Re: [asterisk-users] tls on asterisk 13
On Wed, 2015-07-08 at 15:09 -0400, Ryan, Travis wrote: > Asterisk13 can do native tls with each phone? Nice. Some soft phone support TLS, but does anybody knows a soft phone that support pkcs11? (keys & certs stored on a smart-card) Hans -- _ -- 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] Asterisk Phone ( Telecom feature )
On Tue, 2014-10-07 at 08:37 -0500, Don Kelly wrote: > JG confirmed that "it" is possible, but "it" has not been defined. > > Without knowing what kind of instruments you are using, a possible "it" > would be for a party to dial a 4-digit extension number to talk to someone > internally, completing a call without using the PRI trunks. Indeed, "it" is rather vague. The intercom I came across the last couple of decades, were simple analogue-phones, without a dial-pad, with only one button. If you lift the receiver and press that single button, you'll ring the other phone. And vice-versa. That sounds do-able, for any kind of phone connected to Asterisk. With a wildcard in the dialplan, you can create an truly "any-key" :-) But the O.P. might expect something else -- _ -- 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] Does Asterisk 1.8. Supports Video Calls
On Tue, 2014-09-02 at 13:18 -0500, Khalid Touati wrote: > so it seems Asterisk Versions does not support video I guess > > Used it with jitsi and linphone softphones, works just OK. Just for testing i did a video-call on the loop-back, great test tool for showing the influence of (limited-) bandwith / latency. Ideal for demo's Hans > -- _ -- 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] Attack on Sip server.
On Fri, 2014-06-27 at 22:24 +0530, Anurag Rana wrote: > > iptables -I INPUT 1 -p tcp --dport 5060 -m string > --string "VaxSIPUserAgent" --algo bm -j DROP > > You make a fundamental mistake here. Firewalls (both inline and hostbased) should drop everything by default. And you should specifically accept what you are expecting and capable of handling. Not the other way round. Above rule is something like: The front door is locked between 9:30 AM and 10:15 AM, as you expect burgers to come to your house. -- _ -- 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] High Availability with Asterisk
On Sat, 2014-03-08 at 20:27 +, ad...@3a.hu wrote: > My approach (in theory only, so please correct me if I'm wrong) would be > to run asterisk on multiple boxes (one each). A dedicated monitoring > box (nagios? custom scripts?) would perform frequent checks against the > boxes (one of my previous projects one asterisk was using call files to > demonstrate its health to another one). > > If a box fails, I would simply redirect/reroute its traffic to another > one, using network solutions. Such as shutting down the production > interface of a suspectedly failed asterisk box, having an idle one pick > up its IP address, or using load balancing / routing / NAT to redirect > the client's traffic to a standby box. > > My approach is based on the experience that linux based HA tools are > often not free, or don't scale well, or engineered to circumvent an > error in a slower manner (eg. booting a second VM takes too much time). > However in the network world, there are well known protocols that were > designed to take over in a matter of miliseconds. > > I do understand that this would not provide 'session' data, so failing > over to a different box would mean the need to re-register, could cause > calls to drop etc. This might be unacceptable for you. As I said in > the beginning, I haven't been building such systems, in my experience a > dropped call is not that big of a deal, if it happens because the > network cuts over to a different box. This could be handled with a pair > of frontend load balancers, where the number of asterisk boxes can be > transparent. > > hope this helps > adam === Hi Adam, Don't confuse "high availability" with "load balancing", as these two are not related. These two have totally different objectives and are achieved in different ways. Either/both of them can very well be achieved with opensource tools. Even with commercial software is maintaining call when a intermediate PABX breaks down nearly impossible -- _ -- 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] Lots of calls, less memory
On Mon, 2014-02-10 at 10:39 -0500, Tech Support wrote: > Rather than speculate, take a look at the output of "top". If you're > running out of memory, shut down useless processes. You'd be surprised what > processes get started by default that you don't need. You should also check > the Asterisk logs and look at the last few things Asterisk did right before > it restarted. You may also want to consider not loading Asterisk modules > that you are never going to use. Just a suggestion. > Regards; > John > > > -Original Message- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Justin > Sherrill > Sent: Monday, February 10, 2014 10:27 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Lots of calls, less memory > > On 14-02-10 9:46 AM, Mike wrote: > > What log entries are leading you to think that you're running out of RAM? > > None. It's just my guess. The log doesn't show anything except Asterisk > restarting. how about running "free" or "vmstat" inside cron every hour or so? If you do a "vmstat 1 10" each hour on the hour, it tells you 10 times with one second interval the amount of mem you got. If you do that within cron, you can see the difference during a couple of days. Running out of mem, will cause unexpected results (to be found in syslog), though rebooting should not be one of them. for unintended reboots there are a lot of hardware related causes though Some are easier to detect (like high temp) some are harder. My most favorite is a moron-co-worker, touching sensative parts (cpu, mem, mobo) with his ESD-unprotected hands. Problems might show up even after months or years after the crime has been commited. hw -- _ -- 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] Multiple IAX2 Trunks Load balancing
On Fri, 2013-12-13 at 06:20 -0600, Don Kelly wrote: > On Fri, 2013-12-13 at 12:48 +0500, Muhammad Usman wrote: > > Hi - I have 2 Asterisk servers connected using 05 IAX2 trunks. I want > > to load balance incoming calls over IAX2 trunks. If any trunk goes > > down the calls traffic will be shared with other available trunks. > > When it gets Up the script is supposed to perform as desired i.e in > > load balance mode. > > > Thanks in advance. > > > > Hans said: > > Perhaps it is possible to do the L.B. at the O.S. or network level, and let > all trunks appear to asterisk to one single trunk. > > Don asks: > > What's the value of load balancing multiple IAX trunks between the same > system pair? What resources are being balanced? > ++ Perhaps the O.P. can explain about his intentions... In some situations it makes sense though: If you have to connect two servers, and use different kind of infrastructure / multiple providers... hw -- _ -- 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] ISDN outgoing caller id
-Original Message- From: Gergo Csibra Reply-to: Gergo Csibra , Asterisk Users Mailing List - Non-Commercial Discussion To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] ISDN outgoing caller id Date: Tue, 27 Aug 2013 21:28:36 +0200 Tuesday, August 27, 2013, 8:41:18 PM, Patrick wrote: > On 08/27/2013 08:04 PM, Gergo Csibra wrote: >> Hi, >> >> is anybody out there who can set the outgoing caller id on ISDN (CAPI >> or misdn) channels? I've tryed everything what I found in forums, os >> voip-info.com but no luck. I use a fritz card with CAPI in my first >> installation (1 BRI), and a hfc 4 port bri card with misdn on other. >> The first installation have p-t-mp configuration, the second one is >> p-t-p. Both configuration is EuroISDN in Hungary. >> >> So, can anybody help me? > Have you checked with your Telco if they allow you to change the > callerid? If yes, are you setting the callerid to a number that you are > allowed to use? You can't just set callerid to any number you like. You > must "own" the number which you want to set callerid to. I have no > problem setting the callerid on outgoing calls via chan_capi to one of > the numbers that the telco assigned to me. Yes, of course I want to set our assigned numbers, becuse the called party sees "Unknown" now. -Original Message- It's been a while ago for me, but: Besides the item mentioned above (hit that one also) two things come to mind.. 1) is CLI-Display activated on that line? For some telco's it is a fascility that has to be enabled.. (you can check it by plugging in a isdn-handset, and try to make a call) 2) Perhaps accidentally activated the "HIDE CLI" activated? hw -- _ -- 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] Performance Asterisk large installation on Vmware/Xen
-Original Message- From: Rafael dos Santos Saraiva Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Performance Asterisk large installation on Vmware/Xen Date: Sat, 18 May 2013 15:01:06 -0300 Hi I would like the opinion of you and if anyone has a similar scenario. I have a project for installation of a Asterisk server in a client with about 400 extensions. My question is whether this scenario carry an Asterisk virtualized. Will be used only extensions and trunks sip sip, 1 queue with 2 agents, without call recording. It is best to use XEN or VMware? Which best version of Asterisk for this scenario? _ Use XEN in paravirtualized mode: NOT hardware/full virtualized! Even when using specialized drivers, you get a considerable performance hit. When virtualizing Linux, hw-virtualization is an unneeded waste of cpu-cycles. Acceptable for windows clients, not otherwise. -- _ -- 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] НА: asterisk-users Digest, Vol 105, Issue 40
From: virus.c...@mail.ru Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion To: asterisk-users@lists.digium.com Subject: [asterisk-users] НА: asterisk-users Digest, Vol 105, Issue 40 Date: Tue, 07 May 2013 07:53:53 +0600 help -Original Message- exten => 911,1,Answer() -- _ -- 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] looking for a way to do appointment reminders
-Original Message- From: jg Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] looking for a way to do appointment reminders Date: Fri, 26 Apr 2013 09:33:42 +0200 Hi Brandon! I have a "wakeup call" system based on "call files" that are generated by an external C program. The call files can be triggered by dialing a phone number (e.g. for waking up the hotel guest in room 333 at 6:15 am: *77*3330615) or from outside via a web interface, or whatever. It looks like your task has the same basic requirements. Setting up a call file based system is not very difficult, but details like pronunciation of the guest's or patient's name may involve some additional work. -Original Message- I dare to disagree. Phone call as a reminder won't work. When would you call them? An hour in advance? They will probably never make it in time, or you disturb them at an inconveniant moment. Only reasonable option is to send them an SMS. hw -- _ -- 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] Asterisk Calendar integration suggestions
Might have a look at tine: http://www.tine20.org/wiki/index.php/Admins/Asterisk_integration hw -Original Message- From: Steve Totaro Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk Calendar integration suggestions Date: Thu, 25 Apr 2013 10:43:52 -0400 Without knowing requirements, Sugar CRM seems to be the most supported. Thanks, Steve Totaro On Thu, Apr 25, 2013 at 9:22 AM, j...@millican.us wrote: Hello all, I am looking into building a calendar server (due to business requierments I can not use public hosted calender like Google), and am looking for suggestions based on experience with different calendar applications/servers available for Linux that you have integrated with Asterisk. If you can give a quick, simple list of what worked and what didn't I would be very grateful. Thank You, John -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] Installing Asterisk on Virtual Machine
Could it be distro-related? I have various versions of asterisk (from 1.4 upto 11.3) running paravirtualized or HW-virtualized with XEN. Normally i use the pre-build packages from suse, only when i want to try a release-candidates i need them myself. hw -Original Message- From: Sandeep Raju Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Installing Asterisk on Virtual Machine Date: Tue, 23 Apr 2013 10:18:00 +0530 Hi Tzafrir, I have installed Asterisk 11.2 on ubuntu 12.04 64 bit server instance running on my private openstack cloud. My bare machine is Intel® Core™ i7-2600K CPU @ 3.40GHz × 8 with 8GB ram and running at 64 bit ubuntu 12.04 desktop edition with Kernel Linux 3.2.0-23-generic. output of uname -a on my ubuntu cloud instance where i'm trying to setup asterisk.. Linux asterisk 3.2.0-23-virtual #36-Ubuntu SMP Tue Apr 10 22:29:03 UTC 2012 x86_64 x86_64 x86_64 GNU/Linux Here is my backtrace.. http://paste.kde.org/730316/ Sorry for the late reply... On Mon, Apr 22, 2013 at 5:13 PM, Tzafrir Cohen wrote: On Mon, Apr 22, 2013 at 03:44:45PM +0530, Sandeep Raju wrote: > Hi, > > I'm trying to install Asterisk 11.2 on a virtual machine in my private > opestack cloud.. When I compile Asterisk 11.2 from source (./configure, > make, make install) as specified in the Asterisk book and run it, it gives > me the error: "Illegal instruction (core dumped)". > > Any ideas how I can solve this? What operating system do you have installed there? What CPU? What is the output of: uname -a Illegal instruction means that you tried running an instruction that the CPU cann't run. Maybe an incorrect choice of optimization flags? Maybe this is due to libraries not matching your architecture? Next thing to do: get a trace from the core file that was dumped using gdb. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IPv6
Hi all, I had to re-install a new machine and noticed that by default, ip was only listening on 0.0.0.0, thus ipv4 only. Easily changed. However, when looking at iax.conf, I found here the same, but it looks like iax is still ipv4 only? If i change "bindaddr=192.168.0.1" towards "bindaddr=::", and look with "lsof -i" iax is still not listening on V6. Is iax/ipv6 still on the "TODO-list" ? Hans -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] chan_mobile: FXS
Hi all, Finally i got hold of some bt-dongles that seems p[retty stable, the asus-bt211. After installing them, i rebuild 11.3-rc1 added mobile.conf (bt-addres and blackberry address) and "mobile show devices" is showing me that the BT-link is up, and remains stable up. Seems good, but it looks like asterisk is seeing the BB as a trunk/FXO. However, i want to use the phone as an FXS. Before ending up in trying something that was never foreseen and perhaps even impossible, i was hoping that i could use the BB as an oridinary "audio device" and still use the keys on the phone for starting/ending calls, and the dialpad for selecting phone numbers. And having the connections go (via BT) through asterisk instead of GSM. Is this possible at all, or am i embarking on a "mission impossible" ;-) Hans -- _ -- 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] SIP account registration fails after upgrade to 1.8
-Original Message- From: Jaap Winius Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] SIP account registration fails after upgrade to 1.8 Date: Fri, 22 Mar 2013 02:46:43 + (UTC) On Thu, 21 Mar 2013 16:35:16 +, Jaap Winius wrote: > Hopefully, my ISP will see fit to squash this bug ASAP. Well, I got my answer from them quickly enough: Nope. Luckily, somebody was kind enough to suggest a workaround. Unfortunately, it involves, downloading the source code and making a few changes to it to prevent Asterisk from adding '@' to the end of the Call-ID string. Nevertheless, it's easy enough to do. The idea is to look for this string that appears twice in ./channels/chan_sip.c: ast_string_field_build(pvt, callid, "%s@%s", generate_random_string(buf, sizeof(buf)), host); And to change it to: ast_string_field_build(pvt, callid, "%s", generate_random_string(buf, sizeof(buf))); Now my Call-IDs look like this: Call-ID: 63935a8d2144d4f1309024fd7612f608 Instead of this: Call-ID: 4991f57656d159925b296e5b3b06496b@[2001:888:abcd:1::a] Still, I'd much prefer that my ISP fixed the problem instead, because now every time a security update becomes available for Asterisk, I'm going to have to download the source code, make the same changes, recompile it and install it all over again and again. Ho hum. Of course, an even better solution would be if Asterisk had a variable with which to alter the Call-ID string format so that I could omit the IP address. :-) Cheers, Jaap -Original Message- Hi Jaap, just wondering, might this perhaps be an IPv6 quirk? By altering '@' you got rid of : '@[2001:888:abcd:1::a]' Does the dame happen with V4-only? I presume you didn't activate V6 at your end lately? Other idea (perhaps pointless), you got the numeric address, would the same issue still exists if '2001:888:abcd:1::a' could be translater back into a dns-name? (include it in your /etc/hosts ?) Sometimes the '[]' cause some side-effects (specially if some regex are used unseen) Groet, hw -- _ -- 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] chan_mobile
-Original Message- From: Emiliano Vazquez Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] chan_mobile Date: Tue, 12 Mar 2013 18:01:34 -0300 El 10/03/13 13:18, Hans Witvliet escribió: > Hi, > > I've been looking at the list at: > http://www.voip-info.org/wiki/view/chan_mobile > > But when googling of any of the "known working" devices, there ain't any > for sale anymore, probably replaced by more recent types. > > So, anyone around here who bought recently an BT-dongle that is working > with asterisk? > > hw Hi Hans! You can try with chan_dongle [1]. Whit this you can use USB Modems from Huawei to get a full trunk running with your asterisk box. It's nice to use, can send and receive sms from your company. The quality is good and really better than chan_mobile. Best regards. [1] http://wiki.e1550.mobi/doku.php -Original Message- Hi Emiliano, thanks for your reply, I think i might use it for a different project, I got an huawei-E1820 But at the moment i have to look at something else: The issue is contacting people not currently in the office. I've been trying to accomplish secure voice with a softphone through a vpn-tunnel, but the choosen softphone turns out less reliable then expected. While still working on that thread, other option is to equip each laptop with a _proper_ blue tooth dongle, and use their dumb/smart phone as an USB-audio device. If they are near their laptop, presence should allow me to use chan_mobile. (with an additional advantage not having to pay GSM-providers abroad) So, main issue is stability, reliability and usability for "end users". Unless i can use a huawei as a single-channel BTS, i'll have to stick to use a BT-dongle. Hans -- _ -- 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] digium card and virualbox
-Original Message- From: bilal ghayyad Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] digium card and virualbox Date: Sun, 10 Mar 2013 20:18:52 -0700 (PDT) I am not mixing. I need this for LAB testing. How? This PCI passthrough, how to enable it on virualbox? --- > > Hi All; > > > > How to let the virualbox (ubuntu OS) to be able to see > the digium card? Because when I install elastix or asterisk > with dahdi, it is not able to see the digium card if the > installation though the virualbox .. What is the solution? > The solution is to run Ubuntu and Asterisk on your hardware > natively, > not through VirtualBox. > > Virtualisation and high-performance hardware such as > telephony cards > (it will be creating 8000 interrupts per phone line per > second) do not > mix, I am afraid. What you might do, is running an very elementary asterisk on the iron, just acting as an PSTN-gateway. And run your experimental asterisk as a virtual client. HW -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] chan_mobile
Hi, I've been looking at the list at: http://www.voip-info.org/wiki/view/chan_mobile But when googling of any of the "known working" devices, there ain't any for sale anymore, probably replaced by more recent types. So, anyone around here who bought recently an BT-dongle that is working with asterisk? hw -- _ -- 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] asterisk with 1000 extensions
-Original Message- From: Carlos Alvarez Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] asterisk with 1000 extensions Date: Thu, 7 Mar 2013 09:30:31 -0700 On Thu, Mar 7, 2013 at 1:44 AM, Duncan Turnbull wrote: > This is not school assignment or home work :) We need to > setup in society buildings. Each flat will have SIP extension > (hard phone) registered on asterisk server. Calling > between SIP extensions is required. No PSTN / ITSP SIP > trunking. Just like inter-com feature. > > One way is to install 1000 IP Phones one at each flat > Secondly, install multiple-line SIP gateways with RJ-11 > cabling. > > Is there any other low budget solution for this setup? > Grandstream makes some inexpensive phones that are still very good. "Cheapest" hasn't been defined yet. What's the budget? Is there existing networking at these locations? Will you need switches? PoE? -Original Message- I think Carlos said it properly. Anything related to asterisk is insignificant compared to the rest. I dare to say, that the requirements if for 1000 people to communicate between themselves. So why SIP-phones? Why VOIP at all? Look at it a bit broader: network, maintenance (people), power, ... -- _ -- 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] Error to install Asterisk
-Original Message- From: termo termosel Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Error to install Asterisk Date: Tue, 5 Mar 2013 14:30:05 + Hi, if I write du -sh the response is 271M. I don't know that it means. Thanks, Jordi -Original Message- Hi Jordi, The "du" utility will show you the Disk Utilisation (hence the abbriviation du) What might be more relevant, is how much space is free. That you can examine with: df -h hw -- _ -- 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] GSM Sip Gateway
Thought so. make me wonder, as the O.P. specifically mentioned 4G. But all the functionalities offered by 4G hardly seems to be relevant for an asterisk-GW. Not? If only for speech, first generation should be enough ;-) hw -Original Message- From: John Novack Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] GSM Sip Gateway Date: Sun, 24 Feb 2013 09:15:37 -0500 >From the Freq. list given on eBay, I don't think they are. The listed freqs. are worldwide GSM since the mid 90's, but not 4G John Novack Hans Witvliet wrote: > Are these 4G comaptible > > > -Original Message- > From: Frank > Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: Re: [asterisk-users] GSM Sip Gateway > Date: Sun, 24 Feb 2013 07:40:19 -0500 > > USA, this will be use with a 4G network. > > On Feb 24, 2013, at 5:24 AM, longst wrote: > > > where are you from by the way > > > > Sent from Shitian Long > > > > > > On Feb 24, 2013, at 1:54 AM, Frank wrote: > > > > > Hi all, > > > > > > Anyone ever used GoIP GSM SIP Gateways ? > > > If yes, what was your experience with those ? > > > > > > I'm looking at this: > > > http://www.ebay.com/itm/HOT-GSM-VOIP-GoIP-Gateway-SIP-Trunk-to-Asterisk-iP-PBX-/280736774012?pt=US_VoIP_Business_Phones_IP_PBX&hash=item415d37377c > > > > > > If anyone has any (good) experience with another brand, I'll take the > > > names and models. -- _ -- 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] GSM Sip Gateway
Are these 4G comaptible -Original Message- From: Frank Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] GSM Sip Gateway Date: Sun, 24 Feb 2013 07:40:19 -0500 USA, this will be use with a 4G network. On Feb 24, 2013, at 5:24 AM, longst wrote: > where are you from by the way > > Sent from Shitian Long > > > On Feb 24, 2013, at 1:54 AM, Frank wrote: > >> Hi all, >> >> Anyone ever used GoIP GSM SIP Gateways ? >> If yes, what was your experience with those ? >> >> I'm looking at this: >> http://www.ebay.com/itm/HOT-GSM-VOIP-GoIP-Gateway-SIP-Trunk-to-Asterisk-iP-PBX-/280736774012?pt=US_VoIP_Business_Phones_IP_PBX&hash=item415d37377c >> >> If anyone has any (good) experience with another brand, I'll take the names >> and models. >> >> Thanks >> >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] Asterisk SMS()
-Original Message- From: A J Stiles Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk SMS() Date: Tue, 19 Feb 2013 16:50:10 + On Tuesday 19 February 2013, Nicholas Johnson wrote: > Thanks for the help. Right now I'm running asterisk on a raspberry pi > using a phone number from flowroute. Is using a company like flowroute > the same as connecting to the PSTN? Also i've tried to install smsq but I > couldn't find any good documentation to get it setup properly. So no, I'm > not using smsq. The bad news: You need a GSM modem to send SMS messages. The good news: It is not so. You can send SMS messages on POTS or ISDN lines See the voip-wiki about it -- _ -- 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] Asterisk calls between 2 private networks
-Original Message- From: Carlos Alvarez Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk calls between 2 private networks Date: Thu, 7 Feb 2013 10:36:36 -0700 On Thu, Feb 7, 2013 at 10:26 AM, Frank wrote: AJS, That is a solution that I am envisaging. But I would really love to try to work out with my issue first. It will allow me to deploy more phones in separates buildlings in the future. If I do the IAX solution, it means that for every building, I need a box.. Which I would like to prevent. Adding more points of failure and more devices to maintain without any real benefit is always the wrong thing to do. IAX is also flaky as hell. -- _ Carlos, with regards to your comment about IAX, where can i find your bug-report? -- _ -- 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] OT - Chan-mobile -Bluetooth dongle on remote LAN workstation
-Original Message- From: Olivier Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] OT - Chan-mobile -Bluetooth dongle on remote LAN workstation Date: Thu, 31 Jan 2013 08:25:42 +0100 Hello, On a LAN, is it possible to install a bluetooth dongle on one workstation (at this time, this workstation OS is not specified) and use it with chan_mobile ? I've read some USB over IP (or Ethernet) middleware exist but I'm not certain I'm looking at the right direction. Regards -- _ Hi Oliver, I've been trying to do this for a while. Been using latest blackberries and oldest nokia, and a laptop with build-in and also an external BT-dongle. What i noticed, is that the presence is highly unstable. Even without walking along (kept the phone 10 cm from the BT-dongle) it kept bouncing: found-gone-found-gone Unworkable. I wanted not only presence, but speech-patch also via BT. Idea was, that if co-workers are located at the other end of the world, they can still be reached on they handy, even when no GSM-roaming is acceptable (due to costs). hw -- _ -- 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] IAX2 support of video
On Tue, 2013-01-08 at 08:21 -0600, Danny Nicholas wrote: > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry > Geis > Sent: Monday, January 07, 2013 6:53 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] IAX2 support of video > > > > > > According to this: > https://wiki.asterisk.org/wiki/display/AST/Video+Telephony > yes. > > > > > > > I have a local server with two video phones - running SIP to each > phone. Works. > Then I have an IAX2 connection from that local machine to another > machine. > then a SIP connection from that machine to another machine where the > same model > video phone is in use. A call to that phone does not show video only > audio. > > All machines have in sip.conf:videosupport=yes > > Is there something else to get SIP/IAX2/SIP video call to work? > > Thanks > > Jerry > > > > Make sure you have the H.26X codec enabled at all points. > > Video is hard, but to make life easier, it is handy to add an extension that does the echo-function (after an optional announcement) That takes video-codec-mismatch out of the equation, as you are talking to yourself. Other benefit is that there is always someone that will answer the phone ;-) And won't complain doing video-cakk when dialing at 3am. hans -- _ -- 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] Doubt regarding jabber
On Tue, 2012-12-18 at 10:01 -0600, Jonathan Rose wrote: > Harish Mandowara wrote: > > I have Asterisk server 1.8.19 with jabber enabled. > > > > On the other side i have openfire server with asterisk-im enabled. > > > > I have a doubt, whether my sip client connected with asterisk can > > send message to other sip client, which is connected to same > > asterisk server. > > > > > > I have jitsi as a sip client. > > > > If its possible. Than please suggest any documentation regarding > > this. > > > > any help?? > > > > THanks a lot > > As far as I'm aware, SIP clients are generally incapable of using > XMPP to send and receive messages. I'm aware Jitsi can act as an > XMPP client, but its functionality as one has basically nothing to > do with Asterisk. Asterisk can use XMPP send and receive messages > to/from an XMPP server (also clients on that server by relay). > Jabber is also used for Google Talk and Google Voice, but I'm not > sure which versions those features work best in. I'd imagine 11 > would be your best bet if you wanted that functionality since it > has a bunch of Jabber improvements as well as chan_motif. > > So if you want Asterisk to send jitsi an IM, you need to set up > account on an XMPP server for them to use (as well as profiles to > connect with). Once you are sure you have Asterisk and the Jitsi > client connected to the XMPP server, you can send the message with > the dialplan application 'JabberSend' which takes arguments of > account (which is the account you are using to send), jid (who > is receiving the message) and the message itself. You can similarly > receive messages on Asterisk by using the JABBER_RECEIVE function > with similar arguments except no message and an optional timeout. > Hi, probably Jonathan is correct. firstly, much changes/improvements has been made since 1.8.19 so you might give asterisk-11 a try, in which xmpp has got a lot of attention. Secondly, as you use Jitsi, why don't you use jitsi's xmpp capabilities? afaicr, the SIP-part of jitsi is only capable of simple, not xmpp. Hans. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sip-user status
Hi all, I'm caught up in a struggle between people how can not make up their mind... Half way implementing a asterisk farm and they come up with another feature they've seen in kamaillo. What he showed me was this: three registered sip users, a) one changes his presence status on his softphone, and all see the status change. b) one calls another, and the third person see the status of the other two change to "busy". I've seen code/dialplan snippets where you could change your status by dialling a specific extension, on which asterisk will react (and change some variables accordingly), but that is not what i'm looking for. It seems that kamaillo has build-in features to react on sip-simple changes. Can i perform the same trick with asterisk? if so, how? Hans. -- _ -- 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] [asterisk] Guide for setup a server for end2end video call
On Tue, 2012-12-11 at 23:02 +0800, Barco You wrote: > Dear List, > > > Where can I find a guide for setup an Asterisk server which can > eastanblish a simple video call from two sip clients? > > > Thank you! > > > Regards, > Barco Hi Barco, I don't think there is a specific guide for this. >From the top of my head.. In /etc/asterisk/sip.conf, you will find the the default setting is _not_ to have video enbled. So either you enable it gloabally (for all sip-users) or individually for specific sip-users. Greatest pit-fall. you have to analyse the codecs for all hard-phones and/or softphones. You should have atleast one common codec enabled. If not, you will only get an audio connection (unfortunately, without any warning) Most safe option (atleast to begin with) is to use same clients at both sides, and configure them identically. Second suggestion, is to define a echo-function. If you dial the defined extension, you get not only audio echo, but also video-echo. I've been testing with a couple of clients, with various results. hans -- _ -- 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] disabling regular expressions
On Thu, 2012-11-15 at 12:13 +0100, Frederic Van Espen wrote: > On Thu, 2012-11-15 at 08:53 +0100, Hans Witvliet wrote: > > In stead of "12345678" i would like to use "b.c.o.gr...@minoss.nl" > > But afaicr the dots will cause problems > > If your extension does not start with an underscore, it is not > considered as an extension pattern. Correct me if I'm wrong please! > yes, indeed. Just tried it: OK. Stupid of me [blush] -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] disabling regular expressions
Hi all, Is there a simple way of disabling regular expressions in the dialplan? Reason for asking, is that people hate to remember numbers. So i want to use there full smtp address as as their extension. In stead of "12345678" i would like to use "b.c.o.gr...@minoss.nl" But afaicr the dots will cause problems Hans -- _ -- 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] Managing complex setups with Asterisk
On Thu, 2012-11-08 at 10:07 +0100, martin f krafft wrote: > also sprach Jeff LaCoursiere [2012.11.07.2049 +0100]: > > Just to chime in, if you REALLY want multi-tenant, it is super > > easy and surprisingly efficient to use kernel level virtualization > > to run multiple instances of asterisk (and even FreePBX). We use > > LXC to do this. The "host" runs an instance that has the dahdi > > hardware, drivers, and upstream connections. The "clients" have > > SIP connections to the host for all inbound/outbound > > Yes, separation into logical units is one way forward, but then you > will necessarily have redundant configuration between the instances. > It's nice to have clear separations (unless you cannot clearly > separate), but I am not convinced that this decreases complexity. Actually, i would suggest breaking it up and store most of your data into mysql (realtime). By breaking up, you can separate distinctive parts, like pstn-gateways, GSM-gateways, internal-proxies, external-proxies, voice-mail, conference-server, etc etc. If you store the user-specific data into a database, it doesn't matter on which proxy you register, the configuration is shared them all. Same for your dialplan. If you use LXC, the overhead will be less compared with using XEN. And if you keep each asterisk-container stupid, it is easier to maintain/replace. hw -- _ -- 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] asterisk as IVR using 3g usb modem
On Thu, 2012-10-18 at 15:45 +, Mahendra Dobariya wrote: > hi, > I want to use asterisk as IVR system , > but to make and receive GSM call, i want to use 3g usb modem.(voice > enabled) > http://www.huaweidevice.co.in/Products/MobileBroadband/E303c.php > > > and i want to install this system on two different machine > 1> on mac os x - > 2> raspberry pi- (debian wheezy)-->http://www.raspberrypi.org/ > > > thanx in advance.. Are you very sure about the last one (i.e. the r-pi)? These have a very few resourses (cpu, mem) If looking for something small, how about latest pandaboard, a bit more expensive, but less limited: http://www.hardware-modules.com/index.php?page=Browse&product_type=SBC&designer=Texas%20Instrument&module=Pandaboard%20ES%20(Texas%20Instrument%20-%20OMAP4460)&lang=en And still cheaper than most intel-based sff-boards. hw -- _ -- 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] asterisk as IVR using 3g usb modem
On Thu, 2012-10-18 at 17:18 +0100, Steven Howes wrote: > > On 18 Oct 2012, at 16:50, Mitul Limbani wrote: > > U would have to write a dahdi module for this 3G modem to help > > asterisk understand it as standard gsm channel. > > > Look up chan_datacard (i think that's what it's called from memory). > > > Steve > It got renamed, and is now: http://code.google.com/p/asterisk-chan-dongle/ It's just a tgz, no docu, no wiki.. chan_dongle-1.1.r14.tgz chan_dongle version 1.1 revision 14 sources Featured 6 days ago 6 days ago 184 KB 159 Looks a bit fresh... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RC2, was: Motif/XMPP for Google Voice
Hi, With regards to: On Mon, 2012-10-15 at 09:09 -0500, Joshua Colp wrote: > asterisk asterisk wrote: > > Dear all, > > Hola, > > > I wish to ask a question of the new Motif Channel in asterisk 11. > > > > I successfully compile the binary and run without error. However, when > > dialing out, no external connection only ringing. > > During testing some issues were uncovered with the Motif channel driver, > but unfortunately they did not make the last release candidate. My > suggestion is to get Asterisk 11 from SVN or if you are not comfortable > with that wait until the official Asterisk 11 release. > > Cheers, > And to: "Asterisk 11.0.0-rc2 Now Available" skimming through http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.0.0-rc2 I did not see any reference towards Motif/XMPP. So your code is still only in SVN, not in the RC2? Hans -- _ -- 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] conversion?
On Wed, 2012-10-10 at 18:09 -0300, Joshua Colp wrote: [snip] > Yes, there is no capability for video transcoding in any version of > Asterisk. Thanks for pointing out! So in case my managers starts nagging about it, they have two options: A) use hard/soft-clients with comparable codecs, B) raise enough funds for implementing video-stream transcoding ;-) [tough job, but do-able] hw -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] motif load
Hi, Are there any thoughts about how "cpu-expensive" motif is? Does it only translate SIP <--> jingle (during call-setup) if so, impact will probably neglectible. or does asterisk remains constantly in between the data-stream? In that case, it might be something to pay serious attention to, when doing multiple call conversions simultaneously... hw -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] conversion?
Hi, Perhaps can someone tell me if i had the wrong expectancies If one sip-clinet only supports GSM-codec, and another only supports g711-U, they still can call each other and asterisk does the transcoding Correct? If i try to do the same with an AV-call, (one only h264, the other only h263) there is no video-transcoding, and you get an audio-call only. Sure, video transcoding is probably horrible CPU-expensive, but is what i see correct (on a 1.8 system)? Hans -- _ -- 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] Case-sensitivity of Dialplan variables.
On Tue, 2012-10-02 at 17:11 -0700, Ira wrote: > At 02:19 PM 10/1/2012, you wrote: > >So respond here and let me know what you think. I got a couple of replies on > >the -dev list and they said that this would be good to put out on the -users > >list too. > > > >Mark Michelson > > > >In true Republican fashion, I'm going to vote for case-insensitivity. > > Given that many of the users were not programmers and didn't likely > grow up in a case sensitive world I'd also vote for case > insensitivity. I fall into that category, I grew up with dBase, > Clipper and VB and case issues get me all the time when I program in C. I would vote for case-sensitivity. True, i grew up in the early day's of PDP11, flex, uniflex and so-on, where case-sensitivity was default. I think it is a bad habit to write something else, from what you expect. More important is, that you get a un-avoidable error, when you try to read a variable, that isn't initialised (due to mixed case). Like in the old fortran/pascal/C days, where you just get a compilation error, that you had to solve before you could continue There is already too much insensitivity in this world, let's get rid of (at least) case insensitivity! hans -- _ -- 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] RealTime table fields ordering
On Fri, 2012-09-28 at 01:33 -0700, Vieri wrote: > Hi, > > According to http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip: [snip] > So it seems that the contrib directory and the asterisk.org wiki are > inconsistent and incomplete. > Of course I understand that these are 'contributed' files but they should be > proof-read by the Digium devs before packing them up into the official source > tarball. Or am I wrong about my observations concerning field order and field > omissions? > > Thanks, > > Vieri > how about the line: `ipaddr` varchar(15) DEFAULT NULL, Wonder how they try to squeeze an IPv6 address in it... should be: `ipaddr` varchar(50) DEFAULT NULL, hans -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] chan_motif, xmpp, jabber, jingle
Hi all, For one of my inverstigations it looks like i'm back to "square one" I'm trying to accept an incoming xmpp call and forward it conditionally to a sip, isdn, or voicemail. No google is involved as i use a local xmpp server (ejabberd) I was experimenting on 1.8.15.1 (with jabber.conf, jingle.conf), but some suggested me to have a look at asterisk11,so i did... I downloaded and built 11-beta1. Edited (according to the asterisk11 wiki-page) extensions.conf, chan_motif.conf, jingle.conf and restarted. Same behavior, except for minor details. As soon as I start, ejabberd tells me that the defined user becomes online. >From jitsi I can send a text-message, which I see as I enabled "debug" in motif.conf (This is actually progress, as in 1.8.15.1 I saw only empty strings coming along ;-) But when starting an audio or an AV-call, I only see the xmpp-debug message (used to be jabber-debug-message). Within de xmpp-message I see the capabilities (samplerate, codecs, address, port) from the jitsi-client. Although I made a separate context in the dialplan, it seems never to get there: hence no answer :( Eventhough I explicitly point to them in xmpp.conf and jingle.conf So my client remains in "connecting" for ever Other suggestion were to look at the xmpp server. However, i got three client-machines (two Ubuntu-12.04, one XP) all are installed with Jitsi (xmpp-client) and multiple accounts that are registerd on ejabberd. All of them can place jingle-calls to each other: so nothing wrong with the xmpp-clients or the xmpp-server. The fact that i see on the cli xmpp-debug messages ariving when trying to connect, or sending text messages indicates that asterisk is seen by xmpp-client/server. But i seem to be be missing something some vital config, as asterisk does not respond at all at those incoming trafic. Hans -- _ -- 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] chan_mobile
On Tue, 2012-09-18 at 17:43 +0100, Sebastian Arcus wrote: > Hi Hans, > > > The following page has some useful info: > > http://www.voip-info.org/wiki/view/chan_mobile > > Sebastian Indeed. Didn't realise it was so picky. just bought a couple of bt-adapters. Will try again tomorrow and feed the results into the wiki.. Tnx. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] chan_mobile
Hi all, In one of my other project i had a look at chan_mobile. I build 1.8.15.1 with the apropiate module. (in my distro asterisk is build without chan_mobile ;-) After i filled in the mac-addresses of the BT-adapter and the one from my phone, i see it is recognized, got connected, and immediate gets disconnected. Same behaviour if i use a completely different phone (BB). BT on either phone (one at the time) is constantly "on" and "visable'. Distance between dongle and phne just some centimeters Any suggestions? Hans -- _ -- 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] multiple users for jabber.conf
On Wed, 2012-09-12 at 00:01 -0500, Vladimir Mikhelson wrote: > Hans, > > I did not try 10 or 11 as I run 1.8.15. Following are the related > conf files. > > gtalk.conf > > [General] > context = default > allowguest = yes ; Required if you want to accept calls > from people Not on your contact list. > bindaddr= ;; These two settings are very critical for > getting > externip= ;; gtalk audio with Asterisk server behind > NAT > disallow=all > allow=ulaw > > [guest] ;;special account for options on > guest account > disallow=all > allow=ulaw > context=from-trunk > connection= > > jabber.conf > > [general] > debug=no ;;Turn on debugging by default. > autoprune=no ;;Auto remove users from buddy list. > autoregister=yes ;;Auto register users from buddy list. > > [Jab01] ;; Label > type = client ;; Client or Component connection > serverhost = talk.google.com > ;; Route to server > username = google-user-nam...@gmail.com>/asterisk;; Username with > optional resource. > secret = >;; Password > priority = 1 ;; Resource priority > port = 5222 ;; Port to use, defaults to 5222 > usetls = yes;; TLS is required by talk.google.com, > you'll get a 'socket read error' without > usesasl = yes ;; Use sasl or not > timeout=100 ;; Timeout on the message stack > status=available;; One of: chat, available, away, > xaway, or dnd > statusmessage = "Connected via Asterisk" ;; Custom status message > > [Jab02] > type = client > serverhost = talk.google.com > username = google-user-nam...@gmail.com/asterisk > secret = > priority = 1 ;; Resource priority > port = 5222 ;;Port to use, defaults to 5222 > usetls = yes ;;TLS is required by talk.google.com, > you'll get a 'socket read error' without > usesasl = yes ;;Use sasl or not > buddy=@gmail.com ;;Manual addition of buddy to list. > buddy=@gmail.com ;;Manual addition of buddy to list. > timeout=100 > status=available > statusmessage = "Connected via Asterisk" > > [Jab03] > > [Jab04] > > and so on. > > Reagrds, > Vladimir > Thanks Vladimir, Will digg up an 1.8 machine and give it a try! afaics the only diference is that i am using a local xmpp server (ejabberd) instead of google, but that should only make things easier i think... Hans -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] multiple users for jabber.conf
Hi all, Been reading about chan_motif / chan_xmpp in the wiki's for 1.8, 10 and 11 version of asterisk. In each example i got the impression that the asterisk server is registering on a XMPP server as a single user with the credentials as specified in jabber.conf. Instead of a single xmpp-user, could that also be multiple users? For instance, for each sip-user an xmpp-user? When i skim through most of the examples, the asteriskbox is used for making an outbound call with the jingle protocol. But how about incoming calls? I presume you need multiple xmpp-accounts, in order to differentiate multiple destinations. Not? Or to describe it in an other way: If you just do a single xmpp-registration, how can you become a destination for different end-users? how about multiple presence-states? [utterly confused] Hans -- _ -- 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] asterisk on arm
On Tue, 2012-09-04 at 13:58 +0500, qasimak...@gmail.com wrote: > How about stripping it down to bare minimum's? > How about an other ARM-board? http://gooseberry.atspace.co.uk/?page_id=13 Specifically the more mem (4GB) will help.. hw -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] xmpp / sip
Hi all, After making a nice demo-setup for one of our innivationmanagers, he came up with a completely different stratagy ;-( They want to have an Ejabberd server, with xmpp-clients. When you see a contact coming online, just point and click for making a phone call. Sounds/looks nice and do-able, but there is one catch: -incoming / outgoing call towards corporate pstn (E1) -incoming / outgoing call towards public pstn (E1) -incoming / outgoing call towards GSM (nanobts) So does anybody any any thoughts about mixing/translating XMPP and SIP? firstly: sip -> xmpp If i get an incoming call for user-A, i should be able to see the status of user-A, and if available, pass the call on to him. SecondlyL xmpp -> sip So if i can define a "external-user-A" and let him login on ejabberd, he should indicate "xmpp-available", and local people should be able to call him in that way. Sounds rather complicated, i know... Hans -- _ -- 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] Asterisk hangs while starting in OpenSuse 12.2
On Thu, 2012-08-23 at 15:01 +0530, Gopalakrishnan N wrote: > Hi, > > > Again I stuck up with OpenSuse 12.1 also, Installed OpenSuse 12.1 > (32bit) version in virtualbox. Downloaded Asterisk 1.8.15. Installed, > installation went fine. > > Have you tried the versions from the OBS? Or perhaps a virtualbox issue? Its notorious for vapourizing cpu-cycles... hw -- _ -- 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] HP DL360 G5 better than HP DL360 G7 ?
On Wed, 2012-08-01 at 19:39 +0800, D Tucny wrote: > > For reference... In my opinion HP servers should never be bought > without the battery or alternative, they shouldn't even be offered for > sale without it... In my case, our purchase department changed our order. They thought in their infinite wisdom, that they could strike a better deal (without informing us of course). During the two years to find who to blame, the all the G7 were replaced by a couple DELl's with fusion-IO cards, 64 cores and 256GB mem. Don't think we go back to the G7's anymore ;-) hw -- _ -- 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] PBX, IVR and Conferencing Platforms From the Same Installation of Asterisk
On Tue, 2012-07-24 at 11:07 +0530, Kannan wrote: > Hi Stelios, > > > Thanks for the response. > > > I take the following excerpt from your response. --- "You can, but > usually for virtual/hosted pbx's you need an additional > layer of management software or a lot of copy paste" > > > Could you please elaborate on that? Do need to modify Asterisk or > there exists some software that does the job? > > > Regards. If you have a (large) number of asterisk-servers, it might be handier not to duplicate configs,(to avoid mismatches) but have sip and exension config centralised in ldap or mysql. And when using dns-round-robin, you can almost add/remove machines on the fly. hw -- _ -- 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] HP DL360 G5 better than HP DL360 G7 ?
On Sun, 2012-06-03 at 23:23 -0400, Tom Browning wrote: > Any tips on solving the following performance conundrum: > > Asterisk 1.8.12.2 running on HP DL360 G5 and G7s > > tcpdump running to capture UDP 5060/SIP signaling to .pcap files > > All calls are ultimately B2BUA client -> asterisk -> PSTN > > Media stays on Asterisk at all times > > AGI script has exit handler that connects and updates an external > database upon BYE from either side. > > I know that if exit handler hangs around too long, Bad Things (tm) will > happen. > > Oddly, under load (60-100 B2BUA calls), the G7s start complaining: > > Autodestruct on dialog '' with owner in place (Method: BYE) > > I/O wait is actually higher on the G5s, the G7s have fancy disk cache > cards and never get above 1% i/o wait > > turn off the tcpdump process on the G7s and Autodestruct warnings go > away. The G7s should have > much more capacity than the G5s but we never, ever get Autodestruct > ... Method: BYE on the G5s. > > OS is identical CentOS in both cases. Every other environmental > config is the same (network, subnet, DNS etc). > > Architecture/bus/network card difference? tcpdump starving some other > resource to cause stuff to slow down? Hi Tom, Regarding G5's and G7's We have them both (all though not for asterisk) at work. and found a little snag. The purchase department interferred with our initial ordering of the hardware. As a concequence, they did not order the battery for the raid controller. (seems you have to order that explicitly) After it arrived, we were able to install linux on it and use it, but disk-io was way slower that the original G5. You might want to check/compare disk-io & throughput on your G5 vs G7. Just a thought Hans -- _ -- 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] asterisk 1.8 on Solaris/sparc
On Wed, 2012-07-18 at 02:27 -0400, Jeremy Kister wrote: > I've got the latest asterisk 1.8 running on a Netra X1 with Solaris 10 u10. > > The system itself is happy and phone calls (between two parties) seem fine. > > Unfortunately, when a caller listens to a Playback recording, there > seems to be moments of stutter - perhaps 1 second of stutter for every > 10 seconds of Playback. The stutter is not consistent at the same point > of the playback file. > > To eliminate encoding as an issue, I have only codec_ulaw/format_pcm > loaded and the recording is ulaw. I've niced down the asterisk process > to -19 even though I don't see asterisk taking more than 3% cpu. > > > Is this behavior indicative of a timing problem? loading > res_timing_pthread.so makes things horribly worse. i don't believe any > other software timer is available for Solaris/sparc, right ? > > other thoughts ? Perhaps system too busy, disk not fast enough? before doing a play-back, run "iostat 1" in another window Incase iowait is too high, try moving the files with the playback sound/speech upon tmpfs (thus eliminating the hard disk) hw -- _ -- 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] AMR - Segmentation Fault
On Wed, 2012-07-04 at 10:15 +0530, Chandrakant Solanki wrote: > So, is http://sourceforge.net/projects/aterisk-amr/files/ same patch > also works in 1.8.13.0?? > > On Wed, Jul 4, 2012 at 3:18 AM, Hans Witvliet > wrote: > On Tue, 2012-07-03 at 17:13 +0530, Chandrakant Solanki wrote: > > Hi All, > > > > OS : Cent OS 5 64Bit > > Asterisk : 1.8.0-rc2 > > > > AMR Source Link : > http://sourceforge.net/projects/aterisk-amr/files/ > > > > When I tried to call or start asterisk, I found > "Segmentation Fault". > > > Without trying to be pedantic, but "1.8.0-rc2" > Ever considered upgrading? To 1.8.13.0 or so.. > Hm, i see. Looks like somebody is seriously hibernating: 1.8.0-rc2_asterisk_amr_patch.diff 2010-10-15 Almost two years old! If it is only the codec itself, you might try: http://ftp5.gwdg.de/pub/linux/packman/suse/12.1/Multimedia/src/amrnb-10.0.0.0-1.1.src.rpm or http://ftp5.gwdg.de/pub/linux/packman/suse/12.1/Essentials/src/amrwb-10.0.0.0-1.1.src.rpm As these are source packages, you might be able to turn then into deb's or so Hans -- _ -- 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] AMR - Segmentation Fault
On Tue, 2012-07-03 at 17:13 +0530, Chandrakant Solanki wrote: > Hi All, > > OS : Cent OS 5 64Bit > Asterisk : 1.8.0-rc2 > > AMR Source Link : http://sourceforge.net/projects/aterisk-amr/files/ > > When I tried to call or start asterisk, I found "Segmentation Fault". Without trying to be pedantic, but "1.8.0-rc2" Ever considered upgrading? To 1.8.13.0 or so.. hans -- _ -- 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] port 5060 is blocked by ISP
On Sun, 2012-07-01 at 13:04 +0530, alok srivastava wrote: > dear > i have configured properly asterisk. At the one end i am using x-lite > soft ph and another end twinkle. call is going properly from both end > but after picking the phone not able to listen other one. > when i checked the port 5060 on the asterisk server it is always > showing closed while i have flushed all the rules from iptables > (iptables -F) > > PORT STATE SERVICE VERSION > 5060/tcp closed sip > > telnet localhost 5060 (could not connect) > > regards > alok Hi Alok, telnet is a very crude tool to test with. Try hping or nmap instead. Hans -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] fritzbox
Hi, Couple of moments ago my asteriskbox with a bri-card went down. (burn-out) I've heard that it seems to be possible to use an fritz!box as an isdn-gateway (isdn <--> sip) Anyone around who has good/bad experiences with those AVM-boxes? (yeah, i know it is tech overkill, but i'll get an dualband wifi router, that is Ipv6-ready with it) Hans. -- _ -- 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] looking for "solid state" like PC suitable for Asterisk
On Thu, 2012-05-10 at 07:40 -0500, Tim Nelson wrote: > - Original Message - > > On Thursday 10 May 2012, Bart Coninckx wrote: > > > I'm looking for a smaller, > > > appliance-type like PC, preferably solid state and fanless PC. > > > Since it's only going to run Asterisk for a couple of extensions I > > > don't > > > think CPU and RAM need to be maxed out. > > > > > > Does anyone have inspiration/experience for/about such a model? > > > > Raspberry Pi would be the obvious choice, surely? > > > > The hype around the Raspberry Pi is enormous. I would not consider it a real > option for production voice until it's had a chance to mature and be > available for some time to iron out the bugs, both hardware and software > related. > > My $0.02 USD. > Another couple of cents: the "pi" comes only with arm-cpu and limited amount mem - no upgrade possible. Might be an issue for asterisk... have a look at: http://www.fit-pc.info/ As long as you don't need to plug in a pci-board it is nice small and uses hardly any amps. hw -- _ -- 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] using Wifi smartphones as SIP clients
On Mon, 2012-05-07 at 19:03 +0100, Roger Burton West wrote: > On Mon, May 07, 2012 at 12:03:17PM +0200, Bart Coninckx wrote: > >What about phones like the Unidata WPU-7800 ( > >http://www.udcsystems.com/product/wpu7800.php) ? Does anyone have > >experience with those? Would these also suffer from connection > >losses? > > I've been using a UTStarcom GF-210 for the last year and more as my > personal phone - dual-mode 2G GSM and SIP/802.11. Sound quality on SIP > is slightly better than 2G, getting it to talk to Asterisk is no problem > at all, but certainly if you're moving from one wifi device to another > you will get dropped calls. If that's your use case, it's going to be > that way whatever hardware you use - I haven't seen any implementations > of 802.11F or 802.11r in the field. > Hope that these are better that the utstar F1000: Keep on re-chargibg as battery is empty in no-time, and security is lousy; just wep, no wpa. hw -- _ -- 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] Commercial SSL certs on Asterisk 1.8.10.0 with Polycom phones for encrypted calls using TLS and SRTP?
On Thu, 2012-03-08 at 16:50 +, Gavin Henry wrote: > >> > >> Ah, this makes sense now. So as of today the status of TLS and SRTP in > >> anything > >> other than 1.4.X is unknown? > > > > > > Umm... no :-) > > OK, sorry :-) > > > Asterisk 1.4 did not have support for SRTP or SIP/TLS. Thus, neither of > > these were tested with Polycom phones the last time we did interop testing > > with those phones. > > Ah, I forgot when it was added. > afaicr, it was in 1.6.2 > > The status of SIP/TLS and SRTP support in the Asterisk releases that have > > them are not 'unknown'; they are there and expected to be working. I was > > just pointing out that Digium has not specifically tested Polycom phones for > > interop with these features, and certainly has not specifically tested usage > > of TLS certificates issued by any particular CA. > btw, "commercial" certs are not so special. Somewhere in the chain (root-ca), there is a self-signed cert. You can make such chain yourself, root-ca -> sub-ca -> sub-ca and finally a server+client cert. Or, you can get a free cert from cacert.org hw -- _ -- 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] SIP hardware phones
On Mon, 2012-02-13 at 09:32 +0100, Benny Amorsen wrote: > "Jason W. Parks" writes: > > > I can move my voice infrastructure to an IP-based one running 10Mbps, > > utilize existing wiring infrastructure, with the only cost outlay > > being low cost PoE managed switches (48 ports for about a grand), and > > it ends up a lot cheaper than upgrading the data network to support > > the phones. ...and I can still stay within standard. > > You can, but not all phones will link up at 10Mbps. > > > /Benny > > -- > _ Are you realy shure you want to do that? I mean _existing_ infra (with probably a number of other (non-voip) machines connected to it? Even on a 100Mbps network, if one of the machines on the same network is doing a rsync-job (no saturation), I notice a drop in voip-quality. Adding voip to existing infra might work, if your network is good enough, like Gb with enough unused bandwith and low latency. Or if you can tell complaining users, that it is a temporary solution. hw -- _ -- 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] Proposed changes to Asterisk release and support cycles
On Tue, 2012-01-31 at 15:52 -0500, John Knight wrote: > > I like the idea of LTR release more often that would have the > > feature patches baked in. Case in point the new conference app > > requires a jump to version 10 while the 1.8 conference app is quite > > useless but 1.8 is my LTR version so I am stuck without the > > conference app in my mainline systems for two years. > > Well said! This is also true of any type of long term supported > release whether if it's an operating system, application, etc. In the > "LTS" name, it conjurs up thoughts of Ubuntu, but comparisons to > RHEL/Fedora are far more appropriate I would think as Ubuntu focuses > nearly exclusively on new point releases while backporting new > features is what a company like Red Hat excels at and should be the > prime example of how to run dual software channels (enterprise release > in RHEL vs. hobby release in Fedora). > > > I know distros and applications are two fundamentally different > things, with entirely different goals and requirements, but I still > think Red Hat provides the best example because 1) they have turned it > into a science how smooth their development process goes in ratio to > satisfied customers and 2) it's the only other open source software > project I can think of that can accurately compare. In a past meeting > I had with Digium while working for another company, they too directly > drew a correlation between the new LTS idea and ubuntu lts/non-lts and > rhel/fedora. > > The conference app changes since 1.4 I haven't been thrilled with, but > in the whole time I've been supporting 1.8.x for my customers, I've > come up with a very stable solution building on it and I haven't had > any surprises come my way. > very well said indeed. Some (...) distro's think dat LTS implies a complete feature freeze. Others are more flexibel about it, that besides current versions of applications, they are willing to support both elder _and_ newer versions. (as example, i'm refering to the fact that hours after the anouncement, firefox10 became available for sles11) As said, re-written features like conference, are that important that one shouldn't have to wait years for the next LTS. So this overlap of multiple LTS-versions looks very much attractive Having said that, i do understand that multiple versions of features/applications puts an huge extra burden on the people who have to maintain both versions, as the original version (as the term LTS implies) should be maintained with all its limitations also. hw -- _ -- 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] best softphone for 2012?
On Fri, 2012-01-06 at 16:00 -0600, Tom Poe wrote: > Just installed asterisknow 1.6. I can access freepbx. I need to test > system on my LAN. Which softphone is best to use? I'm running ubuntu > on Dell optiplex G260 desktop at home. I'm hoping to setup basic IP PBX > for incoming/outgoing calls. No video. > Tom > Hi, Our requirements were different, so we came to three candidates; linphone, ekiga and jitsi Linhphone is easy to "pre-configure" from a script and the buttons are easier to use, but lacks the possility for an ldap-adres-book. With ekiga you have the adresbook, but you have to use the mouse everywhere (the return-button gives unexpected results) And with jitsi (java-based) you are independant of Qt/GTK. hw -- _ -- 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] A new hack?
On Mon, 2011-12-05 at 18:51 -0800, Steve Edwards wrote: > Your security needs depends on your environment. At this point in time, > all of the hosts I manage for my clients exist in very limited > environments and have very small attack surfaces. They are racked in > secure data centers. They only accept SIP from clients with static IP > addresses that we have an existing business relationship with. They only > accept SSH connections from me. They only accept HTTP connections from me > and my boss. That's about it. I don't see where F2B adds much value for > me. > > *) Lots of admins think they can't limit access to servers because they > have 'mobile' users. Your users probably don't need to access your servers > from every single place on the Internet. If your users don't come from > China, North Korea, Iran, etc, you can block entire regions with a few > rules and eliminate 80% of probes and attacks from reaching your servers > in the first place. Apologies in advance if you happen to live in some of > these regions -- feel free to `s/China, North Korea, Iran/United States, > Canada, England/g` > Perhaps an other suggestion. If they are "true road warriors", i presume they are capable of setting up an vpn to the company. In that case, only allow registrations/calls through the secured tunnel. Then it's not any concern to asterisk. And if they can breach your tunnel, you have something else to worry about. hw -- _ -- 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] Simple Generic IVR to get us up an running Quick
On Sat, 2011-12-03 at 15:36 -0500, Nick Khamis wrote: > Hello Everyone, > > Are there any descent generic IVR recordings, that we can > use to quickly get our PBX up and running? It will obviously > not include the company name. It's easy enough to make your own recordings. Word of caution though. It might be advisable to ask somebody outside the company to record the phrases, Wonder why? At home i did it my self, and i still hear people stating that they have been talking at me, totaly unaware that it was just the voicemail anouncements. Peope just hear a voice, but seldom listen. And not just 90-old aunts, But people from helpdesks and even CEO's. Sometimes i wonder, if i should ask/test the callers I.Q. , And adapt the IVR's accordingly >;=) hw -- _ -- 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] Skype For Asterisk (SFA): any replacement?
On Thu, 2011-12-01 at 14:02 +, A J Stiles wrote: > On Thursday 01 December 2011, gincantalupo wrote: > > Hi all, > > > > any idea about how to replace Skype For Asterisk? > > > > Thank You. > > > > Giorgio > > 1. Migrate your Skype users over to a better product which supports proper > open standards. perhaps you missed it, but the installed base of skype is unfortunately slightly (,,,) larger than the amount of peope that are using a decent product. Alas > 2. Write to your elected representatives asking that they order Skype to > release documentation on their protocols to allow third party > interoperability > (as is already required under EU law). 3. make it a offence to use any closed source products like skype. >;-) Huge fines, jail centences or worse. [How about an appendice to the Thora, Quran or Bible, even better, forbid it by the sharia] -- _ -- 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] how to find out one way latency
On Wed, 2011-11-30 at 20:03 -0500, Adam Moffett wrote: > > You can make a pretty good prediction with ping. > "sudo ping -f -i .02 -s 180 -Q 0xb8 [ip]" gives a tolerable simulation > of voip traffic. let it run for awhile, then press ctrl+c and see how > many packets were dropped and also check the mdev number. If mdev is > low and packet loss is almost nothing then you can expect decent voice > quality. It may not be a 100% perfect test, but I'll bet you a vast > majority of the time I can do that test and tell you whether it's going > to suck. > > latency by itself with low jitter and no packet loss just means delay. > It's a matter of opinion and circumstance how tolerable delay is, but I > think your 230ms ping is at the upper edge of what most people can live > with. Much more than that and you'll be tempted to say 'over' at the > end of sentence. > > -- Fully agree, Actually, you can do better than just a ping, but it takes some time, equipment and experience: What you can do, is adding an extra box inbetween your voip-client and voip-server, and introduce all kinds of "real-life" circumstances. I mean artificial delay, loss, resequencing, duplicating packages, reduced bandwith. We've done it some time ago as an "satelite simulator" You can build it aroud any *bsd/linux box with multiple nics. The basic idea's you can find at http://lartc.org/ If you combine it with the echo function from asterisk, you can decide for yourself what it acceptable and what not. For one of my projects i push the echo destination as the "default" sip connection to their soft phone, as i noticed that people at the other side of town regularly have a worse connection then people using umts or satelite. Main culprit (in my case) is ill-configured WIFI-setup. Latencies of over 10,000 ms and loss of 80% are daily events. And people complaining hw -- _ -- 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] how to find out one way latency
On Thu, 2011-12-01 at 04:52 +0530, NaJIm wrote: > Is there anything else that I should be concerned about, when looking > to signup for a SIP provider. ?? Latency is important, but packet loss also, likewise packet re-ordering. hw -- _ -- 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] State of Asterisk+Virtualization+Timing
On Wed, 2011-11-09 at 16:10 +0300, Anton Kvashenkin wrote: > Is anybody using pci-passthrough? > Yes, though quite a while ago. About three years ago, i used pci-passthrough to give a dom-U access to a localy mounted smartcard. But i have a vague feeling that you are up to something else... I know that forwarding has been done for ethernet and even VGA-cards, the mere idea of forwarding a analogue or PRI card is quite something else: Timing is here essential.. hw -- _ -- 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] State of Asterisk+Virtualization+Timing
On Mon, 2011-11-07 at 11:45 -0500, Nick Khamis wrote: > That sucks! What about KVM or XEN? > > Nick. No problems here with XEN. (Perhaps i should mention, that i use paravirtualsisation to get the best performance. Distro: mix of SLES11sp1 /open_11.4) hw -- _ -- 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] State of Asterisk+Virtualization+Timing
On Tue, 2011-11-01 at 12:08 -0500, Tim Nelson wrote: > Greetings- > > I'm about to dive into the process of virtualizing some of my Asterisk > (primarily 1.4.x) infrastructure. In the past, when looking at virt > solutions, the primary issue preventing me from moving was the lack of proper > timing. We do not need it for MeetMe but rather for IAX2 trunking. I'd like > to use either OpenVZ or KVM, but each seem to have independent "issues" that > need to be addressed: > > OpenVZ - Better resource usage, lower overhead. Primary issue is how to grant > access to host node timing source (physical device, or dahdi_dummy in > /dev/dahdi/) to the containerized Asterisk process. > > KVM - Higher overhead, easier installation, 'true virtualization'. Primary > issue is not timing per se, but KVM scheduling. Timing source, while present > from dahdi_dummy natively may still not get proper scheduling by KVM process. > This could also affect general call quality (even non IAX2 trunked voice), > DTMF, etc. > > I have to believe there are others running virtualized Asterisk installations > with some degree of success on OpenVZ or KVM. Care to share your thoughts? > You mist out one more mature virtualization technique: XEN Virtual machines can use both hardware- or paravirtualization. I have used both asterisk (1.4, 1.6.x and now 1.8) to separate machines where people should do their sip-registration (internet / intranet / pstn-gateway) and the actual dial-beast. Main advantage for virtualization is (besides easy scaling) that you can perform an upgrade in no-time (one VM-machine down, other up) Don't like it: back in seconds! Migration with an asterisk on real hardware takes much more resources. Both in iron and in time. hw -- _ -- 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] Asterisk Security: Allow only one phone per sip registration
On Fri, 2011-10-14 at 10:02 +0300, Muro, Sam wrote: > Hi there > > Consider this. You have three SIP extension 200, 201 and 202 and you have > configured your phones, say Polycom 331 to those accounts. 200 being one > very sensitive individual. > > Lets say, an insider, get a new phone or perhaps an xlite and configure it > with the same extension, 200. Asterisk will register it as 200 to the new > IP address. Now extension 202 call 200. The hacker answers it and pretend > is the same person. Do what he want to do and thats it. > > Question; > How can i stop this type of threat > > Regads > Peter > Perhaps use secrets? afaicr the secrets you have to provide for hardphone and softphone are readonly. If you avoid something like "secret" or "welcome" or the involved hostname, but instead use a 15 char long generated pwd, he'll have a long time trying all the possibilities And different pwds for each phone. hw -- _ -- 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] realtime goto/gotoif/dial
On Wed, 2011-09-14 at 09:00 +0100, Ishfaq Malik wrote: > On Wed, 2011-09-14 at 00:44 +0200, Hans Witvliet wrote: > > Hi all, > > > > I presume i made a silly mistake while filling a database > > > > But while googling on the results, i came across a lot of messages about > > the layout of app_data in case of goto and dial statements. > > (mostly about using the old "|" seperator instead of the "," separator. > > > > So i was wondering, is this issue been solved? (I presume so, but can > > not find any confirmation about it) > > > > Hans > > > > -- > Hi > > It's pipe in 1.4 and comma in 1.6 and 1.8 > Thanks for confirming it. As i'm using 1.8.5 i'm now certain that it is a case of "PEBKAC" The symtoms are the same, but obviously a different cause. ;-) Hans -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] realtime goto/gotoif/dial
Hi all, I presume i made a silly mistake while filling a database But while googling on the results, i came across a lot of messages about the layout of app_data in case of goto and dial statements. (mostly about using the old "|" seperator instead of the "," separator. So i was wondering, is this issue been solved? (I presume so, but can not find any confirmation about it) Hans -- _ -- 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] Distributed device state / presence info??
On Fri, 2011-09-02 at 11:33 -0500, Kevin P. Fleming wrote: > On 09/01/2011 04:39 PM, Hans Witvliet wrote: > > >> From the asterisk-bible and the wiki's i learned that it is possible to > > let asterisk do some of the presense-info by means of the jabber.conf > > file and a seperate xmpp-server. > > You are misunderstanding a bit; Asterisk can use an XMPP server and > PubSub to *distribute* presence information among a cluster of Asterisk > servers. This information is not intended to be directly sent to XMPP > clients. > > > What i assume (please correct me if i am wrong) is that when a client > > registers/deregisters, asterisk will update the presence info towards > > the XMPP-server. Correct? > > Yes, in order to let other Asterisk servers in the cluster know about it. > > > But otoh, what people would like to see is who is "on line". > > And not only on the asterisk-server that they are connected to, but also > > from other possible asterisk servers. > > And furthermore, each registered user might want to set their > > presencse-status to either free/busy/away/what-ever. > > Asterisk does not support 'user' presence; it supports device and > extension presence. In some applications these can be used > interchangeably, but in others they don't match up very well. > Ok, so that should mean that the "presence-status" is controlled by the fect wether a sip-user is registered or not? If so, i'm still making a mistake somewhere: I tried to simplify my configuration: Just one asterisk machine, And two users (my and myself) with two linphones, one from a XP-machine and the other from a SuSE-machine. They can both register and call each other, and in linphone presence is enabled. However if i manually add on each side the corresponding info (name + sip-address) in the linphone app, they always remain grey / "away" Even while in the middle of an connection to each other. (Probably not relevant, but i'm using 1.8.3) hw -- _ -- 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] Anyone using Asterisk on VirtualBox ?
On Thu, 2011-09-01 at 21:32 +0530, RSCL Mumbai wrote: > > > My main interest of being on Virtual platform is portability / Backup. > In case of any h/w issues, or crashes, simply copy the VM on to > another box and you are up in minutes. > > > Sanjay > -- Doing that right now, although in my case i use XEN. Besides being hw independant, it is easier to play with a different version for a while (1.4 / 1.6.0 / 1.6.1 / 1.6.2 / 1.8.0) and being able to switch back in minutes. hw -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Distributed device state / presence info??
Hi all, Last couple of days i've arguing with my colleges about presence-info. >From the asterisk-bible and the wiki's i learned that it is possible to let asterisk do some of the presense-info by means of the jabber.conf file and a seperate xmpp-server. On the other hand, most soft-phones are capable of "doing something" with presence, allthough most of them use SIMPLE-protocol, instead of XMPP. So if when should one use the presence info from asterisk and when use the presence info from the softphones. It looks to me like doing the same job twice. What i assume (please correct me if i am wrong) is that when a client registers/deregisters, asterisk will update the presence info towards the XMPP-server. Correct? But otoh, what people would like to see is who is "on line". And not only on the asterisk-server that they are connected to, but also from other possible asterisk servers. And furthermore, each registered user might want to set their presencse-status to either free/busy/away/what-ever. So if the changing/reading is to be done on a softphone, what is the point of having asterisk doing someting with the device-status??? While writing, i've got a distinct feeling i'm understanding less by the minute ;( Anyway, what i'm building is a central server and a number of asterisk-boxes that act as proxy/six-iax-converter. All of the registered users should be able to see the presence of all the users on either proxy. Hans -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] phone + video
Hi all, I know that a lot of people have negative experiences with grandstream-2000, but personally. i'd only the repace one poweradapter after three years... So, can anybody give some comment on one of their recent models, the GXV-3175 (the one with the 7" display) I'm looking for a phone with video capabilities, as i don;t want to limit my self to testing with softphone.. HtH, Hans -- _ -- 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] Looking for ideas for nice **Asterisk** home phone system
On Fri, 2011-08-26 at 19:03 -0400, Eric Wieling wrote: > >-Original Message- > >From: asterisk-users-boun...@lists.digium.com > >[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Hans Witvliet > >Sent: Friday, August 26, 2011 6:09 PM > >To: Asterisk Users Mailing List - Non-Commercial Discussion > >Subject: Re: [asterisk-users] Looking for ideas for nice **Asterisk** home > >phone system > > > >On Fri, 2011-08-26 at 07:41 -0700, Steve Edwards wrote: > >> On Fri, 26 Aug 2011, linux guy wrote: > >> > > > How much power does the home asterisk box need ? > > > >I use a small box (like those hp thin clients) But these are a bit stronger > >aluminium housing, instead of plastic, and better foor cooling. > > > >Power consumption: 8 Watt under full load > >CPU: Model: 6.28.2 "Intel(R) Atom(TM) CPU Z530 @ 1.60GHz" > >Memory Size: 1 GB > >Disk /dev/sda: 64.0 GB, 64023257088 bytes This model has just one ethernet > >port, others have two > >Size: 10x10 cm > > Is this a custom build box or does a company sell them preassembled?We > are always on the lookout for potential boxes we can use for small > installations. > It is pre-assembled, You can opt for either no internal disk small (8GB) sdd, larger (64GB) sdd or ordinary disk, And either no, MS, or ubuntu pre-installed. Also with/without wifi antenna. hw -- _ -- 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] Looking for ideas for nice **Asterisk** home phone system
On Fri, 2011-08-26 at 07:41 -0700, Steve Edwards wrote: > On Fri, 26 Aug 2011, linux guy wrote: > > > How much power does the home asterisk box need ? I use a small box (like those hp thin clients) But these are a bit stronger aluminium housing, instead of plastic, and better foor cooling. Power consumption: 8 Watt under full load CPU: Model: 6.28.2 "Intel(R) Atom(TM) CPU Z530 @ 1.60GHz" Memory Size: 1 GB Disk /dev/sda: 64.0 GB, 64023257088 bytes This model has just one ethernet port, others have two Size: 10x10 cm hw -- _ -- 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] Lightning and thunder
On Wed, 2011-07-27 at 09:44 -0400, Claude Hayn wrote: > We are frequently losing power during lightning storms. (Yes we have > UPS, but often by the time power comes back up the UPS has run out of > juice) > > > Does anyone know of a solution for this issue? Having to get up in > the late night to manually reboot the Asterisk box is getting old! > Perhaps an other suggestion... Re-install asterisk on a other piece of hardware. There are small boxes that consume less than 5 Watt. If you put that on your UPS, it will last longer. Other one, ever thought of an alternative power source? Either solar of conventional? hw -- _ -- 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] Strange network issue
On Fri, 2011-07-22 at 10:58 -0700, Dave Platt wrote: > > They've got a bunch of Grandstreams that seem to be rock solid... until > > 7:00pm. At 7:00, some of the phones become unavailable, and stay down. > > Call > > quality is solid almost all the time. But right at 7:00, things go bad. > > Only > > some of the phone lines go down and they stay down until the phone is > > rebooted. > > > > I'm not even sure what to look for when I go to the site. Any ideas? > > I'd look to see if there are any electrical circuits (lights, > fans, etc.) which are on a timer of some sort, and are automatically > powered off at 7 PM. > > If somebody mistakenly plugged a piece of network kit into such a > circuit, it would lose power at that time, and your network might > end up being partitioned, or routing (switch or IP-level) might > change abruptly. > Hi, Even if there is no equipment you own controlled by a timer, you still can suffer from it. Some power companies have different rates for power you use during daytime or at night. So even if _you_ don't have equipment on a timer, your neighbours might have. Something like electrical boilers or so, or other "heavy equipment". Switching them on/off can cause huge spikes on the electrical wires. A couple of neigbours at work have their own micro-power-generators. About one in ten times, when they start delevering power to the grid, all of our test-systems go down. Only the systems behind the re-generated UPS (that removes spikes from the powerlines) are protected against them. So nasty litte spikes are harder to detect/tracedown than a full blackout. hw -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] dialplan: all extern, except
Hi all, Perhaps a no-brainer, but i think i am making my dialplan on my proxy too complicated. Normally, what you find in the examples is that you have to dial a specific number, other "9" or "0" for an external line. What i want to do is this: If you pre-pend a number with something like "*" then you can dial local defined numbers, otherwise everything goes through my iax-trunk-line. So for instance: "*#1" gives you a local welcome text "*#2" gives you the local echo function while "#1" gives you a remote welcome text "#2" gives you the remote echo function And ordinary numbers or sip's go straight extern: "0174539053" or "j.witvl...@a-domani.nl" should go to my main asterisk-server. Currently i'm doing it pattern-matching all numbers, and each upper +lower case character, but i wonder if it can be done simpler. Hans -- _ -- 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] realm question: solved
On Tue, 2011-07-05 at 12:33 +0500, Faisal Hanif wrote: > The problem you are reporting is not related to realm but can be context or > domain. > Tnx, It was indeed a domain issue. In some cases static definitions in /etc/hosts is not a good replacement for DNS... hw -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] realm question
Hi all, Trying to find where i got wrong in my config Is the "realm" parameter in sip.conf only used for possible autentication? The thing is, i got my box more-or-less working as i wanted, but i can only reach internal functions (like echo-test and so on) and other sip-clients if i dial "1234@fqdn", while i was expected to be able to just dial "1234" I presume i have either a mismatch between how the softphones register, and my asterisk conf. Kind regards, Hans -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Solved, was: Permanent restart after upgrade
On Fri, 2011-06-10 at 05:52 -0400, Steve Totaro wrote: > On Fri, Jun 10, 2011 at 1:48 AM, Hans Witvliet wrote: > > On Thu, 2011-06-09 at 16:32 -0700, Steve Edwards wrote: > >> On Thu, 9 Jun 2011, Hans Witvliet wrote: > >> > >> > I went originally from a almost working machine running: > >> > asterisk180-1.8.3.2-87.1 > >> > > >> > To a machine that continuously restarts asterisk (+core dumps) running: > >> > asterisk180-1.8.3-85.2 > >> > >> Any chance you have a mix of Asterisk and module versions? Was > >> Zaptel/Dahdi compiled with the proper set of headers for your kernel? > >> > >> Can you start Asterisk from the command line instead of the usual startup > >> script? What do the first couple of errors look like? Capturing the output > >> via the 'script' command will help. > >> > >> For example*, > >> > >> script foo > >> sudo /usr/sbin/asterisk -C /etc/asterisk/asterisk.conf\ > >> -c -d -d -d -f -g -n -p -q -v -v -v > >> exit > >> > >> Can you turn off auto module loading and start with no modules? > >> > >> *) I'm a 1.2 Luddite, so the command line arguments may have changed... > >> > > No dahdi/zaptel involved. > > I'll be off to work in a while, report back later. > > > > > > hw > > > > It amazes me when people run into a problem but refuse to post logs or > verbose when you start Asterisk. Nothing meaninful. > > I would wager a gentleman's bet that I can have your system working > just fine in a half hour or less (unless your bandwidth sucks). > > If I do it, then you have to post to the list and you owe me a favor, > plus, in the future you have to help someone else. > > If I don't, I have to post my failure to the list and I owe you a favor. > > I have spare cycles, just let me know. > > Thanks, > Steve Totaro Hi Steve, thanks for your time and consideration. Hadn't a chance to report back, as i just returned from work ;-( I think i found the reason behind it; a missing file from the update. As the machine involved is not connected to Internet, each and every file has to be put on a portable medium, checked, and only then i'm allowed to put it on our corporate lan. It turned out, that not all required files were transferred on the usb-disk, or removed by someone. Anyway it, i copied the missing file ( ../repo/network:/telephony:/asterisk/SLE_11_SP1/x86_64/asterisk180-1.8.4.2-90.1.x86_64.rpm ) manually, updated it again and: voila So regarding not posting config/log/trace/core's.. Initially i just put the symptom's on the list. Next step would have been any specific log's or config files. As you know, any most of the logfiles for *, can be rather long, same for logfiles. If you are still interested in it (educational purposes) i can still get the /var/log/asterisk/messages file, and put the relevant section on the list Bottomline is however, when doing an upgrade, and only some of the asterisks RPM's are processed, you get funny results And what you see on the asterisk console/logfile does not indicate directly what went wrong. If i run into a situation, it is mostly that i can not get something working in the first place, or i made an incorrect change. In those cases the remedy is obvious. In this case however, the cause is not PEBKAC or asterisk issue, but something with the prebuild packages, i'll guess a missing dependancy, which allowed some asterisk files to be updated, while not all were present, resulting into an unstable result. So, i'll try to find the person responsible for packaging, and try to convince that he ahs some work to do. Kind regards, Hans -- _ -- 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] Permanent restart after upgrade
On Thu, 2011-06-09 at 16:32 -0700, Steve Edwards wrote: > On Thu, 9 Jun 2011, Hans Witvliet wrote: > > > I went originally from a almost working machine running: > > asterisk180-1.8.3.2-87.1 > > > > To a machine that continuously restarts asterisk (+core dumps) running: > > asterisk180-1.8.3-85.2 > > Any chance you have a mix of Asterisk and module versions? Was > Zaptel/Dahdi compiled with the proper set of headers for your kernel? > > Can you start Asterisk from the command line instead of the usual startup > script? What do the first couple of errors look like? Capturing the output > via the 'script' command will help. > > For example*, > > script foo > sudo /usr/sbin/asterisk -C /etc/asterisk/asterisk.conf\ > -c -d -d -d -f -g -n -p -q -v -v -v > exit > > Can you turn off auto module loading and start with no modules? > > *) I'm a 1.2 Luddite, so the command line arguments may have changed... > No dahdi/zaptel involved. I'll be off to work in a while, report back later. hw -- _ -- 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] Permanent restart after upgrade
On Fri, 2011-06-10 at 07:21 +0800, Larry Moore wrote: > On 10/06/2011 5:32 AM, Hans Witvliet wrote: > > Hi all, > > > > I got three asterisk-machines, two of them acting as proxies. > > On one machine (sles11sp1) i got iritating messages about not bing able > > to find codec's and other stuff, so i thought it might be time for an > > update: Stupid! > > > > I went originally from a almost working machine running: > > asterisk180-alsa-1.8.3.2-87.1 > > asterisk180-dahdi-1.8.3.2-87.1 > > asterisk180-1.8.3.2-87.1 > > asterisk180-odbc-1.8.3.2-87.1 > > > > To a machine that continuously restarts asterisk (+core dumps) running: > > asterisk180-alsa-1.8.4.2-90.1 > > asterisk180-dahdi-1.8.4.2-90.1 > > asterisk180-1.8.3-85.2 > > asterisk180-odbc-1.8.4.2-90.1 > > > > > > Jun 9 16:35:44 kc3004 kernel: [ 713.970342] asterisk[5122]: > > segfault at 7fedcb450716 ip 7fedeff89c8a sp 7fff4b4d5d38 error 7 in > > libpthread-2.11.1.so[7fedeff7f000+17000] > > > > No change in config or other settings. > > > > Any suggestions are very much welcome... > > (only thing that puzzles me is that the repo contains 1.8.3 for main > > asterisk, and 1.8.4 for the rest) > > > > > > One thought comes to mind, I'm not sure if you are using the same > computer with the same IP address or if you have set up a different > computer with the new installation which probably has a different IP > address, if the latter I would suggest you check your configuration file > for bindings to specific IP addresses and make sure they match the new > machine. > > Larry. Hi Larry, No, identical machine, just did a "zypper up -y" Because of a kernel patch (main reason for updating), i had to reboot. Just a sip and IAX machine, no harware involved. hw -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Permanent restart after upgrade
Hi all, I got three asterisk-machines, two of them acting as proxies. On one machine (sles11sp1) i got iritating messages about not bing able to find codec's and other stuff, so i thought it might be time for an update: Stupid! I went originally from a almost working machine running: asterisk180-alsa-1.8.3.2-87.1 asterisk180-dahdi-1.8.3.2-87.1 asterisk180-1.8.3.2-87.1 asterisk180-odbc-1.8.3.2-87.1 To a machine that continuously restarts asterisk (+core dumps) running: asterisk180-alsa-1.8.4.2-90.1 asterisk180-dahdi-1.8.4.2-90.1 asterisk180-1.8.3-85.2 asterisk180-odbc-1.8.4.2-90.1 Jun 9 16:35:44 kc3004 kernel: [ 713.970342] asterisk[5122]: segfault at 7fedcb450716 ip 7fedeff89c8a sp 7fff4b4d5d38 error 7 in libpthread-2.11.1.so[7fedeff7f000+17000] No change in config or other settings. Any suggestions are very much welcome... (only thing that puzzles me is that the repo contains 1.8.3 for main asterisk, and 1.8.4 for the rest) Hans -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] half sip registration at 1.8.3
Hi all,
I've got something strange, that got me searching for quite awhile.
Configuration as followed:
Linphone on a laptop, that is connected via openvpn to a proxy.
That proxy is connected with iax to another asterisk.
On the second one i have several hard and softphones.
Behaviour at first glance:
>From the softphone i can allways set up a connection,
But the otherway round fails 9 out-of 10 times.
However, if i stop-and-start linphone, the connections is allways
succesful.
First conclusion was, that if i got a diffrent (dynamic) ip-adress from
openvpn, i got to restart linphone, to force a re-registration.
Sounds reasonable, but why is linphone able to place calls, but not able
to accept them? (guests are off) I mean, if the phone is registered with
different values, also the outgoing call should fail. Not?
To avoid this behaviour, should i drastically drop the registration
duration at the softphone side? I still uses the default one (3600s).
Or should i tweak the min/max/default expiry-timers at asterisk?
Currently they are (also the default) 60/3600/120 seconds.
Hans
ps these are the lines from the console:
-- Executing [0277611@from_iax:1] noop("IAX2/kc3004-6511",
",0277611")
-- Executing [0277611@from_iax:2] answer("IAX2/kc3004-6511", "")
-- Executing [0277611@from_iax:3] dial("IAX2/kc3004-6511",
"SIP/0277611 ") [Jun 6 19:03:32] WARNING[23015]: app_dial.c:2039
dial_exec_full: Unable to create channel of type 'SIP' (cause 20 -
Unknown)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [0277611@from_iax:4] hangup("IAX2/kc3004-6511", "")
== Spawn extension (from_iax, 0277611, 4) exited non-zero on
'IAX2/kc3004-6511'
-- Hungup 'IAX2/kc3004-6511'
corresponding lines from the ARA-dialplan:
| 118 | from_iax | 0212676 |1 | noop | ${CALLERID},${EXTEN} |
| 119 | from_iax | 0212676 |2 | answer | |
| 120 | from_iax | 0212676 |3 | dial | SIP/0212676 |
| 121 | from_iax | 0212676 |4 | hangup | |
--
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Re: [asterisk-users] benefits of asterisk 1.8
On Fri, 2011-06-03 at 09:07 +0100, Ishfaq Malik wrote: > Are you suggesting that there are no bugs in 1.4 or 1.6? I presume that you are aware of the fact that it is impossible to prove the absence of "bugs" in any piece of software You might not have detected them yet. Furthermore behaviour that might have been coded on purpose, can be considered "eroneously" some time later. > Currently there seems to be a fear of 1.8. We're about to put it into > production and yes, we've had issues with it, mostly due to the fact we > use RealTime, but before you change anything it is always advisable to > test the hell out of it. > > To anyone who is thinking of moving to 1.8 the question is not, 'is it > stable?'. The question is, 'have I comprehensively tested it to show > that it is suitable for my needs?' If you put it into production, test at least the functions that you are going to use. There might (and probably will) problems in the code, but as long as it does not bother you, so what? And don't stop testing after you put it into production: have a shadow system (with representative configuration). According to Murphy, side-effects will probably rise to the survice after going into production End-users will come up with situations you never enticipated in your worst nightmares. -- _ -- 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] asterisk fails when DNS or internet fails
On Tue, 2011-05-31 at 10:29 -0400, Eric Wieling wrote: > As far as I can tell it is trying to do a reverse lookup on the IPs > configured on the system. With the internet down, does the command "host > 10.10.10.1" (or whatever IPs you have on the system) take a while to come > back? Unless you can do a reverse lookup of all the IPs on the system don't > expect Asterisk to be able to. If your /etc/hosts is set up correct, you > should be able to look up any IP configured on any interface on the system > without delay. > > I'm sure there are other places Asterisk tries to do DNS lookups, but the > above info has solved this issue for me in the past. > I'm not sure if that's all is true. Sure, if you add a line in /etc/hosts, that works for most applications, as not all commands follow /etc/resolv.conf i just tried, adding a line to /etc/hosts. ping hostname works, but host hostname fails, just as host ip-address. So even when you only put ip-addresses (brrr) into your config files, the reversed-lookup will still spoil the party. -- _ -- 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] asterisk fails when DNS or internet fails
On Mon, 2011-05-30 at 23:15 -0400, Jeff LaCoursiere wrote: > > On Mon, 30 May 2011, Sherwood McGowan wrote: > > > True, but with all due respect, if the cache's TTL expires and the OP's > > PBX cannot reach an external DNS server, they have bigger problems ;-) > > > > Slainte all! > > The Mick > > > > I couldn't disagree more. In fact I think this problem is more serious > than it is getting credit for, when asterisk is in use in places where > Internet connectivity is far from stable. I have several hotels that have > gone without Internet connectivity for days, and somewhere between one and > three days down they can only spottily call within the system, and can't > make outbound calls on their voice T1. Its certainly true that they were > suffering without Internet access, but it is very hard to explain to the > owners why they can't use their phones. In fact the symptoms are very > strange - inbound calls on the T1 get the auto-attendant, but internal > transfers fail. No one can call outbound, and only *sometimes* do > internal extension to extension calls fail. > > I still scratch my head about what exactly asterisk is trying to lookup > that keeps it from being able to place internal SIP calls from extension > to extension, and sadly the few times this has occurred I wasn't around to > debug. > > Hasn't anyone managed to solve this with something better than a caching > DNS server, which seems to only last a short while? What exactly is going > on that is failing? > What kind of info is it about? If it is the hostname of _local_ machines/clients, you should be authoritive. That should keep asterisk happy. If it is about remote nodes, well if your isp-connection is lost, you can not contact them anyway ;-( So run locally your bind-server, authoritive for your own addresses, and caching for external ones. hw -- _ -- 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] Why PRI not BRI ?
On Mon, 2011-05-30 at 13:57 +0530, virendra bhati wrote: > Thanks a lot all, > Now my view is clear ... > > On Sun, May 29, 2011 at 3:15 PM, Gordon Henderson +aster...@drogon.net> wrote: > On Sun, 29 May 2011, virendra bhati wrote: > > Hi List, > > I have stupid question but I want to know it. Why we > use the PRI insted of > BRI ? Just for the sake of number of lines or any > thing else ? > > > It probably depends on your country. > > In mainland europe (or maybe just Germany), ISDN2e (BRI) is > very popular - not uncommon in home installations too. > > In the UK, it's almost the standard in small businesses - the > migration path seems to be from a single line to 3 lines > sharing the same number to ISDN2e... > > There was a push in the UK to support BRI in the home (~10 > years back, under the name Home Highway), but it came at a > time when ADSL was almost upon us, and BT in their infinite > wisdom removed a lot of the ISDN features that make it > actually useful... > > I don't think BRI ever caught on in the US - It was analogue > or PRI (or channelised/fractional T1 or whatever it's called) > Probably made it much easier for the telcos to support (and > afford) Only reason for using bri instead of pri in the number of voice chanels and costs. It took ages before telco's realised that with fractured-E1 they could save a lot of costs (telco/customers) while offering a cheap upgrade path. At that time that ISDN was introduced, the costs in installing a pri-interface in the local-exchange was identical to installing a bri-interface. Only reason nowadays for using bri instead of pots, is that you get the incoming speech channel already digitialised. > And why SIP is used for making calls rather then IAX? > Even we know IAX takes > 1 channel for making calls? > > > SIP is an open standard that's been around since the late > 90's. IAX, which is also open and free was only just accepted > as a standard last year, but even so, there's inertia. Very > few phone manufacturers are using it - why should they, when > they've been using SIP for years, and the same PBX that works > with IAX also works with SIP... (And does any other PBX > support IAX yet?) > Freepbx is the only other afaicr. Only a limited number of clients. hw -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk18 realtime/mysql - extconfig.conf - res-mysql.conf
Right, Here is its (res_mysql.conf) content: [general] dbhost = 127.0.0.1 dbname = asterisk dbuser = voipadmin dbpass = secret dbport = 3306 dbsock = /tmp/mysql.sock dbcharset = latin1 requirements=warn ; or createclose or createchar The file didn't exists on the systemm so i had to create it. I kept it identical to res_config_mysql.conf Hans On Mon, 2011-05-23 at 14:03 +0100, Andrew Thomas wrote: > It's the contents of res_mysql.conf that's really needed - as > extconfig.conf has to match what's in it. > > -Original Message- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Hans > Witvliet > Sent: 23 May 2011 13:42 > To: asterisk-users@lists.digium.com > Subject: [asterisk-users] [Fwd: FW: extconfig.conf] > > Hi Andrew, > > OK, (the simple fact that those machines are not connected to internet > makes that i have to go to those machines and copy them on a usb-stick, > so it causes some delay each time...) > > > > Forwarded Message > Sorry - I meant extconfig.conf - not cdr_mysql.conf (my mistake). > > I use (and done for a long time) mySQL for realtime storage - and it's > never let me down (touch wood). > Forwarded Message > > > ; > > ; Static and realtime external configuration ; engine configuration ; > ; Please read doc/extconfig.txt for basic table ; formatting > information. > > ; > > [settings] > > ; > > ; Static configuration files: > > ; > > ; file.conf => driver,database[,table[,priority]] > > ; > > ; maps a particular configuration file to the given ; database driver, > database and table (or uses the ; name of the file as the table if not > specified) ; ;uncomment to load queues.conf via the odbc engine. > > ; > > ;queues.conf => odbc,asterisk,ast_config ;extensions.conf => > sqlite,asterisk,ast_config ; ; The following files CANNOT be loaded from > Realtime storage: > > ; asterisk.conf > > ; extconfig.conf (this file) > > ; logger.conf > > ; > > ; Additionally, the following files cannot be loaded from ; Realtime > storage unless the storage driver is loaded ; early using 'preload' > statements in modules.conf: > > ; manager.conf > > ; cdr.conf > > ; rtp.conf > > ; > > ; > > ; Realtime configuration engine > > ; > > ; maps a particular family of realtime ; configuration to a given > > database driver, ; database and table (or uses the name of ; the > > family if the table is not specified ; ;example => > > odbc,asterisk,alttable,1 ;example => mysql,asterisk,alttable,2 > > ;example2 => ldap,"dc=oxymium,dc=net",example2 ; ; Additionally, > priorities are now supported for use as failover methods ; for > retrieving realtime data. If one connection fails to retrieve any ; > information, the next sequential priority will be tried next. This ; > especially works well with ODBC connections, since res_odbc now caches ; > when connection failures occur and prevents immediately retrying those ; > connections until after a specified timeout. Note: priorities must ; > start at 1 and be sequential (i.e. if you have only priorities 1, 2, ; > and 4, then 4 will be ignored, because there is no 3). > > ; > > ; "odbc" is shown in the examples below, but is not the only valid > realtime ; engine. There is: > > ;odbc ... res_config_odbc > > ;sqlite ... res_config_sqlite > > ;pgsql ... res_config_pgsql > > ;curl ... res_config_curl > > ;ldap ... res_config_ldap > > ; > > ;iaxusers => odbc,asterisk > > ;iaxpeers => odbc,asterisk > > ;sipusers => odbc,asterisk > > ;sipusers => mysql,asterisk,sip_devices ;sippeers => > > mysql,asterisk,sip_devices ;sipusers => mysql,general,sip_devices > > ;sippeers => mysql,general,sip_devices sipusers => > > mysql,default,sip_devices sippeers => mysql,default,sip_devices > > ;sippeers => odbc,asterisk ;sipregs => odbc,asterisk ;voicemail => > > odbc,asterisk ;extensions => odbc,asterisk ;meetme => mysql,general > > ;queues => odbc,asterisk ;queue_members => odbc,asterisk ;musiconhold > > => mysql,general ;queue_log => mysql,general ; ; ; While most dynamic > > realtime engines are automatically used when defined in ; this file, > 'extensions', distinctively, is not. To activate dynamic realtime ; > extensions, you must turn them on in each respective context within ; > extensions.conf with a switch statement. The syntax is: > > ; switch => R
[asterisk-users] [Fwd: FW: extconfig.conf]
Hi Andrew, OK, (the simple fact that those machines are not connected to internet makes that i have to go to those machines and copy them on a usb-stick, so it causes some delay each time...) Forwarded Message Sorry - I meant extconfig.conf - not cdr_mysql.conf (my mistake). I use (and done for a long time) mySQL for realtime storage - and it's never let me down (touch wood). Forwarded Message > ; > ; Static and realtime external configuration ; engine configuration ; ; > Please read doc/extconfig.txt for basic table ; formatting information. > ; > [settings] > ; > ; Static configuration files: > ; > ; file.conf => driver,database[,table[,priority]] > ; > ; maps a particular configuration file to the given ; database driver, > database and table (or uses the ; name of the file as the table if not > specified) ; ;uncomment to load queues.conf via the odbc engine. > ; > ;queues.conf => odbc,asterisk,ast_config ;extensions.conf => > sqlite,asterisk,ast_config ; ; The following files CANNOT be loaded from > Realtime storage: > ; asterisk.conf > ; extconfig.conf (this file) > ; logger.conf > ; > ; Additionally, the following files cannot be loaded from ; Realtime storage > unless the storage driver is loaded ; early using 'preload' statements in > modules.conf: > ; manager.conf > ; cdr.conf > ; rtp.conf > ; > ; > ; Realtime configuration engine > ; > ; maps a particular family of realtime > ; configuration to a given database driver, ; database and table (or uses the > name of ; the family if the table is not specified ; ;example => > odbc,asterisk,alttable,1 ;example => mysql,asterisk,alttable,2 > ;example2 => ldap,"dc=oxymium,dc=net",example2 ; ; Additionally, priorities > are now supported for use as failover methods ; for retrieving realtime data. > If one connection fails to retrieve any ; information, the next sequential > priority will be tried next. This ; especially works well with ODBC > connections, since res_odbc now caches ; when connection failures occur and > prevents immediately retrying those ; connections until after a specified > timeout. Note: priorities must ; start at 1 and be sequential (i.e. if you > have only priorities 1, 2, ; and 4, then 4 will be ignored, because there is > no 3). > ; > ; "odbc" is shown in the examples below, but is not the only valid realtime ; > engine. There is: > ;odbc ... res_config_odbc > ;sqlite ... res_config_sqlite > ;pgsql ... res_config_pgsql > ;curl ... res_config_curl > ;ldap ... res_config_ldap > ; > ;iaxusers => odbc,asterisk > ;iaxpeers => odbc,asterisk > ;sipusers => odbc,asterisk > ;sipusers => mysql,asterisk,sip_devices > ;sippeers => mysql,asterisk,sip_devices > ;sipusers => mysql,general,sip_devices > ;sippeers => mysql,general,sip_devices > sipusers => mysql,default,sip_devices > sippeers => mysql,default,sip_devices > ;sippeers => odbc,asterisk > ;sipregs => odbc,asterisk > ;voicemail => odbc,asterisk > ;extensions => odbc,asterisk > ;meetme => mysql,general > ;queues => odbc,asterisk > ;queue_members => odbc,asterisk > ;musiconhold => mysql,general > ;queue_log => mysql,general > ; > ; > ; While most dynamic realtime engines are automatically used when defined in > ; this file, 'extensions', distinctively, is not. To activate dynamic > realtime ; extensions, you must turn them on in each respective context > within ; extensions.conf with a switch statement. The syntax is: > ; switch => Realtime/[[db_context@]tablename]/ > ; The only option available currently is the 'p' option, which disallows ; > extension pattern queries to the database. If you have no patterns defined ; > in a particular context, this will save quite a bit of CPU time. However, ; > note that using dynamic realtime extensions is not recommended anymore as a ; > best practice; instead, you should consider writing a static dialplan with ; > proper data abstraction via a tool like func_odbc. > so i have: sipusers => engine, context, table_name sippeers => engine, context, table_name (tried both 'default' as 'general' as context value. I previously tried: sipusers => engine, db_name, table_name sippeers => engine, db_name, table_name but with same results Hans -- _ -- 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] [Fwd: FW: realtime mysql - p4]
On Fri, 2011-05-20 at 10:05 +0100, Andrew Thomas wrote: > Post your cdr_mysql.conf and res_mysql.conf and we'll take it from > there. > > Don't forget to remove any 'private' info first (like passwords). > > Cheers Tnx for the offer, Wil get the files when got back at the office. I presume that cdr_mysql.conf is only relevant for storing call-data-records? Perhaps that is something for later on. For now, i have to show a working *, with all sip-details in a mysql-DB. Other people pointed out that other means (postgres, ldap) might work better, but that's not an option for me. Hans -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [Fwd: FW: realtime mysql - p4]
Ok, i tried the suggestion: Instead of: sippuser => resource, database_name, table_name sippeer => resource, database_name, table_name I put in: sippuser => resource, context, table_name sippeer => resource, context, table_name Unfortunately, with the same results. btw i tried both "general" as "default" Besids the commands i tried below, isn't there any other way to see what's going on? Perhaps it is totally unrelated, but if i perform a mysql-login on the prompt, i first have to select the database manualy, ie it isn't selected by default for the created mysqluser [in this case: voipadmin] Other wild idea, is there a minimum number of fields that haved to be filled? And why is asterisk complaining about not being able to find the databse, when trying to fill it from the asterisk-CLI? My database _is_ named "asterisk".. > kc3054*CLI> realtime update sipusers set SET port = 4343 WHERE name = > 0277611 Failed to update. Check the debug log for possible SQL > related entries. > Command 'realtime update sipusers set SET port = 4343 WHERE name = > 0277611' failed. > [May 18 18:47:16] WARNING[16718]: res_config_mysql.c:559 update_mysql: > MySQL RealTime: Invalid database specified: 'asterisk' (check res_mysql.conf) I mean, is that silly or what? > > > # grep mysql extconfig.conf |grep sip > ;sipusers => mysql,asterisk,sip_devices > ;sippeers => mysql,asterisk,sip_devices > ;sipusers => mysql,general,sip_devices > ;sippeers => mysql,general,sip_devices > sipusers => mysql,default,sip_devices > sippeers => mysql,default,sip_devices > > > kc3054*CLI> module show like mysql > Module Description Use > Count > cdr_mysql.so MySQL CDR Backend0 > > res_config_mysql.soMySQL RealTime Configuration Driver 0 > > app_mysql.so Simple Mysql Interface 0 > > 3 modules loaded > kc3054*CLI> > kc3054*CLI> sip show users > Username Secret Accountcode Def.Context > ACL ForcerPort > j.witvliet geheimdefault > No Yes > 027761125b06d3a0b5ef73 default > No Yes > kc3054*CLI> > kc3054*CLI> sip show peers > Name/username HostDyn > Forcerport ACL Port Status Realtime > 0277611(Unspecified)D N > 0Unmonitored > j.witvliet (Unspecified)D N > 0Unmonitored > 2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 2 offline] > kc3054*CLI> kc3054*CLI> > > kc3054*CLI> > kc3054*CLI> realtime mysql cache > kc3054*CLI> realtime mysql status > general connected to asterisk@127.0.0.1, port 3306 with username voipadmin > for 18 seconds. > kc3054*CLI> > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk18 - realtime/mysql - take 3
Still a couple of questions.. I did configure extconfig.conf ... ;iaxusers => odbc,asterisk ;iaxpeers => odbc,asterisk ;sipusers => odbc,asterisk sipusers => mysql,asterisk,sip_devices sippeers => mysql,asterisk,sip_devices ;sippeers => odbc,asterisk ;sipregs => odbc,asterisk ;voicemail => odbc,asterisk ;extensions => odbc,asterisk ;meetme => mysql,general ;queues => odbc,asterisk ;queue_members => odbc,asterisk ;musiconhold => mysql,general ;queue_log => mysql,general So only defining sipusers & sippeers for mysql And noticed two files for configuring mysql-stuff: file: res_config_mysql.conf database access config: host, user, pwd file: res_odbc.conf in section [mysql2]: mysql database config: host, user, pwd So, i configured them both... Quick check:kc3054*CLI> sip show users Username Secret Accountcode Def.Context ACL ForcerPort j.witvliet geheim default No Yes 0277611 25b06d3a0b5ef73default No Yes kc3054*CLI> kc3054*CLI> sip show peers Name/username Host Dyn Forcerport ACL Port Status Realtime 0277611 (Unspecified) D N 0Unmonitored j.witvliet (Unspecified) D N 0Unmonitored 2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 2 offline] kc3054*CLI> Here i see at for both users and peers ONLY the statis entries from sip.conf file, nothing from mysql... kc3054*CLI> realtime mysql status general connected to asterisk@127.0.0.1, port 3306 with username voipadmin for 5 seconds. kc3054*CLI> =>No warnings/errors but nothing else either... kc3054*CLI> kc3054*CLI> realtime mysql cache kc3054*CLI> =>No warnings/errors but nothing else either... the module res_config_mysql.so is loaded, Try todo something else: kc3054*CLI> realtime update sipusers set SET port = 4343 WHERE name = 0277611 Failed to update. Check the debug log for possible SQL related entries. Command 'realtime update sipusers set SET port = 4343 WHERE name = 0277611' failed. [May 18 18:47:16] WARNING[16718]: res_config_mysql.c:559 update_mysql: MySQL RealTime: Invalid database specified: 'asterisk' (check res_mysql.conf) kc3054*CLI> ==> here the system talkes about _another_ config file! <== So which file should i configure: A) res_config_mysql.conf B) res_odbc.conf C) res_mysql.conf But even when i put my credentials in all three of them, still no show! DB check: mysql -h localhost -u voipadmin -p Enter password: Server version: 5.0.67 SUSE MySQL RPM mysql> use asterisk; select name,username,secret,host,nat from sip_devices; Database changed +-+-++-+-+ | name| username| secret | host| nat | +-+-++-+-+ | 0031756 | 0031756 | geheim | dynamic | Yes | +-+-++-+-+ 1 row in set (0.00 sec) mysql> According to *,TDG, page 349: Also filled the file /etc/unixODBC/odbcinst.ini, and the command odbcinst -q -d produced the required result: "[MySQL]" I presume i made a silly mistake/omission, but i fail to see how i can detect that, or other steps to test the correct configuration of ARA. So it looks that i'm stuck. Can not imagine that i'm the first here! But even from "the definitive guide", chapter 16 and onwards, it isn't clear if you should use the mysql-stuff directly of through the odbc-routines Kind regards, Hans -- _ -- 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] iptables for Asterisk - Any good guides out there?
On Sat, 2011-05-14 at 19:51 -0400, Bruce B wrote: > Hi everyone, > > > I want to issue the command: > > > iptables -F > > > and then rebuild everything from the beginning with a very limited > scope and then without locking myself block all other traffic. Can you > suggest what I should put in the shell that would get me this: > > > Allow traffic from subnet 172.16.0.0/24 (my VPN tunnels) - All > traffic including those of Asterisk and HTTP - I trust this network > Allow traffic from subnet 192.168.1.0/24(other side of VPN > network) - All traffic including those of Asterisk and HTTP - I trust > this network > Allow traffic from single IP of DID provider - 5060 TCP/UDP and > 1-10200 UDP > Allow VPN access on port 1194 UDP --- I have that figured out to be > (iptables -A INPUT -p udp -m udp --dport 1194 -j ACCEPT) works for > this. > > > BLOCK all other traffic <- Important most of all > > > Please note that from the subnets I want to allow every single port > possible and all traffic. I specially have problems with getting a > whole subnet be able to access everything. > > > Thanks It's a bit more complicated Firstly you have to set the default rules FIRST $IPT -P INPUT DROP $IPT -P OUTPUT ACCEPT $IPT -P FORWARD ACCEPT And then do the flusing, not the otherway round After that you can add rules to accept trafic after the last rules, it is handy to put: $iptables -A INPUT -i $EXTERNAL_DEV -j LOG --log-prefix " EXT; INC " iptables -A OUTPUT -o $EXTERNAL_DEV -j LOG --log-prefix " EXT; OUT " iptables -A FORWARD -i $EXTERNAL_DEV -j LOG --log-prefix " EXT; FWD " So can can see in the syslog what you are missing ;-) I'll guess, you would also like to accepts ntp,dhcp, domain-dns from your isp-provider. Perhaps also http, https, pop, pops, imap, imaps. And probably some more, depending on your need So'll see them soon enough in your logfiles hw -- _ -- 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
