Re: [asterisk-users] Asterisk as a skinny/sccp client?
On 13:33, Wed 17 Mar 10, Brian J. Murrell wrote: On Wed, 2010-03-17 at 10:56 -0500, Jason Parker wrote: No, this isn't currently possible. Damn. I did ponder this for a while, but my conclusion was that the effort required to do so would far outweigh any benefit you'd gain from it. How about having something -- anything on Linux, able to connect to a Cisco PBX using Skinny, where one has no other options? Cisco has been moving to SIP for a very long time. There aren't any phone features that Asterisk could emulate that would make this any better than SIP (or even anything approaching parity). Perhaps not, but there are still installations where all they are using is Skinny and nothing else. You don't need to tell me about the pros/cons of this as I have no control over what they are doing. I am just a consumer of it. Unfortunately, I think Asterisk and skinny is about as close as anything is on Linux to utilizing SCCP. :-( At the moment the primary focus for chan_skinny is to make it a better server for the phones. Like Jason said, it would take a great ammount of time and effort to implement 'client side' code. And as most of the ppl involved in chan_skinny mostly care about fixing stuff so their phones work correctly this would be near to impossible with the current setup. Of course we are open for patches, and we're accepting them as well as licenses for callserver to test them. Like Jason I've been looking into this, but lack of resources and lack of interrest and lack of usage I stopped looking at it and linked my asterisk to callserver using SIP. -- Michiel van Baak mich...@vanbaak.eu http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7940: showing FWD in display.
On 21:40, Sun 14 Feb 10, Oliver Nittka wrote: Olivier schrieb: Thanks for the suggestion anyway, I'm going to test this just out of curiosity :-) And that's what i get in the CLI: Got SIP response 501 Not Implemented back from XXX.XXX.XXX.XXX Well, I guess I should really give chan_sccp another shot ... Or use the provided chan_skinny -- Michiel van Baak mich...@vanbaak.eu http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Important security alert: update your?dialplans now!
On 08:48, Mon 15 Feb 10, Tilghman Lesher wrote: On Monday 15 February 2010 03:37:24 Rob Hillis wrote: On 02/15/10 20:00, Randy R wrote: Olle, this may be a stupid question, but shouldn't a native santitize function be urgently added to the code base in all versions or change the dialplan comp?ler to ignore dangerous characters? Whilst I agree with this, the unfortunate attitude we seem to get from Digium on most of these issues is you can already do this in dialplan, therefore we don't need to invest any effort in it. The fact that a workaround may be quite difficult to implement properly doesn't come in to it. The most obvious example of this one is the deprecation and removal of chan_agent without any sort of replacement being introduced because it's already possible to do in the dialplan. Uh, chan_agent has been neither removed nor deprecated. He probably means AgentCallbackLogin -- Michiel van Baak mich...@vanbaak.eu http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk IPv6 update - we need an update
is as always o...@edvina.net. Please don't hesitate to mail me with any questions you might have about this project. Thank you for your support. Best regards, /Olle -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michiel van Baak mich...@vanbaak.eu http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Virtual Asterisk Installation
On 23:28, Wed 20 Jan 10, Felix Tiefenthaler wrote: Hi all! I've been reading this list for a few weeks and now this is my first post. :-) I'm planning to build a new VoIP telephone system at our company. It's just a small company with not more than 3-4 employees. The telephone system is not so important for us because each employee has it's own mobile phone. Because our company is a small one, we don't want to/we can't buy an expensive phone system. So we are going to use Asterisk. Additionally we don't want to obtain extra hardware. We have already a Server running with Linux (for Network monitoring). Because it's a waste to use this Server just for monitoring I thought about virtualizing. Now I want to run a machine with monitoring and a machine with Asterisk on this Server. I already bought a ISDN Card (berofix 400) with a S0 module. Now my big question: What kind of virtualization should I run on the Server? I have already used VMware ESXi and Proxmox. It would be very nice if there was a way to make snapshots (for backup purposes). I read about clock problems (physical time != virtual time) and so on. If I'm right this does not matter when using OpenVZ but when using KVM, XEN, ESX, ... Please tell me your opinion. I definitely want to run the Asterisk via virtualization - so we have to find a solution for this ;-) Forget about virtualization! This system is running linux as base os (I conclude by the tone of your mail) Just install asterisk on it besides the monitoring software and be done with it. What do you gain by running virtualisation on it ? Nothing. snapshots are not bound to virtualisation. Just redo the box with lvm and you can make snapshots with that. Virtualisation is nice for test-setups, but thats it. for any real job it's a major pain in the ass and makes stuff bork beyond imagination. -- Michiel van Baak mich...@vanbaak.eu http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ChanIsAvail querry
On 14:59, Wed 25 Nov 09, ABBAS SHAKEEL wrote: Hello We need to know if a channel is not in use and can be used to dial a number etc.. I have tried the ChanIsAvail function with different parameters. ie ChanIsAvail(DAHDI/1DAHDI/2) ,ChanIsAvail(DAHDI/1,s) etc no matter the channel is busy or not it always return 0 . Please suggest As the documentation will tell you: This application sets the following channel variable upon completion: AVAILCHAN - the name of the available channel, if one exists So check the contents of that variable after running ChanIsAvail() -- Michiel van Baak mich...@vanbaak.eu http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How many lines do you use.
On 10:18, Wed 25 Nov 09, Julian Lyndon-Smith wrote: Just for some information really : How many of you use multiple sip lines on a phone ?. I'm sitting here looking at my 7960, with it's 6 lines. I've every only used one line, and I was wondering if I was a weirdo ;) The only time I've ever found a use was when I had two systems (production and test) and it caused so much grief (could have been asterisk or cisco) I simply use a softphone for testing now. Curious minds are wanting to know ... I use three lines on my cisco 7960 (not sip, but that's not really relevant here) 1 - Private home number 2 - Daytime job number I got from work and is redirected to my home asterisk box from the office pbx 3 - number for my private business. The other three buttons are speeddial. -- Michiel van Baak mich...@vanbaak.eu http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ChanIsAvail querry
On 16:54, Wed 25 Nov 09, ABBAS SHAKEEL wrote:
Dan I have reverted to 1.4.27 but got no success. Same behaviour
Do anyone has any success with it ?
This ael snippet is working great for me on current -trunk.
I have been using this for some time now, it's from before 1.6 got
branched so it should work there as well I think.
Verbose(1,Routing call from ${CALLERID(num)} (${CALLERID(name)}) to
${EXTEN} on channel ${CHANNEL});
ChanIsAvail(Skinny/6000Skinny/6002SIP/michiele71,a);
if ( x${AVAILORIGCHAN} != x ) {
Verbose(1,Calling available channels: ${AVAILORIGCHAN});
Dial(${AVAILORIGCHAN},45,htxk);
}
On Wed, Nov 25, 2009 at 3:54 PM, ABBAS SHAKEEL
shakeel.abbas@gmail.comwrote:
Thanks Michiel and Dan
@ Michiel i have checked the variables but they dont contain any value.
@Dan I am using 1.6.1.2 May be some issue with it ... In the mean while
let me test with an older version of asterisk
On Wed, Nov 25, 2009 at 3:19 PM, Dan Journo
d...@keshercommunications.comwrote:
What version of Asterisk are you using?
I think this might be related to an issue that was resolved in version
1.4.27
http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-1.4.27-summary.html-
look in the list of Closed Items, second one down.
https://issues.asterisk.org/view.php?id=14426 ? link to the issue
Hope that helps.
Dan Journo
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *ABBAS SHAKEEL
*Sent:* 25 November 2009 09:59
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] ChanIsAvail querry
Hello
We need to know if a channel is not in use and can be used to dial a
number etc..
I have tried the ChanIsAvail function with different parameters.
ie ChanIsAvail(DAHDI/1DAHDI/2) ,ChanIsAvail(DAHDI/1,s) etc
no matter the channel is busy or not it always return 0 .
Please suggest
FYI
ChanIsAvail(Technology/resource[Technology2/resource2...][,options]):
This application will check to see if any of the specified channels are
available.
Options:
a - Check for all available channels, not only the first one.
s - Consider the channel unavailable if the channel is in use at all.
t - Simply checks if specified channels exist in the channel list
(implies option s).
This application sets the following channel variable upon completion:
AVAILCHAN - the name of the available channel, if one exists
AVAILORIGCHAN - the canonical channel name that was used to create the
channel
AVAILSTATUS - the status code for the available channel
--
Best Regards
Shakeel Abbas
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Best Regards
Shakeel Abbas
--
Best Regards
Shakeel Abbas
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Re: [asterisk-users] How many lines do you use.
On 17:03, Wed 25 Nov 09, Robert Lister wrote: On Wed, 2009-11-25 at 08:46 -0500, David Gibbons wrote: I use two ???lines??? though ???Line appearances??? would be a better term, though still confusing in my book. One line for incoming, one line that auto-answers for paging. Cisco really has so many line appearances on their phones to enable BLF using SIP over TCP. Cisco 7960 does not do BLF (at least not on the SIP firmware) but the 7961 might. It's a shame they haven't added such features, but there we go.) It does with the skinny firmware :) If you enable two line keys with the same user/pass then the phone will automatically put a second call/call waiting onto the second line key (assuming you have call waiting enabled.) But personally I preferred the way it presented the second call before, on a single line, and found the way it displays it with two lines a bit confusing. (I can't remember exactly why now, something like it would flash the second line icon but not show you the call information until that key was pressed, or you scrolled to it.) I could see users not getting on with this, so I didn't configure it like that. The rest can be used for speed dials, but these were of limited use to me since for some reason, although the line keys can be provisioned remotely over TFTP, the speed dials cannot. It's okay for personal use though. With the skinny firmware you configure the lines and speeddials in asterisk skinny.conf :) Personally moved off my 7960 in favour of the SNOM 370 as this supports far more features than the Cisco SIP image, which is only really a piece of migration fluff to enable Cisco to migrate customers away from competitors SIP systems onto Call Manager with the dual-boot/application loader. Asterisk has chan_skinny. The SNOM perhaps doesn't look as fancy as the Cisco handset, but it wins hands-down on SIP features. (the remote provisioning system was a little complicated to set up, but once set up it's okay.) It's a shame since the Cisco is a very capable (and expensive) handset, just let down by no development in the software other than small bug fixes for many years. If you dont like it, send the cisco to wedhorn or me so we can make chan_skinny even better. ;) -- Michiel van Baak mich...@vanbaak.eu http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multi Tenant Asterisk Server ?
On 17:54, Fri 13 Nov 09, John A. Sullivan III wrote: On Fri, 2009-11-13 at 19:55 +, Gavin Spurgeon wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi List, What I hope is a simple question... As the subject states, I would like to know if anyone has setup a Multi Tenant Asterisk Server ? If so, what would I need to do to get to a Multi Tenant setup (preferably an Open Source solution) ? Any suggestions/comments/pointers/URLs ? snip Entirely doable and reasonably well documented in the literature. Pay particular attention to the use of contexts. If I recall correctly, the followme and meetme applications do not support contexts. I believe you also have to be careful with SIP ids even in different contexts (someone correct me on that if I'm wrong as Asterisk is only a small part of my job and so the details are not always fresh in my mind). For those, we rely upon some other globally unique attribute, e.g., in our environment, all tenants have a unique posix uid and username. We use that username for the SIP ID and the uid for the meetme and followme identifiers. Hope this helps - John PS - Ah - multi-tenant parking - it is broken as recently as 1.6.1.7. There is a patch which works perfectly. I do not know if that patch was included in 1.6.1.8. In fact, if someone knows, please respond as we need to do that upgrade for security purposes and are concerned about breaking multi-tenant parking. That patch is not yet in. I'm planning to get it in this weekend. -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michiel van Baak mich...@vanbaak.eu http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multi Tenant Asterisk Server ?
On 18:55, Fri 13 Nov 09, John A. Sullivan III wrote: On Sat, 2009-11-14 at 00:30 +0100, Michiel van Baak wrote: On 17:54, Fri 13 Nov 09, John A. Sullivan III wrote: On Fri, 2009-11-13 at 19:55 +, Gavin Spurgeon wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi List, What I hope is a simple question... As the subject states, I would like to know if anyone has setup a Multi Tenant Asterisk Server ? If so, what would I need to do to get to a Multi Tenant setup (preferably an Open Source solution) ? Any suggestions/comments/pointers/URLs ? snip Entirely doable and reasonably well documented in the literature. Pay particular attention to the use of contexts. If I recall correctly, the followme and meetme applications do not support contexts. I believe you also have to be careful with SIP ids even in different contexts (someone correct me on that if I'm wrong as Asterisk is only a small part of my job and so the details are not always fresh in my mind). For those, we rely upon some other globally unique attribute, e.g., in our environment, all tenants have a unique posix uid and username. We use that username for the SIP ID and the uid for the meetme and followme identifiers. Hope this helps - John PS - Ah - multi-tenant parking - it is broken as recently as 1.6.1.7. There is a patch which works perfectly. I do not know if that patch was included in 1.6.1.8. In fact, if someone knows, please respond as we need to do that upgrade for security purposes and are concerned about breaking multi-tenant parking. That patch is not yet in. I'm planning to get it in this weekend. snip Thanks for the update. How will it be available at that point? Will there be an immediate 1.6.1.9 release or will it only be via SVN? - John not sure yet. Will have a look at it tomorrow and get back to you here ok ? -- Michiel van Baak mich...@vanbaak.eu http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] allowguest defaults to yes for SIP
On 11:16, Thu 12 Nov 09, Lee Howard wrote: Danny Nicholas wrote: Gentlemens clubs usually don't have any. While LH probably has a valid point, jumping on Til isn't the way to bring it home. You can't protect the stupid or lazy from themselves. If you can't do this right, pay someone else to. You're suggesting that if I pay someone they'll be able to get the default setting for allowguest changed to no ? No, he was saying that if you dont know the system you are going to setup, and dont have the time/resources to read up on how it works, you can always hire someone who knows how stuff works. I could be wrong, but I don't generally consider myself stupid or lazy... and yet this default setting as yes took me by surprise, obviously. No-one told you you are stupid or lazy. It's just that this option only allows unwanted stuff if the configuration is made to do that. So either I am stupid or lazy or there is a risk here that can catch even others off-guard. I've been down this contribution road-path a half-dozen times before with Asterisk. So forgive me if I don't play it out to the final futile note. In ESR's CatB there's the idea where the maintainer encourages (and wants) bug reporting, feedback, and other non-code forms of contribution (as well as code contributions). He refers to it as grooming co-developers. That's not how Asterisk development works... here you can contribute if you're already in the meritocracy, but if you're not, then you have more than a difficult time in trying to even contribute in small non-monetary ways. This is so untrue. When I started working with asterisk, and found my first issue, I created a patch, put it on the tracker, followed up on the comments, and stuff got in. Sometimes it takes some time before the first review of your patch is happening. This is mainly because the developers are really busy, and only part of the developers is being paid to do this stuff for asterisk, all the others are doing it in their free time. If you read the page about contributing code to asterisk, it clearly states that the dev mailinglist is the place to discuss development. If you post comments there, people will read it, comment on it, and if more people agree with the ideas it will get implemented. It's how all OpenSource projects work. -- Michiel van Baak mich...@vanbaak.eu http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Request for Review: Building Queues with Asterisk
On 17:19, Thu 12 Nov 09, Leif Madsen wrote: I have been working on some documentation for how to build queues for Asterisk. This is an introduction for getting device state working for queues, and building queues. It contains the documentation file (text format) and also has the .tar.gz file of the /etc/asterisk/ directory I was using for testing. The modules.conf file has autoload=no enabled, and just loads the modules that were required for the example (along with probably a couple extra modules, but the list of modules has been toned down). Please review and let me know how it goes for you! Where can we find all of this ? -- Michiel van Baak mich...@vanbaak.eu http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] allowguest defaults to yes for SIP
On 12:38, Fri 13 Nov 09, Matt Riddell wrote:
On 13/11/09 12:33 PM, Tzafrir Cohen wrote:
On Fri, Nov 13, 2009 at 12:19:54PM +1300, Matt Riddell wrote:
Maybe the best way would be to make it that the default context only
provides the info from the examples unless you provide an option:
read_security_document=yes
Asterisk used to require that you set have 'TELEPHONY=yes' in
/etc/{sysconfig,default}/asterisk to start running. This is no longer
the case. Such requirements are not the thing that will make the user
read the documentation, and they get in the way of automating the
installation.
Yeah, but would you automate an install with additional contents in the
default context?
We do. It's the only way to get ENUM running on new boxen ;)
and yes I know, I'm not the beginning user anymore.
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Re: [asterisk-users] Cisco 7921
On 17:25, Fri 23 Oct 09, Torintino T wrote: How can i please register Cisco 7921 (Skinny) wireless phone on Asterisk. You will have to configure chan_skinny using /etc/asterisk/skinny.conf Have a look at the sample file, it has plenty of docs in it. -- Michiel van Baak mich...@vanbaak.eu http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7921
On 10:55, Tue 20 Oct 09, Torintino T wrote: Is available to setup chan_sccp on Asterisk 1.2.28 and register Cisco 7921 wireless phone on it? Yes, but we cannot support it here because chan_sccp is not part of asterisk. Contact the chan_sccp project for help. If yes, Can anybody please post his example. Thanks _ Keep your friends updated?even when you?re not signed in. http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/social-network-basics.aspx?ocid=PID23461::T:WLMTAGL:ON:WL:en-xm:SI_SB_5:092010 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michiel van Baak mich...@vanbaak.eu http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best Firewall Suggestions?
On 10:04, Tue 13 Oct 09, David Wathen wrote: Hi, My customer has a outdated firewall that is also presenting a NAT nightmare for getting the Asterisk server reachable from the internet. What firewalls work good with VOIP? I really want to steer away from any ALG supported firewall. I just want a good firewall that works well with Asterisk. We use OpenBSD with the built-in pf and it works great. -- Michiel van Baak mich...@vanbaak.eu http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best Firewall Suggestions?
On 23:52, Tue 13 Oct 09, Hans Witvliet wrote: On Tue, 2009-10-13 at 14:42 -0500, Karl Fife wrote: I think one of the very best options is pfSense. Free Open-source, but it's BSD based, rather than LINUX based. As such it has a lower risk of external exploits. The user-interface makes it incredibly simple to set up and maintain. There is an embedded versions of it available to run on affordable/reliable solid-state, diskless, fanless Soekris/PCEngines embedded system boards. It's incredibly powerful, and It's ROCK SOLID. I find the traffic shaping engine to work without a hitch. PFSense can do anything you want including VPN (PPTP, IPSec, OpenVPN), failover (Multi-WAN), IDS/IPS (snort) The NEWEST embedded version 1.2.3 rc3 (1.2.3-release is very close) can run the sipproxd package as well as many other packages that previously required the FULL version. Goodbye one-way audio! :-) -Karl pfsense with FreeBSD is a very powerfull combination, period. However, it is compared with a 64-character password from a generator. Darn-difficult to use, and often written on a post-it and a plague for the help-desk (and thus a security risc in itself). If you are familiar with BSD, good, fine. If not you probably are not aware that you're exposing yourself somewhere (if you got it working anyway). A good *NIX admin will only need like 2 or 3 hours to get over it and understand how BSD works when they work with linux. That's how things work with the admins I have met. In the end they all choose for the elegance and clean code and good documentation of BSD before linux. -- Michiel van Baak mich...@vanbaak.eu http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] app_hackblock to prevent SIP/IAX reg trolling
On 14:42, Fri 02 Oct 09, Michelle Dupuis wrote: Has anyone written an app that monitors SIP/IAX registration attempts? A couple of clients are being flooded with SIP registrations (but the source IP changes every few hours so IPtables won't do).. I would think that any attempt to reg 5 times with a bad password should cause a 5 minute timeout until reg is considered again. Has anyone written such an app? The name app_hackblock is my contribution to the project :) Right now, there's no such thing in asterisk. fail2ban comes to mind to read the logs and automagically create iptables/pf rules. There has been a lot of discussion and brainstorming about this type of things during astricon 2008. Maybe a google search will get you some slides/ideas. As far as I know, no code has been written yet. -- Michiel van Baak mich...@vanbaak.eu http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:Asterisk server remote access
On 09:41, Sat 26 Sep 09, hadi motamedi wrote: Dear All Can you please do me favor and let me know if there is an facility in Asterisk server that can be used to have remote access to the server ? Please be informed that we have installed commissioned our Asterisk server at remote site with DECT telephony service provisioning for our subscribers . Can you please let me know if there is an facility in Asterisk pbx that can be used to provide remote access to the server for our maintenance duties ? Thank you in advance ssh usern...@server -- Michiel van Baak mich...@vanbaak.eu http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GTalk functionality Asterisk
On 14:24, Thu 03 Sep 09, ABBAS SHAKEEL wrote: Hello Previous context :- After Looking up sip and IAX2 that require configuration at router level which may cause some problems like connection break etc... so i left them . and start wondering if there is some thing that dont require configuration at router layer. The task to accomplish to make and recieve calls from outside local network using any protocol whose soft phones are available. Problem :- In order to Gtalk work with Asterisk ... We need to have make some changes in menu config.. ie in channel drivers we need to select chan_gtalk. but when i execute make manuconfig . it appears Applications [*] chan_agent Call Detail Recording [*] chan_alsa Channel Drivers XXX chan_console Codec Translators [*] chan_dahdi Format Interpreters XXX chan_gtalk Dialplan FunctionsXXX chan_h323 PBX Modules [*] chan_iax2 Resource Modules XXX chan_jingle Test Modules [*] chan_local the XXX makes me worry. how to remove this ?? include chan_gtalk also Use your arrow key to select chan_gtalk and chan_jingle. It will show on the bottom of your screen what you need. -- Michiel van Baak mich...@vanbaak.eu http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] regcontext regexten
On Aug 10, 2009, at 9:52 AM, harry R wrote: Anyone know how to use regcontext et regexten parameter from sip.conf and can give an example ? Sure... let's say I have a phone with the following configuration in sip.conf: [myphone] type=friend context=inside host=dynamic ; phone will register w/ Asterisk secret=mysecret regcontext=some-context regexten=6123 Thank Jared. So I have one more and last question about regcontext. Where do asterisk create context some-context ? I see context by taping dialplan show some-context in CLI but I dont know in which config file it's created. Harry, The context is created in the running dialplan. It's not stored in a configuration file. You can however create this context in extensions.conf or extensions.ael and asterisk will use that. Basically how it works is: On a sip register asterisk checks if the regcontext exists. if not it will create it, if it exists asterisk will add the regexten to it. Michiel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to match no callerid in 1.6 ?
On 10:17, Fri 24 Jul 09, Louis-David Mitterrand wrote:
Hi,
This used to work fine in 1.4:
exten = 2131/,1,NoOp(reject3: ${CALLERID(num)})
exten = 2131/,n,Playback(no_unknow_callerid_here)
exten = 2131/,n,Hangup
And now, after upgrading to 1.6.1.x it matches every callerid.
Did something change?
Yes, it's now working as it supposed to work.
Use something like this:
exten = 2131,1,GotoIf($[${CALERID(num) = ]?nocallerid,1)
exten = 2131,n,Dia(SIP/Something); or whatever you want to do
exten = nocallerid,1,Playback(no_unknown_callerid_here)
exten = nocallerid,n,Hangup()
--
Michiel van Baak
mich...@vanbaak.eu
http://michiel.vanbaak.eu
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD
Why is it drug addicts and computer aficionados are both called users?
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[asterisk-users] Mexican ITSP needed
Hey all, I was wondering if anyone knows about a Mexican ITSP I can connect to to route calls from and to my * boxen. If it matters: I'm located in The Netherlands and one of our customers is in Mexico so if we need a Mexican presence that is not an issue. Thanks. -- Michiel van Baak mich...@vanbaak.eu http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mexican ITSP needed
On 11:39, Thu 16 Jul 09, Carlos Chavez wrote: Try http://www.inext.com.mx they can provide DIDs in several cities in Mexico. Thanks. I asked the customer to have a look (I'm only capable of reading English and Dutch ;)) You have any experience with them ? On Thu, 2009-07-16 at 09:16 +0200, Michiel van Baak wrote: Hey all, I was wondering if anyone knows about a Mexican ITSP I can connect to to route calls from and to my * boxen. If it matters: I'm located in The Netherlands and one of our customers is in Mexico so if we need a Mexican presence that is not an issue. Thanks. -- Michiel van Baak mich...@vanbaak.eu http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Segmentation Faults Using Skinny (v1.6.0.10)
On 19:07, Fri 10 Jul 09, Steve Totaro wrote: And convert those phones to SIP, forget chan_skinny. Opinion time for me as well: Dont. without bugreports chan_skinny will never be on par with chan_sip. I know there are some segfaults with it here and there, but it's being worked on. -- Michiel van Baak mich...@vanbaak.eu http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Segmentation Faults Using Skinny (v1.6.0.10)
On 08:58, Sat 11 Jul 09, Steve Totaro wrote: On Sat, Jul 11, 2009 at 8:53 AM, Steve Totaro stot...@asteriskhelpdesk.comwrote: On Sat, Jul 11, 2009 at 3:43 AM, Michiel van Baak mich...@vanbaak.infowrote: On 19:07, Fri 10 Jul 09, Steve Totaro wrote: And convert those phones to SIP, forget chan_skinny. Opinion time for me as well: Dont. without bugreports chan_skinny will never be on par with chan_sip. I know there are some segfaults with it here and there, but it's being worked on. -- Michiel van Baak mich...@vanbaak.eu http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? My advice was to use chan_skinny, get a core dump and then convert to SIP. Result = chan_skinny gets fixed and OP has a usable phone system. What good is a phone system if it core dumps everytime you make a call. Some really off-topic opinion. Stop spending time on chan_skinny and work create chan_nbx and chan_megaco. Then you could connect Asterisk to 3Com and NEC products without messing with T1s or analog ports. Contact Digium to find out where to send the hardware and wireshark protocol dumps so we can get to it. -- Michiel van Baak mich...@vanbaak.eu http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Segmentation Faults Using Skinny (v1.6.0.10)
On 08:53, Sat 11 Jul 09, Steve Totaro wrote: On Sat, Jul 11, 2009 at 3:43 AM, Michiel van Baak mich...@vanbaak.infowrote: On 19:07, Fri 10 Jul 09, Steve Totaro wrote: And convert those phones to SIP, forget chan_skinny. Opinion time for me as well: Dont. without bugreports chan_skinny will never be on par with chan_sip. I know there are some segfaults with it here and there, but it's being worked on. -- Michiel van Baak mich...@vanbaak.eu http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? My advice was to use chan_skinny, get a core dump and then convert to SIP. Result = chan_skinny gets fixed and OP has a usable phone system. Indeed. What good is a phone system if it core dumps everytime you make a call. That's why we need bugreports with backtraces so we can fix it. -- Michiel van Baak mich...@vanbaak.eu http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Centrale FastAgi server down
On 10:42, Fri 26 Jun 09, Arjan Kroon | Mobillion wrote: Hi, How do you all handle the situation when a centrale fastagi server process(es) are down? AGI(..) prints Unable to locate host and the dailplan jumps to extension h. I'd like to handle the return value and keeping the caller in the dailplan and not to the hangup extension. Any tips about how to handle a AGI(..) returns -1 condition? Let it connect to localhost and use balance to handle the connection to a set of fastcgi servers so you have redundancy :) thx Arjan Kroon Mobillion BV ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michiel van Baak mich...@vanbaak.eu http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AstDB wildcards
Hi,
On 12:09, Wed 27 May 09, Geoff Lane wrote:
Hi All,
I need to use partial matches on the CIDNAME family I have stored in
AstDB. For example, an organisation might have several numbers with
the same area code and the same first few digits:
1234 567890
1234 567889
1234 567824
...
I'd like to store these (e.g.) as CIDNAME/12345678* (where * is a
wildcard) so that I can retrieve the organisation name from
extensions.conf with:
Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})})
Does AstDB support this (I'm using Asterisk 1.4.22.1)?
Nope.
I know that I can create a function to iterate backwards through the
number until a partial match is met, but I'd rather use built-in
functionality should it exist.
It's your only option.
--
Michiel van Baak
mich...@vanbaak.eu
http://michiel.vanbaak.eu
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD
Why is it drug addicts and computer aficionados are both called users?
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Re: [asterisk-users] Pressing number 2 in dialplan
On 14:49, Wed 27 May 09, Elliot Murdock wrote: Hello! I am having an odd problem in that when the caller dials extension 2 in a dialplan, the system waits 3 to 4 seconds before proceeding. This doesn't happen when any other other extensions are dialed, including an identical dialplan on other another extension! Do you have other extensions that start with a two and are longer then 1 digit ? If so, that's the reason. Is this a bug? Later, Elliot ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michiel van Baak mich...@vanbaak.eu http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: SIP hardphone with multi-color BLF
On 13:00, Tue 19 May 09, Gordon Henderson wrote: On Tue, 19 May 2009, jonas.kell...@telenet.be wrote: To feed your curiosity... I'm about to implement it. I have several GXP2020 and GXP1200 Grandstream telephones. I'm reading documentation to know how to start and what to expect. I'm hoping that implementing BLF on these Grandstreams in combination with Asterisk is easier then configuring sla.conf in Asterisk :-). BLF on Grandstreams is easy - it works well in the later firmware (yes, there were issues in early firmware) and I have many sites using it to good effect. Just follow the docs on the Grandstream website. However this multi-colour thing is intersting - Grandstreams do have red+green LEDs - right now my desk phone has a row of green LEDs, indicating phone is online, but no calls. When I call a phone, that particular one flashes red, or is constant red when the phone is in-use, *or* offline. But it does that all on it's own - I've done nothing special in the dialplan, just added the relevant hints and set the buttons to BLF in the phone GUI. This is asterisk 1.2 too... Same here. It does work on 1.0 like that as well. (although you should not use asterisk 1.0.X anymore) Gordon Jonas. - Oorspronkelijk bericht - Van : Olivier [mailto:oza-4...@myamail.com] Verzonden : dinsdag , mei 19, 2009 09:17 AM Aan : 'Asterisk Users Mailing List - Non-Commercial Discussion' Onderwerp : Re: [asterisk-users] OT: SIP hardphone with multi-color BLF 2009/5/19 jonas.kell...@telenet.be Check out the Grandstream GXP-serie also... http://www.grandstream.com/gxp2020.html This feature is called Dual color LED indicator You can program the line buttons to support BLF (red, red blinking, green) For curiosity's sake, do you use this feature ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michiel van Baak mich...@vanbaak.eu http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Provisioning GXP 2000
On 13:45, Thu 26 Mar 09, Lutgring, Sam wrote: My preferred method is to use my own TFTP server. This makes changes to accounts/phones very fast and easy. The whole process takes me about 5 minutes to deploy an entirely new phone. 1) I modified the Grandstream template to contain my own information. This is a simple TXT document and can be edited in your favorite editor. I once counted that I am down to 8 lines in my template that need adjusting for a new user. 2) I open the above mentioned template and change the appropriate lines for the users phone and then save it to a directory utilizing a naming convention of EXTENSION-USERNAME.txt (this allows me ease of changing if ever required). 3) Then I use the Grandstream config generator to compile that into a bin file in the appropriate tftp directory. 4) Then (first time phone is ever used, not required on a redeploy) I log into the web interface on the phone and change 1 line that tells the phone where to find the config file. The phones by default allow you to use DHCP option 66 to provide the tftp server address. That will remove step 4 from your list :) 5) Reboot the phone and all done. Hope this helps. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Ruggles Sent: Thursday, March 26, 2009 11:41 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Provisioning GXP 2000 I've done some googling and searched voip-info but I'm not able to find a good answer about how to provision the GXP 2000. Based on questions I've asked before it seems like a lot of people are using the grandstream phones so I figure provisioning can't be that hard. Is everyone using the web interface for *every* phone? Or is there a better, more automatic, way? TIA!!! Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 da...@safedatausa.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michiel van Baak mich...@vanbaak.eu http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Provisioning GXP 2000
On 11:41, Thu 26 Mar 09, David Ruggles wrote: I've done some googling and searched voip-info but I'm not able to find a good answer about how to provision the GXP 2000. Based on questions I've asked before it seems like a lot of people are using the grandstream phones so I figure provisioning can't be that hard. Is everyone using the web interface for *every* phone? Or is there a better, more automatic, way? Checkout http://www.grandstream.com/configurationtool.html -- Michiel van Baak mich...@vanbaak.eu http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] chan_sip and database integration
On 10:56, Tue 24 Feb 09, Klaus Darilion wrote: Hi! I tried to understand how chan_sip can be configured by means of a database. I found these 2 different approaches (please correct me if I am wrong): static configuration: the sip.conf file is mapped to a database table. The table contains one line for each line in sip.conf. realtime configuration: the peers/users are stored in the database using a single line for each peer/user. Static does not eases provisioning as configuring a SIP peer/user using this approach is really complicated - it is just a method to store .conf files in database. realtime really eases provisioning of SIP peers/users. You only have to insert/update/delete a single line. But functionality is different - there are limitations as these objects are not stored in memory (can be cached), for example device status information. What I am looking for is a method to provision peers/users with a single line in the database, but without limitations. Thus, the peers need not to be realtime but are loaded on sip reload. So is there a possiblity to have static peer/users configuration using a nice and easy way? Store them in a database and use a combination of cron and some scripting to generate the configuration files. Some advice: keep track if an update has been done to the database since last reload and only regen files and issue a reload when this is true. thanks klaus ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michiel van Baak mich...@vanbaak.eu http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No such command 'core stop now'
On 13:06, Sun 15 Feb 09, Jim Boykin wrote: This happens mysteriously randomly. If asterisk was killed and restarted, it often gives this error myast*CLI core stop now No such command 'core stop now' (type 'core show help core' for other possible commands) If you wait a bit, does it work then ? It's possible asterisk is not fully loaded yet (dns resolution being the main thing that can take some time). -- Michiel van Baak mich...@vanbaak.eu http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 32 bit server is ok?
On 16:15, Thu 29 Jan 09, David fire wrote: hi i have a 32 bits server asterisk 1.6 will work ok in it? is just for a demo server no more than 4 to 10 calls at the same time. and a tdm board. waht do you think? Yup, no problem. Any X86 machine with a pci slot will be able to handle this. even 486 machines will be able to do that. -- Michiel van Baak mich...@vanbaak.eu http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Document with differences between 1.2, 1.4 and 1.6?
On 16:28, Mon 26 Jan 09, Carlos Chavez wrote: Is there a bullet type document with the features each version of Asterisk has? I know you can read the CHANGES file but that is not something you give a customer. I just need a one or two page document with bullet points showing the features added from 1.2 to 1.4 and from 1.4 to 1.6. Anyone know of an existing document or it this a make your own moment? Because it depends on what your customers use and what your customers need it's basically a 'make your own' task. Reading the UPGRADE-1.X.txt files gives you a nice overview of what you have to do differently and what happened between major releases. -- Michiel van Baak mich...@vanbaak.eu http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? pgpFzZXvpYZ5b.pgp Description: PGP signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie in Cisco Phone
On 15:56, Fri 23 Jan 09, Sam Tam wrote: Well does it matter if the asterisk server is not located in the same network? No, I used to have my phones at home and my asterisk in Denmark in a colocating facility. I am willing to spend a bit of cash to get someone help me to set it up . Since I need it quite done before end of this month If it's ok for you that I'm not in the same country I'm willing to help you a bit. The next 8 to 9 hours are for my boss, but after that I can help. Contact me off-list if you want. Sam -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michiel van Baak Sent: Friday, January 23, 2009 3:35 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Newbie in Cisco Phone On 05:39, Fri 23 Jan 09, Sam Tam wrote: Yes I know too. Is there anyway to make it work with asterisk without using Callmanager? Sam Asterisk does have chan_skinny. Featureset is not as good as CCM, but it's handling my phones and some customers phones as well. Check it out before returning the phone. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jason Aarons (US) Sent: Friday, January 23, 2009 5:10 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Newbie in Cisco Phone The 7936G/ 7937G Data Sheet says SCCP only which is a shame. It really is a great sounding phone. I have several customers with them as SCCP. http://www.cisco.com/en/US/prod/collateral/voicesw/ps6788/phones/ps379/ps875 9/product_data_sheet0900aecd806e021a.html From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sam Tam Sent: Thursday, January 22, 2009 3:54 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Newbie in Cisco Phone Hello all I have used some low end cisco phones in the past and had no problem setting up SIP on it. But today, I have made a big mistake. Buying Cisco Conference phone without even looking whether it supports SIP on not. And yes it is the nice 7937G that I am talking about. Damn this is annoying. So wondering is there anything I can do to make it work with Asterisk or am I good to send back to exchange another item? Sam Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michiel van Baak mich...@vanbaak.eu http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michiel van Baak mich...@vanbaak.eu http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Long Delay after sip reload command
On 13:46, Fri 23 Jan 09, Christopher Gray wrote: Hello: I am experiencing long delays, minutes not seconds, after issuing sip reload or /etc/init.d/asterisk restart commands. When reloading Asterisk, for the first minute or more, sip show registry says there is no such command. When sip show registry begins to provide information, registration can take another 3-4 minutes. Sometimes, timeouts occur as well, and sometimes these timeouts are never resolved. sip.conf has three register commands: two to different SIP carriers, vitelity and voicepulse; and one to another computer domain address. Any thoughts on what I should do to begin resolving this? I'm relatively new to Asterisk, running Version 1.4 on Ubuntu Linux. Check your /etc/resolv.conf Does the first nameserver there work ? Your problem is probably a resolving issue. -- Michiel van Baak mich...@vanbaak.eu http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie in Cisco Phone
On 05:39, Fri 23 Jan 09, Sam Tam wrote: Yes I know too. Is there anyway to make it work with asterisk without using Callmanager? Sam Asterisk does have chan_skinny. Featureset is not as good as CCM, but it's handling my phones and some customers phones as well. Check it out before returning the phone. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jason Aarons (US) Sent: Friday, January 23, 2009 5:10 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Newbie in Cisco Phone The 7936G/ 7937G Data Sheet says SCCP only which is a shame. It really is a great sounding phone. I have several customers with them as SCCP. http://www.cisco.com/en/US/prod/collateral/voicesw/ps6788/phones/ps379/ps875 9/product_data_sheet0900aecd806e021a.html From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sam Tam Sent: Thursday, January 22, 2009 3:54 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Newbie in Cisco Phone Hello all I have used some low end cisco phones in the past and had no problem setting up SIP on it. But today, I have made a big mistake. Buying Cisco Conference phone without even looking whether it supports SIP on not. And yes it is the nice 7937G that I am talking about. Damn this is annoying. So wondering is there anything I can do to make it work with Asterisk or am I good to send back to exchange another item? Sam Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michiel van Baak mich...@vanbaak.eu http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] lock SIP Account after too many failed logins
On 11:04, Fri 09 Jan 09, Matthew Nicholson wrote: On Fri, 2009-01-09 at 16:49 +, Steve Howes wrote: On 9 Jan 2009, at 16:36, Klaus Darilion wrote: Hi! I want to detect brute-force password hacking attacks - thus if there are too many failed login attempts for a SIP account I want to lock this account. Does somebody have any ideas how this could be implemented? Bad plan? Could quite easily turn into a DoS. Could this be done at the IP tables level? Or maybe you could write a script that monitors the asterisk logs and detects failed login attempts then adds problematic IP address to hosts.deny. I know of several ssh blocking scripts that work this way. I think fail2ban can do this. It has a configuration file where you can list your logs and regexp matches in this logfile. I use fail2ban on linux to detect those types of attacks on my ftp, imap, pop3, smtp+sasl, ssh etc etc It can take action by blocking the ip for a specified period. The block can be configured. iptables, hosts.deny, pf, ipfw, custom-script-to-send-block-rule-to-cisco-pix,whatever. http://www.fail2ban.org/wiki/index.php/Main_Page -- Matthew Nicholson Digium, Inc. | Software Developer ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michiel van Baak mich...@vanbaak.eu http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CISCO 7940 United_States/7960-tones.xml
On 12:55, Thu 08 Jan 09, Mikel Lindsaar wrote: Thanks Mark, The phone starts, I can get to settings and such... just it keeps looking for this file. I just opened a TAC ticket and getting it handled. As far as I know it only looks for this file when the phone boots and does a couple of retries and then gives up (at least that's what my 7960 does) I cant remember where I got the file but I have it, and noticed totally no difference in the phone's behaviour. Same for a couple of other files. Like the fonts, ringlist, dialplan etc. With or without them the phone behaves the same. PS: this is with chan_skinny, no idea what impact it has on SIP. On Thu, Jan 8, 2009 at 3:26 AM, Mark G. Thomas m...@misty.com wrote: Mikel, On Thu, Jan 08, 2009 at 12:52:02AM +1100, Mikel Lindsaar wrote: I have a smartnet contract for this phone, and have searched high and low for this file on the Cisco website. I need: United_States/7960-tones.xml English_United_States/7960-font.xml Every road seems to lead to the Call manager express downloads... I don't have a CME, so that's basically useles. Can anyone point me in the right direction? Those files aren't directly included in the CME downloads. I think their contents must be included in the binary phone load or internal to CME. Using CME, if one sets cnf-file location flash:, then does a create cnf-files, they are then written out to the CCME flash, however they default to being on system:, not the tftp server flash. Have you tried resetting your phone to factory defaults -- **# to unlock the settings menu? You might not actually need these files. Mark -- Mark G. Thomas (m...@misty.com) http://mail-cleaner.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- http://lindsaar.net/ Rails, RSpec and Life blog ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michiel van Baak mich...@vanbaak.eu http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Installing Asterisk v1.6 on Ubuntu Intrepid?
On 20:14, Tue 16 Dec 08, Christian wrote: Hi all, I am trying to isntall the v1.6 version of Asterisk on my Intrepid system, but I get an error after I have typed make: [CC] manager.c - manager.o manager.c: In function ‘action_getvar’: manager.c:1732: error: ‘SENTINEL’ undeclared (first use in this function) manager.c:1732: error: (Each undeclared identifier is reported only once manager.c:1732: error: for each function it appears in.) make[1]: *** [manager.o] Error 1 make: *** [main] Error 2 Hi, What version of asterisk 1.6 is this ? This error (my fault) has been fixed in svn shortly after it appeared and the -rc version that's listed as download on http://www.asterisk.org dont have this problem anymore. If you did an svn checkout please run 'svn up' and if you downloaded a .tar.gz please download the -rc listed on http://www.asterisk.org and you should be fine. Sorry for the trouble. -- Michiel van Baak mich...@vanbaak.eu http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.6.1: iax trunk needs dahdi timing ??
On 20:24, Sun 14 Dec 08, sean darcy wrote: starting 161.1-beta3: chan_iax2.c:10925 build_user: Unable to support trunking on user 'iax-out' without DAHDI timing But I have these timing modules: ls /usr/lib/asterisk/modules/res_tim* /usr/lib/asterisk/modules/res_timing_dahdi.so /usr/lib/asterisk/modules/res_timing_pthread.so Do I need to do some magic to get these loaded? modules.conf is set to auto. Is this what iax is looking for? If you dont have any dahdi hardware installed and configured, make sure to load dahdi_dummy. That will provide you the timers. -- Michiel van Baak mich...@vanbaak.eu http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Please explain the meaning of the output of lsmod | grep ztdummy?
On 20:30, Sat 13 Dec 08, Shaun Wingrin wrote: lsmod | grep ztdummy ztdummy38856 0 zaptel231496 3 ztdummy it means you have ztdummy loaded so you can do Meetme rooms and IAX2 trunking without actual zaptel hardware. -- Michiel van Baak mich...@vanbaak.eu http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FBI issues VoIP security warning on Asterisk --which version?
On 14:51, Fri 12 Dec 08, Khaled Chehab wrote: Dear All FBI issues VoIP security warning on Asterisk -- but which version? Any one know which version ? Regards Hi, See this listpost: http://lists.digium.com/pipermail/asterisk-users/2008-December/223172.html -- Michiel van Baak mich...@vanbaak.eu http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SVN
On 09:06, Wed 26 Nov 08, Alex Montoanelli wrote: Hello, everyone. Anybody know when that svn will be available again? Regards Hey, I can checkout stuff fine from svn.digium.com. Maybe you can provide some more info about how it's not working for you. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cli commands missing
On Oct 12, 2008, at 6:31 AM, Eric Fort wrote: autoload is enabled but all the modules are not loaded. Why would this be? what should I look at? The cli output and modules.conf are below. Eric modules.conf: ; ; Asterisk configuration file ; ; Module Loader configuration file ; [modules] autoload=yes noload=chan_oss.so noload=chan_alsa.so noload=chan_phone.so [global] --- A couple of things to check: Is your DNS lookup on the box working? Are all the configfiles readable by asterisk? Can you try to stop asterisk and start it with: asterisk -vvgcd That will show you what it's doing and may give you pointers why it's not working. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7960 sccp, Skinny and 1.4
On 08:26, Fri 10 Oct 08, David Gibbons wrote: You need to check out the chan_sccp-b mainling lists on sourceforge. There is active development in SVN but not in tarball releases. http://sourceforge.net/mailarchive/forum.php?forum_name=chan-sccp-b-discussion It is very stable. Or, if you dont want to use outside modules use Asterisk 1.6 (which has been released as well) with the chan_skinny driver. A lot of development went into it and it's much more useable then the 1.2 version. Myself uses chan_skinny in production without too much trouble. Specially when you use the 7960 phones it's a nice setup. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wayne Sent: Thursday, October 09, 2008 6:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Cisco 7960 sccp, Skinny and 1.4 Hi All, I'm thinking of creating a new asterisk server using the latest 1.4 stable release to replace my ageing Asterisk SVN-branch-1.2-r7231 (its been a while!). My only concern - my phones are Cisco 7960's (with sccp firmware 7.2 loaded) and to support them better, I remember compiling in a skinny(?) driver to replace the (from what I could tell) basic in built sccp support. After digging around a little it would appear that the original creator of the skinny driver has not done any development for ages. What driver are you referring to ? It must be something outside of the core asterisk, because a lot of commits went into chan_skinny the last year or so. Simple question, has 1.4 got better native support for sccp now without having to add in anything extra to make everything work ok?, if not, is there a version that someone may have carried forward of the skinny driver that will work with 1.4? Yes, chan_skinny in 1.4 is better then the 1.2 version, but the real stuff happened in the 1.6 version. 1.6.0 is released, so why not use that one instead of 1.4? Thank you, Wayne. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7960 sccp, Skinny and 1.4
On 21:28, Fri 10 Oct 08, Wayne wrote: Thanks both, The only thing I have a little concern over is that 1.6 is that its still a development release (if I understand things correctly). No, 1.6.0 has been released. This is indeed the first public 'final' release of the 1.6 series. But it's not in beta or release-candidate anymore. Basically, it's the latest and greatest version that should be stable. Stability is the main thing for me (its only a very small set up) but there are no technical people around if something were to go wrong through the day. You do know it's just another daemon an a linux box right ? If you cant afford downtime you should not bet on one server, but make every part of your network redundant. That means at least: connectivity power hardware locations backups all the other stuff I forgot -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Update (IAX Trunking Help)
On 10:21, Thu 09 Oct 08, Steve Anness wrote:
First off thank you for your help, using your help in conjunction with a
couple of my own changes it partially worked. I got rid of the iax-incoming
context, it seemed useless. I may be wrong in that assumption.
Looking back at what I have now:
Extensions.conf on server A
[vvfarm-extensions]
exten = _1XX,1,Dial(SIP/${EXTEN}-1,20)
exten = _1XX,n,Voicemail(${EXTEN:0:3}|su)
exten = _1XX,n,Dial(SIP/${EXTEN}-1)
exten = _17XXX,1,Dial(iax2/colo/${EXTEN},20)
Extensions.conf on Server B
[remote-extensions]
exten = _17XXX,1,Dial(SIP/17${EXTEN}-1,20)
exten = _17XXX,n,Voicemail(${EXTEN:0:3}|su)
exten = _17XXX,n,Dial(SIP/${EXTEN}-1)
exten = _11XXX,1,Dial(iax2/vvfarm/${EXTEN:2},20)
exten = _11XXX,n,Voicemail(${EXTEN:2:3}|su)
I can call from server B to Server A I added the voicemail line, however;
it isn't working like it should. We have things set-up, as you can see, if
someone dials 327 they get the voicemail box for 127. When I dial 11127
from Server B it rings but when it gets time for voicemail to pick up it
tries calling 127 instead of 327 looking for a voicemail box. I was under
the impression ${EXTEN:2:3} should cut off the 11 (that part works) and
change the first variable to a 3 (that part doesn't work)
I still can't make calls from Server A to Server B I still get the same
error
[Oct 9 10:26:05] NOTICE[3118]: chan_iax2.c:7773 socket_process: Rejected
connect attempt from 64.194.211.170, request '[EMAIL PROTECTED]'
does not exist
17110-1 does exist, I can pick up the phone connected on server B and dial
17110 and get 17110-1.
No, it doesn't.
You only have 17XXX
Get rid of the -1 or add the -1 to the exten on serverB
Thanks again for your help.
Steve
On 10/8/08 7:04 PM, Alejandro Kauffmann [EMAIL PROTECTED] wrote:
Steve Anness wrote:
I posted earlier in the day about needed help with IAX trunking. I did
some more reading and made some more changes.
Here is what I have thus far:
Iax.conf on one server
[general]
bindport = 4569
bindaddr = 0.0.0.0
disallow=all
allow=ulaw
allow=alaw
allow=gsm
mailboxdetail=yes
[vvfarm]
type=friend
username=colo
secret=testpassword
auth=plaintext
host=64.194.211.170
context=iax-incoming
peercontext=vvfarm-extensions
qualify=yes
trunk=yes
Extensions.conf on the same server
[iax-incoming]
exten = _###,1,Dial(SIP/17${EXTEN}-1,20)
[remote-extensions]
exten = _1,1,Dial(SIP/17${EXTEN}-1,20)
exten = _1,n,Voicemail(${EXTEN:0:3}|su)
exten = _1,n,Dial(SIP/${EXTEN}-1)
exten = _11XXX,1,Dial(iax2/vvfarm/${EXTEN:2}-1,20)
Iax.conf on server B
[general]
bindport = 4569
bindaddr = 0.0.0.0
disallow=all
allow=ulaw
allow=alaw
allow=gsm
mailboxdetail=yes
[colo]
type=friend
username=vvfarm
secret=testpassword
auth=plaintext
host=72.249.129.91
context=iax-incoming
peercontext=remote-extensions
qualify=yes
trunk=yes
Extensions.conf on server B
[vvfarm-extensions]
exten = _1XX,1,Dial(SIP/${EXTEN}-1,20)
exten = _1XX,n,Voicemail(${EXTEN:0:3}|su)
exten = _1XX,n,Dial(SIP/${EXTEN}-1)
exten = _17XXX,1,Dial(iax2/colo/${EXTEN}-1,20)
[iax-incoming]
exten = _XXX,1,Dial(SIP/${EXTEN}-1,20)
The error I am getting when trying to call from Server A to Server B is
[Oct 8 17:13:00] NOTICE[3616]: chan_iax2.c:7367 socket_process:
Rejected connect attempt from 72.249.129.91, who was trying to reach
'[EMAIL PROTECTED]'
The error I am getting when trying to call from server B to Server A is
[Oct 8 17:26:46] NOTICE[3115]: chan_iax2.c:7332 socket_process:
Rejected connect attempt from 64.194.211.170, who was trying to reach
'[EMAIL PROTECTED]'
What have I done wrong? Why won?t it dial 17119-1 and 127-1, respectfully.
Steve Anness
Your patterns don't match. You are sending [EMAIL PROTECTED], but
vvfarm-extensions has no pattern xxx-1. Same problem in the other
direction. Try changing the dial statement in server A from:
exten = _11XXX,1,Dial(iax2/vvfarm/${EXTEN:2}-1,20)
to:
exten = _11XXX,1,Dial(iax2/vvfarm/${EXTEN:2},20)
and in server B from:
exten = _17XXX,1,Dial(iax2/colo/${EXTEN}-1,20)
to:
exten = _17XXX,1,Dial(iax2/colo/${EXTEN},20)
Alex
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Re: [asterisk-users] Amazing show uptime
On 09:59, Fri 12 Sep 08, Stephen Davies wrote: xx-montague-gardens*CLI show uptime System uptime: 38 years, 37 weeks, 4 days, 10 hours, 47 minutes, 11 seconds Amazing. Especially considering: [EMAIL PROTECTED]:/var/log uptime 09:58:14 up 18:42, load average: 0.21, 0.09, 0.02 Steve Did ntp/rdate set the clock forward for 38 years right after boot ? -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SCCP - max lines per phone limit
On 12:51, Fri 12 Sep 08, OCG Technical Support wrote: I'm setting up a 7921 and now want to add a second line to the phone. In my SCCP.conf file I have: autologin = 235,299 However, on reloading SCCP the phone fails to login to the second line with this error: [Sep 12 12:46:49] WARNING[12224]: sccp_actions.c:185 sccp_handle_register: SEP001BD457F8B1: Failed to autolog into 299: Max available lines phone limit reached 299 You are better off asking on the chan_sccp mailinglist. Asterisk has chan_skinny which works differently in assigning lines to devices. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Setup speed dials on Cisco 7921
On 13:15, Fri 12 Sep 08, OCG Technical Support wrote: I've added lines like this: speeddial = 123,test speeddial = 260,Bob in the [device] section for my 7921, but the speed dials do NOT appear on the menu (click right from the main screen). Am I missing something obvious here? chan_skinny or chan_sccp ? -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Setup speed dials on Cisco 7921
On 15:37, Fri 12 Sep 08, OCG Technical Support wrote: Chan_sccp again... From what I read chan_sccp is the successor to chan_skinny. No, it's a fork that never contribute back anything to asterisk. The last year there have been activity in chan_skinny again, and I can say it works ok for my home system now. There are some interesting patches on the bugtracker, and I know that at least wedhorn is putting effort into chan_skinny to make it even better. The biggest problem with chan_sccp is that there are already around 4 different branches of it, and they all go another way. None of them is as close to the asterisk development as chan_skinny. MD -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michiel van Baak Sent: September 12, 2008 2:08 PM To: Asterisk Users List Subject: Re: [asterisk-users] Setup speed dials on Cisco 7921 On 13:15, Fri 12 Sep 08, OCG Technical Support wrote: I've added lines like this: speeddial = 123,test speeddial = 260,Bob in the [device] section for my 7921, but the speed dials do NOT appear on the menu (click right from the main screen). Am I missing something obvious here? chan_skinny or chan_sccp ? -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Network Monitoring
On 14:50, Tue 09 Sep 08, Jacobus van Niekerk wrote: Dear Asterisk Users I'm looking for a solution that can be used to monitor Asterisk and the Telco lines aswell as the network (Servers, WAN LAN links, Router Switches) We use nagios for that. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems with 2 Asterisk servers on same LAN
On 08:24, Sun 07 Sep 08, Steve Totaro wrote: On Sun, Sep 7, 2008 at 7:47 AM, Tim Panton [EMAIL PROTECTED] wrote: This is one of those cases where it is almost certainly simpler to use IAX2 not SIP. You will need zero config on the router and it will 'just work' - assuming your provider supports IAX that is. It may be simpler to get working but will it be simpler to diagnose the audio issues that will invariably come down the pipe? How about the rather popular error I should never be called!? Google it with the quotes. It seems to get called quite a bit for something that should Never be called. OT: I have a couple of asterisk boxen, and yes, I see that error from time to time. But still, in my opinion IAX2 is a nice protocol and it works great for us. Troubleshooting is not as hard as you think. In my experience, audio issues are either codec related, or connectivity related. If it's the connection you will see your IAX peers going OFFLINE or UNREACHABLE a lot. With SIP you have all this trouble with RTP and routers that dont understand it etc. /OT I think I would spend a day or two getting SIP working properly, now, rather than spending days trying to figure out audio issues and having to revisit and get SIP working properly in the future. After people are actually relying on the system and already have a bad experience/opinion associated with the New Phone System. I think they should go with the Tech their DID provider prefers. That way you will get the best support from them if something goes wrong. Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] lists.digium.com monthly reminders
On 10:43, Mon 01 Sep 08, Tony Mountifield wrote: I am subscribed to several of the mailing lists hosted at lists.digium.com. However, the memberships reminder that I receive on the first of each month only lists asterisk-biz, and none of the others. Just curious whether this was intentional or a mis-configuration. Just a confirmation from here. I'm not on the -biz list, and I did not get any reminder. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to query a remote MySQL DB from dialplan
On 13:12, Mon 25 Aug 08, Rich wrote: I want to query an existing MySQL DB from my Asterisk Dialplan. This to check one field in a table in a database on a remote DB server. Is this possible using 'app_addon_sql_mysql' from asterisk-addons pkg? I would like to use an ODBC connector eg. unixODBC. I would like it to be 'stock', ie. part of the standard release and supported into the future. I have researched this for a couple days now (Asterisk Wiki and maillist) and am confused by the many and varied 'solutions' I find online. I so far haven't been able to navigate thru all the online docs to a real solution. If I am finally successful... I will contribute a step by step HOWTO to the Wiki. Any hints and pointers will be greatly appreciated. Thanks, Rich If you want to use ODBC, there's no need to use app_addon_sql_mysql. This addon uses native mysql, instead of odbc. If you want something stock that is in default asterisk base package have a look at func_odbc. It matches your requirements (odbc, stock, standard release, supported in the future) and is a very nice tool to get used to. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] I used to use an Asterisk server, but now it is overkill, ...
On 01:30, Wed 13 Aug 08, Ronald Wiplinger wrote: I had installed in the office an Asterisk server, but the company is gone and I could keep the server. However, for my family with three members and two phone lines this server is overkill. I am looking for a compact solution, which is more suitable for me. I want a small silent box, which can connect two phone lines and 6 internal VoIP phones and about 6 external VoIP phones. I would like to have: 1. Announcements for callers (dial the extension number) 2. voice mail with mail forwarding 3. wakeup call 4. pickup group 5. call forwarding after 20 seconds, ... 6. ISN support, Sipbroker support 7. remote gateway support Get a Soekris 5501 Small, silent, low power consumption, stable -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] intercom/paging with grandstream gxp2000
On 10:59, Thu 07 Aug 08, Fidel Garcia wrote:
Thanks for your reply!
Just so you have a better understanding of what I am trying to accomplish.
The distinctive ring is working fine with Family, however, the intercom
configuration that I am currently testing makes all my calls and intercom
call. It does not matter if I call using Dial or Page on the GXP2000, the
call is always and intercom call. For some reason the GXP2000 is receiving
the SipAddHeader when I do Dial and Page. Can you tell what is wrong with
the configuration by looking at the configuration below?
exten=s,1,SIPAddHeader(Alert-Info: http://127.0.0.1\;info=Family)
exten=s,2,GotoIf($[${SIP_HEADER(Call-Info)}=answer-after=0]?2:3)
exten=s,3,SIPAddHeader(Call-Info: answer-after=0)
if the sip header Call-Info has value answer-after=0 it goes to prio 2,
otherwise 3
Now let's have a closer look at those.
Hhmm, prio two is the gotoif, prio three adds the answer-after=0 ...
I think you mean:
exten=s,2,GotoIf($[${SIP_HEADER(Call-Info)}=answer-after=0]?3:4)
exten=s,4,Dial(${ARG2},20)
exten=s,5,Goto(s-${DIALSTATUS},1)
exten=s-NOANSWER,1,Voicemail(${ARG1},u)
exten=s-NOANSWER,2,Goto(default,s,1)
exten=s-BUSY,1,Voicemail(${ARG1},b)
exten=s-BUSY,2,Goto(default,s,1)
exten=_s-.,1,Goto(s-NOANSWER,1)
exten=a,1,VoicemailMain(${ARG1})
what would you do differently?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gordon
Henderson
Sent: Thursday, August 07, 2008 7:32 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] intercom/paging with grandstream gxp2000
On Wed, 6 Aug 2008, Fidel Garcia wrote:
Guys I have been reading for days on how to get this to work with asterisk
and for some reason every time I call the call goes to intercom. I know I
must be doing something wrong with the way I am adding the steps to my
call;
I am not familiar with variables and flags.
What *exactly* are you trying to achieve?
I have used both paging and intercom mode in the Grandstreams with good
results.
You do need the settings in the phone set ON - ie.
Allow Auto Answer by Call-Info: No Yes
Turn off speaker on remote disconnect: No Yes
These both need to be set to YES or ON.
That won't affect normal calls to that account on the phone - although the
turn off speaker one does make the phone easier to use IMO...
So call the phone and the person answers normally, as before, but if you
rhen add the SIP header:
SIPAddHeader(Call-Info: answer-after=0)
The phone will auto-answer - when the next Dial or Page command is
directed to it.
What next? If you want to Page the phone, use the Page() application.
So if the phone is SIP/100 then to Dial the phone normally..
exten = 100,1,Dial(SIP/100)
but to page it:
exten = 200,1,SIPAddHeader(Call-Info: answer-after=0)
exten = 200,n,Page(SIP/100)
and to intercom to it:
exten = 300,1,SIPAddHeader(Call-Info: answer-after=0)
exten = 300,n,Page(SIP/100,d)
So this has added 3 new extensions, 100, 200 and 300 - which all 'call'
SIP/100, but in 3 differet ways.
Gordon
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Checked by AVG - http://www.avg.com
Version: 8.0.138 / Virus Database: 270.5.12/1596 - Release Date: 8/6/2008
4:55 PM
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Why is it drug addicts and computer aficionados are both called users?
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Re: [asterisk-users] 2000+ user Asterisk PBX
On 16:20, Sun 03 Aug 08, Darren Sessions wrote: I can speak first hand to this having gone through it just a few months ago . . After being spoiled with all the features and standard compliance in Postgres, I was put in a position with a new project to setup a redundant (Master-Slave) database cluster. I immediately jumped to Postgres to do the job (using 8.3). My biggest gripe at the time was that there was really nothing built IN postgres to do the replication as I soon found out. Everything was third party and there were several replication modules suggested to me that seemed stagnant or un-maintained or required an older version of Postgres (bypassing the massive performance increase of the 8.3 release). Of those that I did try that were opensource, all of them seemed fairly complex to get up and running - to say the least. Also having used MySQL extensively, I decided to give it a test run on a separate set of boxes. I'm not exaggerating when I say the replication was up and running in about 10 minutes. While I do appreciate (a lot) how standards compliant Postgres is, MySQL was an absolute clear winner in my book with regards to the replication. Amen. been there and been bitten by the same stuff. We are now using a 4 node mysql master-master setup which works great. Ok, the total setuptime was closer to an hour then two minutes, but that's because we wanted write access to all nodes. make sure to setup the primary key start and increment config params correctly, and you're done. Just my two cents . . My two cents and two weeks of investigation+testing+redoing_it_over_and_over_again - Darren Hmmm... is that really the case? -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7970, CTLSEPmac.tlv
On 15:02, Fri 01 Aug 08, Jason Parker wrote: I just wanted to post this so that it was out there and Googleable. Hopefully it will save other people a bit of time. If you have a Cisco phone (I was testing with a 7970, though presumably it would affect 7960 and others as well) that is looping trying to fetch the CTL tlv file - it may be because you are using Debians 'tftpd' (should be netkit-tftpd...*cough*hey, Debian developers*cough*) package, which is apparently not RFC 783 (tftp) compliant with file not found responses. The whopping 18 page RFC states that Error Code should be 0x00,0x01 for file not found errors, but netkit-tftpd returns 0x00,0x00 which is Not defined - causing the phone to ignore it and request the file again a few seconds later. Solution: Switch to any other tftpd. The moment I switched to tftpd-hpa or atftpd, the phone stopped looping, picked up the SEPmac.cnf.xml file, and immediately registered to Asterisk. Hopefully in the future Debian will rename, remove, or fix this package so it is no longer the default tftpd. Thanks for the write-up. I tried with the latest 7960 firmware, and it did work with the default debian tftpd (had to install a new VM) For googleable stuff: The default tftpd on OpenBSD works fine ;) -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk loosing IAX users's registration??
On 11:13, Sat 05 Jul 08, [EMAIL PROTECTED] wrote: Can anyone tell me why my * is loosing the IAX user's registration? Before, I had asterisk installed with root, and after a hardware failure I reinstalled it under the user asterisk I'm using exactly the same configuration as before, exactly the same version as before but now when I try to call any IAX users it fails right the way and send me to the voicemail, I have to tray 3 or 4 times until asterisk succeed to contact the user. lnxca*CLI core show version Asterisk 1.4.16.2 built by asterisk @ lnxca Running on Fedora 7 First of all, update to the latest version, 1.4.21.1 Are you having other network issues with the box? Check the iax show peers output. set qualify=yes for all the entries so you can see how much they are lagged. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [asterisk-dev] Locking, coding guidelines addition
On 14:26, Sat 05 Jul 08, Steve Totaro wrote: On Fri, Jul 4, 2008 at 10:15 PM, Tilghman Lesher [EMAIL PROTECTED] wrote: On Friday 04 July 2008 19:59:55 Steve Totaro wrote: FreeSwitch will be the clear winner, or at least the heart of large scale systems with a few Asterisk boxen here and there until it becomes more mature. If you want to be a Freeswitch fanboy, that's fine, but please keep it off this list. This list is for usage questions of Asterisk, not for fanboyism of other software projects. -- Tilghman Again, your ability to miss the point is astounding. I never said I was a Freeswitch fan boy. I am just suggesting using a similar method of locking with FreeSwitch. Ideas, obviously Digium doesn't care enough to listen to it's users. Why do you keep repeating this ? It's very clear that you totally dont like the way asterisk is run etc. Why dont you move away from asterisk if it annoys you so damn much ? -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to change http port on appliance?
On 17:53, Wed 02 Jul 08, Fidel Garcia wrote: Well, most dangerous guys out there (noobs) scan port 80 to begin their attacks. As you may know port numbers go from 1 to 65535 and scanning all of them takes a while. I am using 65531 on my box just to stay away from ip ranges scans on default ports. Not big deal, but I feel safer. Next I will change the ssh service to a different port. The best firewall out there is still scissors. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP/IAX2 Provider with fallback dialing?
On 17:36, Thu 26 Jun 08, Steve Finkelstein wrote: Hi all, I was curious if anyone can recommend a company that would work with small businesses, and capable of using a fallback number (mobile phone, home number etc) in the event SIP or IAX2 peering was to terminate because of some outage. This could be useful when you do not have a backup T1 PRI, etc. Thanks all, I dont know where you are, but here in .nl you can use Speakup. They route calls using IAX2 and/or SIP and in the case that wont work they will route it to another number you tell them (in my case, our support mobile number) -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco Presence
On 14:59, Wed 25 Jun 08, Peder @ NetworkOblivion wrote: Does anybody have the settings that you use on a Cisco 7970/79x1 to get presence? I see the * side settings, but I can't find the Cisco side settings anywhere. Sip or Skinny ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP vs. SKINNY
On 14:16, Wed 25 Jun 08, Joe Carroll wrote: Can anyone comment on the performance benefits when comparing sip to skinny ? Most cisco phones work better with the skinny firmware. That is not true when connecting to asterisk though. It all depends on the version of asterisk you are running. I have a setup with over 20 skinny phones on asterisk -trunk and that works great. Specially after today, now that chan_skinny supports transfers. If you are running 1.4 I'm not sure what is best. It basically depends on what you are doing with the phones. In my home setup it worked great, but in my business I have to run trunk for the phones to be as workable as the sip variant. The skinny firmware has some neat stuff like XML push etc. Dont know how the current SIP firmware is doing, as I have not run it in over 2 years now. YMMV -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Loose connection with MySql.
On 09:54, Tue 24 Jun 08, Catalin S. wrote: Hello, I configured asterisk to use mysql for CDR. Well when i check from time to time I realize that asterisk loose connection with mysql (i use phpmyadmin and i watch the processes). Can anybody tell me how can i solve that problem? I want to have all cdr statistics logged in mysql, is very important for billing. Thank you for support. Use cdr_adaptive_odbc backport for 1.4. That one does a check if the connection is still working, and if not it will reconnect. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Lockups with IAX2 and cdr_odbc in 1.4.21 is it my weird config ?
On 12:44, Tue 24 Jun 08, Tim Panton wrote: Is anyone using 1.4.21 with cdr_odbc and IAX channels successfully? I'm getting lockups where asterisk stops responding (to anything). Foolishly I've built a box with 2 new things on it, 1.4.21 and Oracle as the odbc server. If others are running it fine against MySql or postgres , I'll focus on the oracle side. I was just wondering if it was a side effect of the new IAX threading in 1.4.21. Looks like this one: Issue 12925 [Channels/chan_iax2] IAX2 channel gets stuck, causes CLI to get stuck, * won't restart http://bugs.digium.com/view.php?id=12925 -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Website callback
On 15:45, Wed 18 Jun 08, Mark Hamilton wrote: Hi, I have a website where customers enter their phone numbers to be called. I'd like them to have to put in information and 'schedule' a call. 1) Call Immediately 2) Call in the next _ minutes 3) Call me tomorrow, same time. So, Asterisk will pull two variables from this php websites, $phonenumber and $timetocall. $timetocall will need to be calculated as to exactly what time Asterisk will need to call. Then, Asterisk calls it (by way of call files? Either putting the call file in at the time it needs to be called, or I don't know what else) and then if the call is has a human on it, plays a message saying We're now transferring you to an agent. Please wait. And transfer that call to a queue. How can I do this? Is there something prebuilt like this? I would store the info in a database (RDBMS, flat file, whatever) and have a cronjob running every minute that processes this info, creating call files when needed. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] features.conf not working
On 08:36, Sat 07 Jun 08, Russell Bryant wrote: On Jun 6, 2008, at 4:33 PM, Manolet Gmail wrote: i have this on my features.conf: [applicationmap] testfeature = *9,callee,Playback,tt-monkeys extensions.conf: [globals] DYNAMIC_FEATURES=testfeature trunk_1 = Zap/g1 trunk_2 = Zap/g2 what else i have to add in order to make this works? im using 2 xlite, Just a hunch ... if you're using xltite, it's likely that you're not pressing the digits fast enough to satisfy the default timeout. The default featuredigittimeout is 500 ms. Change this option in features.conf and increase it to 2000 ms and try again. another tip: Make sure you have the dtmfmode for the xlite sip stanza set correctly. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dutch Asterisk mailing list?
On 15:34, Mon 19 May 08, Jaap Winius wrote: Quoting Michiel van Baak [EMAIL PROTECTED]: I can start it if you want. Yeah, why not. Eventually, it may help to solve this and other KPN-related problems, which can only be good for Asterisk adoption in the Netherlands. I'll sign up for it if you let me know where. Feel free to join the brandnew [EMAIL PROTECTED] Mailman webinterface is here: http://lists.three-dimensional.net/mailman/listinfo/asterisk-nl -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dutch Asterisk mailing list?
On 22:46, Sun 18 May 08, Jaap Winius wrote: Hi folks, Would anyone here happen to know of the existence of a Dutch Asterisk mailing list? If so, where can it be found? Not that I know off. I can start it if you want. It's not that I'm unable to pose my questions here in English, but I'm hoping that I may sooner find an answer there to the following question: What is the most reliable method for Asterisk to detect the Called ID for incoming calls on an analog line in the Netherlands? So far, I've tried using a Linksys SPA3000 and an SPA3102, as well as a Digium Wildcard TDM401BF for this purpose, but all to no avail. I suspect that there is a solution, but perhaps the people who are familiar with it like to hang out somewhere else. At least, I hope that's the case. Cant help you with this. Callerid in .nl is done with dtmf on analog lines. Maybe that is of some help ? Can anyone help? Thanks Jaap ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk virtualization on VMWARE SX infrastructure
On 14:42, Sat 17 May 08, Steve Totaro wrote: On Sat, May 17, 2008 at 2:18 PM, nik600 [EMAIL PROTECTED] wrote: Hi what about asterisk virtualization on VMWARE XS infrastructure? The system installed will manage a call center with 50 operator, queues, CDR logging on external database. the protocol used is SIP, probably with G711 codec. Virtualization of Asterisk i a risk regarding performance? Thanks to all I wouldn't do it. Maybe in a lab but certainly not for a 50 seat call center. I would ;) We run asterisk under vmware in production and have no problem with it. This is in a pure voip setup. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk virtualization on VMWARE SX infrastructure
On 16:18, Sat 17 May 08, Steve Totaro wrote: On Sat, May 17, 2008 at 2:56 PM, Michiel van Baak [EMAIL PROTECTED] wrote: On 14:42, Sat 17 May 08, Steve Totaro wrote: On Sat, May 17, 2008 at 2:18 PM, nik600 [EMAIL PROTECTED] wrote: Hi what about asterisk virtualization on VMWARE XS infrastructure? The system installed will manage a call center with 50 operator, queues, CDR logging on external database. the protocol used is SIP, probably with G711 codec. Virtualization of Asterisk i a risk regarding performance? Thanks to all I wouldn't do it. Maybe in a lab but certainly not for a 50 seat call center. I would ;) We run asterisk under vmware in production and have no problem with it. This is in a pure voip setup. Production = 50 seat call center? What would an hour or two of downtime cost you in your production setup? That would be: one of the many customers we host. We have a hosted setup with over 100 companies, so an hour or two will be a massive claim I'm sure. At the moment we have 4 asterisk servers under vmware that act like one freaking big and stable machine to the outside world. I dont think this is because of vmware, could have done the same setup with asterisk dedicated hardware but why bother when it works this way as well ? -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk concurrent calls count
On 16:20, Sat 17 May 08, Steve Totaro wrote: On Fri, May 16, 2008 at 8:37 PM, Jay R. Ashworth [EMAIL PROTECTED] wrote: On Fri, May 16, 2008 at 10:11:11AM -0400, Steve Totaro wrote: It seems any constructive criticism offered, you take as an attack against Digium. That is not a good attitude. I dunno, Steve; I wouldn't call Digium needs to 'man-up' constructive criticism, myself. I'd call it an ad-hominem. Tilghman *does* seem to be a bit of a cheerleader, but there's nothing wrong with that... unless you're an *employee*, and you're going out of your way to hide it. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) He is an employee and he does not post from a Digium account or include that fact in his signature. Not that it is to hide the fact, but it certainly is obfuscated. I think it just shows that his opinions are his, and in no way are linked to the 'digium opinion' -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk - CRM Integration
On 18:16, Tue 29 Apr 08, Fernando Berretta wrote: Is Sugar CRM the best Free CRM to be integrated with Asterisk ? Is Asterisk VoiceRD Integration the best integration patch to be used with Sugar CRM ? Is any other ? Have a look at Covide: http://sourceforge.net/projects/covide /shameless_plug -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GoToIfTime problem
On 15:43, Thu 24 Apr 08, Lee Jenkins wrote: I'm having a problem at a custom site where GotoIfTime doesn't seem to be working for some reason. I had putty running and logging CLI output and below is the call data: -- Executing Answer(Zap/3-1, ) in new stack -- Executing Ringing(Zap/3-1, ) in new stack -- Executing Wait(Zap/3-1, 0) in new stack -- Executing SetMusicOnHold(Zap/3-1, default) in new stack -- Executing Goto(Zap/3-1, check_time|s|1) in new stack -- Goto (check_time,s,1) -- Executing GotoIf(Zap/3-1, 0?set_no_callerid|s|1) in new stack -- Executing NoOp(Zap/3-1, CallerID: 443866 Cell Phone MD) in new stack -- Executing GotoIfTime(Zap/3-1, 08:30-17:00|mon-fri|*|*|?daytime_ivr|s|1) in new stack -- Executing Goto(Zap/3-1, after_hours|s|1) in new stack -- Goto (after_hours,s,1) This call came in at about 3:10 PM EDT today (Thursday). I did a date command at the linux prompt and the date and time of the computer is set correctly. Now, I have had problem with this particular computer in that the date/time gets changed somehow, although I'm not sure exactly how. I've changed it back several times using the commands (copied from command line history): # date -s 23 APR 2008 1:42:00 # hwclock --utc --systohc Install ntp so it will sync with the internet all the time. It's the first package I install on servers, no matter if it's brandnew or not-so-brandnew hardware. Time IS important for a lot of applications. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7961 + 7914, speeddials, BLF Asterisk 1.4?
On 16:12, Tue 22 Apr 08, Patrick wrote: Hi, Does anyone have any experience with a Cisco 7961 + 7914 Operator Console setup and speeddials/BLF on Asterisk 1.4? Would appreciate feedback if this works reliably. I have a 7961 on skinny registering on an 1.4.19 box with chan_sccp and speeddials work fine so that part seems ok. I have no experience with the BLF part. From googling around it seems that BLF on the 7914 only works with skinny. Is that the case? If so, anyone know if chan_skinny as part of Asterisk 1.4 will do or will I need to use chan_sccp? Any other requirements or patches needed? Your feedback is most appreciated. chan_skinny in 1.4 is very limited. It does not support hints/speeddials on the phones. The 1.6 version has this, and should work fine in your setup. I'm using chan_skinny at home with both an 7905 and 7960 and hints/speeddials just work. I dont know how good chan_sccp is these days. I used it in the 1.0 time but went with chan_skinny a long time ago (I only use it at home, and there I run trunk) -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] need examples of asterisk and mysql integration
On 13:13, Tue 22 Apr 08, Eric Fort wrote: I'm presently working on a project to build a scheduling system accessible by both web and phone. on the web side one can query what items are available when by using the time or the item as a key then reserve for an available time slot. reservations may also be modified by the user that made them or an admin. Where may I find examples of doing similar things with asterisk? all I've been able to find thus far is examples of how to store call detail records and voicemail using a database. Hi, With AGI you can write logic in whatever language fits you best. if you like perl, use the asterisk-perl stuff. if you like php, you can use the native mysql stuff if you like bash, you can use the mysql commandline client etc etc You can even use the MYSQL dialplan functions. Not sure where they are, I think in -addons. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unable to load module chan_zap.so
Make sure /usr/lib/asterisk/modules/chan_zap.so is on your system.
If not, my best guess is you compiled asterisk before zaptel.
You'll need to recompile asterisk with the zaptel channeldriver enabled.
Check with: make menuselect
On 17:02, Mon 14 Apr 08, Jeremy Malcolm wrote:
I am having trouble with chan_zap.so not loading. When I load it from
modules.conf, Asterisk bails out without any error message. When I
load it from the console, it just says Unable to load module
chan_zap.so no matter what verbose level I am using.
dmesg says:
Zaptel Version: 1.4.4
Zaptel Echo Canceller: MG2
Freshmaker version: 73
Freshmaker passed register test
Module 0: Installed -- AUTO FXS/DPO
Module 1: Installed -- AUTO FXO (FCC mode)
Module 2: Installed -- AUTO FXO (FCC mode)
Module 3: Installed -- AUTO FXO (FCC mode)
Found a Wildcard TDM: Wildcard TDM400P REV I (4 modules)
Registered tone zone 1 (Australia)
lsmod says:
wctdm 30912 0
wcfxo 9344 0
zaptel180388 2 wctdm,wcfxo
ztcfg -vv says:
Zaptel Version: 1.4.4
Echo Canceller: MG2
Configuration
==
Channel map:
Channel 01: FXO Loopstart (Default) (Slaves: 01)
Channel 02: FXS Kewlstart (Default) (Slaves: 02)
Channel 03: FXS Kewlstart (Default) (Slaves: 03)
Channel 04: FXS Kewlstart (Default) (Slaves: 04)
4 channels to configure.
cat /proc/zaptel/* says:
Span 1: WCTDM/0 Wildcard TDM400P REV I Board 1
1 WCTDM/0/0 FXOLS
2 WCTDM/0/1 FXSKS
3 WCTDM/0/2 FXSKS
4 WCTDM/0/3 FXSKS
/etc/zaptel.conf is:
# Span 1: WCTDM/0 Wildcard TDM400P REV I Board 1
fxols=1
fxsks=2
fxsks=3
fxsks=4
# Global data
loadzone = au
defaultzone = au
I have Googled for help but not found anything. Does anyone have any
suggestions?
TIA
--
Jeremy Malcolm LLB (Hons) B Com
Internet and Open Source lawyer, IT consultant, actor
host -t NAPTR 1.0.8.0.3.1.2.9.8.1.6.e164.org|awk -F! '{print $3}'
Luxury Perth apartment for sale!
http://www.yourestate.com.au/sresult.php?property_id=8581
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GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD
Why is it drug addicts and computer aficionados are both called users?
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Re: [asterisk-users] best way for call detail logging
On 10:00, Thu 10 Apr 08, Pete Kay wrote: Hi, I would like to be able to log call details in Asterisk. The kind of logs that I like to generate is like this: From To Forward Time Incoming Call604-343-3334 503-233-4454 13:33:32 Extension Routing 503-233-4454 Extension 403 13:33:32 Forwarding 503-233-4454 454-444-2334 13:33:32 where 503-233-4454 is my DID number. Basically, I would like to log how calls are being handled in Asterisk. I understand I can use AGI to log the information in database, but I am wondering if this is scalable enough for large number of users. I am using realtime CDR but it does not record the kind of detail that I am looking for. If I don't use AGI, what would be the best way to do it? Can someone please give me some advice or inputs? Thank you very much in advance for your suggestion. Thanks, Pete Maybe write something that connects to the AMI and listens to what happens there. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ring back when free?
On 10:35, Fri 04 Apr 08, Tony Mountifield wrote: Has anyone here implemented Ring back when free in Asterisk? The way it works in the UK is as follows: 1. A calls B. B is engaged (busy). 2. A hears The number you called is busy. To use ringback, press 5 3. A presses 5, and hears Your ringback request has been accepted. 4. A hangs up. 5. Later, B hangs up. The system then calls A (if A is now busy, it waits until A is clear again). 6. If/when A answers, the system calls B on A's behalf and A hears ringing. Any implementation has to cater for the fact that when B is busy, he could be either the calling or the called party on his current call. If he is the calling party, he will execute 'h' when he clears, but if he is the called party, he won't be in the dialplan to execute 'h', so we need some other way to invoke the ringback (step 5). Thoughts? Have a look at this: http://bugs.digium.com/view.php?id=10689 -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Howto connect to Cirpack softswitch with Asterisk ?
On 10:11, Wed 02 Apr 08, Robert Rozman wrote: Hi, has anyone connected Asterisk to Cirpack softswitch sucessfully ? Any howto or more info about needed Asterisk SW and setup ? Yes, it works fine. Where do you get stuck ? It's basically a normal sip connection setup. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] show uptime and last reload
On 01:40, Wed 02 Apr 08, Vieri wrote: Hi, I just upgraded from 1.2 to 1.4. In 1.2, when I did a show uptime I used to see a second line telling me the time since the last reload. Has this been removed in 1.4? The following is the output of my two test boxes: Connected to Asterisk 1.4.18.1 currently running on voip2 (pid = 10605) Verbosity is at least 3 voip2*CLI show uptime System uptime: 15 hours, 55 seconds voip2*CLI sip reload Reloading SIP == Parsing '/etc/asterisk/sip.conf': Found voip2*CLI show uptime System uptime: 15 hours, 1 minute, 28 seconds voip2*CLI - Connected to Asterisk 1.2.27 currently running on voip1 (pid = 26496) -- Remote UNIX connection Verbosity is at least 3 voip1*CLI show uptime System uptime: 4 days, 23 hours, 55 minutes, 1 second Last reload: 1 day, 4 minutes, 23 seconds voip1*CLI Can you try with: reload instead of just a sip reload ? -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Had it with Dell Garbage - HP Question
On 00:05, Mon 31 Mar 08, Al Baker wrote: Could you elaborate a bit more on : For example, if I install zaptel from source, your support contract with them is void. Does this mean it is impossible to run Asterisk on Vendor Supported versions of RedHat or Suse ? Installing zaptel from source means you use a kernel module that is not tested/supported by RedHat/Suse. So if you call them for support they wont help you unless you unload this module and then reproduce the problem. Thanks Michiel van Baak wrote: On 02:34, Sat 29 Mar 08, Al Baker wrote: Helps a bunch !!! One follow up question - out of all of your possible choices for the OS how did you pick *Debian*. I 'm not saying is bad, I just know nothing about the particular disto. and and very curious what it brought to the table that made you pick over say *RedHat* - where you can *buy support *or *SUSE* - where you can *buy support*. My fear from hell is that I' get 50 or 60 of these boxes in, start having kernel panics, and have no damn body to help except the folks on mailing lists. Mind you these are often really smart people, very generously giving of their time, but not quite the say as a manned/paid support organization. I choose Debian because I was already using it. And because there are people out there that can help me. I dont want the support from suse or redhat because they wont help me when running anything that's not in their repositories. For example, if I install zaptel from source, your support contract with them is void. I also really like the Open and Free mindset of Debian. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] e164.org
On 00:36, Sun 30 Mar 08, Armin Schindler wrote: On Sat, 29 Mar 2008, Grey Man wrote: Does anyone know if the e164.org ENUM service is still active? If anyone who has anything to do with the e164.org ENUM site monitors this list could you check your signup page as the Captcha's (the test to see if you are human) fails for both the text and audio tests every time. I'd post a message on the e164.org forums but the signup page there has the test missing altogether. I don't really know the 'official' status, but I use it and it does work without problems. Same here -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Had it with Dell Garbage - HP Question
On 02:34, Sat 29 Mar 08, Al Baker wrote: Helps a bunch !!! One follow up question - out of all of your possible choices for the OS how did you pick *Debian*. I 'm not saying is bad, I just know nothing about the particular disto. and and very curious what it brought to the table that made you pick over say *RedHat* - where you can *buy support *or *SUSE* - where you can *buy support*. My fear from hell is that I' get 50 or 60 of these boxes in, start having kernel panics, and have no damn body to help except the folks on mailing lists. Mind you these are often really smart people, very generously giving of their time, but not quite the say as a manned/paid support organization. I choose Debian because I was already using it. And because there are people out there that can help me. I dont want the support from suse or redhat because they wont help me when running anything that's not in their repositories. For example, if I install zaptel from source, your support contract with them is void. I also really like the Open and Free mindset of Debian. Thx for sharing !!! Michiel van Baak wrote: On 08:02, Thu 27 Mar 08, Al Baker wrote: How do you get notifications ? Is this thru one of the add on packages HP sells for the box ? Which One ? Could you be more specific about what you mean by a recovery CD and hod do you get console access below multi used to do recovery ?? What is integrated ILO BIOS Access sounds cool. What O/S you usin and what made you pick it ? What kind and how many RAIDS are you using. The HP site gave like 8 different RAID controllers and like 20 CPUs to chose from. How did you chose ? Thx for sharing !!! I'm not the op, but sending a reply anyways. The notifications come from the HP tools you can download for free from their website. The recovery cd is probably a selfmade installer for their setup. At least that's what we have. the ILO stuff is to give you access to the box like you were sitting right in front of it with a physical keyboard and monitor, but over IP. You can boot the machine, access the cd in your local machine etc, even if the box is on the other side of the moon. We use Debian. HP even supports it on their DL380 boxen. We use the P400 raid controller. Setup RAID5 with 3 disks. CPU we use right now is the Intel E5405 Hope this helps a bit. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Had it with Dell Garbage - HP Question
On 08:02, Thu 27 Mar 08, Al Baker wrote: How do you get notifications ? Is this thru one of the add on packages HP sells for the box ? Which One ? Could you be more specific about what you mean by a recovery CD and hod do you get console access below multi used to do recovery ?? What is integrated ILO BIOS Access sounds cool. What O/S you usin and what made you pick it ? What kind and how many RAIDS are you using. The HP site gave like 8 different RAID controllers and like 20 CPUs to chose from. How did you chose ? Thx for sharing !!! I'm not the op, but sending a reply anyways. The notifications come from the HP tools you can download for free from their website. The recovery cd is probably a selfmade installer for their setup. At least that's what we have. the ILO stuff is to give you access to the box like you were sitting right in front of it with a physical keyboard and monitor, but over IP. You can boot the machine, access the cd in your local machine etc, even if the box is on the other side of the moon. We use Debian. HP even supports it on their DL380 boxen. We use the P400 raid controller. Setup RAID5 with 3 disks. CPU we use right now is the Intel E5405 Hope this helps a bit. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Druid Open Source Edition
On 18:12, Wed 12 Mar 08, Joshua Wilson wrote: How did you like the gui interface? For a standalone PBX it's the best gui I've seen so far. On 3/12/08, Michiel van Baak [EMAIL PROTECTED] wrote: On 15:32, Wed 12 Mar 08, Joshua Wilson wrote: I don't believe it supports multi-tenant as of yet. It could be requested I am sure. I created a new VM and installed it. Guess what, no multi tenant support. Too bad all them good GUI tools never come with multi-tenant features On 3/12/08, Michiel van Baak [EMAIL PROTECTED] wrote: On 11:43, Wed 12 Mar 08, Joshua Wilson wrote: I have recently noticed that druid @ http://www.voiceroute.org has created an open source edition of their platform. I downloaded it today and installed it on a play system where I have about 20 ip phones ranging from cisco, polycom and aastra phones. I didn't even have to configure them as the system automatically did it for me. I have been using trixbox/freepbx combination for over that last year and I will now be making the switch to druid. It came with a user portal that was easy to use and had alot of great features. Has anyone downloaded this system today?? Can you please let me know what you think as well. -Josh Does it implement the ability to run more than 1 PBX in asterisk ? (Virtual PBX) To be clear: more then 1 company using the same physical asterisk -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] does the meetme module still require an external timing source?
On 16:27, Wed 12 Mar 08, Steve Totaro wrote: Try Callweaver. Thanks, Steve Totaro or app_conference for asterisk. That does the trick for me on OpenBSD where you dont have ztdummy. On Wed, Mar 12, 2008 at 4:12 PM, Dennis Christopher [EMAIL PROTECTED] wrote: Thanks Matt, However I am looking to see if Asterisk with meetme is viable on OS X, and I believe that ztdummy will not compile on that platform. If so, I would need an alternative to meetme to do conferencing...? Dennis On 12-Mar-08, at 4:02 PM, Matt Riddell wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Dennis Christopher wrote: All, Can anyone confirm if the meetme module still requires an external timing source, such as a card and or driver? Correct, but insofar as a driver, you can just use ztdummy, which will be loaded by default when starting up zaptel if you have no hardware installed. - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFH2DbpDQNt8rg0Kp4RArouAKCF0D36feiSxokdOx8UzF2gGOhonACgou4K WIAhdj/PUrOx5Z4N0fePRqM= =xfLA -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Druid Open Source Edition
On 11:43, Wed 12 Mar 08, Joshua Wilson wrote: I have recently noticed that druid @ http://www.voiceroute.org has created an open source edition of their platform. I downloaded it today and installed it on a play system where I have about 20 ip phones ranging from cisco, polycom and aastra phones. I didn't even have to configure them as the system automatically did it for me. I have been using trixbox/freepbx combination for over that last year and I will now be making the switch to druid. It came with a user portal that was easy to use and had alot of great features. Has anyone downloaded this system today?? Can you please let me know what you think as well. -Josh Does it implement the ability to run more than 1 PBX in asterisk ? (Virtual PBX) To be clear: more then 1 company using the same physical asterisk -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Druid Open Source Edition
On 15:32, Wed 12 Mar 08, Joshua Wilson wrote: I don't believe it supports multi-tenant as of yet. It could be requested I am sure. I created a new VM and installed it. Guess what, no multi tenant support. Too bad all them good GUI tools never come with multi-tenant features On 3/12/08, Michiel van Baak [EMAIL PROTECTED] wrote: On 11:43, Wed 12 Mar 08, Joshua Wilson wrote: I have recently noticed that druid @ http://www.voiceroute.org has created an open source edition of their platform. I downloaded it today and installed it on a play system where I have about 20 ip phones ranging from cisco, polycom and aastra phones. I didn't even have to configure them as the system automatically did it for me. I have been using trixbox/freepbx combination for over that last year and I will now be making the switch to druid. It came with a user portal that was easy to use and had alot of great features. Has anyone downloaded this system today?? Can you please let me know what you think as well. -Josh Does it implement the ability to run more than 1 PBX in asterisk ? (Virtual PBX) To be clear: more then 1 company using the same physical asterisk -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dead Air on PF firewall
On 17:31, Tue 11 Mar 08, NOC ph wrote: Hi Mich, I added the following line for the RTP its still the same, I can hear ring but no voice when answer from the other side. Any more ideas? Firewall rules look ok now. Like I said, did you set externip and localnet settings in asterisk sip.conf ? -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dead Air on PF firewall
On 19:56, Tue 11 Mar 08, NOC ph wrote: Yes, here's my sip.conf [general] register = 1000:[EMAIL PROTECTED]/1000 callevents = yes port = 5060 nat = yes canreinvite = no #bindaddr = 172.16.1.1 - if I en able this call cannot go out... localnet = 172.16.1.0/24 externip = 203.172.25.11 Thanks... Ok, try to enable all logging in pf and 'set loginterface' etc. After that, run: tcpdump -n -e -x -i pflog0 There you will see the blocked traffic. Maybe that will give you an idea. Michiel van Baak wrote: On 17:31, Tue 11 Mar 08, NOC ph wrote: Hi Mich, I added the following line for the RTP its still the same, I can hear ring but no voice when answer from the other side. Any more ideas? Firewall rules look ok now. Like I said, did you set externip and localnet settings in asterisk sip.conf ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dead Air on PF firewall
On 07:00, Mon 10 Mar 08, NOC ph wrote:
Hi All,
I have an asterisk box on my DMZ, and I'm using a PF for my firewall, I
can make a call but some reasons I have a dead air.
Any Ideas? below are my rules...
ext_if = bce0
int_if = bce1
altitude = 172.16.1.0/24
machines
vbox = 172.16.1.1
uci = 172.16.1.4
voices = 203.172.x.1
ipc = 203.172.x.2
default deny
set block-policy return
set loginterface $ext_if
set skip on lo
scrub in
nat
nat on $ext_if from !($ext_if) - ($ext_if:0)
nat on $ext_if inet proto { udp tcp } from $vbox to any port 5060 -
$ext_if port 5060
nat on $ext_if inet proto tcp from $uci to any port 1500 - $ext_if port
1500
Why those two rules ? The first nat rule already takes care
of that
rdr on $ext_if proto { udp tcp } from any to $ext_if port 5060 - $vbox
port 5060
rdr on $ext_if proto udp from any to $ext_if port 5100 - $vbox port 5100
you have to forward the rtp ports as well
rdr on $ext_if proto udp from any to $ext_if port
1:2 - $vbox
filtering section
pass out on { $int_if, ext_if } inet proto { udp tcp } from $altitude to any
pass in on $ext_if inet proto { tcp udp } from $ipc to any port 5060
pass in on $ext_if inet proto tcp from $ipc to any port 1500 flags S/SA
keep state
And you should allow the rtp ports as well
pass in on $ext_if inet proto udp from any to any port
1:2 keep state
pass in on bce0 proto tcp from $ipc to any port ssh flags S/SA keep state
pass in inet proto icmp all icmp-type echoreq keep state
pass in quick on bce1
For reference, here are my pf rules for my internal pbx:
##
# Macros #
##
ext_if = rl0
ext_ip = 82.95.XXX.XXX
int_if = wb0
int_net = 192.168.2.0/24
voip_server = 192.168.2.4
voip_ports = { 4569, 5060, 1:2 }
# NAT rules: rdr, nat, binat #
nat on $ext_if from $int_if:network to any - $ext_ip
# asterisk server
rdr on $ext_if proto udp from any to any port $voip_ports -
$voip_server
#
# Filtering #
#
# voip always goes in the priority class
pass out quick on $ext_if inet proto udp from any to any
port $voip_ports keep state queue q_pri
pass in quick on $ext_if inet proto udp from any to any port
$voip_ports keep state queue q_pri
Also, make sure in asterisk sip.conf you have the externip
and localnet config parameters set.
--
Michiel van Baak
[EMAIL PROTECTED]
http://michiel.vanbaak.eu
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD
Why is it drug addicts and computer aficionados are both called users?
___
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Re: [asterisk-users] display time on Cisco 79xx
On 10:59, Mon 10 Mar 08, Don Smith wrote: I am running Asterisk 1.4.5 on a debian Linux server. Saturday night/Sunday Morning Daylight Savings time occurred. The server shows Mon Mar 10 10:59:42 PDT 2008 when I do a date command, but the Cisco 7940 and 7960 show 09:59 10/03/08. How do I update the time display on the telephones please? I guess they are not running Skinny right ? I have no idea how they work with the SIP image, but Skinny image gets the time from asterisk just fine -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
