 at 11:34 -0700, bilal ghayyad wrote: Hi list; Does asterisk work with fedora because redhat enterprise is licensed and costly. Regards Bilal __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --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] SHSU asterisk installation?
Maybe I should just hold a conference call about all this stuff. On Sat, 2006-09-16 at 18:21 +0200, Matt Riddell (IT) wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Roy Sigurd Karlsbakk wrote: Linked from /. today, http://www.networkworld.com/news/2006/091206-von-sam-houston.html talks about the Sam Houston State University (SHSU) migrating a rather large amount of users to asterisk. The article describes the installation in rather vague terms, so I was wondering if someone know how they plan to do this, in detail. Yes, there have been several threads about this. Obviously I _have_ tried to google about this, so if you could point me to one of those threads, I'd be grateful - From 3 days ago: http://www.sineapps.com/news.php?rssid=1509 Thread here: [asterisk-users] University switches to Asterisk - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://wap.sineapps.com (Daily Asterisk News for your cellphone) http://feeds.feedburner.com/AsteriskNews (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFFDCScS6d5vy0jeVcRAmQAAKCOcWU04FK6txwH1NfduA0QsFYaogCfb3a4 KojTXtiqQLK7YNtXp+4Qh+I= =T9d+ -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] Response to KP Flemming...
On Thu, 2006-09-07 at 02:31 -0700, Joe Shmoe wrote: You say its not your code. But yet, why would you actually admit to one of your own leaking it. Well some research has been done one the code.. here's what we found.. the g723.1 library code that was posted matches the library code distributed by Digium and committed to CVS by Mark in March 2003, with the one exception that the tab_lpc.c file that was distributed by the poster had CRLF line endings in it, where the one from Digium CVS had only LF endings. The module code was identical to: http://svn.digium.com/view/asterisk/trunk/codecs/codec_g723_1.c?rev=5869view=markup Also if you want to know if Digium fully complies the the GPL no. They dont. Digium has added a paragraph of text under the symbol ASTERISK_GPL_KEY in include/asterisk/module.h which every Asterisk module must return when a function *key() is called by the module loader. This paragraph makes a claim that modules must only be released under the GPL license, not any other license, which excludes GPL compatible licensing and thereby constitutes an additional restriction which is explicitly prohibited by section 7 of the GPL. see http://www.eff.org/legal/cases/Lexmark_v_Static_Control/20041026_Ruling.pdf for additional information on this type of activity and generally why that paragraph cant even be legally copyrighted (at least in America, where digium is based). Missed the link for the Codec's? Here ya go! http://s6.quicksharing.com/v/6876458/_codec.tgz.html If you're going to cause flamewars and be a general ass on the mailing list, you might as well be an adult and become an active participant in the discussion. Better yet, I think I'll do what you did and create a fake email with a fake name so no one will know when I send a real email asking how to push the power button :) -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco MWI
If you want to leave the @default off, if I remember correctly, you have to set searchcontexts=yes in voicemail.conf. On Wed, 2006-09-06 at 23:16 +0200, Michiel van Baak wrote: On 17:04, Wed 06 Sep 06, Doug Lytle wrote: Steve Kennedy wrote: But my mailbox (5200) is in default. I'm pretty sure that you'll still need to include the @context for the MWI to work correctly. Or switch the phone to SCCP. It will give you a lot of extra power :) -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7960 won't download dialplan.xml
Put this line in your SIPDefault.cnf file (or the individual phone's): dial_template: dialplan Just cut the .xml off the filename and the phone will pull that particular dialplan :) On Fri, 2006-09-01 at 16:39 -0400, Peter Pauly wrote: I'm monitoring my tftp servers' logs and my Cisco 7960 test phone won't download dialplan.xml to the phone. I know this from the logs and from the behavior of the phone. I see it downloading other files like the ring tone file, etc. Is there something that needs to be set in the cnf files to download the dialplan? I thought it is included automatically. I've also tried reseting the phone to factory presets. I'm running POS3-08-2-00. ___ --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 -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7970 8.0.4 SIP firmware
I tried that image for about 5 minutes. Kept getting errors in asterisk from the phone and it wouldn't stay registered. Rolled back to 8.0.2 and that works fine for us for now. On Thu, 2006-08-31 at 15:30 +0200, Tomislav Parčina wrote: Does anybody use 8.0.4 SIP firmware for Cisco 7970 IP phone? I have upgrade my phone and now it doesn't register with Asterisk. In full.log file I don't see any reason why phone doesn't register. Has anybody head problems like this one? -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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 -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7970 8.0.4 SIP firmware
MWI has been working on our (2) 7970's, as far as I can tell. My boss usually complains when his doesn't work, so it seems to be working fine as far as that's concerned. The 8.0.4 firmware attempted to register, but asterisk threw an error on a response it got back from the phone (I don't remember exactly which one), but I could make calls from it, just not to it. Aaron On Thu, 2006-08-31 at 14:33 -0500, Lacy Moore - Aspendora wrote: Aaron, was the MWI working for you on 8.0.2? I've got a 7970 and 7961 sitting on a shelf because the MWI doesn't work. On the 8.0.4, it never registered, but I was able to make calls with it. I didn't try calling it, since I never saw it register. It appeared it was authenticating for outgoing calls. On 8/31/06, Aaron Daniel [EMAIL PROTECTED] wrote: I tried that image for about 5 minutes. Kept getting errors in asterisk from the phone and it wouldn't stay registered. Rolled back to 8.0.2 and that works fine for us for now. On Thu, 2006-08-31 at 15:30 +0200, Tomislav Parčina wrote: Does anybody use 8.0.4 SIP firmware for Cisco 7970 IP phone? I have upgrade my phone and now it doesn't register with Asterisk. In full.log file I don't see any reason why phone doesn't register. Has anybody head problems like this one? -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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 -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --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 -- Lacy Moore Aspendora, Inc. ___ --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 -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --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] SER Dispatcher Load Balance How-To?
On Wed, 2006-08-30 at 08:34 -0600, Douglas Garstang wrote: -Original Message- From: Aaron Daniel [mailto:[EMAIL PROTECTED] Sent: Tuesday, August 29, 2006 11:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] SER Dispatcher Load Balance How-To? Well, it really depends on what he's using the asterisk servers for. If it's for voicemail or apps, he'll just have to make sure that certain apps land on certain servers, and voicemail can be distributed for various things. If ser can do what I've heard/read it can do, it can handle all the basic call functions (i.e. forwarding) for plenty of calls. Also, if the asterisk servers are just acting as gateways (i.e. t1, e1, etc), then they will have no problem handling a load balanced configuration. To do that, you'd need to use the avpops module in OpenSER. You think Asterisk documentation is bad? Wait until you try and get that stuff to work. LOL, I've never gotten further than installing SER, so yeah, I understand ;) -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --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] Using HINT with Cisco 7940/SIP
On Wed, 2006-08-30 at 16:25 +0100, Conrad Wood wrote: On Sat, 2006-06-17 at 15:49 -0500, Lacy Moore - Aspendora wrote: Can't be done using the 7960 with SIP, unless you are talking about just monitoring that phone. You can monitor a 7960, but you can't show the status of other phones on a 7960 with SIP. Do you know wether it can be done with a 7940(SIP)? Can it display status of (for example) 4205,hint,SIP/phone1 ? Conrad No, not running SIP. -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --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] 911 versus 9.911
On Wed, 2006-08-30 at 20:10 -0700, George Pajari wrote: I'd rather pay the fine than the liability settlement when found negligent in a lawsuit because someone panicked, repeatedly dialled 911, and could not reach Emergency when their coworker had a major myocardial infarction right beside them. We configure all our systems, regardless of whether or not they have a dial-9 for an outside line dialplan, to route both 911 and 9911 to an outside line and 911. We also log every call so when someone does dial and hangup, we send Big Eric to their cube to rearrange a few fingers on their dialling hand :-) Most people are going to attempt to dial 911 regardless of where they are, especially if they're in a panic... We use both 911 and 9911 (our nortel expects 9911, but allows 911), which seems to be better for users that aren't used to the dial 9 to get out mentality. 100 accidental calls is worth the 1 time that someone could die because they don't realize that they're supposed to dial 9911 instead. We've actually had on several occasions people in my office dial 911 on accident when dialing something like 91800. and ended up hitting the 1 twice, and usually dispatch just calls back and asks what was up. Just my 2 cents. Aaron ___ --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] SER Dispatcher Load Balance How-To?
Well, it really depends on what he's using the asterisk servers for. If it's for voicemail or apps, he'll just have to make sure that certain apps land on certain servers, and voicemail can be distributed for various things. If ser can do what I've heard/read it can do, it can handle all the basic call functions (i.e. forwarding) for plenty of calls. Also, if the asterisk servers are just acting as gateways (i.e. t1, e1, etc), then they will have no problem handling a load balanced configuration. As an fyi, we have two primary call servers, and two gateways that are basically exact replicas of each other. If either of the gateways die, it's almost completely transparent to the users (I say almost because any calls in progress will be lost due to the loss of a t1 or two). It can be done, it's all in how you design it. On Tue, 2006-08-29 at 21:42 -0600, Douglas Garstang wrote: Not really. You need to make sure that a phone always uses the same primary asterisk system under normal circumstances. You can simulate load balancing my staggering your phones to use different asterisk systems. -Original Message- From: Andy Chung (Power-All) [mailto:[EMAIL PROTECTED] Sent: Tue 8/29/2006 9:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: Re: [asterisk-users] SER Dispatcher Load Balance How-To? Hi Douglas, Thanks for your advice. So is there any alternatives? Thanks! Andy Douglas Garstang wrote: That might not be a good idea. If you transfer or forward calls on your phones, you MUST make sure the transferred or forwarded call goes back to the same Asterisk box it was handled on. If you use the dispatcher, and load balance, there is no guarantee of that, and transfers and forwarding will break. Doug. -Original Message- From: Andy Chung (Power-All) [mailto:[EMAIL PROTECTED] Sent: Tue 8/29/2006 7:49 PM To: asterisk-users@lists.digium.com Cc: Subject: [asterisk-users] SER Dispatcher Load Balance How-To? Hi all, I have three Asterisk servers behind a SER. I want to know how to configure the Dispatcher module of SER to achieve load balance for the Asterisk servers. I have visited http://www.openser.org/docs/modules/1.1.x/dispatcher.html, is there any web sites have more detail information on that? Thanks! Andy ___ --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 ___ --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] Realtime and hints
I seem to remember someone posting somewhere (was it the list or some site I was browsing...) where someone had created a hint lookup table and just put a db lookup in the dialplan for the hint priority. Then you just need one line to cover all your hints and the db will handle everything else. You just have to figure out the best realtime - static combination that works for you, just like ALL of us have had to. Trial and error does wonders. Aaron On Thu, 2006-08-24 at 11:00 -0600, Douglas Garstang wrote: I don't see how that helps. If you have a portion of the hint still in extensions.conf, then what use is the database? -Original Message- From: Aaron Daniel [mailto:[EMAIL PROTECTED] Sent: Thursday, August 24, 2006 10:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Realtime and hints That's what he was gettin at. Take the second line out, and put the first priority in the database. On Thu, 2006-08-24 at 10:17 -0600, Douglas Garstang wrote: But... you need _both_ in your dialplan. My extensions.conf has: exten = 2944054,hint, SIP/2944054 exten = 2944054,1, Dial(SIP/2944054) ie two lines for the hint. -Original Message- From: Dovid Bender [mailto:[EMAIL PROTECTED] Sent: Thursday, August 24, 2006 9:32 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Realtime and hints Doug I have Exten = 10,hint,SIP/11010 and in mysql I have exten = 10,1,Dial(SIP/11010) - Original Message - From: Douglas Garstang [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, August 21, 2006 3:37 PM Subject: [asterisk-users] Realtime and hints Can realtime be used with hints? How would you get the following into the database given that the priority column is numeric, and that there is no application for the first entry? exten = 2944006,hint,SIP/2944006 exten = 2944006,1,Dial(SIP/2944006) Every time I touch realtime I hit obstacles. How are others getting around this limitation? Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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 -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --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 -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --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] Realtime and hints
That's what he was gettin at. Take the second line out, and put the first priority in the database. On Thu, 2006-08-24 at 10:17 -0600, Douglas Garstang wrote: But... you need _both_ in your dialplan. My extensions.conf has: exten = 2944054,hint, SIP/2944054 exten = 2944054,1, Dial(SIP/2944054) ie two lines for the hint. -Original Message- From: Dovid Bender [mailto:[EMAIL PROTECTED] Sent: Thursday, August 24, 2006 9:32 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Realtime and hints Doug I have Exten = 10,hint,SIP/11010 and in mysql I have exten = 10,1,Dial(SIP/11010) - Original Message - From: Douglas Garstang [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, August 21, 2006 3:37 PM Subject: [asterisk-users] Realtime and hints Can realtime be used with hints? How would you get the following into the database given that the priority column is numeric, and that there is no application for the first entry? exten = 2944006,hint,SIP/2944006 exten = 2944006,1,Dial(SIP/2944006) Every time I touch realtime I hit obstacles. How are others getting around this limitation? Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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 -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --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] voicemailmain
Not sure about that Doug. It should read:
exten = a,1,VoicemailMan([EMAIL PROTECTED])
If you put it in the brackets, it becomes part of the variable name
instead of part of the argument.
On Thu, 2006-08-24 at 16:57 -0400, Doug Lytle wrote:
existx wrote:
Cristian,
The only other line in extensions.conf that references VoicemailMain
is this:
exten = a,1,VoicemailMain(${ARG1})
This should read:
exten = a,1,VoicemailMain([EMAIL PROTECTED])
Doug
--
Aaron Daniel
Computer Systems Technician
Sam Houston State University
[EMAIL PROTECTED]
(936) 294-4198
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Re: [asterisk-users] Direct to Voicemail
Since you're using the variables to decide what to do next
(VMBOXEXISTSSTATUS), you can go ahead and set priorityjumping=no in the
general section of extensions.conf, unless you're using the n+101
priority jumping elsewhere.
On Wed, 2006-08-23 at 08:39 -0400, Doug Lytle wrote:
Hey everybody,
I've set up an extension that allows users to send a call directly to
voice mail. Yesterday, someone accidentally sent a call to an extension
that didn't exist and the call was dropped. I found the option to check
if a mailbox exists and it works fine, but I get the following 'warning':
Spawn extension (sip, 04258, 0) exited non-zero on 'Zap/3-1'
-- Executing Set(Zap/3-1, _direct_vm=4258) in new stack
-- Executing MailboxExists(Zap/3-1, [EMAIL PROTECTED]|) in new stack
Aug 23 08:26:30 WARNING[8313]: app_voicemail.c:5697 vm_box_exists: VM
box [EMAIL PROTECTED] exists, but extension 04258, priority 103 doesn't exist
Is there a way to avoid this warming? Code fragment below:
[direct-to-voicemail]
; **
; Allow anybody to send a call directly to voicemail
; by pre-pending a 0 to the destination extension.
; Checks to see if voice mail box exists, if not
; Tells the callee that no such vm box exists and
; then transfers them to the operator
; **
exten = _04XXX,1,Set(_direct_vm=${EXTEN:1})
exten = _04XXX,2,MailboxExists([EMAIL PROTECTED])
exten = _04XXX,3,Goto(s-${VMBOXEXISTSSTATUS},1)
exten = s-FAILED,1,SayDigits(${direct_vm})
exten = s-FAILED,2,Playback(vm-nobox)
exten = s-FAILED,3,Playback(pbx-transfer)
exten = s-FAILED,4,Goto(incoming,s,1)
exten = s-SUCCESS,1,Set(CALLBACK=${DB(vmcallback/${direct_vm})})
exten = s-SUCCESS,2,GotoIf($[${CALLBACK} =
YES]?s-SUCCESS,3:s-SUCCESS,4)
exten = s-SUCCESS,3,System(/usr/local/bin/vm-callout.sh ${direct_vm})
exten = s-SUCCESS,4,Voicemail([EMAIL PROTECTED])
exten = s-SUCCESS,5,Hangup()
--
Aaron Daniel
Computer Systems Technician
Sam Houston State University
[EMAIL PROTECTED]
(936) 294-4198
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RE: [asterisk-users] Realtime Extensions -- Comments?
On Tue, 2006-08-22 at 10:14 -0600, Douglas Garstang wrote: -Original Message- From: Jason Parker [mailto:[EMAIL PROTECTED] Sent: Tuesday, August 22, 2006 9:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Realtime Extensions -- Comments? - Douglas Garstang [EMAIL PROTECTED] wrote: The unofficial docs on the voip wiki for the realtime extensions table structure is: CREATE TABLE `extensions_table` ( `id` int(11) NOT NULL auto_increment, `context` varchar(20) NOT NULL default '', `exten` varchar(20) NOT NULL default '', `priority` tinyint(4) NOT NULL default '0', `app` varchar(20) NOT NULL default '', `appdata` varchar(128) NOT NULL default '', PRIMARY KEY (`context`,`exten`,`priority`), KEY `id` (`id`) ) TYPE=MyISAM; Uhm... what abouts comments? What if I wanted to temporarily deactivate a couple of extensions? Without a comment flag, I'd have to completely remove those entries from the extensions table! That's not very friendly is it... Is there a better way? Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I cheated, and just added a comments field to the table. Asterisk only reads fields by name, so extra columns don't hurt at all. How did an extra field that Asterisk doesn't know anything about, change it's behaviour? Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- I think he meant a comments field, to describe the extension. Asterisk doesn't care about extra fields in the db, but it won't use them to it's benefit unless it's programmed in. -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --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] Realtime Extensions -- Comments?
Just a thought, but try putting a ; in the exten field before the extension... then asterisk won't match any extensions with it, and it'll effectively become commented. Don't know if it'll work, but from looking at the code and the table structure, it shouldn't break anything. On Tue, 2006-08-22 at 08:20 -0600, Douglas Garstang wrote: The unofficial docs on the voip wiki for the realtime extensions table structure is: CREATE TABLE `extensions_table` ( `id` int(11) NOT NULL auto_increment, `context` varchar(20) NOT NULL default '', `exten` varchar(20) NOT NULL default '', `priority` tinyint(4) NOT NULL default '0', `app` varchar(20) NOT NULL default '', `appdata` varchar(128) NOT NULL default '', PRIMARY KEY (`context`,`exten`,`priority`), KEY `id` (`id`) ) TYPE=MyISAM; Uhm... what abouts comments? What if I wanted to temporarily deactivate a couple of extensions? Without a comment flag, I'd have to completely remove those entries from the extensions table! That's not very friendly is it... Is there a better way? Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How To NOT Generate A CDR For A Call?
exten = blah,n,NoCdr() On Fri, 2006-08-18 at 16:55 -0400, Nate Kapi wrote: Can anyone tell me the proper way to NOT generate a CDR record for a call using Asterisk 1.2? I heard about the C option and tried it, but I still see the call details in Master.csv. It would be nice if there was a way to NOT log an incoming call as well. Is that possible? Thanks for any help in advance! ___ --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 -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --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 Real Time and sip.conf file used at the same time
Yes. On Wed, 2006-08-16 at 19:03 +0100, Ricardo Carvalho wrote: Is it possible to use Asterisk RealTime and also config files (like sip.conf) at the same time? As much as I know, only one thing can be used and I need them both working!... Thanks, Ricardo. ___ --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 -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --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] Macro inside macro
Just a thought, but shouldn't the exten = h be priority 1, not priority
6? The h extension is totally separate from the s extension, the server
doesn't think it exists because there's no priority 1 for it. Try
that :)
On Sun, 2006-08-13 at 19:32 +1000, Michael Strelnikov wrote:
I have the same problem even with AEL.
When call is finished from macro dialout asterisk just stops all
futher processing.
I haven't found any solution yet.
On 8/13/06, Dovid Bender [EMAIL PROTECTED] wrote:
Please include what you send to the macro from your
extensions.conf so we
can see what you are sending down to the macro.
- Original Message -
From: Attilla De Groot [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Sunday, August 13, 2006 4:56 AM
Subject: [asterisk-users] Macro inside macro
Hi all,
I'm making a little macro te record conversations if a user
want so and
if it's recorded the recording should be e-mailed.
This is what I have come up with right now:
[macro-record]
exten =
s,1,Setvar(CALLFILENAME=CALL-${ARG1}-${MACRO_EXTEN:4}-$
{TIMESTAMP})
exten = s,2,Monitor(wav,${CALLFILENAME},m})
exten = s,3,GotoIf($[${ARG1} =
conference]?macro-record|s|
4:macro-record|s|5)
exten = s,4,Macro(conference|${ARG2})
exten = s,5,Macro(dialout|${ARG2}|${ARG3})
exten = h,6,System(/etc/asterisk/mail.sh
[EMAIL PROTECTED] $
{CALLFILENAME} )
The conversation gets recorded perfectly and it's also
possible to dial,
but somehow the script doesn't get executed and I don't
know why (the
script works manually, I already tested this).
Can anyone helpyp me with the solution ?
Best regards,
Attilla
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Best regards,
Michael Strelnikov
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Aaron Daniel
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Sam Houston State University
[EMAIL PROTECTED]
(936) 294-4198
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Re: [asterisk-users] DUNDi with SIP
Using the SECRET variable for sip doesn't work. On Wed, 2006-08-02 at 13:11 -0600, Douglas Garstang wrote: I've trying to use DUNDi with SIP to see if it works around some limitations of IAX2. I do a DUNDi lookup, get my SIP path, and try to dial it. Asterisk immediately says 'No such host', eventhough that's the path is just returned! [Aug 2 13:07:05] == Spawn extension (global_vmdeposit, u9220371, 3) exited non-zero on 'SIP/3254101-eb7d' [Aug 2 13:07:13] -- Executing NoOp(SIP/3254101-6373, *** OnNet originated call Chocolate Chip 3254101 - 9220371) in new stack [Aug 2 13:07:13] -- Executing AGI(SIP/3254101-6373, ipt/originator.py) in new stack [Aug 2 13:07:13] -- Launched AGI Script /var/lib/asterisk/agi-bin/ipt/originator.py [Aug 2 13:07:13] -- AGI Script Executing Application: (SetAccount) Options: (9220371) [Aug 2 13:07:13] -- AGI Script Executing Application: (ChanIsAvail) Options: (SIP/9220371) [Aug 2 13:07:14] -- AGI Script Executing Application: (Dial) Options: (SIP/dundisip:[EMAIL PROTECTED]/9220371) [Aug 2 13:07:14] WARNING[5429]: chan_sip.c:1980 create_addr: No such host: xxx.yyy.142.163/9220371 [Aug 2 13:07:14] NOTICE[5429]: app_dial.c:1040 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) [Aug 2 13:07:14] == Everyone is busy/congested at this time (1:0/0/1) Not sure what is going on. I can see the query at the other end, but it doesn't look like it ever receives the call. Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --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] DUNDi with SIP
I'm talking about the rotating DUNDi secret that is stored in dbsecret in iax.conf. It doesn't exist in the SIP channel. On Wed, 2006-08-02 at 13:43 -0600, Douglas Garstang wrote: Secret? Do you mean sbsecret in sip.conf? -Original Message- From: Aaron Daniel [mailto:[EMAIL PROTECTED] Sent: Wednesday, August 02, 2006 1:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DUNDi with SIP Using the SECRET variable for sip doesn't work. On Wed, 2006-08-02 at 13:11 -0600, Douglas Garstang wrote: I've trying to use DUNDi with SIP to see if it works around some limitations of IAX2. I do a DUNDi lookup, get my SIP path, and try to dial it. Asterisk immediately says 'No such host', eventhough that's the path is just returned! [Aug 2 13:07:05] == Spawn extension (global_vmdeposit, u9220371, 3) exited non-zero on 'SIP/3254101-eb7d' [Aug 2 13:07:13] -- Executing NoOp(SIP/3254101-6373, *** OnNet originated call Chocolate Chip 3254101 - 9220371) in new stack [Aug 2 13:07:13] -- Executing AGI(SIP/3254101-6373, ipt/originator.py) in new stack [Aug 2 13:07:13] -- Launched AGI Script /var/lib/asterisk/agi-bin/ipt/originator.py [Aug 2 13:07:13] -- AGI Script Executing Application: (SetAccount) Options: (9220371) [Aug 2 13:07:13] -- AGI Script Executing Application: (ChanIsAvail) Options: (SIP/9220371) [Aug 2 13:07:14] -- AGI Script Executing Application: (Dial) Options: (SIP/dundisip:[EMAIL PROTECTED]/9220371) [Aug 2 13:07:14] WARNING[5429]: chan_sip.c:1980 create_addr: No such host: xxx.yyy.142.163/9220371 [Aug 2 13:07:14] NOTICE[5429]: app_dial.c:1040 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) [Aug 2 13:07:14] == Everyone is busy/congested at this time (1:0/0/1) Not sure what is going on. I can see the query at the other end, but it doesn't look like it ever receives the call. Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --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 -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --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: DUNDi with SIP
On Wed, 2006-08-02 at 15:52 -0600, Douglas Garstang wrote:
Well yes, it looked dubious to me too, although I can't find the syntaxt
documented anywhere.
However, that's what DUNDis giving me as a path to the phone!
Something is screwed with DUNDi and SIP. Has ANYONE actually implemnted it?
I can't find it documented anywhere
Doug.
DUNDi gives you only what you give it to give you. You're the one that
needs to set the dial string correctly in DUNDi to get one back that
works. DUNDi is only as automatic as you let it be.
This is what ours looks like. We don't use the iax versions (mainly
cause I want a homogenous SIP system), but we have entries in sip.conf
include files for each of the servers so we just dial
${server}/${number}. This has been working for us for about 2 months
now, pretty much flawlessly as long as the phone's registered.
e164 = dundi-extens,0,SIP,scm1/${NUMBER}
e164-iax = dundi-extens,0,IAX2,dundi:[EMAIL PROTECTED]/${NUMBER}
internal = dundi-extens,0,SIP,scm1/${NUMBER}
internal-iax = dundi-extens,0,IAX2,dundi:[EMAIL PROTECTED]/${NUMBER}
[scm1]
type=friend
secret=p4ssw0rd
insecure=very
context=incoming
host=scm1.shsu.edu
qualify=yes
nat=no
--
Aaron Daniel
Computer Systems Technician
Sam Houston State University
[EMAIL PROTECTED]
(936) 294-4198
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RE: [asterisk-users] Re: DUNDi with SIP
You don't have to have every host in your dundi.conf files. The way
we've got ours set up, each server tells the other servers how to reach
it. For example, our primary call server (scm1) publishes what numbers
it can handle, so I only list it's contexts in it's own file. Then,
sgw1 publishes what IT can handle. There's a matching e164 and internal
context on each server to tell the others what it can take.
On Wed, 2006-08-02 at 16:35 -0600, Douglas Garstang wrote:
-Original Message-
From: Aaron Daniel [mailto:[EMAIL PROTECTED]
Sent: Wednesday, August 02, 2006 4:02 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Re: DUNDi with SIP
On Wed, 2006-08-02 at 15:52 -0600, Douglas Garstang wrote:
Well yes, it looked dubious to me too, although I can't
find the syntaxt documented anywhere.
However, that's what DUNDis giving me as a path to the phone!
Something is screwed with DUNDi and SIP. Has ANYONE
actually implemnted it?
I can't find it documented anywhere
Doug.
DUNDi gives you only what you give it to give you. You're
the one that
needs to set the dial string correctly in DUNDi to get one back that
works. DUNDi is only as automatic as you let it be.
This is what ours looks like. We don't use the iax versions (mainly
cause I want a homogenous SIP system), but we have entries in sip.conf
include files for each of the servers so we just dial
${server}/${number}. This has been working for us for about 2 months
now, pretty much flawlessly as long as the phone's registered.
e164 = dundi-extens,0,SIP,scm1/${NUMBER}
e164-iax = dundi-extens,0,IAX2,dundi:[EMAIL PROTECTED]/${NUMBER}
internal = dundi-extens,0,SIP,scm1/${NUMBER}
internal-iax = dundi-extens,0,IAX2,dundi:[EMAIL PROTECTED]/${NUMBER}
[scm1]
type=friend
secret=p4ssw0rd
insecure=very
context=incoming
host=scm1.shsu.edu
qualify=yes
nat=no
Thanks Aaron, but I don't understand how that can work. Don't you have more
than one host in your DUNDi domain?
Doug.
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RE: [asterisk-users] Re: DUNDi with SIP
Yeah, they're both in the sip.conf files on each server in the system.
That's how I got the SIP/${server}/${exten} to work.
On Wed, 2006-08-02 at 17:01 -0600, Douglas Garstang wrote:
Ok, but don't you need to have [scm1] AND [sgw1] in sip.conf?
-Original Message-
From: Aaron Daniel [mailto:[EMAIL PROTECTED]
Sent: Wednesday, August 02, 2006 4:56 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Re: DUNDi with SIP
You don't have to have every host in your dundi.conf files. The way
we've got ours set up, each server tells the other servers
how to reach
it. For example, our primary call server (scm1) publishes
what numbers
it can handle, so I only list it's contexts in it's own file. Then,
sgw1 publishes what IT can handle. There's a matching e164
and internal
context on each server to tell the others what it can take.
On Wed, 2006-08-02 at 16:35 -0600, Douglas Garstang wrote:
-Original Message-
From: Aaron Daniel [mailto:[EMAIL PROTECTED]
Sent: Wednesday, August 02, 2006 4:02 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Re: DUNDi with SIP
On Wed, 2006-08-02 at 15:52 -0600, Douglas Garstang wrote:
Well yes, it looked dubious to me too, although I can't
find the syntaxt documented anywhere.
However, that's what DUNDis giving me as a path to the phone!
Something is screwed with DUNDi and SIP. Has ANYONE
actually implemnted it?
I can't find it documented anywhere
Doug.
DUNDi gives you only what you give it to give you. You're
the one that
needs to set the dial string correctly in DUNDi to get
one back that
works. DUNDi is only as automatic as you let it be.
This is what ours looks like. We don't use the iax
versions (mainly
cause I want a homogenous SIP system), but we have
entries in sip.conf
include files for each of the servers so we just dial
${server}/${number}. This has been working for us for
about 2 months
now, pretty much flawlessly as long as the phone's registered.
e164 = dundi-extens,0,SIP,scm1/${NUMBER}
e164-iax = dundi-extens,0,IAX2,dundi:[EMAIL PROTECTED]/${NUMBER}
internal = dundi-extens,0,SIP,scm1/${NUMBER}
internal-iax = dundi-extens,0,IAX2,dundi:[EMAIL PROTECTED]/${NUMBER}
[scm1]
type=friend
secret=p4ssw0rd
insecure=very
context=incoming
host=scm1.shsu.edu
qualify=yes
nat=no
Thanks Aaron, but I don't understand how that can work.
Don't you have more than one host in your DUNDi domain?
Doug.
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Re: [asterisk-users] Server redundancy
On Tue, 2006-07-11 at 14:34 -0400, Mike Lynchfield wrote: #3b.. we need multiple listening addies.. since asterisk can only listen to one ip its sucks for now Incorrect. Asterisk most definitely listens on multiple interfaces. We've got several asterisk boxes that are multi-homed... one public and one private interface, so that we can have external phones and internal phones. Works fine. I'm thinking this is a misconception. We even have heartbeat set up to switch ip's around. The server actually listens on the fly to the new ip address that comes up under it. -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --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] Server redundancy
Well, if you want to get into the nitty gritty of it, why not set up a firewall that allows only traffic from certain areas, and let asterisk do what it was designed for... handle phone calls. Am I wrong for thinking that's the more logical way of doing it? On Tue, 2006-07-11 at 17:17 -0400, Mike Lynchfield wrote: that what i meant as in /16 etc.. but That was case for asterisk 1.x is wrong too .. since 1.2.9.1is 1.x ;) my bad On 7/11/06, Douglas Garstang [EMAIL PROTECTED] wrote: I think it can listen either on a specific address, or on ALL addresses, not on a subset of available addresses. -Original Message- From: Aaron Daniel [mailto:[EMAIL PROTECTED] Sent: Tuesday, July 11, 2006 2:38 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Server redundancy On Tue, 2006-07-11 at 14:34 -0400, Mike Lynchfield wrote: #3b.. we need multiple listening addies.. since asterisk can only listen to one ip its sucks for now Incorrect. Asterisk most definitely listens on multiple interfaces. We've got several asterisk boxes that are multi-homed... one public and one private interface, so that we can have external phones and internal phones. Works fine. I'm thinking this is a misconception. We even have heartbeat set up to switch ip's around. The server actually listens on the fly to the new ip address that comes up under it. -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --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 -- Mike Sales Manager http://www.theclubvoip.com Making it happen 1.888.470.7253 ___ --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 -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --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] Text priority labels not working for me
The problem is in the space. You've got it as ? scid)... In order
for the label to work, you need to get rid of the space. Make it
?scid) and it should work fine.
The error's in the details:
pbx_extension_helper: No such label ' scid' in extension 's' in
context 'macro-dosomething'
On Tue, 2006-07-11 at 08:32 +0300, Tzafrir Cohen wrote:
On Mon, Jul 10, 2006 at 08:54:37PM -0700, Wes Santee wrote:
Greetings all,
I'm on 1.2.9.1, and I'm trying to get a dialplan working that uses text
labels, but it's not working. For instance, take the following macro
snippet:
[macro-dosomething]
exten = s,1,GotoIf($[${MACRO_EXTEN:1:1} != 1] ? scid)
exten = s,n,Set(MACRO_EXTEN=1${MACRO_EXTEN})
exten = s,n(scid),SetCallerId(${MY_CID})
exten = s,n,Dial(...)
When I call this macro, I get the following:
-- Executing Macro(SIP/1000-66b0, dosomething) in new stack
-- Executing GotoIf(SIP/1000-66b0, 1 ? scid) in new stack
Jul 10 20:05:52 NOTICE[99803]: pbx.c:1753 pbx_extension_helper: No such
label ' scid' in extension 's' in context 'macro-dosomething'
Jul 10 20:05:52 WARNING[99803]: pbx.c:6514 ast_parseable_goto: Priority
' scid' must be a number 0, or valid label
For starters, ask asterisk if there is.
In the CLI: show dialplan macro-dosomething
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Re: SV: [asterisk-users] Cisco 7941/7961/7971 wont register with asterisk
The 79X1 phones don't use the same configuration setups as the 79X0's. They're the upgraded versions, using the SEPmac.cnf.xml files instead of the SIPmac.cnf files. On Thu, 2006-07-06 at 10:44 -0400, Doug Lytle wrote: Per Møller wrote: Hey Doug, Yes my 7940 and 7960 using the 7.4 or 7.5 SIP firmware works fine and does not use xml style config files. I was looking for: Asterisk side: sip.conf Cisco side: SIPDefault.cnf SIPMacaddress.cnf Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco SIP Firmware
On Thu, 2006-07-06 at 10:55 -0500, Peder @ NetworkOblivion wrote: What is the current recommended version of firmware for SIP on 7960/7940's. I was reading through some of the stuff on voip-info and it looks like the 8.x's have pretty serious bugs in regards ti *. Thanks. PA We stick with the 7.4 firmware. It does exactly what we need, doesn't decide it doesn't want to forget about registration if the server falls out from under it, and doesn't have the server name in the caller id. -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --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] Looking for an asterisk guru
LOL, that's descriptive :) On Wed, 2006-07-05 at 16:03 -0400, Sean Cook wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 We are looking for an asterisk guru / linux geek for full time employment in the South West Virigina area. Please email me off list for more details. Sean -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.3 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFErBr11Kolm8VQlAURArW1AKDUitc4lbwCD665gMD+G5y7LJl5SQCfQMQ4 WlIZmCPlB6Gai0lkHuX13YY= =NWil -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --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] Looking for an asterisk guru
Whoops, sorry about that :) all I got in my client was Sean... ignore that message I sent. On Wed, 2006-07-05 at 15:10 -0500, Aaron Daniel wrote: LOL, that's descriptive :) On Wed, 2006-07-05 at 16:03 -0400, Sean Cook wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 We are looking for an asterisk guru / linux geek for full time employment in the South West Virigina area. Please email me off list for more details. Sean -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.3 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFErBr11Kolm8VQlAURArW1AKDUitc4lbwCD665gMD+G5y7LJl5SQCfQMQ4 WlIZmCPlB6Gai0lkHuX13YY= =NWil -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --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] Realtime SIP Registrations
On Thu, 2006-06-29 at 10:04 +0100, Tim Panton wrote: Yes, except, if I understand you correctly, you would also need to write insert and update triggers on the view, so that when asterisk writes to the compiled config, the correct changes are applied to your separate tables. That might limit your choice of databases a bit. The way I designed the second table, you wouldn't have to update any other tables with information from the sipregs table. The only information in there is information that asterisk needs to contact phones and such. So, for example, unless you need the ip address listed somewhere else in your database, you can leave it in sipregs. The other thing to watch is that you have to ensure that the resulting view behaves exactly the way that asterisk expects it to, unless you get the join right, you can get duplicate (apparently identical) records back which would confuse asterisk. That's something that you have to be careful about anyhow :) The way I'm looking at it, you can either use a view (we use 3 different tables for actual phone configuration... so a view makes sense). Or for smaller systems, use an actual sippeers table and put the info in there. -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --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] Realtime patch
On Thu, 2006-06-29 at 08:39 +0200, Patrick wrote: If I get some spare time I wouldn't mind playing around with the patch for 1.2.9.1. Can you please stick that one on bugs.digium.com too. I've uploaded the 1.2.9.1 patch as well. Let me know if you find anything I did wrong (I'm not much of a coder, so I'm sure I screwed something up). -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --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] Realtime SIP Registrations
On Thu, 2006-06-29 at 09:15 -0600, Douglas Garstang wrote: How about fixing realtime SIP so that multiple Asterisk boxes can reference the same database? Doug. That's kinda what I'm hoping to work towards :) -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --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] username in Real-time changes all the time
What kinda phone is it? That shouldn't affect the actual calls to the phone, I would expect. On Thu, 2006-06-29 at 23:49 +0800, Ronald Wiplinger wrote: I cannot explain that: One of my users shows up in sip show peers as 654200/Elmit_Unl I can set it back to 654200/654200 but it will change back to 654200/Elmit_Unl Why? bye Ronald Wiplinger ___ --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 -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Voicemail volume adjustment
The other problem is that if you add the gain to the original message, it seems to me the volume on the phone will be too loud as compared to the volume of the emailed message. Just a thought. On Wed, 2006-06-28 at 15:41 -0400, Cullin J. Wible wrote: Because: usg(2)[EMAIL PROTECTED] causes the app to exit with a non-zero status, Because: [EMAIL PROTECTED]|g(2) causes asterisk to hard crash. And trying to use g2 in either case doesn't work either. Cullin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dustin Wildes Sent: Wednesday, June 28, 2006 1:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Voicemail volume adjustment Why use an application like sox - when you can make the voicemail application do it natively: exten = s,1,Dial(SIP/100,10) exten = s,2,Voicemail([EMAIL PROTECTED]|ug(10)) The key is the g(10) parameter: From the 'show application voicemail': g(#) - Use the specified amount of gain when recording the voicemail message. The units are whole-number decibels (dB). -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Technical Support Sent: Tuesday, June 27, 2006 3:51 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Voicemail volume adjustment I frequently find voice messages are emailed to users with insufficient volume - barely audible. I would like to have asterisk run a sox command to adjust the volume of each message before emailing (perhaps once the message has been left). Has anyone done this? Care to share the steps? Thanks, MD ___ --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 -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --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] Realtime Voicemail
maxlogins =3D 3;
if ((s =3D ast_variable_retrieve(cfg, general, maxlogins))) {
if (sscanf(s, %d, x) =3D=3D 1) {
maxlogins =3D x;
} else {
ast_log(LOG_WARNING, Invalid max failed login attempts\n);
}
}
And yes, that should be documented somewhere... not a bad option to see
about writing in for a per user setting :)
Douglas Garstang wrote:
-Original Message-
From: Douglas Garstang=20
Sent: Tuesday, June 27, 2006 11:55 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Realtime Voicemail
I'm noticing that the documentation on the voip wiki for=20
voicemail and realtime voicemail hasn't kept up with reality.
I just created a column called maxmsg in my table. I set it=20
to 1 for the user. I can leave more than once voicemail message.
Why?
=20
Weird. Maxmsg suddenly worked on the next call. I tried setting maxlogi=
ns for the user to 1, and it's letting me put the wrong pin in 3 times be=
fore disconnecting me. What am I missing here? Are the supported options =
documented somewhere, that matches up with what's really in the code?
=20
Doug.
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[Asterisk-Users] Realtime SIP Registrations
Has anyone considered the idea of splitting the sip registration information in a realtime database from the actual configuration of the peers? I mean, instead of having a table full of the configuration information (i.e. name, regexten, secret, etc) and registration information (i.e. ipaddr, fullcontact, etc), you have separate tables with their own information. This way, you can have separate tables with config information, and use a view for the actual compiled configuration, instead of how it is now, where there may be repeating info all over the database. Does any of that make sense? -- Aaron Daniel signature.asc Description: OpenPGP digital signature ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Realtime patch
If anyone's interested, I've just put together a sip realtime patch, figured anyone that uses realtime may want to have a look at it. The patch basically takes the stuff asterisk updates (fullcontact, ipaddr, port, regseconds, and username) out of the sippeers table and puts it in it's own table. For those that are using multiple tables, this allows you to create a view of those tables that munges it together in a manner that makes sense to asterisk, while alleviating some of the management from you, as well as letting you make a table structure that makes sense to you ;) http://bugs.digium.com/view.php?id=7443 The one on the bugs site is for SVN, but I do have a version that works on 1.2.? (only tried on 1.2.9.1, so it may work on older versions). So far it seems to work well. -- Aaron Daniel signature.asc Description: OpenPGP digital signature ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemail volume adjustment
There's a patch up on the bugs site that uses sox to raise/lower the volume of wav's. We've been using it for about 5 months now without any glaring issues (we also convert to mp3 and do a few other things, but this patch is only the volume part). http://bugs.digium.com/view.php?id=6237 On Tue, 27 Jun 2006, Mojo with Horan Company, LLC wrote: I know this isn't what you asked for, but I think I achieved the same effect by changing the format= line to put wav first, not wav49. Of course, if you've already modified this line to use gsm or something else, then this really wont' help :P Moj Technical Support wrote: I frequently find voice messages are emailed to users with insufficient volume - barely audible. I would like to have asterisk run a sox command to adjust the volume of each message before emailing (perhaps once the message has been left). Has anyone done this? Care to share the steps? Thanks, MD ___ --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 !DSPAM:500,44a18bde194498152433266! -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --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 load balance
On Fri, 16 Jun 2006, Douglas Garstang wrote: Unless you can guarantee that the system that is currently processing a call will be the system that handles a transfer request from a phone, are the same, then transfers will not work. Incorrect. Transfers work fine between multiple asterisk boxes. Round robin DNS won't work at all. Every time you send out a SIP message, your going to be sending it to a different Asterisk box. For example, your initial INVITE will go to asterisk server 1. Asterisk server 1 will then send back a message requesting authorisation. Your phone does another lookup, and gets Asterisk server 2 this time. The phone sends the new INVITE with the auth info to Asterisk server 2. Asterisk server 2 will probably be ok with this, but when it sends a TRYING back to the phone, depending on the phone you are using, everything will fall in a heap on the floor. I know polycoms do. They get this TRYING from an asterisk server they didn't send and they go 'huh?'. This is entirely phone dependant, and usually the phones that fall in a heap (like the phrase much?) also handle secondary server configurations MUCH better than the phones that don't. Polycoms and sipura's handle SRV and backup server settings better than cisco's, but cisco's won't jump from server to server. I'm sure most other stuff will fail too. The Asterisk boxes share no state information. It's all in how you program the dial plan. The main thing that doesn't share state information that may cause problems is hinting. Everything else is programmable somewhere in the system :) -Original Message- From: unplug [mailto:[EMAIL PROTECTED] Sent: Fri 6/16/2006 9:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: [Asterisk-Users] asterisk load balance Hi, I am designing a asterisk load balancing model as follow. There are 3 asterisks connected to a single DB and a single server storing all the configuration file and voicemail. Round Robin DNS will distribute the request to asterisks. DNS round robin ---+ asterisk1--+ DB and file server +---asterisk2---+ +---asterisk3---+ Your design would work just fine as long as you have your dialplan is configured right. Keep in mind though that if asterisk1 dies, you just lost your db. Does anyone has load balancing experience implemented in asterisk that can share? Does my design work? Does any conflict will happen in my design? Any comment? 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 -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --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 load balance
On Sat, 17 Jun 2006, Douglas Garstang wrote: Good grief I hate Outlook webmail. I can't reply inline. Switch to thunderbird ;) Anyway, I disagree that all state info except hinting can be replicated. What about call transfers? If a call is sitting on pbx1, and the user transfers a call, if it goes to pbx2, Asterisk will complain that it cannot transfer the call as it doesn't know anything about it Well, I'm not sure what the problem with call transfers is. We have two registration servers, in which the phones can and do register with either server. If one phone makes a call on one server, they can complete the call with anyone else on their server, plus anyone on the other servers. The server just treats the transfer and bridge like any other phone call. If the phone is on another server, it hands off the conversation to that server after the transfer. And I think I'll address your NFS problems. Are you doing that for redundancy's sake or just for MWI? If it's just for MWI, then you might be better off setting up some scripts that drop some msg.txt files in the user's voicemail box on the registration servers. No need to replicate registration to the voicemail server, that's just extra unneeded traffic. Plus, with something like that, you don't have to worry about the voicemail nfs share dying and bringing down the asterisk network. If it's for redundancy, set up another voicemail server or two, and use DRBD or some sort of sync tool between them, with the MWI script and you'll have fixed the redundancy problem. -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Voicemail with NFS
On Sat, 17 Jun 2006, Douglas Garstang wrote: Other applications can handle it. Don't see why Asterisk can't. Mount the nfs volume with the -soft option. Do a 'df -k' and you will see that the df command will time out in a couple of seconds. Why can't Asterisk do the same? Just gonna throw gas on the fire. df -h doesn't continuously poll the drive for data, asterisk is (for mwi). So each timeout turns into another timeout. Didn't you already test changing the time on checkmwi? And did it not change the behavior (not necessarily for the better)? -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --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] Anyone see this?
Dunno if anyone else has seen this yet: http://www.scmagazine.com/us/news/article/563800/vulnerabilities+put+asterisk+telephone+systems+risk/ -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --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] DUNDi Not Able to Handle Complex FailoverSituations
On Thu, 15 Jun 2006, Douglas Garstang wrote: It isn't as simple as that. When a failure occurs, we only want to use a DUNDi route when it's the primary for a queue. Then don't use DUNDi for queues, use it just for the phones. Seriously, you obviously know exactly which servers you want to be primary for a certain queue, program it into the dialplan. DUNDi should only be used for DYNAMIC extensions, i.e. phones that may or may not be registered at the time of the call, phones that move, phones that register with different servers at different times. If you're deadset on using DUNDi for it, set up different DUNDi contexts so that you can say these queues are available here and these queues are available there. Honestly, it seems like a waste of server time to use DUNDi for something that you know is going to be on a particular server regardless of what happens. -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --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] DUNDi Not Able to Handle Complex FailoverSituations
On Thu, 15 Jun 2006, Douglas Garstang wrote: DUNDi does not handle the situation of phone failover as well as static numbers (ie queues), which is what we are trying to acheive. I'm confused, explain the phone failover not working to me. -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --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] DUNDi Not Able to Handle ComplexFailoverSituations
On Thu, 15 Jun 2006, Douglas Garstang wrote: We need our queue application to follow the primary pbx server for a set of phones within a company. See my 'ACD Distributed Scenario' post made a little earlier for a full explanation. OK, let me get this straight. You want the phones on the SAME server to hit the queues on THAT server only. Right? If that's right, then why use DUNDi for the queues, just set up an extension (i.e. the queue entry point) that goes straight into the queue instead of using DUNDi for it, which adds more logic to something VERY simple. Since the phones are registered to that server, obviously they will drop into the local queue and not some random one. You're making something dynamic that really shouldn't be dynamic at all. When the failover happens, the new primary server will have the queue set up, and anyone calling in will be calling into the queue on that server. Now, if you're calling in from another server, i.e. someone outside calling in, you can then use DUNDi with weights to drop them onto the right server, but that's another story. Finally, in order for the LOCAL server's DUNDi response to show up, you have to add the server to dundi.conf. So, so pbx1 has to be in pbx1's file, just like the other servers do. Make sense? -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --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] DUNDi Not Able to HandleComplexFailoverSituations
On Thu, 15 Jun 2006, Douglas Garstang wrote:
No... this last bit doesnt. My dundi.conf has:
180q = global_dundi_q_pbx1,100,IAX,dundi1:[EMAIL PROTECTED]/${NUMBER},nopartial
180q = global_dundi_q_pbx2,200,IAX,dundi2:[EMAIL PROTECTED]/${NUMBER},nopartial
180q = global_dundi_q_pbx3,300,IAX,dundi3:[EMAIL PROTECTED]/${NUMBER},nopartial
What are you suggesting I change it to? Something like this?
180q = global_dundi_q_pbx1,100,IAX,dundi1:[EMAIL PROTECTED]/${NUMBER},nopartial
180q = global_dundi_q_pbx2,200,IAX,dundi2:[EMAIL PROTECTED]/${NUMBER},nopartial
180q = global_dundi_q_pbx3,300,IAX,dundi3:[EMAIL PROTECTED]/${NUMBER},nopartial
I really don't follow.
Here's an example. We have two primary call servers, both are capable of
handling the call volume if one fails out. They're scm1 and scm2.
scm1 has a peer section for itself, so it shows up during lookups. scm2
has a peer section for itself as well. They also have peer sections for
each other and for the gateways:
scm1:
[00:E0:81:25:28:D3]
model = symmetric
host = sgw1
inkey = sgw1
outkey = scm1
include = all
permit = all
qualify = yes
[00:14:22:13:90:8D]
model = symmetric
host = scm1
inkey = scm1
outkey = scm1
include = all
permit = all
qualify = yes
[00:14:22:13:B6:B6]
model = symmetric
host = scm2
inkey = scm2
outkey = scm1
include = all
permit = all
qualify = yes
[00:13:72:4E:D7:54]
model = symmetric
host = sgw2
inkey = sgw2
outkey = scm1
include = all
permit = all
qualify = yes
scm2 will be exactly the same except it has an outkey of scm2. This
should fix your issue with having dundi lookup on the local machine.
I'm not gonna try to understand your ACD stuff right now, so I'll just
figure you need DUNDi for that and give up on it :) Too busy fixing the
voicemail app.
--
Aaron Daniel
Computer Systems Technician
Sam Houston State University
[EMAIL PROTECTED]
(936) 294-4198
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RE: [Asterisk-Users] DUNDi Not Able to HandleComplexFailoverSituations
On Thu, 15 Jun 2006, Douglas Garstang wrote: Ahh this reminds me too. If I am going to be getting the local system first always, then I need to be able to return ALL the Dundi paths with the DUNDILOOKUP function. It only returns one. How can I get DUNDILookup to return every single path? It'd be great if they could return the weights for each too, and then I could do my own logic. But you're still not getting the point of DUNDi. DUNDi is designed to pick the best route. When you do a dundi lookup on the CLI, it shows you what possible jumps there are, in the order that it would choose them. DUNDi will ALWAYS use the first one. Also, the logic you want to do would involve an unknown size array of return values. For example, let's say you can get to an extension on 4 different servers. DUNDi will return something like: 1. 0 SIP scm1/44198 2. 1 SIP scm2/44198 3. 100 SIP sgw1/44198 4. 101 SIP sgw2/44198 If you return that to the dialplan, you're going to have to do some array lookups and parsing in order for any of it to make sense. Correct me if I'm wrong, the dialplan doesn't have any array functionality. Does that make sense? -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --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] DUNDi Docs
On Wed, 14 Jun 2006, Douglas Garstang wrote: The examples in dundi.conf are pretty much useless. I still can't figure out why Digium can't write some good documentation. It's their 'baby' after all. This really drives me nuts and pisses people off in general. I've been dicking around with DUNDi for over 6 months and still can't figure it out past the most basic application. What are you trying to do? -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --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] DUNDi Users
If you do a reload pbx_dundi.so, it'll reload the dundi configuration. If
you're talking about the strings it returns, if you want to get an
immediate result and not use the cache, use something like dundi lookup
num bypass.
Also, if you have separate entry points for each section of the dundi
numbers, you're going to have to have separate users to identify where the
call's coming from. If you only use one iax user, you can only use one
context. That's like trying to put a phone in two different contexts...
where is it supposed to start it's dialing attempts? If you really want,
create a context in extensions.conf that includes the other three, because
that seems to be the functionality you are attempting.
Seems to make sense to me, not sure what's horrible about it :)
On Wed, 14 Jun 2006, Douglas Garstang wrote:
It has also just become glaringly apparent to me that a 'reload' does not
always reload the DUNDi configuation.
How can I reload DUNDi without stopping/starting Asterisk?
-Original Message-
From: Douglas Garstang
Sent: Wednesday, June 14, 2006 11:00 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] DUNDi Users
I have three Asterisk boxes.
Each has the following in dundi.conf:
180net =
dundi_local,0,IAX,dundi:[EMAIL PROTECTED]/${NUMBER},nopartial
180q =
dundi_q_pbx1,1,IAX,dundi:[EMAIL PROTECTED]/${NUMBER},nopartial
180q =
dundi_q_pbx2,2,IAX,dundi:[EMAIL PROTECTED]/${NUMBER},nopartial
180q =
dundi_q_pbx3,3,IAX,dundi:[EMAIL PROTECTED]/${NUMBER},nopartial
My iax.conf on all three Asterisk boxes has this:
[dundi]
type=user
dbsecret=dundi/secret
context=dundi_local
disallow=all
allow=ulaw
allow=g729
I can do a lookup on pbx2 to find where a number is:
hermes*CLI dundi lookup [EMAIL PROTECTED]
1. 1
IAX2/dundi:[EMAIL PROTECTED]/oe_main (EXISTS)
from 00:0e:0c:a1:92:6f, expires in 0 s
2. 1
IAX2/dundi:[EMAIL PROTECTED]/oe_main (EXISTS)
from 00:0e:0c:a1:92:4d, expires in 0 s
DUNDi lookup completed in 53 ms
However, when I dial the DUNDi path, this is what pbx1 logs
on the console:
Jun 14 10:51:39 NOTICE[22424]: chan_iax2.c:7215 socket_read:
Rejected connect attempt from xxx.187.142.204, request
'[EMAIL PROTECTED]' does not exist
I tried adding the contexts to [dundi] in iax.conf:
[dundi]
type=user
dbsecret=dundi/secret
context=dundi_local
context=dundi_q_pbx1
context=dundi_q_pbx2
context=dundi_q_pbx3
disallow=all
allow=ulaw
allow=g729
However, the call on pbx1 is still routed to the dundi_local
context instead of dundi_q_pbx1.
Do I have to go and modify dundi.conf, so that every dundi
entry uses a different DUNDi user, like this?
180q =
dundi_q_pbx1,1,IAX,dundi1:[EMAIL PROTECTED]/${NUMBER},nopartial
180q =
dundi_q_pbx2,2,IAX,dundi2:[EMAIL PROTECTED]/${NUMBER},nopartial
180q =
dundi_q_pbx3,3,IAX,dundi3:[EMAIL PROTECTED]/${NUMBER},nopartial
And then add users dundi1, dundi2 and dundi3 to iax.conf?
I sure hope not. What a horrible way to have to do it.
Doug.
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--
Aaron Daniel
Computer Systems Technician
Sam Houston State University
[EMAIL PROTECTED]
(936) 294-4198
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Re: [Asterisk-Users] DUNDi Not Able to Handle Complex Failover Situations
On Wed, 14 Jun 2006, Douglas Garstang wrote: Why doesn't the DUNDILOOKUP function return the weight of a path to a number? The CLI 'dundi lookup' command does. What about the mac address and expiry period? The CLI command returns those, but the DUNDILOOKUP function does not. Why? Correct me if I'm wrong, but DUNDi is doing all the failover work for you. It decides based on the weights what route is best. If you want one route to be higher than another, set it up that way. That's the benefit of using DUNDILOOKUP to handle it, no more work for you after the initial routing. If that doesn't work for you, program the routes directly into the dialplan instead of using DUNDi, it seems like you'll get better results that way. We did that for a while until we decided to move to DUNDi. Some people will find it more suited to their needs, some won't. -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --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] Unlock / install of Cisco 7940 IP Phone ?
You have to press settings, then **#, and wait a moment to make sure it unlocks. Then you can configure a tftp server to use. The alternative is to configure your dhcp server with a tftp server. On linux, that would be next-server ip/host in the subnet section. TFTP server on windows. On Wed, 7 Jun 2006, Mateo Meier wrote: Hello Guys, I just got my new Cisco 7940* IP Phone Unfortunately I can't find out how to setup this phone. Am I really that stupid ;-) ? I've read the Cisco Manuel, but everything I tried did not help anything. 1. When I turn on the phone it will display Configuring VLANConfiguring IP.. This message will not disappear. 2. I can see that the phone has a local IP. I can also access the IP over my LAN with http (only http, telnet does not work) 3 My Main menu will this show Configuring VLANConfiguring IP.. But if I click on settings, network settings it will show me the local IP of the phone Now, my question, what do I do wrong ? how can I get that phone installed with a sip image ? I tried to unlock the phone with **# but that does not do anything. Also, there is no unlock function in the phone menu (phone settings) This is a new Cisco phone, no sip image on it. Thank you for the help Matt ___ --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 -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --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: AW: [Asterisk-Users] Unlock / install of Cisco 7940 IP Phone ?
Here's a suggestion :) Depending on your phone, there's about a thousand different methods to upgrade them. Take a look at this, it explains how to do it depending on model, current firmware version, future firmware version. http://www.cisco.com/en/US/products/sw/voicesw/ps4967/products_upgrade_guides09186a008022a968.html On Wed, 7 Jun 2006, Mateo Meier wrote: Thx guys for the help. I was able to unlock it. I see the following under Network Configuration : 8 TFTP Server 1 On 32 I can see 32 Alternate TFTP NO Shall I type in the IP of the TFTP Server there ? if yes, how ? BTW: Im assuming that TFTP has to be installed on my local windown pc, and then let my phone access it over the local ip, correct ? Thx again for ur help! Grüsse / Best Regards Matt -- « hola! » see...habla espanol :-) -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von [EMAIL PROTECTED] Gesendet: Mittwoch, 7. Juni 2006 23:29 An: Asterisk Users Mailing List - Non-Commercial Discussion; Asterisk Users Mailing List - Non-Commercial Discussion Betreff: Re: [Asterisk-Users] Unlock / install of Cisco 7940 IP Phone ? To convert the phone to SIP you have to unlock and set the TFTP server to your TFTP server address as explained below. But... you also need to make sure you have a SIP image in the root directory of your TFTP server and also edit the OS79XX.TXT file to contain only the SIP image name (no .bin or .sbin extension). This should tell the phone what image it should be running and will then start a tftp transaction to download the SIP image. Once the SIP image is loaded on your phone, you will have to reset the TFTP Server addresses again so it can then download the SIP configuration files (SIPdefault.cnf and SIPMAC.cnf) good luck! -- Original message -- From: Aaron Daniel [EMAIL PROTECTED] You have to press settings, then **#, and wait a moment to make sure it unlocks. Then you can configure a tftp server to use. The alternative is to configure your dhcp server with a tftp server. On linux, that would be next-server ip/host in the subnet section. TFTP server on windows. On Wed, 7 Jun 2006, Mateo Meier wrote: Hello Guys, I just got my new Cisco 7940* IP Phone Unfortunately I can't find out how to setup this phone. Am I really that stupid ;-) ? I've read the Cisco Manuel, but everything I tried did not help anything. 1. When I turn on the phone it will display Configuring VLANConfiguring IP.. This message will not disappear. 2. I can see that the phone has a local IP. I can also access the IP over my LAN with http (only http, telnet does not work) 3 My Main menu will this show Configuring VLANConfiguring IP.. But if I click on settings, network settings it will show me the local IP of the phone Now, my question, what do I do wrong ? how can I get that phone installed with a sip image ? I tried to unlock the phone with **# but that does not do anything. Also, there is no unlock function in the phone menu (phone settings) This is a new Cisco phone, no sip image on it. Thank you for the help Matt ___ --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 -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --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 -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Busy Signals after hangup
On Mon, 5 Jun 2006, Andrew Kohlsmith wrote: I'll bet a donut it's not a busy signal but rather a fast busy which is known as a congestion signal. I'll be a jelly filled donut that it's the device he's using and not asterisk sending the signal :) We have a few ATA's that don't automatically hang up even though the call has ended, they just do the congestion symbol. It's caught me off guard a couple times. -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --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] Config Revision Control
While I was playing with svn, it was driving me nuts. It would ALWAYS re-create the current directory, even if I said to check out all files from inside that directory. Means if you went to /etc/asterisk and checked out asterisk, you'd get /etc/asterisk/asterisk. Yuk. Doug. Ahem. cd /etc/asterisk svn update -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --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] Config Revision Control
No, if you do an svn co http://svn.server.com/svn/configs/trunk asterisk in /etc, it'll make a folder called asterisk in your /etc directory. Once that's done, any modifications made that are committed to the server can be downloaded into /etc/asterisk by running svn up inside the directory. Might need to get your brakes checked if you keep hitting walls :) On Fri, 2 Jun 2006, Douglas Garstang wrote: Ok, does anyone know if anyone has already created a guide for using subversion with Asterisk? I've hit a wall already, where the subversion docs say that your files _must_ go into a directory called trunk(huh? What's with that?). That's going to break Asterisk, who obviously wants conf files in /etc/asterisk. Gr. -Original Message- From: Watkins, Bradley [mailto:[EMAIL PROTECTED] Sent: Friday, June 02, 2006 3:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Config Revision Control The first situation you mention can be solved by creating separate files that contain the unique elements, and then including them in the main files where all the commonality is. That is how we do things, and it works well for us. It may be a little cumbersome if you have a *lot* of uniqueness, but if you really want to share a significant portion of the configs this is the only way I know of to do it. As for revision control, we use Subversion with a branch for each server containing the unique files. All of our configuration scripts also include automatic checkins of changed files (we can always revert if need be). It also makes it easy to spot changes if something goes wrong, as an svn diff will tell you. Regards, - Brad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Douglas Garstang Sent: Friday, June 02, 2006 4:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Config Revision Control Has anyone got any neat solutions for Asterisk .conf file revision control? We have multiple Asterisk boxes here, that we'd like to maintain a _mostly_ common set of conf files on. They aren't all the same though. There's subtle differences. For example, in sip.conf, iax.conf etc, the bindaddr setting is different. Dundi.conf is very different between each system. At the moment I have a file tree on a separate server, and I use the m4 processor to replace certain unique sections of the files. I have a bunch of scripts to build sip.conf etc and then rsync the files out to the servers. It works, mostly, but it isn't elegant. I'd like to revision control all this. I don't know how it could be done with revision control though. As I said, not all the files are the same. I don't know if we'd run a version control client on each Asterisk box, or if we'd run it centrally, and then use rsync again, to copy the files out. Doug. =00The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --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] Config Revision Control
Read this: http://subversion.tigris.org/faq.html#repository http://svn.collab.net/repos/svn/trunk/README That'll link you to the README that comes with subversion, which has a very detailed explanation on how to get a repo set up and running :) If it says anything in there about using trunk, it's just a suggestion. Ours is split out by server name inside a configs folder. On Fri, 2 Jun 2006, Douglas Garstang wrote: Aaron, I'm trying to check-in (is that the right term?) the files for the first time. There's nothing in the repository yet. Doug. -Original Message- From: Aaron Daniel [mailto:[EMAIL PROTECTED] Sent: Friday, June 02, 2006 3:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Config Revision Control No, if you do an svn co http://svn.server.com/svn/configs/trunk asterisk in /etc, it'll make a folder called asterisk in your /etc directory. Once that's done, any modifications made that are committed to the server can be downloaded into /etc/asterisk by running svn up inside the directory. Might need to get your brakes checked if you keep hitting walls :) On Fri, 2 Jun 2006, Douglas Garstang wrote: Ok, does anyone know if anyone has already created a guide for using subversion with Asterisk? I've hit a wall already, where the subversion docs say that your files _must_ go into a directory called trunk(huh? What's with that?). That's going to break Asterisk, who obviously wants conf files in /etc/asterisk. Gr. -Original Message- From: Watkins, Bradley [mailto:[EMAIL PROTECTED] Sent: Friday, June 02, 2006 3:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Config Revision Control The first situation you mention can be solved by creating separate files that contain the unique elements, and then including them in the main files where all the commonality is. That is how we do things, and it works well for us. It may be a little cumbersome if you have a *lot* of uniqueness, but if you really want to share a significant portion of the configs this is the only way I know of to do it. As for revision control, we use Subversion with a branch for each server containing the unique files. All of our configuration scripts also include automatic checkins of changed files (we can always revert if need be). It also makes it easy to spot changes if something goes wrong, as an svn diff will tell you. Regards, - Brad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Douglas Garstang Sent: Friday, June 02, 2006 4:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Config Revision Control Has anyone got any neat solutions for Asterisk .conf file revision control? We have multiple Asterisk boxes here, that we'd like to maintain a _mostly_ common set of conf files on. They aren't all the same though. There's subtle differences. For example, in sip.conf, iax.conf etc, the bindaddr setting is different. Dundi.conf is very different between each system. At the moment I have a file tree on a separate server, and I use the m4 processor to replace certain unique sections of the files. I have a bunch of scripts to build sip.conf etc and then rsync the files out to the servers. It works, mostly, but it isn't elegant. I'd like to revision control all this. I don't know how it could be done with revision control though. As I said, not all the files are the same. I don't know if we'd run a version control client on each Asterisk box, or if we'd run it centrally, and then use rsync again, to copy the files out. Doug. =00The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --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 -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --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] NFS and voicemail
Has anyone successfully gotten two separate servers to handle voicemail on an NFS shared mount? The main thing I'm concerned about at this point is keeping both systems from writing the voicemail file to the same filename... any thoughts? -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --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
