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| 00:35.38 | k-man | any anyone recommend a softphone that works well with asterisk? |
| 00:35.43 | k-man | err... for OSX that is |
| 00:36.31 | SunTsu | k-man: any soft phone that talks sip will do |
| 00:37.07 | k-man | yeah - just wondered if anyone knew of a particularly good one |
| 00:37.25 | SunTsu | that's a matter of preferences |
| 00:41.53 | k-man | yeah ok |
| 00:47.24 | p3nguin | I doubt that you have very many choices, so test them all. |
| 00:50.35 | JerJer | k-man: Zoiper Communicator hasn't crashed on me |
| 00:51.15 | JerJer | don't use it a whole lot, but it works with any of the sound devices i've thrown at it |
| 00:57.01 | k-man | thanks JerJer ill have a look at it |
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| 00:59.01 | JerJer | np |
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| 01:09.32 | Juggie | JerJer: word. |
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| 01:17.43 | JerJer | meep meep |
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| 01:36.02 | *** join/#asterisk misc-- (~misc@202.171.160.4) |
| 01:36.13 | misc-- | hi all - is there a way to set "keep-alives" in asterisk, for sip? |
| 01:36.47 | p3nguin | Yes. You use the qualify setting in each peer definition. |
| 01:36.58 | misc-- | ahhh |
| 01:36.59 | p3nguin | qualify=yes is the same as qualify=2000 |
| 01:37.02 | misc-- | so, qualify=900 for example? |
| 01:37.06 | p3nguin | yes |
| 01:37.09 | misc-- | ahhh ok |
| 01:37.21 | p3nguin | That'll send an OPTIONS packet every .9 seconds. |
| 01:37.33 | misc-- | oh |
| 01:38.43 | p3nguin | I normally use yes rather than another value. |
| 01:38.51 | p3nguin | 2 seconds is usually good enough. |
| 01:41.15 | misc-- | ah ok. Because we have four trunks to our provider, and according to their faq, the "registration timeout" should be 900 seconds. I'm wondering if that's the same as the qualify setting |
| 01:42.06 | p3nguin | They aren't behind NAT, I'm quite certain, so you don't really need to qualify your provider. |
| 01:42.34 | misc-- | oh ok |
| 01:42.44 | p3nguin | Keepalives are used to keep the phones' NAT open. |
| 01:43.02 | misc-- | oh I see. What if asterisk is behind NAT though? |
| 01:43.19 | p3nguin | Configure it for NAT and enjoy. |
| 01:43.28 | p3nguin | What branch are you using? |
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| 01:44.14 | misc-- | 1.6.2.16.1 |
| 01:44.37 | misc-- | I don't have any nat options set though in my sip trunk to the voip provider. Maybe that's the issue then |
| 01:45.06 | p3nguin | You'll want to make sure you put in your sip.conf general section, canreinvite=no, nat=yes, set a value for externhost or externip, set a value for localnet. |
| 01:45.24 | p3nguin | There is no "trunk," so let us stop using that term. |
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| 01:45.53 | p3nguin | If you mean the provider, let's say provider or ITSP. |
| 01:47.48 | p3nguin | Your ITSP isn't going to be behind NAT, so the sip entry for the ITSP will have nat=no and canreinvite=no in it. |
| 01:48.18 | misc-- | ah ok, sorry, provider. So, in my sip.conf in the general, I have: registertimeout=900, Defautlexpirey=900, Maxexpirey=3600. I initially did have externip and localnet set but only one way conversations could be heard. Maybe it was because I didn't have the nat=yes in there though. I will try that (with canreinvite=no as well) and asee what happens |
| 01:48.47 | misc-- | ohhh ok, so I have nat=yes in my sip conf, but nat=no for the sip ITSP |
| 01:49.28 | p3nguin | That sounds like it is correct. |
| 01:49.53 | p3nguin | Asterisk does a pretty good job at working around NAT. It's not perfect, but it's not bad. |
| 01:50.18 | misc-- | ok. Yeah it's weird because it worked for the past two weeks, then just today for no reason (well I'm sure some reason), I get all these registration timeout issues all of a sudden |
| 01:50.37 | p3nguin | You don't know what changed? |
| 01:50.58 | misc-- | no nothing changed at all, at least not on our side |
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| 01:51.31 | misc-- | so then I speak to the voip provider and they say that it looks fine on their end... which of course it does go fine but a few minutes later, it will just drop out |
| 01:51.34 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 01:51.37 | p3nguin | Registration timeouts, huh? That seems like it could be networking related. |
| 01:52.19 | p3nguin | One-way audio is almost always caused by NAT settings being wrong. |
| 01:52.20 | misc-- | well you would think that. We have four lines to them. One of them would be stable, the others would go up and down. |
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| 01:52.40 | p3nguin | Four lines? How does that work? |
| 01:52.43 | misc-- | yeah the one-way audio was fixed - that only occurred when I put this asterisk server in |
| 01:53.17 | misc-- | well we are using freepbx - that's why I was saying "trunks" before (that's how they are set up in there). So each phone number has its own setup file |
| 01:53.27 | misc-- | which is a bit silly. I can't see why they can't all be on the one setup |
| 01:53.44 | p3nguin | A typical configuration between an ITSP and Asterisk involves one peer definition on Asterisk and multiple channels will be created AS-NEEDED for calls. |
| 01:54.07 | misc-- | actually this is a customer of ours. *Our* setup works fine, which is a different provider. One ITSP setup for all of our lines. We are not behind nat however (at least not for asterisk) |
| 01:54.27 | misc-- | that's what I thought |
| 01:56.02 | misc-- | it seems to be stable at the moment. No config change has been done (I haven't done your settings yet but I will put that in later if they have problems again) |
| 01:56.57 | p3nguin | My Asterisk, for example, has one register statement and one peer entry for my ITSP. I have multiple DIDs and can get multiple simultaneous calls per DID through that one peer entry in my sip.conf. I can also send multiple simultaneous outbound calls through the ITSP via that same peer entry. |
| 01:57.17 | misc-- | we are the same |
| 01:57.31 | misc-- | but the other voip provider that our customer uses (Engin) doesn't appear to work like that |
| 01:57.56 | p3nguin | Then I have no flippin' idea what you mean by "multiple lines" to the provider. |
| 01:58.05 | p3nguin | If it weren't VoIP, it might make sense. |
| 01:58.24 | p3nguin | Because physical lines to an analog device would be possible. |
| 02:00.14 | misc-- | in my sip_additional.conf, I have four separate configs in there. One per phone number. Sorry, just that freepbx calls them "sip trunks" which get set up in the sip_additional.conf file. So it's one ITSP config per phone number |
| 02:00.38 | p3nguin | That sounds ridiculous to me. |
| 02:02.05 | p3nguin | One account on the provider, one peer definition. One account on the provider can have multiple phone numbers associated with it. |
| 02:02.38 | misc-- | yes you would think that one config for multiple phone numbers would be better. Not sure why it's one separate config per line (phone number). So when you do a "sip show registry", there are four entries, one per phone number |
| 02:04.06 | p3nguin | no |
| 02:04.11 | p3nguin | one account, one entry |
| 02:04.25 | p3nguin | All my DID numbers are sent via the one account. |
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| 02:05.16 | misc-- | that is the same with our setup too with our provider. Just not with this other voip provider. Oh well |
| 02:05.43 | p3nguin | Sounds like it needs some adjustments. |
| 02:07.18 | misc-- | yeah but they were the ones that sent the account information to be set up like that. Doesn't make sense though |
| 02:07.53 | misc-- | anyway, it appears to be more stable. Still going up and down but not as much. However I don't get any "Registration timed out" issues like I used to. |
| 02:08.01 | p3nguin | I've never met an ITSP that knows how to configure a user's system. It's rather stupid. |
| 02:08.21 | misc-- | well this one is no different, because they don't know asterisk either |
| 02:09.03 | p3nguin | It doesn't make any sense that a group in such a position wouldn't be able to provide a sensible configuration sample for the user's system. |
| 02:09.30 | p3nguin | You would think that a phone company would know how to configure phone equipment. |
| 02:09.35 | misc-- | no it doesn't make sense =) |
| 02:09.53 | misc-- | well especially software that is as widespread as asterisk |
| 02:10.05 | misc-- | brb coffee |
| 02:26.07 | misc-- | thanks for your help today p3nguin. |
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| 04:31.01 | p3nguin | Is it normal to have a ton of "warning: '__lineno' may be used uninitialized in this function" spewing out when I run make on 1.8.3.2? |
| 04:32.27 | p3nguin | For example: app_dial.c:864:4: warning: '__lineno' may be used uninitialized in this function |
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| 04:45.46 | gruvfunk | How can I Park to a specific place/variable so that I can re-connect to that Parked call later in a script? |
| 04:47.51 | gruvfunk | ideally, parking to the CALLERID(num) would be great |
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| 05:23.40 | WiretapSeven | p3nguin, note how its a 'warning' |
| 05:23.41 | WiretapSeven | ignore it |
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| 05:47.21 | p3nguin | There is obviously some reason it exists, therefore I am inquiring if such warnings are normal. |
| 05:52.44 | Corydon76-home | p3nguin: Sounds highly suspicious. What compiler are you using? |
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| 05:53.38 | p3nguin | gcc 4.5.2 |
| 05:54.49 | Corydon76-home | That might be why |
| 05:55.03 | Corydon76-home | What distro is packaging 4.5.2? |
| 05:55.20 | p3nguin | Arch |
| 05:55.35 | Corydon76-home | Warnings are not normal. In -dev-mode, compiler warnings are fatal |
| 05:55.55 | p3nguin | What version of gcc do you recommend I use to compile Asterisk 1.8.3.2? |
| 05:57.00 | Corydon76-home | 4.4.x, if you can |
| 05:57.26 | Corydon76-home | That appears to be an errant warning, as we use __LINE__ for line numbers |
| 05:57.33 | p3nguin | I've compiled other versions in the 1.8 branch without seeing those warnings. Let me check to see if I can determine which gcc version I used. |
| 05:58.17 | Corydon76-home | Probably an uncaught upstream bug, but... not something we can generally do something about |
| 05:59.19 | Corydon76-home | You should be able to use any version of gcc. |
| 05:59.29 | Corydon76-home | Well, >=3.0 |
| 05:59.58 | Corydon76-home | I'm on 4.4.3 here |
| 06:00.34 | p3nguin | It looks like it was gcc 4.5.0 and asterisk 1.8.2.3. |
| 06:00.54 | p3nguin | I didn't see the warnings, but they could have printed while I was doing something else away from the screen. |
| 06:02.57 | p3nguin | Should I worry about those warnings? I'm not a programmer, so I don't know what they mean or how they affect anything. |
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| 06:22.08 | gavimobile | folks, how do I get an older kernel |
| 06:22.25 | gavimobile | I did an update and now I have a lot of digium packaes which won't update |
| 06:27.11 | p3nguin | What OS are you using? |
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| 06:32.29 | gavimobile | p3nguin: I believe centos5 |
| 06:32.55 | p3nguin | What kernel version do you have now and what version do you want to have? |
| 06:35.10 | gavimobile | p3nguin: 2.6.18-238.5.1.el5, and I want 2.6.18-194.31.1.el5 |
| 06:35.14 | p3nguin | Did you simply want to roll back any updates you've recently applied, or roll back only the kernel and any dependency of it? |
| 06:35.28 | gavimobile | if I can rollback, I think that might be good |
| 06:35.35 | gavimobile | but what will I do next time I want to update |
| 06:35.41 | p3nguin | Don't. |
| 06:36.21 | p3nguin | If you have packages that depend on a specific version of the kernel, make sure there is a compatible version of that before you update the dependency. |
| 06:37.01 | gavimobile | im not sure how, but I won't do an update unless I check with others next time |
| 06:38.01 | gavimobile | so am I rolling back, or downgrading kernel? |
| 06:41.16 | wdoekes2 | downgrading sounds fine |
| 06:41.36 | gavimobile | wdoekes2: you make the calls dide. |
| 06:41.42 | gavimobile | how do I do it |
| 06:41.50 | wdoekes2 | I don't know if your (any) package manager supports rolling back |
| 06:42.03 | gavimobile | well I sure as hell don't know |
| 06:42.07 | wdoekes2 | <package-manager> install linux-kernel-<version> |
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| 06:42.25 | wdoekes2 | I don't run centos.. but it's probably yum |
| 06:42.26 | gavimobile | im using asterisknow |
| 06:42.33 | gavimobile | ahh yes.. its yum |
| 06:43.18 | wdoekes2 | do a 'yum search linux' | grep kernel |
| 06:43.22 | wdoekes2 | or something |
| 06:43.28 | p3nguin | Do you have the previous kernel version in package cache? |
| 06:44.17 | gavimobile | yum sear linux | grep kernel, came back a new line blank |
| 06:44.26 | gavimobile | p3nguin: possibly. |
| 06:44.35 | gavimobile | how can I check |
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| 06:46.24 | p3nguin | Look under /var/cache/yum |
| 06:46.24 | Iiiak | plop |
| 06:47.10 | p3nguin | find /var/cache/yum/ -iname \*2.6.18-194.31.1.el5\* |
| 06:48.32 | p3nguin | If it's there, maybe it is under /var/cache/yum/base/packages/ or something. |
| 06:48.39 | gavimobile | this is in /var/cache/yum addons base digium-current rpmforge updates |
| 06:48.39 | gavimobile | asterisk-current c5-media extras timedhosts.txt |
| 06:48.59 | p3nguin | Had you used the 'find' command that I gave you, you won't see that stuff. |
| 06:49.25 | gavimobile | /var/cache/yum/base/packages seems to be empty |
| 06:49.46 | gavimobile | p3nguin: 'yum search linux' | grep kernel without the '' came back with nothing |
| 06:51.48 | p3nguin | I have no idea which build version this is, but maybe it's the one you want: http://vault.centos.org/5.5/os/i386/CentOS/kernel-2.6.18-194.el5.i686.rpm |
| 06:52.14 | gavimobile | so what do I do? wget http://vault.centos.org/5.5/os/i386/CentOS/kernel-2.6.18-194.el5.i686.rpm |
| 06:52.23 | gavimobile | doesn't it have to match my hardware or something |
| 06:52.28 | gavimobile | than afterwards how do I install it |
| 06:52.43 | p3nguin | You can. Then you can use rpm to upgrade (downgrade) it. |
| 06:52.57 | p3nguin | rpm -Uvh kernel-2.6.18-194.el5.i686.rpm |
| 06:53.06 | gavimobile | so wget http://vault.centos.org/5.5/os/i386/CentOS/kernel-2.6.18-194.el5.i686.rpm |
| 06:53.10 | gavimobile | then rpm -Uvh kernel-2.6.18-194.el5.i686.rpm |
| 06:53.14 | gavimobile | and than im done? |
| 06:53.49 | p3nguin | That should upgrade your current kernel package to the one you've just downloaded. |
| 06:54.05 | p3nguin | But I don't know if that will solve your problem. It could even cause new problems. |
| 06:54.05 | gavimobile | I thought we wanted to downgrade |
| 06:54.20 | gavimobile | p3nguin: maybe I should paste bin to you my original problem |
| 06:54.40 | gavimobile | maybe you have a different diagnosis from the person who gave me a diagnoiss yesterday |
| 06:55.01 | p3nguin | There is no downgrade term in rpm. You install or upgrade. Upgrading can be to a lower version number. |
| 06:55.54 | gavimobile | p3nguin: http://pastebin.com/JsuUZ337 |
| 06:55.59 | gavimobile | check that out.. this is my original problem |
| 06:56.35 | gavimobile | p3nguin: ahh then that makes more sense. so I downloaded the rpm. now I just run rpm -Uvh kernel-2.6.18-194.el5.i686.rpm? |
| 06:57.31 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 06:57.33 | schmidts | good morning |
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| 07:03.31 | p3nguin | Did you try the first three out of the four provided suggestions at the end of your yum update? |
| 07:04.20 | gavimobile | I tried the skip |
| 07:04.33 | gavimobile | p3nguin: and that's how I was able to download all the updates |
| 07:04.38 | gavimobile | but these few packages won't update |
| 07:05.22 | gavimobile | I originally did an update cause (I don't remember what) but some package/update required a higher version of asterisk |
| 07:05.30 | gavimobile | I think it is an update of my freepbx |
| 07:12.59 | p3nguin | In the future, remember that updating isn't always a good idea. |
| 07:17.32 | gavimobile | p3nguin: king solomon says mistiakes are great to learn from |
| 07:17.42 | gavimobile | :-) |
| 07:17.57 | gavimobile | p3nguin: so should I run rpm -Uvh kernel-2.6.18-194.el5.i686.rpm |
| 07:20.14 | p3nguin | You still haven't done that? |
| 07:20.51 | *** join/#asterisk jacc0 (~jacco@D522448D.static.ziggozakelijk.nl) |
| 07:20.54 | jacc0 | hi all |
| 07:20.56 | jacc0 | good morning |
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| 07:24.50 | gavimobile | p3nguin: im waiting for the ok |
| 07:24.54 | gavimobile | I guess that was the ok |
| 07:25.42 | gavimobile | p3nguin: error:http://pastebin.com/wkVxZbC4 |
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| 07:26.43 | gavimobile | http://pastebin.com/9D5CGUvc |
| 07:26.49 | gavimobile | the second one is more clear.. |
| 07:26.55 | gavimobile | do I need to put the rpm in a special directory? |
| 07:27.04 | p3nguin | no |
| 07:28.00 | p3nguin | It looks as if your current version is 2.6.18-194.11.1.el5, which is needed by another package. |
| 07:28.46 | gavimobile | p3nguin: you make the call captain |
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| 07:39.21 | p3nguin | I would probably remove the package that is giving me trouble. I would then update the other packages. Then try installing the problematic package using yum just to see what happens. |
| 07:39.58 | p3nguin | That will, of course, not guarantee the issue is fixed, nor will it guarantee that you will be able to install all necessary packages. |
| 07:40.21 | p3nguin | That's when you learn how to check the log and revert package versions. |
| 07:41.47 | gavimobile | how do I know which package is giving me trouble. and how do I remove it |
| 07:41.53 | gavimobile | p3nguin: ? |
| 07:43.09 | p3nguin | It's printed in plain text on your screen. You've copied and pasted it more than once. You remove it with yum or rpm. |
| 07:44.21 | p3nguin | You're probably going to break something, but you've already broken something or we wouldn't be discussing it. |
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| 07:46.24 | gavimobile | p3nguin: im guessing your refering to the kernel itself |
| 07:46.29 | gavimobile | to remove the kernel |
| 07:46.43 | p3nguin | Don't do that. |
| 07:47.07 | gavimobile | kmod-dahdi-linux-fwload-vpmadt032-2.4.1-2_centos5.2.6.18_194.32.1.el5.i686 from digium-current has depsolving problems |
| 07:47.29 | verywiseman | i want to buy T1/E1 card , which cable i must purchase cross-over cable or straight-through cable where i will connect T1 card directly to PRI device? |
| 07:47.30 | p3nguin | Your system won't boot without a kernel. If you leave the kernel and remove the dadhi package that's giving you trouble, at least you can still boot up if something causes the box to go down suddenly. |
| 07:48.04 | gavimobile | yea, but the whole point of the server is for dahdi/asterisk |
| 07:48.13 | gavimobile | so I remove dadhi and reinstall? |
| 07:49.13 | gavimobile | the broken package hasn't been installed yet. so how can I remove it |
| 07:49.17 | gavimobile | im missing the logic here |
| 07:49.28 | gavimobile | which package of mine which is installed is broken |
| 07:50.20 | *** join/#asterisk Tim_Toady (~moi@79.103.44.106.dsl.dyn.forthnet.gr) |
| 07:50.31 | p3nguin | You don't have a kmod-dahdi-linux-whatever package installed? yum update isn't going to pull that package name out of its ass and say that it is causing trouble. |
| 07:52.29 | *** join/#asterisk davlefou (~david@41.225.9.81) |
| 07:52.54 | gavimobile | so basicly im hiding the update |
| 08:05.25 | *** part/#asterisk Iiiak (~Iiiak@AMontpellier-551-1-9-198.w92-133.abo.wanadoo.fr) |
| 08:06.34 | *** join/#asterisk oej (~olle@static-213-115-251-100.sme.bredbandsbolaget.se) |
| 08:07.03 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 08:09.22 | *** join/#asterisk creativx (~creadurex@226.62-97-205.bkkb.no) |
| 08:09.41 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 08:11.26 | *** join/#asterisk dimm (~appleworm@unaffiliated/dimm) |
| 08:13.47 | *** join/#asterisk nosbig (~nosbig@cpe-65-25-22-206.neo.res.rr.com) |
| 08:15.31 | *** join/#asterisk bmg505 (~leon@196-209-99-25.dynamic.isadsl.co.za) |
| 08:16.16 | *** join/#asterisk Cesare (~Adium@creati59.lnk.telstra.net) |
| 08:21.54 | gavimobile | p3nguin: ? |
| 08:22.52 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 08:23.37 | p3nguin | Yeah? |
| 08:25.24 | *** join/#asterisk mzb (~mzb@ppp108-88.static.internode.on.net) |
| 08:26.51 | gavimobile | sorry to bug ya. im still not clear as what I need to do. |
| 08:29.48 | *** join/#asterisk mzb (~mzb@ppp108-88.static.internode.on.net) |
| 08:31.56 | p3nguin | I'm still not clear on what you have done, what packages you do and do not have installed, what packages you want to install... |
| 08:32.13 | *** join/#asterisk jg1234 (~jan@dslc-082-082-037-188.pools.arcor-ip.net) |
| 08:32.50 | jg1234 | hi |
| 08:35.13 | p3nguin | Can you show me "rpm -qa|grep dahdi"? |
| 08:35.15 | *** join/#asterisk Denial (Denial@drgi.co.uk) |
| 08:35.57 | gavimobile | sure |
| 08:36.51 | gavimobile | p3nguin: http://pastebin.com/Sp223nnp |
| 08:37.49 | nosbig | What is the current best practice for integrating Asterisk 1.8 and Festival? Modify festival.scm and run unpatched using the Festival() app, run a patched Festival copy and use the Festival() app, or use text2wave and use Playback() to play back the audio? Or is there something I might be missing? |
| 08:40.34 | p3nguin | As I said, I would remove kmod-dahdi-linux-fwload-vpmadt032-2.3.0.1-1_centos5.2.6.18_194.11.1.el5 since it is giving you a problem, then update the rest of the system, then try to fix that jacked up package. |
| 08:42.05 | p3nguin | I think I said that, anyway. |
| 08:42.24 | p3nguin | Yep, over an hour ago. |
| 08:42.58 | *** join/#asterisk bjornts (~BTS@it010226.klientdrift.uib.no) |
| 08:44.52 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 08:46.20 | jg1234 | what am i doing wrong http://pastebin.com/05C0XBud |
| 08:51.13 | gavimobile | p3nguin: ok ill give it a try |
| 08:51.13 | gavimobile | sorry for any ttrouble |
| 08:53.44 | *** join/#asterisk lost_soul (~noymfb@cpe-74-78-191-114.twcny.res.rr.com) |
| 08:55.45 | gavimobile | p3nguin: I ran it, it removed these. http://pastebin.com/CUC7ehR1 . I then tried yum update and yum upgrade and yum install update and yum install upgrade (cause I don't remember if it needs an install. and they all say nothing to do.) |
| 08:55.58 | gavimobile | It looks like everything is working |
| 08:56.56 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 09:14.20 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 09:16.01 | *** join/#asterisk mintos (~mvaliyav@114.143.161.197) |
| 09:18.31 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 09:20.13 | killown | I have a analogic pabx with 20 extensions, I'd like integrate it with asterisk, do I need 20 ports fxo or there is a solution more cheaper? |
| 09:20.31 | jg1234 | anyone an idea why my updateconfig doesnt work ? |
| 09:20.57 | tzafrir | killown, can that PBX talk in any other channel? |
| 09:21.31 | tzafrir | What to you want to keep it for? What functionality? |
| 09:21.52 | tzafrir | (and what do you want Asterisk to do?) |
| 09:21.53 | killown | tzafrir I don't know so much about analogic pabx, what means any other channel? wich case do it not talk in any other? |
| 09:22.43 | gavimobile | hi tzafrir :-) |
| 09:23.00 | killown | tzafrir I need asterisk to record my calls |
| 09:23.03 | tzafrir | gavimobile, hi |
| 09:26.13 | killown | I think the only way to do that is with 20 fxo ports :( |
| 09:26.48 | *** join/#asterisk casix (~casix@144.165.219.87.dynamic.jazztel.es) |
| 09:26.51 | casix | hello |
| 09:28.06 | hensema | killown: if all 20 extensions of the analogue pbx are connected to asterisk, what's the use of the analogue pbx then? |
| 09:28.20 | hensema | only line interface to the outside world? |
| 09:28.23 | hensema | no phones connected? |
| 09:28.31 | killown | hensema will not be used |
| 09:29.02 | killown | it will be replaced unless there is a solution with less fxo ports |
| 09:29.36 | hensema | asterisk connected to an isdn30? |
| 09:30.56 | killown | analogue lines |
| 09:31.18 | hensema | migrate them to isdn? |
| 09:31.26 | hensema | migrate them to sip? |
| 09:31.48 | killown | hensema, no, I will keep using analogue lines |
| 09:32.00 | casix | I have problems with regexp using operators : and =~ . The operator =~ don't work and the : works but I don't know how to find quotes " . I have tried to escape with one or two \ I have tried to do things like [^"]* or [^\"]* or [^\\"]* but nothing works. Any ideas? thanks |
| 09:32.43 | henk | casix: can you paste an example of what you are doing, expecting and what actually happens? |
| 09:34.20 | hensema | killown: I'd *really* try to migrate your public phone numbers to a SIP provider, it'll really save you a lot of money and pain |
| 09:34.26 | kaldemar | casix: what version of asterisk are you using? |
| 09:35.47 | killown | hensema, you're right, but there is no reliable voip services in my country (Brazil) |
| 09:35.58 | casix | kaldemar: 1.4.26.2 |
| 09:38.20 | casix | henk: http://pastebin.com/hTBDVupG |
| 09:39.18 | casix | henk: sorry in the line 5 change : for =~ sorry :P |
| 09:40.05 | casix | henk: http://pastebin.com/y3rVbjPz -> this is the good one |
| 10:13.48 | *** join/#asterisk Tim_Toady (~moi@188.4.65.193.dsl.dyn.forthnet.gr) |
| 10:16.21 | *** join/#asterisk SiNGLer (~singler@81-7-123-162.static.zebra.lt) |
| 10:27.17 | jacc0 | I'm experiencing great instability with several asterisk 1.8 installations on debian 6 |
| 10:28.06 | jacc0 | these are asterisk realtime configurations |
| 10:28.42 | tzafrir | jacc0, please be more specific |
| 10:28.52 | jacc0 | asterisk stops responding |
| 10:28.57 | jacc0 | I have the astcanary running |
| 10:29.15 | jacc0 | I'm still looking in the logfiles |
| 10:29.37 | jacc0 | i've made some changes to the mysql configuration in my.cnf: |
| 10:29.57 | jacc0 | interactive_timeout= 864000 |
| 10:30.08 | jacc0 | to try and fix the instabilitys |
| 10:30.35 | jacc0 | logfiles don't give me a clue on why asterisk stoped |
| 10:31.16 | jacc0 | there is just a big gap in the logging |
| 10:33.02 | *** join/#asterisk niekniek (~niekniek@92.70.112.34) |
| 10:41.46 | *** join/#asterisk wonderworld (~ww@port-92-201-66-14.dynamic.qsc.de) |
| 10:42.28 | jacc0 | I have some coredumps; are they usefull? |
| 10:43.30 | *** join/#asterisk jenna (~jjones@unaffiliated/jenna) |
| 10:44.49 | jenna | hey all. Anyone else getting dependency errors while trying to install asterisk on centos 5.6 from the official repos ? http://pastebin.com/GQGyUVzY |
| 10:51.13 | *** join/#asterisk dimm (~appleworm@unaffiliated/dimm) |
| 10:53.17 | *** join/#asterisk mzb (~mzb@ppp108-88.static.internode.on.net) |
| 11:02.29 | casix | jenna: http://www.voip-info.org/wiki/view/Asterisk+Linux+Redhat |
| 11:02.47 | *** part/#asterisk TehRabbitt (~TehRabbit@pool-71-172-89-155.nwrknj.fios.verizon.net) |
| 11:04.55 | jenna | casix, that link is too outdated |
| 11:05.35 | casix | sorry, I use debian |
| 11:06.03 | jenna | casix, squeeze ? |
| 11:06.16 | casix | yes |
| 11:06.36 | casix | but compiling, no repository |
| 11:06.44 | jenna | casix, u compile from source or use the official asterisk.org debs ? |
| 11:07.00 | casix | compiling |
| 11:07.08 | jenna | casix, don't get it why would you want to compile for a production system ? |
| 11:07.42 | henk | and why 1.4? |
| 11:08.34 | casix | and... why not? what problems can I have? |
| 11:08.38 | jenna | yeah. I think the wise choice is to stick with 1.6.X, at least for production systems. no ? |
| 11:08.56 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 11:11.58 | jenna | anyway, I have seen a bug report has been submitted. https://issues.asterisk.org/view.php?id=18992 as patch has also been issueed |
| 11:12.52 | jenna | .. official repo packages haven't been patched/update. Any asterisk.org representative lurking around here ? |
| 11:13.23 | casix | i thing 1.4 (or some versions) are more stable than 1.6 and is better to wait to a good one 1.8 than go to 1.6 |
| 11:14.23 | *** join/#asterisk phix (~threat@123-243-44-131.static.tpgi.com.au) |
| 11:14.29 | casix | jenna compile it by youself, its easy and give you the oportunity to change the version whenever you want. You just have to download, ./configure, make menuconfig, make, make install |
| 11:14.34 | casix | and thats all you need |
| 11:14.54 | *** part/#asterisk gavimobile (~user@84.108.104.165) |
| 11:15.45 | *** join/#asterisk guilhermebr (~Guilherme@200.103.96.98) |
| 11:16.00 | phix | Evening! |
| 11:16.19 | jenna | casix, I have done it off n on for the last 5-6 years. but it seem logical to go with binaries in production. |
| 11:16.31 | casix | why? |
| 11:16.50 | casix | why leave the control to someother if you can have the control? |
| 11:17.19 | jenna | casix, having a production server. laced with compilers is not a wise thing in terms of security. as well as backing up of things etc.. |
| 11:17.36 | casix | okok |
| 11:18.13 | jenna | casix, btw what are you credentials/experience with sys admining etc ? (just asking to gauge about ur advise ;) ) |
| 11:18.40 | casix | well as sysadmin i have been working 7 or 8 years |
| 11:18.44 | casix | with asterisk 4 |
| 11:21.33 | jenna | hmm. in a cave ? |
| 11:22.17 | jenna | casix, anyway I feel its quite a bit risky. and against the advise of SAGE norms. |
| 11:22.20 | casix | no, in my house :P |
| 11:22.50 | jenna | casix, where are from ? tora bora region ;) |
| 11:23.09 | kaldemar | that's have you compile in another machine and install the compiled binaries on production systems. |
| 11:23.46 | casix | spain |
| 11:24.08 | casix | in a consultory and now in a ip telephony company |
| 11:25.02 | jacc0 | in the full log i see : aststerisk -rerisk.c: -- Remote UNIX connection |
| 11:25.15 | jacc0 | normaly it says: asterisk.c: -- Remote UNIX connection |
| 11:25.22 | jacc0 | what is the difference? |
| 11:25.35 | jenna | kaldemar, yeah thats an option. but I was hoping to save some time. |
| 11:25.35 | casix | jenna: for me the risk is to use centos, they release packets without check them! |
| 11:25.49 | jacc0 | wtf is -rerisk.c? |
| 11:26.39 | henk | jacc0: looks like two programs wrote at the same time... |
| 11:26.41 | SunTsu | jenna: you might want to check your attitude, because it's you who wants help, maybe you stop insulting people who try to help |
| 11:27.09 | henk | jacc0: afaict the 'asterisk -r' somehow found its way there. |
| 11:27.32 | jacc0 | ty |
| 11:27.47 | jacc0 | you are right |
| 11:28.05 | jenna | SunTsu, all was/is said without any intention of offending. besides casix seems like a good sport as well. Anyway I suppose I should tone down it a little further. |
| 11:28.14 | jacc0 | I've just been typing in the wrong window ;) |
| 11:28.39 | *** join/#asterisk hehol (~hehol@buero-gw.dortmund.loca.net) |
| 11:28.48 | casix | jenna: no problem for me... just change of ways to do things, no? |
| 11:28.49 | casix | :P |
| 11:29.25 | casix | jenna: no problem for me... just different oppinion how to do things, no? |
| 11:29.30 | casix | that's better :P |
| 11:29.45 | niekniek | hello! can i do something like call-limit=2 and call-wating=no in sip.conf (call-waiting=no does not exist, but something like that :) ) |
| 11:29.55 | SunTsu | jenna: btw. you could buold your own packages that work on a different box. And no, having a compiler on a box is not a security issue. Because as soon as any intruder could use it they could as well bring/build their own |
| 11:29.56 | jenna | casix, hehe. yeah |
| 11:30.11 | *** join/#asterisk [loy] (~nobody@95.73.19.30) |
| 11:31.20 | henk | anything not _needed_ on a server is a potentiel security issue... |
| 11:31.48 | SunTsu | henk: in terms of services that can be reached I agree |
| 11:32.08 | jenna | SunTsu, yeah well potayto potaato. security is debate-able . but ur right I guess I've to build the 1.6 packages. |
| 11:32.14 | SunTsu | anything else only makes it marginally easier once that system is compromised |
| 11:32.27 | niekniek | because call-limit limits my calls outgoing & incoming, but i want to be able to transfer calls but not have a second conversation coming in |
| 11:32.47 | jenna | casix, btw what billing s/w have you got going for you there ? |
| 11:32.48 | niekniek | some phones have a setting, but others don't |
| 11:33.05 | henk | combine 100 things making a hack marginally easier and you have an open relay/proxy/whatever... |
| 11:33.10 | casix | but... put a good firewall and live happy. Some people say that is possible go inside a computer that have no open ports.... |
| 11:33.57 | casix | jenna: there? where? |
| 11:34.18 | SunTsu | casix: well, there's exploits of firmwares, of the tcp/ip stack itself, and stuff |
| 11:35.00 | jenna | casix, in your asterisk box |
| 11:35.49 | casix | and, what I have seen in asterisk world, is force brute attacks to 5060... nothing more. I don't think anyone want to explode anyone of my pbx, they only want to call call and call |
| 11:36.11 | casix | jenna: my english is limited, I don't undertand what are you asking... |
| 11:39.02 | jenna | casix, I meant what billing software are you running on/for that asterisk 1.4 server of yours ? |
| 11:39.34 | casix | heheheheh okok buff the worst possible |
| 11:40.51 | henk | casix: "I don't think" <-- spammers don't care what you think... |
| 11:41.12 | casix | is a propietary system the the company bought so many years ago and everybody hate it. We have to change it but we haven't find any decent one. Maybe we will make one |
| 11:41.24 | jenna | I guess a little bit of attitude should be allowed after all :) keeping in view the diverse international community. |
| 11:42.02 | casix | henk: I don't care if someone use a pbx to send spam... I'm only worried to lose a lot of money because someone call to Norh Korea |
| 11:42.17 | jenna | henk: what about you. whats your take on a good asterisk billing software (possibly open source ) ? |
| 11:43.59 | casix | jenna: and you? |
| 11:44.18 | henk | casix: great attitude... |
| 11:44.28 | henk | jenna: i don't. |
| 11:45.45 | jenna | casix, I still in market to build an experienced opinion of it. So I guess my advise is good as none. |
| 11:46.21 | jenna | casix, been exploring stuff. a2billing , jbilling , easyitsp etc.. |
| 11:46.40 | *** join/#asterisk coppice (~chatzilla@m121-203-193-51.smartone-vodafone.com) |
| 11:47.32 | casix | henk: why? there are a lot of server less secures, a lot of people with exploids in his personal computers... I don't send emails I don't have any port open except 5060, and in 4 years no spammer have used any of my pbx... I have to be so worried about a thing that have never happened?? if there is a bug in one pbx and a spammer start sending emails I will protect all my pbx but until this happens... |
| 11:48.45 | casix | I will no lose time in thinks that I can protect with a simple firewall... |
| 11:49.03 | jenna | casix, ever one of your clients complained of DOS |
| 11:49.54 | casix | yes, a lot of them but only in 5060 port |
| 11:49.57 | henk | casix: you're off the topic and it looks like this would become a loooong discussion so i don't care |
| 11:50.05 | casix | hehehehehe |
| 11:50.13 | casix | yes it can be a log discussion |
| 11:50.25 | *** join/#asterisk asterisk-learner (~chatzilla@77.42.241.114) |
| 11:50.51 | casix | jenna: for this we have installed fail2ban |
| 11:50.55 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 11:50.55 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 11:51.40 | henk | brrr 'shudder' |
| 11:51.51 | henk | iptables > fail2ban imho :) |
| 11:51.52 | jenna | casix, its not the same but. I guess we can go on & on on this topic & invite the wrath of the chOp |
| 11:52.45 | leifmadsen | fail2ban is iptables... |
| 11:53.20 | casix | henk: yes is iptables. fail2ban only make regexp over log files and execute scripts when detect a pattern |
| 11:53.49 | casix | and this scripts ban ips in iptables for amount of time |
| 11:54.56 | henk | nope, fail2ban is a script for parsing logs. which is a really bad idea imho and when you look at the problems such solutions have had in the beginning, the impression doesn't get better. it _uses_ iptables, that's true, but iptables could do (almost) the same thing without having to parse logs which is error-prone. |
| 11:55.19 | leifmadsen | welcome to stupid debate monday! |
| 11:56.25 | casix | ok, henk. its true. I'm agree with you, but I can change that I'm not the boss |
| 11:56.37 | casix | s/can/can't/ |
| 11:57.23 | henk | casix: too bad... does your boss have the better arguments or is his single argument "i'm the boss"? :-p |
| 11:58.41 | casix | no, is like billing software, we don't have time to change it. We have a lot things to change and no time... we need more hands and we have only 2 by person |
| 11:59.09 | henk | know what you mean :-/ |
| 12:01.02 | *** join/#asterisk m_tadeu (~quassel@89-180-156-4.net.novis.pt) |
| 12:02.19 | *** join/#asterisk davevg-btwtech (~davevg__@24.115.249.195.res-cmts.senj.ptd.net) |
| 12:04.16 | m_tadeu | hi...I'm looking for which privileges are needed for asterisk in a mysql database...when googling, all I can find is to grant all privileges, but are all really needed? |
| 12:08.25 | *** join/#asterisk benngard (~mabe@213.88.138.230) |
| 12:22.56 | jacc0 | @m_tadeu: I think only SELECT & UPDATE are needed |
| 12:23.15 | m_tadeu | jacc0: thanx a lot |
| 12:24.25 | *** join/#asterisk bn-7bc (bjarne@macbook-pro.lan-sx.noare-1.holmedal.net) |
| 12:24.29 | jacc0 | @m_tadeu: if you are talking about realtime installation (reading accounts from a table) |
| 12:25.02 | *** join/#asterisk dimm (~appleworm@unaffiliated/dimm) |
| 12:25.02 | jacc0 | @m_tadeu: if you are using the database to store cdr and voicemail you would need insert also |
| 12:26.11 | m_tadeu | jacc0: thanx again....apreciate your help |
| 12:26.49 | *** join/#asterisk Dovid (~Dovid@213.8.121.90) |
| 12:27.06 | Dovid | any one have a VPN connection to Verizon? |
| 12:27.22 | *** join/#asterisk daxt (~daxt@112.135.81.142) |
| 12:28.06 | daxt | hi guys , i am trying to install asterisk and got http://pastebin.com/ubc52AJ0 error , Can somebody help me to locate the error ? |
| 12:30.36 | *** join/#asterisk Karen_m (~karen@d66-222-153-231.abhsia.telus.net) |
| 12:31.53 | Karen_m | I want to buy a phone number (one that spells something), run asterisk to call forward and possibly record messages if there is no answer. Is there a way to get a phone number and a voip provider really cheap? like 10 bucks a month or does it require a sizeable investment? |
| 12:34.01 | leifmadsen | Karen_m: VOIP providers are crazy cheap |
| 12:34.11 | leifmadsen | Karen_m: it basically comes down to how many minutes per month |
| 12:34.25 | leifmadsen | unlimitel.ca (for example) is 1 cent per minute |
| 12:34.49 | leifmadsen | you pay $50 pre-paid, and that $50 lasts until you've placed or received enough calls to exhaust it, then it tops back up to $50 |
| 12:35.00 | Karen_m | wow, now when i pick the phone number do they let me own the number or is it always theirs? |
| 12:35.08 | Karen_m | so i cannot port it over to a cell company later if i wanted? |
| 12:35.15 | leifmadsen | depending on the country (Unlimitel is Canada) you can port the number |
| 12:35.35 | leifmadsen | yes, in Canada you can port the number to a cell provider (or vice-versa) |
| 12:35.44 | leifmadsen | I believe the same rules apply in the USA |
| 12:35.56 | jacc0 | I found this error : asterisk[5865]: segfault at 0 ip (null) sp b17bea9c error 4 in asterisk[8048000+1ac000] |
| 12:35.58 | leifmadsen | (may be dependent on geography) |
| 12:36.11 | jacc0 | what could be the cause that asterisk is segfaulting? |
| 12:36.24 | Karen_m | are you sure that is the website? i see nothing about the 1 cent per minute and all that? |
| 12:36.27 | leifmadsen | jacc0: looks like asterisk crashed: https://wiki.asterisk.org/wiki/display/AST/Debugging |
| 12:36.34 | leifmadsen | Karen_m: yes I'm positive |
| 12:36.38 | leifmadsen | I use them all the time |
| 12:36.44 | leifmadsen | oops |
| 12:37.01 | leifmadsen | www.unlimitel.ca |
| 12:37.12 | leifmadsen | not sure why unlimitel.ca brings up something different |
| 12:37.16 | jacc0 | @@liefmadsen: What could be the cause of the segfault? |
| 12:37.30 | Karen_m | leifmadsen, do you ever have troubles where the sound of the call sucks or anything? |
| 12:37.31 | leifmadsen | jacc0: anything |
| 12:37.33 | jacc0 | asterisk logging doesn't show anything wierd |
| 12:37.41 | leifmadsen | Karen_m: only if my network is the problem |
| 12:37.52 | leifmadsen | jacc0: if you got a segfault, asterisk crashed |
| 12:37.54 | Karen_m | like, magic jack, at one point would *suck* ... i'm using skype and I love it, but i want to setup this voip and not have it suck |
| 12:38.00 | leifmadsen | at which point you need to get a backtrace from the core file |
| 12:38.15 | Karen_m | oh my network is solid , using telus (i've had shaw for years and telus beats them all over the place), so thank you I will check that site out |
| 12:38.16 | leifmadsen | Karen_m: then make sure your network doesn't suck -- the problem won't be on the Unlimitel side |
| 12:38.19 | jacc0 | I have a core dump file |
| 12:38.28 | leifmadsen | jacc0: look at the link I gave you |
| 12:38.34 | Karen_m | leifmadsen, does the company allow you to choose your own numbers or do they just give you one? |
| 12:39.23 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 12:39.34 | leifmadsen | Karen_m: usually just give you one, but you may be able to request a vanity number for a fee |
| 12:39.37 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 12:39.38 | leifmadsen | I've never tried |
| 12:40.03 | leifmadsen | Karen_m: FYI -- Unlimitel operates only in Canada, so you'll get a Canadian phone number (or you can request a toll-free) |
| 12:40.05 | Karen_m | leifmadsen, what protocol would I want ? iax2 or sip? |
| 12:40.08 | leifmadsen | sip |
| 12:40.30 | Karen_m | i have never setup asterisk, is there anything I can do that will make them very upset if i make a mistake? |
| 12:40.45 | leifmadsen | not really |
| 12:40.49 | leifmadsen | they are asterisk friendly |
| 12:40.58 | Karen_m | leifmadsen, also I see "voice mail 4 dollars a month" .. can't asterisk do that for me? |
| 12:41.12 | leifmadsen | you'll just get rejected if your auth is wrong |
| 12:41.25 | leifmadsen | Karen_m: yes it will -- that's for people who need voicemail hosted externally from their system |
| 12:41.31 | leifmadsen | you don't need anythign but the DID and service |
| 12:41.54 | Karen_m | are they the cheapest? |
| 12:42.41 | leifmadsen | they are the cheapest with excellent quality that I've found -- you may be able to find cheaper providers, but at 1c/minute that's pretty fuckin' cheap. You might be able to find something cheaper, but I can't say that the quality will be as good |
| 12:42.59 | leifmadsen | there is a reason I only deploy with them (in Canada -- in the USA I use Bandwidth.com) |
| 12:43.24 | Karen_m | lastly, do they have a minimum use per month? what if i only use 30 minutes a month, do they ever expire your 50 bucks after 4 months or something? |
| 12:44.23 | *** join/#asterisk seraphie (~erin@207.98.195.107) |
| 12:44.45 | Karen_m | do I need an analog telephone adapter? |
| 12:45.09 | Karen_m | I don't think I need anything do i? can't i use the computer mic/headphones? |
| 12:45.28 | leifmadsen | Karen_m: just use the 30 mins per month -- the $50 is not based on time |
| 12:45.51 | leifmadsen | DID is $3.50 per month though, but that is the only monthly charge |
| 12:46.07 | leifmadsen | the rest is based on usage (the $3.50 just comes out of your $50 pre-pay credit) |
| 12:46.18 | leifmadsen | yes, you can just use a mic/headphones with a softphone on a laptop or whatever |
| 12:46.21 | leifmadsen | you don't need any adapters |
| 12:46.26 | leifmadsen | ~thebook |
| 12:46.26 | infobot | thebook is probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://astbook.asteriskdocs.org, or http://ofps.oreilly.com |
| 12:46.33 | leifmadsen | at this point you're ready for documentation :) |
| 12:48.06 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 12:48.10 | leifmadsen | infobot: no, thebook is Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342. |
| 12:48.10 | infobot | okay, leifmadsen |
| 12:48.22 | leifmadsen | when we have the HTML version back online then I can update a link to it |
| 12:49.56 | Karen_m | my boyfriend sent them a letter |
| 12:49.58 | leifmadsen | which i should email o'reilly about now... |
| 12:50.02 | leifmadsen | sent who a letter? |
| 12:50.07 | Karen_m | email, hopefully they respond |
| 12:50.10 | Karen_m | unlimitedtel |
| 12:50.16 | benngard | can u in someway write special character in a string like prepending with a \ or something else |
| 12:50.17 | leifmadsen | about what? |
| 12:50.35 | leifmadsen | benngard: yes you can escape with a backslash |
| 12:50.41 | Karen_m | about choosing a number or getting a list of possible numbers and letting me select one .. |
| 12:50.46 | leifmadsen | gotcha |
| 12:51.01 | Karen_m | i don't wnat.. 719-9846 kind of thing, i would prefer.. 770-1000 or something |
| 12:51.07 | tzanger | actually I will be switching away from Unlimitel now that they have bene bought by Primus |
| 12:51.21 | Karen_m | tzanger, where are you switching to? |
| 12:51.22 | tzanger | I was a big supporter of Stephan but Primus and I are not friends |
| 12:51.28 | tzanger | Karen_m: voip.ms |
| 12:51.33 | Karen_m | do they do canada? |
| 12:51.35 | tzanger | yep |
| 12:51.39 | tzanger | (I am Canadian) |
| 12:51.45 | benngard | i would like to insert decimal 229 in a string, how can i escape that? |
| 12:51.49 | tzanger | dammit, I just earwormed myself with those stupid commercials |
| 12:52.31 | benngard | or hex e5 |
| 12:52.34 | leifmadsen | tzanger: heh, well I met with the big boss at Primus and Stephan is staying on, and the network isn't changing at Unlimitel |
| 12:52.45 | leifmadsen | I'll be staying on with them until I have a reason not to |
| 12:53.10 | tzanger | leifmadsen: that's good news, but all the same, I won't support Primus |
| 12:53.16 | tzanger | if I could avoid supporting Rogers I would do so as well |
| 12:53.18 | Karen_m | what is wrong with primus? |
| 12:53.31 | tzanger | speaking of which, teksavvy cable is ordered. :-) |
| 12:53.55 | tzanger | Karen_m: I have had nothing but billing fuckups with them back before I got VOIP and dropped telcos in general |
| 12:54.32 | leifmadsen | tzanger: heh ya same (re: Rogers) -- I switched to WIND and only use Rogers for cable now... |
| 12:54.43 | leifmadsen | and not teksavvy cable in my area :( |
| 12:54.45 | tzanger | it's anecdotal, but I've got a few friends who used them and had similar stories |
| 12:54.51 | tzanger | leifmadsen: really? they just opened up in Kitchener |
| 12:54.58 | jacc0 | 'core show locks' doesn't work in asterisk 1.8; what command shoud I use in 1.8? |
| 12:55.02 | tzanger | although I have heard rumour that Rogers is playing dirty tricks |
| 12:55.03 | leifmadsen | yep, I'm in a brand new area in Caledon though |
| 12:55.10 | tzanger | ah |
| 12:55.11 | leifmadsen | tzanger: what else is new? :) |
| 12:55.24 | leifmadsen | jacc0: it works if you enable it in menuselect |
| 12:55.27 | Dovid | anyone here have a Trunk with Verizon ? |
| 12:55.48 | tzanger | yeah I use Rogers for the cell, billing fuckups there too but until I am comfortable that Bell's HSPA network is everywhere and beyond where I can get GPRS even then I won't switch |
| 12:55.48 | tzafrir | jacc0, have you enabled that debugging option? |
| 12:55.52 | Karen_m | so if you only call local, do these people consider it long distance? |
| 12:56.06 | Karen_m | if you have a calgary number, and call calgary numbers, do they charge you 1 cents per minute? |
| 12:56.13 | *** join/#asterisk JonathanRose (~jonathan@nat/digium/x-vyetkhylgyhehqzx) |
| 12:56.14 | leifmadsen | jacc0: per the wiki, in the big orange box with the exclaimation point at the start: "You need DEBUG_THREADS enabled in the Compiler Flags menu of menuselect. Be sure you recompile, install, and restart Asterisk prior to running 'core show locks'." |
| 12:56.25 | jacc0 | k |
| 12:56.26 | jacc0 | ty |
| 12:56.31 | *** join/#asterisk mintos (~mvaliyav@114.143.162.65) |
| 12:56.32 | jacc0 | will start recompiling |
| 12:56.57 | jacc0 | astcanary restarts asterisk is asterisk stops or am I mistaking? |
| 12:57.20 | tzanger | leifmadsen: not only do you write the documentation, but you read it to them too! |
| 12:57.23 | jacc0 | or should I use safe_asterisk in combination with astcanary? |
| 12:57.28 | leifmadsen | tzanger: I'm kind of a big deal |
| 12:57.42 | tzanger | Karen_m: I pay per minute. there is no LD anymore. |
| 12:58.06 | leifmadsen | tzanger: +1 |
| 12:58.20 | leifmadsen | Karen_m: it doesn't matter where you call, it is 1c/minute |
| 12:58.25 | Karen_m | if i use the asterisk for incoming only, when I make outgoing on my skype is there a way to list the caller id to match so that people think I am calling back from the number they called? |
| 12:58.27 | leifmadsen | there is no local vs long disance |
| 12:58.47 | leifmadsen | Karen_m: just set the CALLERID(num) to whatever you want |
| 12:58.54 | leifmadsen | only national vs international |
| 12:59.00 | Karen_m | also, if I have 2 numbers, one local 403-* and one 800*, does asterisk know which number they called? I want to call them back on the same number |
| 12:59.33 | Karen_m | i was going to get a local phone from telus again and they wanted something like 24 dollars a month. Lol this is so fun |
| 12:59.33 | tzanger | that's one thing I have to check with voip.ms... if I can set my outgoing callerid however I want/need |
| 12:59.35 | Karen_m | and exciting |
| 12:59.51 | Karen_m | leifmadsen, unlimitedtel does not mind you set up your caller id to whatever? |
| 12:59.53 | tzanger | Karen_m: yeah, I love it when they call all breathless to tell me about hte latest pricing |
| 13:00.00 | tzanger | "Can you beat $0.00/min?" <silence> |
| 13:00.01 | leifmadsen | Karen_m: you can enable that functionality, yes |
| 13:00.02 | schmidts | Karen_m asterisk will not recognize your outgoing number, you have to tell it |
| 13:00.19 | Karen_m | if (800, callerid = 800-123-1234) else if (403.. ) { caller_id = "..." } ? |
| 13:00.29 | leifmadsen | basically yes |
| 13:00.35 | Karen_m | schmidts, but does it recongize if you have 2 numbers setup, which the call came from? |
| 13:00.39 | leifmadsen | yes |
| 13:00.47 | leifmadsen | when the call comes in, it'll be on separate accounts |
| 13:00.52 | Karen_m | if someone calls on a 403-* number, and I call them back with 800-* as the caller id, they won't answer :) |
| 13:00.52 | leifmadsen | each account can be handled any way you want |
| 13:01.31 | Karen_m | so for less than 10 dollars a month i can get my own 403* number, plus an 800 number and only use per minute rates... this is amazing, why doesn't everyone do it? |
| 13:01.32 | schmidts | Karen_m if you have two different accounts, yes. If you only use one you have to tell asterisk which one you want to use ;) |
| 13:01.55 | Karen_m | oh so the key is, get unlimitedtel to setup 2 different accounts for the numbers? |
| 13:02.54 | Karen_m | voip.ms is more expensive than unlimitedtel, so I am going to unlimited :) |
| 13:03.20 | Karen_m | i expect, at most, 30 minutes a month on these phone numbers.. |
| 13:03.44 | schmidts | can someone help me with cisco 7940 Phone with SIP firmware? |
| 13:03.51 | leifmadsen | Karen_m: each account is a separate DID |
| 13:04.34 | schmidts | Leifmadsen i allways forget these DID thing, in europe we didnt have something like DID or no DID lines ;) |
| 13:04.43 | leifmadsen | in europe it's a DDI |
| 13:04.52 | *** join/#asterisk kamikazemicrowav (~kamikazem@seraph.contegix.com) |
| 13:04.59 | leifmadsen | you don't have phone numbers in europe? :) |
| 13:05.31 | schmidts | :D no we still use smoke to communicate |
| 13:05.43 | leifmadsen | that's kinda fun :) |
| 13:05.56 | jaytee | I like Flowroute, they're cheap and I've been very satisfied with their reliability. |
| 13:06.10 | Karen_m | what is the best way to get a phone working with asterisk? what hardware do you need? I'm currently using a zoom box 4900 for skype. |
| 13:06.11 | leifmadsen | I like coffee |
| 13:06.17 | jaytee | and their user account portal doesn't suck |
| 13:06.23 | jaytee | I like coffee too |
| 13:06.30 | schmidts | leifmadsen sounds like a project, control smoke pipes with asterisk like chan_smoke :D |
| 13:06.30 | leifmadsen | Karen_m: best is relative.... but I like using the SPA3102 or a Polycom SIP phone |
| 13:06.36 | jaytee | well....actually I'm nuts about coffee |
| 13:06.45 | leifmadsen | I'm out of coffee and I'm going nuts |
| 13:06.48 | leifmadsen | hazelnuts |
| 13:06.51 | jaytee | I've spent over 30 bucks a pound for rare coffees |
| 13:07.01 | leifmadsen | that doesn't seem unreasonable at all |
| 13:07.09 | *** join/#asterisk mawhii (~mawhii@170.220.119.70.cfl.res.rr.com) |
| 13:07.30 | Karen_m | leifmadsen, which polycom model do you have? |
| 13:08.10 | jaytee | yeah, considering some gourmet stores around here sell stale Jamaican Blue Mountain for 50 bucks a pound nad I can get it fresh vacuum sealed from the distributor for 36 bucks a pound roasted or 24 bucks a pound in the green bean and roast it myself. |
| 13:08.41 | jaytee | Only thing in a kitchen that smells better than coffee beans roasting is bread baking. |
| 13:10.05 | psilikon | jaytee, you may be right my friend |
| 13:10.53 | leifmadsen | Karen_m: Polycom 335 |
| 13:11.17 | leifmadsen | jaytee: what about coffee bread baking? |
| 13:11.23 | leifmadsen | or coffee cake?! |
| 13:11.36 | leifmadsen | jaytee: how do you like your coffee? |
| 13:12.10 | *** join/#asterisk mheadd (~Adium@c-69-141-4-36.hsd1.de.comcast.net) |
| 13:14.08 | jaytee | I like my coffee with half&half and 1 sugar |
| 13:14.26 | jaytee | and sometimes I might make cardamom coffee |
| 13:14.27 | psilikon | jaytee, why ruin a good thing? |
| 13:14.36 | jaytee | which is a middle eastern thing |
| 13:15.32 | jaytee | I'll drink it black with nothing if I'm testing out a new roast to see how it tastes. Roasting is kind of science and an art. It's fun :-) |
| 13:15.44 | leifmadsen | jaytee: I like mine crisp |
| 13:16.32 | jaytee | I like a good columbian from Huila province medium roasted. The best ones have a nice malted chocolate finish. |
| 13:18.28 | jaytee | there was a small farm in Kona that produced awesome bean but I haven't been able to get any in almost 2 years :-( |
| 13:19.33 | *** join/#asterisk tasca (~tasca@mail.moldurarte.com.br) |
| 13:20.21 | *** join/#asterisk ariel_ (~chatzilla@unaffiliated/abatista) |
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| 13:20.27 | jenna | excuse me is sruffell around here somewhere ? |
| 13:21.24 | leifmadsen | I don't think sruffell uses IRC all that much |
| 13:21.30 | leifmadsen | might have to email him directly |
| 13:21.56 | jenna | leifmadsen, oh okay |
| 13:22.32 | jenna | leifmadsen, btw are you affiliated with asterisk.org . Just wondering if any official staff hangs around here |
| 13:23.59 | leifmadsen | jenna: I'm more affiliated with issues.asterisk.org than with www.asterisk.org (I'm not sure what you're asking yet :)) |
| 13:24.34 | jenna | leifmadsen, was following this https://issues.asterisk.org/view.php?id=18992 |
| 13:25.02 | leifmadsen | waits for a question |
| 13:25.48 | jenna | leifmadsen, and wondering when would the official binary packages (rhel/centos) be patched/updated with this patch. |
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| 13:32.40 | jenna | leifmadsen, any ideas there ? |
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| 13:38.25 | hensema | hmmm weird, I thought asterisk didn't need any external process to read mp3 as MOH? |
| 13:38.29 | hensema | yet, I get: |
| 13:38.30 | hensema | MOH: exec failed: No such file or directory |
| 13:38.35 | hensema | (asterisk 1.8) |
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| 13:44.26 | Iiiak | Hello, I search a benchmark software you know one ? |
| 13:47.15 | *** join/#asterisk bent_screwdriver (~socain00@c-71-199-214-165.hsd1.fl.comcast.net) |
| 13:47.21 | krion | user, it's a great one with multiple scenario |
| 13:47.27 | krion | ;) |
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| 13:51.21 | tzafrir | hensema, please pastebin the output of: moh show classes |
| 13:53.13 | hensema | tzafrir: http://pastebin.com/DU34MHDe |
| 13:54.31 | tzafrir | hensema, if the mode is not 'files', asterisk tries to execute a program in order to play music |
| 13:56.09 | hensema | ok, now I get: [Apr 11 15:55:19] WARNING[13039]: file.c:644 ast_openstream_full: File /var/lib/asterisk/mohmp3/Funky Dunky does not exist in any format |
| 13:56.16 | hensema | (I do have asterisk-addons installed) |
| 13:57.14 | *** join/#asterisk bent_screwdriver (~socain00@c-71-199-214-165.hsd1.fl.comcast.net) |
| 14:00.33 | hensema | hmmmm, probably it's just opensuse being pedantic about mp3 |
| 14:04.12 | *** part/#asterisk benngard (~mabe@213.88.138.230) |
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| 14:23.58 | jaytee | I just read an article about Red Hat now incorporating patches into the kernel. This may not bode will for CentOS. |
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| 14:27.30 | hensema | actually for centos it may not matter much: they get the fully patched kernel and can simply compile and distribute it |
| 14:28.05 | hensema | it's bad news for companies trying to sell support for redhat because they don't fully understand the kernel they are supposed to support |
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| 14:29.08 | jacc0 | <PROTECTED> |
| 14:29.14 | jacc0 | <PROTECTED> |
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| 14:49.50 | devmod | Is there a way to set maxcallbitrate per call by evaluation some condition? |
| 14:51.47 | kuku | For the first time in 6 years working with asterisk, a new client has it running on a VM machine. The Vmware tools are installed. I moved it to a host with lots of free ram and fast scsi disks. The problem is the voice is still choppy. The choppines is even there when calling from extension to extension ( same location/switch ) so its not the sip TRUNK ( cbeyond ). Any suggestions ? |
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| 14:52.55 | *** mode/#asterisk [+o russellb] by ChanServ |
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| 14:58.47 | tzafrir | kuku, what kernel/distro? What asterisk? |
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| 15:01.06 | pecenipicek | Hokay folks, i'm having some problems getting asterisk to start as a non-root user on boot-time. |
| 15:01.50 | pecenipicek | i've followed the instructions from the book for that, and after a reboot and running ps aux | grep asterisk, it returns that the process has been actually started by root |
| 15:02.42 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 15:02.45 | pecenipicek | i'm running asterisk 1.8.3.2 on Debian, 2.6.32-5-686 kernel. |
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| 15:05.05 | kuku | tzafrir: centos. Tried 2.6.18-53 el5vm kenrnel. asterisk 1.4.36 |
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| 15:46.53 | chazzam | ~newbook |
| 15:46.54 | infobot | Asterisk: The Definitive Guide. A preview of the upcoming release is available at http://ofps.oreilly.com/titles/9780596517342/. The book is being released under a Creative Commons license: http://creativecommons.org/licenses/by-nc-nd/3.0/us/. Please consider supporting the authors generosity by purchasing a copy at http://oreilly.com/catalog/9780596517342. |
| 15:47.29 | chazzam | will that be available for viewing online again or will it be buy only now? |
| 15:47.37 | breardo | okay |
| 15:47.43 | breardo | got my 34-minute phone stuff figured out |
| 15:47.54 | leifmadsen | chazzam: it'll be online again as soon as the data is entered back into subversion |
| 15:48.02 | breardo | of course, it had nothing to do with Asterisk..but the problem was most visible as one-way-audio and/or dropped calls |
| 15:48.04 | chazzam | ahhh |
| 15:48.08 | leifmadsen | o'reilly is taking all the changes and fixes and manually entering them back in, so it'll take some time |
| 15:48.13 | chazzam | ok |
| 15:48.17 | chazzam | cool |
| 15:48.28 | leifmadsen | it's still creative commons, so it'll for sure be available as a web page again |
| 15:48.50 | breardo | I had a Cisco switch that thought it was the STP root of a few vlans, but it was not... for some reason, it would not accept that another switch was the STP root; disabling portfast and changing the STP 'weight' of the vlans solved the problem |
| 15:49.06 | breardo | we did not see any STP-related errors in debug output.. so it was a hard one to track down |
| 15:49.34 | chazzam | is hoping for it to come out in an ebook deal of the day... |
| 15:50.19 | leifmadsen | chazzam: sooooon :) |
| 15:50.26 | russellb | chazzam: you could ask your employer to buy it for you :-p |
| 15:50.32 | russellb | (or just wait until it's back on the web) |
| 15:50.37 | breardo | I work at a college, can I get a desk copy ? :) |
| 15:51.20 | russellb | sure |
| 15:55.14 | drmessano | Can I get 100 copies? |
| 15:55.16 | chazzam | heh |
| 15:56.14 | drmessano | I want a PDF of an autographed copy |
| 16:00.11 | pecenipicek | autoprovisioning doesnt work without stuff in users.conf? |
| 16:00.20 | *** join/#asterisk casix (~casix@145.164.219.87.dynamic.jazztel.es) |
| 16:00.22 | casix | hello |
| 16:01.44 | casix | anyone know how can I match a quotes " in a regexp : ?? I have tried with " or escaping \" or with double scaping \\" but don't work. I'm using asterisk 1.4.26.2 thanks |
| 16:06.08 | jaytee | I was thinking of getting a Kindle edition and a print edition just in case we get hit with an EMP weapon by China. |
| 16:09.17 | jaytee | hmmm, I can't get back into ##vyatta channel, it says invite only? |
| 16:10.04 | jaytee | maybe if I quit and reconnect |
| 16:10.59 | *** join/#asterisk SiNGLer (~singler@78-60-139-121.static.zebra.lt) |
| 16:11.10 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 16:11.59 | leifmadsen | casix: \\\" |
| 16:12.04 | leifmadsen | you have to escape the escape |
| 16:12.15 | leifmadsen | (not necessary on 1.6.2+) |
| 16:12.17 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 16:12.17 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 16:16.57 | casix | leifmadsen: it doesn't work... http://pastebin.com/cvaWezh7 |
| 16:17.34 | Qwell | tzafrir: ping? |
| 16:17.39 | tzafrir | Qwell, pong |
| 16:17.59 | Qwell | tzafrir: Are you aware of any issues compiling xpp stuff in the new kernel in CentOS 5.6? |
| 16:18.14 | tzafrir | Frankly, I didn't yet test it |
| 16:18.22 | leifmadsen | casix: then I have no idea |
| 16:18.42 | casix | leifmadsen: ok, thanks :) |
| 16:19.18 | Qwell | tzafrir: card_bri.c > xpd.h > linux/device.h:407: error: expected identifier or '(' before 'const' |
| 16:19.34 | Qwell | wasn't sure if it was just my install or not. I'm betting not. |
| 16:20.06 | tzafrir | the new bug report? OK, looking into it |
| 16:20.27 | Qwell | oh, yeah, I guess that's the same as 18992 |
| 16:21.00 | *** join/#asterisk jakent (~john@2001:470:8:1fc:226:8ff:fedd:93f6) |
| 16:21.03 | tzafrir | ah. It's the same dev_name issue as https://issues.asterisk.org/view.php?id=19097 |
| 16:22.06 | Qwell | 19097 is a different hing |
| 16:22.15 | Qwell | that's an AsteriskNOW package issue |
| 16:25.22 | *** join/#asterisk serafie (~erin@nat/digium/x-kogfugxninhyhodr) |
| 16:29.47 | *** join/#asterisk bipolar (~bipolar@offsitesysadmin.com) |
| 16:31.47 | *** join/#asterisk dimm1 (~appleworm@unaffiliated/dimm) |
| 16:39.52 | *** join/#asterisk Andrew__M (60fa1394@gateway/web/freenode/ip.96.250.19.148) |
| 16:40.03 | Andrew__M | Hello All! |
| 16:40.31 | carrar | All is here?!!! |
| 16:40.46 | Andrew__M | carrar: awesome! |
| 16:42.06 | Andrew__M | Q: I am trying to pull a value I set up in sip.conf under "accountcode = 6262" with exten => o,1,NoOp(${CDR(accountcode)}) in the dialplan, but getting an empty string. What am I missing? |
| 16:42.54 | Andrew__M | . |
| 16:43.21 | Andrew__M | I want to use accountcode for zero-out value on a per-user basis. |
| 16:46.14 | carrar | perhaps you are not setting it |
| 16:46.15 | tzafrir | Andrew__M, where did you write it? under [general]? |
| 16:46.15 | Andrew__M | If you call user A, the VM zero-out will be to one extension, but if you call user B, the VM zero-out will be to another extension. |
| 16:46.58 | Andrew__M | I entered it under a user, right under the line that specifies what mailbox to use in sip.conf. |
| 16:50.32 | Andrew__M | So I call the user, the VM picks up, I zero-out, and get nothing, because it does not pull my entry. |
| 16:51.45 | Andrew__M | Basically, I want to specify zero-out on a per-user basis, and have a simple way to administer it. |
| 16:52.36 | Andrew__M | Someone gave me the idea of hijacking a channel variable. |
| 16:53.35 | Andrew__M | If I could make up my own entry in sip.conf, then have a ay of accessing it from the dialplan, I would do that instead. |
| 16:53.50 | Andrew__M | ay= way |
| 16:55.23 | Andrew__M | But the only other ways I can think of achieving that is having separate entries in the dialplan for each extension, or using the AstDB. |
| 16:55.30 | paulc | Andrew__M: Why not use AstDB? |
| 16:55.34 | paulc | hehe yeah - there we go |
| 16:55.45 | Andrew__M | Either of which is hard to administer. |
| 16:55.51 | paulc | it's nicer because then you can build dialplan logic to allow users (or your admin) to change the destination |
| 16:56.06 | paulc | I'd argue that AstDB is easier to administer than having to change hard coded variables in sip.conf |
| 16:57.11 | Andrew__M | I agree, but when you add a new user, almost everything is set up in sip.conf and voicemail.conf, so why not this...? |
| 16:59.44 | *** join/#asterisk tasca (~tasca@mail.moldurarte.com.br) |
| 17:00.48 | Karen_m | what does this mean? "VoIP.ms (Swiftvox Inc) Currently does not support Dialer and Call Center traffic on its termination services" |
| 17:01.08 | Qwell | Karen_m: "Stop sending spam through our service." |
| 17:01.38 | Karen_m | does it mean that the service is gimped in a way? I don't spam... |
| 17:01.43 | *** join/#asterisk cyford (~cyford@adsl-074-188-021-226.sip.asm.bellsouth.net) |
| 17:02.10 | Qwell | Stop using them to sell car warranties. |
| 17:02.14 | Qwell | basically |
| 17:02.15 | _Corey_ | Translation: They don't want you as a customer if you're a call center... :( |
| 17:03.02 | Karen_m | Oh not a call center or anything, ok |
| 17:03.58 | cyford | i do anyone know where i can find any sample survey scripts for asterisk? |
| 17:04.13 | cyford | i = hi |
| 17:05.40 | Andrew__M | <paulc> You are probably right. I will use VMAuthenticate and allow changing own zero-out. |
| 17:05.48 | *** join/#asterisk coppice (~chatzilla@9.160.232.220.dyn.pacific.net.hk) |
| 17:06.34 | Andrew__M | <paulc> Thanks! |
| 17:06.39 | paulc | Andrew__M: Yeah, that's a good solution :-) |
| 17:07.17 | paulc | cyford: I did a small survey recently - a couple of digit questions plus a recorded answer, results stored in MySQL via CURL calls in the dialplan... |
| 17:11.26 | Karen_m | does the owner of voip.ms idle in here? |
| 17:12.19 | *** part/#asterisk bn-7bc (bjarne@macbook-pro.lan-sx.noare-1.holmedal.net) |
| 17:13.52 | psilikon | Karen_m, that would be pretty cool. I use voip.ms and I like it a lot. |
| 17:15.09 | *** join/#asterisk Diffen (~diffen@109.58.36.107.bredband.tre.se) |
| 17:15.40 | Karen_m | psilikon, does voip.ms load for you? every page click takes about 2+ minutes, their database is obviously locked up :) |
| 17:20.08 | Qwell | or they blackholed you for spam :p |
| 17:20.29 | Karen_m | Qwell, blackholed me for spam; ok lol |
| 17:20.41 | Karen_m | i think you missed the part where I'm brand new to voip and never spammed, trying to learn |
| 17:20.48 | Karen_m | read up in the channel log |
| 17:24.58 | *** join/#asterisk sahX (~sahX@4.53.128.213) |
| 17:25.43 | *** join/#asterisk sahX (~sahX@4.53.128.213) |
| 17:27.35 | psilikon | Karen_m, lemme try just a sec |
| 17:28.09 | psilikon | Karen_m, yeah it works like a champ for me. |
| 17:28.17 | Karen_m | it seems to have fixed/resolved itself.. maybe they were backing up mysql or their database |
| 17:31.46 | breardo | hey qwell, got that 34-minute issue resolved.. |
| 17:31.55 | Qwell | breardo: yeah I saw |
| 17:31.57 | Qwell | crazy |
| 17:32.01 | breardo | ok, just making sure |
| 17:32.05 | breardo | yeah it was an odd one.. |
| 17:34.23 | nosbig | What is the current best practice for integrating Asterisk 1.8 and Festival? Modify festival.scm and run unpatched using the Festival() app, run a patched Festival copy and use the Festival() app, or use text2wave and use Playback() to play back the audio? Or is there something I might be missing? |
| 17:34.58 | *** join/#asterisk iprouteth0 (~iprouteth@unaffiliated/iprouteth0) |
| 17:44.57 | pecenipicek | is in love with Yealink T20 phones... |
| 17:47.13 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 17:47.35 | pecenipicek | now to tie in the whole thrice damned thing to the autoprovisioning... |
| 17:48.06 | iprouteth0 | I've yet to try asterisk autoprovisioning. |
| 17:48.14 | iprouteth0 | Anyone tried the Nortal 1535s yet? |
| 17:49.20 | pecenipicek | i'm currently just provisioning the phone via tftp, didnt touch asterisks provisioning yet |
| 17:50.18 | iprouteth0 | I've only got one Polycom IP450 to mess with, but havent yet tried autoprovisioning. Just got finished a bit ago upgrading to 1.8 and integrating directly with Google Voice |
| 17:50.24 | iprouteth0 | Works flawlessly |
| 17:51.26 | pecenipicek | heh. |
| 17:52.08 | pecenipicek | i'm trying to solve this relatively painlessly, due to the simple fact that i'm supposed to oversee deployment of 100 or so of the buggers sometime next week. |
| 17:52.53 | iprouteth0 | Yeah, I can see wanting to prove it out first. Wish I had opportunities for doing deployments and VoIP installs..... |
| 17:53.41 | pecenipicek | now for you "knowledgeable types", here's a doozy. How would you go about tying together numbers with phones, when phone names arent the numbers they'll be given? Database approaches also welcome. |
| 17:53.50 | iprouteth0 | Still improving my VoIP and networking skills |
| 17:54.28 | *** part/#asterisk tasca (~tasca@mail.moldurarte.com.br) |
| 17:54.40 | *** join/#asterisk usc911 (~ben@78-105-116-233.zone3.bethere.co.uk) |
| 17:55.22 | usc911 | Hey guys, just wondering if anyone knows how to reset the unavailable voicemail greeting back to default? |
| 17:55.34 | pecenipicek | iprouteth0: i got dragged into asterisk by my dear father half a year ago, then he dumped the whole thing over to me last month. The whole job is basically "develop an asterisk solution to replace our current CCM with possibilities of expansion to cover all our retail stores as well" |
| 17:56.25 | pecenipicek | halfway through it turned from that to "develop a solution that will work with concurrently with our CCM, for retail store use. also, catching transfers in the CDR is a must" |
| 17:56.53 | iprouteth0 | oh wow. It's been awhile since I last worked with CCM. I'm much more familiar with CCME. Right now for me Asterisk is also a side project |
| 17:56.56 | pecenipicek | and, as a PS... "be ready to deploy 500 of these phones some time during the next 4 months" |
| 17:57.19 | iprouteth0 | I am studying for CCNA Voice... I'm doing that to keep moving forward and maintain my current CCNA |
| 17:57.27 | pecenipicek | i dont have any access to anything related to the CCM. 4.1 is the version if i remember correctly. |
| 17:57.33 | pecenipicek | hah. |
| 17:57.55 | pecenipicek | i got no certifications to speak of other than "he's the guy that can do that" |
| 17:58.09 | iprouteth0 | Plus I started network monitoring position with my company not too long ago and have been learning five different class 5 telco switchs.... damn full plate |
| 17:58.44 | pecenipicek | fun. |
| 17:59.03 | iprouteth0 | I've only lightly used CCM4. CCM6 is what I'm most familiar with on the Full call manager. CCME is nice cause I can emulate the system without having any hardware |
| 17:59.45 | pecenipicek | ccm6 is installable via VMWare's stuff if i remember correctly? |
| 17:59.55 | *** part/#asterisk usc911 (~ben@78-105-116-233.zone3.bethere.co.uk) |
| 17:59.56 | iprouteth0 | correct. |
| 18:00.21 | iprouteth0 | although It can be terrible difficult to get working in VMware sometimes |
| 18:00.29 | pecenipicek | never got it to recognise any phones tho... |
| 18:00.42 | pecenipicek | probably fucked up something on the network bridging level. |
| 18:00.50 | iprouteth0 | CCME is much easier to work with when having zero equipment |
| 18:01.06 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 18:01.11 | pecenipicek | well, i actually got it to install and work in VirtualBox, not VMWare's stuff. |
| 18:01.12 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 18:01.36 | iprouteth0 | Really??? I've never gotten CCM6 to work in Virtualbox. I thought only VMware could do it |
| 18:01.48 | iprouteth0 | for CCME I use GNS3 |
| 18:02.09 | pecenipicek | you have to change something in the text file for the VM. |
| 18:02.13 | pecenipicek | gimmie a moment. |
| 18:02.18 | iprouteth0 | One of these days I'd like to connect a virtual call manager express router to my asterisk server |
| 18:02.38 | iprouteth0 | I'm very interested in that info |
| 18:02.49 | pecenipicek | http://ubuntuforums.org/showthread.php?t=1029144 |
| 18:02.52 | pecenipicek | i went by that. |
| 18:03.18 | pecenipicek | lemme check what i used for the wm itself. |
| 18:06.01 | pecenipicek | 1024 MB ram, PIIX3 chipset, enable IO APIC checked, hardware clock in UTC time checked, 2 cpu's, PAE/NX checked, both the options on the acceleration tab ticked, and under networking, put the adapter type to Intel PRO/1000 MT Server |
| 18:06.11 | pecenipicek | if it helps you any, cheers :D |
| 18:09.19 | *** join/#asterisk gopal (~chatzilla@61.12.17.170) |
| 18:14.06 | iprouteth0 | thanks! AFK for now |
| 18:15.25 | *** join/#asterisk brainiac (~brainiac@208.86.215.38) |
| 18:17.35 | *** join/#asterisk teathsch (~chatzilla@207.7.97.18) |
| 18:18.15 | pecenipicek | no worries. |
| 18:21.06 | *** join/#asterisk b14ck (~b14ck@cpe-72-129-70-245.socal.res.rr.com) |
| 18:21.53 | *** join/#asterisk Kobaz (~kobaz@its.kobaz.net) |
| 18:21.55 | Kobaz | holy hell |
| 18:21.57 | Kobaz | [2011-04-11 14:21:34] WARNING[3713]: format_wav_gsm.c:220 update_header: Unable to find our position |
| 18:22.01 | Kobaz | [2011-04-11 14:21:34] WARNING[3600]: format_wav_gsm.c:220 update_header: Unable to find our position |
| 18:22.04 | Kobaz | [2011-04-11 14:21:34] WARNING[2964]: format_wav_gsm.c:232 update_header: Unable to set our position |
| 18:22.07 | Kobaz | i have like millions of those in the logs |
| 18:25.19 | *** join/#asterisk n3hxs (~ed@63.68.135.4) |
| 18:26.06 | *** join/#asterisk Schreiber1337 (cee4b465@gateway/web/freenode/ip.206.228.180.101) |
| 18:26.31 | *** join/#asterisk The_Boy_Wonder (~manbearpi@asterisk/batman-developer/dvossel) |
| 18:27.55 | d_preston215 | Does any one use Cisco 7940/60s and have come across the issue of time and date randomly disappearing? |
| 18:28.17 | Schreiber1337 | I'm running 1.8.3.2, after a few hours of heavy use Asterisk freezes and I see not sip communications in the CLI... what should I be looking at in the log files to find out what it wrong? |
| 18:30.42 | _Corey_ | d_preston215: NTP is a little touchy on those things |
| 18:32.40 | *** join/#asterisk titter (~Justin@c-98-208-153-148.hsd1.fl.comcast.net) |
| 18:33.12 | *** join/#asterisk joshaidan (~brianj@S0106001c1023e838.tb.shawcable.net) |
| 18:33.15 | Kobaz | Schreiber1337: core show locks |
| 18:35.11 | Schreiber1337 | Kobaz: looking in /var/logs/asterisk/messages I don't see that any where.... the system has only been online for a day and the messages files is up to 23mb.... Is that excessive? |
| 18:38.36 | Kobaz | Schreiber1337: it's a console command |
| 18:39.30 | Schreiber1337 | Kobaz: No such command 'core show locks' |
| 18:41.13 | Schreiber1337 | Kobaz: is it an addon? |
| 18:45.49 | Kobaz | no, you'll have to recompile with DEBUG_LOCKS though |
| 18:53.16 | jaytee | do I need special permission to use the Asterisk logo on our company's website if we offer custom Asterisk solutions but aren't a Digium Authorized reseller? |
| 18:57.25 | leifmadsen | jaytee: absolutely |
| 18:57.32 | *** join/#asterisk NightMonkey (debian-tor@pdpc/supporter/professional/nightmonkey) |
| 18:57.38 | leifmadsen | jaytee: talk to malcolmd about it |
| 18:58.52 | jaytee | malcolmd, do you have time for a brief PM chat? |
| 19:00.39 | russellb | jaytee: http://www.digium.com/en/company/view-policy.php?id=Trademark-Policy |
| 19:02.05 | jaytee | thanks russell |
| 19:03.20 | russellb | np |
| 19:10.12 | leifmadsen | ~asterisk-trademark |
| 19:10.40 | Kobaz | [2011-04-11 14:21:34] WARNING[3713]: format_wav_gsm.c:220 update_header: Unable to find our position |
| 19:10.44 | Kobaz | so those errors |
| 19:10.44 | leifmadsen | infobot: asterisk-trademark is reply Information about the use of the Digium and Asterisk trademarks is available at http://www.digium.com/en/company/view-policy.php?id=Trademark-Policy |
| 19:10.44 | infobot | okay, leifmadsen |
| 19:10.51 | Kobaz | are apparently what you get when you run out of disk space |
| 19:11.05 | leifmadsen | Kobaz: running out of disk space on Linux is bad :) |
| 19:11.15 | _Corey_ | d'oh |
| 19:11.24 | leifmadsen | things fail in a spectacular manner |
| 19:12.16 | _Corey_ | usually when my log is filling up with stuff I wrongly assume that BECAUSE my log is filling up then the disk can't be full... |
| 19:18.05 | *** part/#asterisk Poincare (~jefffnode@2001:470:d6b3:4::2) |
| 19:22.41 | Kobaz | well |
| 19:22.50 | Kobaz | when i said log i was actually referring to console output |
| 19:22.59 | Kobaz | the log file was frozen mid-line |
| 19:23.20 | Kobaz | time to bump the disk space watcher script i wrote to be higher on the priority list |
| 19:24.00 | _Corey_ | ahh ;) |
| 19:30.38 | *** join/#asterisk dimm (~appleworm@unaffiliated/dimm) |
| 19:34.56 | *** join/#asterisk retentiveboy (~retentive@74-95-28-33-Atlanta.hfc.comcastbusiness.net) |
| 19:35.01 | *** part/#asterisk retentiveboy (~retentive@74-95-28-33-Atlanta.hfc.comcastbusiness.net) |
| 19:42.31 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v007-216.mobile.uci.edu) |
| 19:47.41 | *** join/#asterisk Defraz (~Defraz@184-155-137-27.cpe.cableone.net) |
| 19:55.49 | *** join/#asterisk Aut0ExeC (~Jack@24.244.156.75) |
| 19:55.54 | Aut0ExeC | anyone here have cisco spa3102? |
| 19:56.07 | Aut0ExeC | or any SPAXXXX ? |
| 19:56.39 | Aut0ExeC | guess not |
| 19:56.41 | Aut0ExeC | :( |
| 19:57.07 | psilikon | Aut0ExeC, yeah I have several |
| 19:57.48 | *** join/#asterisk chandoo (~chandoo@ool-4a5a2824.dyn.optonline.net) |
| 19:57.49 | Aut0ExeC | psilikon: great... please tell me how u stop your caller id from showing the name of your pstn line |
| 19:58.00 | Aut0ExeC | psilikon: and instead show the real caller id name |
| 19:58.29 | Aut0ExeC | mine just shows the number and "pstn" which is what i named my pstn line |
| 19:59.06 | psilikon | Aut0ExeC, what did you enter for callerid in the sip.conf |
| 19:59.20 | Aut0ExeC | i dont have an entry for that |
| 19:59.22 | Aut0ExeC | :( |
| 19:59.26 | Aut0ExeC | should I have? |
| 19:59.34 | psilikon | Aut0ExeC, paste ur sip.conf somewhere |
| 19:59.49 | Aut0ExeC | ok |
| 20:00.03 | psilikon | Aut0ExeC, yeah. All you have to do is add something for callerid in the respective sip.conf section |
| 20:00.04 | *** join/#asterisk sahX (~sahX@4.53.128.213) |
| 20:00.17 | Aut0ExeC | what do you have in yours? |
| 20:00.26 | Aut0ExeC | callerid= ??? |
| 20:01.04 | psilikon | callerid="Tuomas Tammisalo" <1000> |
| 20:01.15 | Aut0ExeC | psilikon: oh no not like that |
| 20:01.29 | psilikon | callerid="Tuomas Tammisalo" <5555551212> |
| 20:01.34 | Aut0ExeC | i mean.. when i receive calls on my local telco line... it shows "pstn" and the number |
| 20:01.42 | Aut0ExeC | i want it to show the actual person name |
| 20:02.25 | psilikon | Aut0ExeC, oh. My bad I misunderstood you. |
| 20:03.11 | psilikon | Aut0ExeC, you want it to just pass the cid thru and not get changed by the spa? |
| 20:03.48 | Aut0ExeC | psilikon: yes |
| 20:03.59 | Aut0ExeC | precisely |
| 20:04.40 | Aut0ExeC | I looked and looked for that feature |
| 20:04.42 | Aut0ExeC | didnt see it |
| 20:05.37 | Aut0ExeC | psilikon: do you know how to do that? |
| 20:07.20 | psilikon | Aut0ExeC, What do you have for Supplementary Service Subscriptions in your SPA? |
| 20:07.34 | psilikon | There should be a few regarding callerid |
| 20:07.38 | Aut0ExeC | damn i'm not home right now |
| 20:07.42 | Aut0ExeC | i cant check |
| 20:07.51 | Aut0ExeC | is that under "pstn line" ? |
| 20:08.24 | psilikon | not sure |
| 20:09.15 | Aut0ExeC | oh wait ... i have access |
| 20:09.16 | Aut0ExeC | remote |
| 20:09.24 | psilikon | Aut0ExeC, You'll probably want to search for SPA documentation as I don't think this is so much of an * issue. |
| 20:09.36 | Aut0ExeC | does urs come thru? |
| 20:09.36 | psilikon | ok good. remote in and let me know what you have for th SSS |
| 20:09.45 | psilikon | Aut0ExeC, yeah |
| 20:10.18 | Aut0ExeC | I dont see SSS under "pstn line" |
| 20:10.29 | psilikon | spa3102? |
| 20:10.52 | Aut0ExeC | yes |
| 20:10.58 | Aut0ExeC | "pstn line" tab |
| 20:12.09 | psilikon | All I have to go on is a 2102 right now. |
| 20:12.33 | Aut0ExeC | ohh ok |
| 20:13.12 | psilikon | Aut0ExeC, http://www.wirelessforums.org/uk-telecom-voip/caller-id-problem-spa-3102-a-57166.html |
| 20:13.56 | Aut0ExeC | k thanks |
| 20:15.47 | psilikon | Aut0ExeC, what happens when you hit *65? |
| 20:16.00 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-155-145.cablep.bezeqint.net) |
| 20:16.27 | psilikon | Aut0ExeC, here is another link: http://forums.whirlpool.net.au/archive/741612 |
| 20:18.19 | Aut0ExeC | thanks reading the links now |
| 20:19.04 | titter | Could Asterisk act as a proxy so to speak and simply forward on all registrations and RTP to another Asterisk install? So client <---> asterisk proxy <---> main asterisk <---> PSTN |
| 20:20.14 | titter | Terminology is more than likely wrong, please correct me if so. |
| 20:22.06 | Aut0ExeC | psilikon: perhaps i'll mess around with the regional stuff when I get home and experiment.. .still noone has a clear solution |
| 20:23.58 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 20:34.41 | Aut0ExeC | psilikon: I never understood the upgrade versions... I'm using 3.3.6(gu) which apparently is the latest version yet... they have version 5.x out which is older |
| 20:34.49 | Aut0ExeC | figured the latest would be the higher number |
| 20:35.20 | psilikon | Aut0ExeC, hmm so would i |
| 20:35.56 | Aut0ExeC | anyways i changed soem regional stuff. i'll check again when I get home... thanks again for everything bud |
| 20:37.01 | Aut0ExeC | sorry last question.... do you know how to back up the settings? |
| 20:41.07 | *** join/#asterisk Preytell (~jerry.win@65.114.21.3) |
| 20:42.10 | Preytell | Hello, I have a problem with phones not registering from time to time. I put a sniffer on the line and I do see the Register packets coming to the server, but the server never replies. If I change the IP address of the endpoint in question it will register just fine. |
| 20:42.22 | Preytell | I am looking for a reason as to why this would happen. |
| 20:45.54 | Preytell | sorry, should mention that I am using Asterisk 1.8 / PIAF/FREEPBX 2.8 |
| 20:50.43 | Qwell | Preytell: You didn't mention what you changed it from/to. |
| 20:53.36 | *** join/#asterisk DrDamnit (~michael@highpoweredhelp.com) |
| 20:53.43 | DrDamnit | Where is the documentation for exten => same? |
| 20:54.40 | Qwell | DrDamnit: configs/extensions.conf.sample |
| 20:54.46 | DrDamnit | awesome. thanks. |
| 20:56.22 | Preytell | sorry, just the last octet, so 10.0.110.131 to 10.0.110.132. |
| 20:56.29 | Preytell | that's enough to fix the problem. |
| 20:56.42 | *** join/#asterisk GTXComm (~John@cpe-72-128-62-30.kc.res.rr.com) |
| 20:57.32 | Preytell | until I do this sip show peers will show this peer unavailable. |
| 20:58.27 | Preytell | I reboot the phone, it gets the config files from the tftp server, comes up as it's ext, can make calls, but of course you cannot call it, nor trans call to it. |
| 20:58.56 | Preytell | the phones are all polycom 550's |
| 21:00.30 | Preytell | I know there is a bug, id 0018075 [Asterisk] Channels/chan_sip/Subscriptionsthat relates to one problem that I am having. |
| 21:00.49 | Preytell | I just wonder if it also creates this issue. |
| 21:01.08 | *** join/#asterisk jong2 (~chatzilla@63.224.204.153) |
| 21:01.44 | Preytell | I am going to come up to 1.8.3 this weekend, picking up that bug fix, I just wondered if anyone else has this issue. |
| 21:02.35 | GTXComm | master |
| 21:02.45 | GTXComm | :) wrong chat window |
| 21:18.13 | *** join/#asterisk carloimperia (~gpt@109.112.68.226) |
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| 21:20.19 | carloimperia | Anyone have used the chan fax from digium ? |
| 21:22.39 | _Corey_ | We use it on a bunch of systems |
| 21:22.46 | _Corey_ | works nicely |
| 21:23.54 | carloimperia | But I have * 1.4: work in this version too ? |
| 21:25.03 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 21:26.35 | _Corey_ | carloimperia: Go to http://www.digium.com/en/docs/FAX/fax_faq.php and read #7 |
| 21:28.00 | carloimperia | Very tanks ! But in ] 1.8 is nth |
| 21:28.03 | carloimperia | sorry |
| 21:28.28 | *** join/#asterisk mheadd (~Adium@c-69-141-4-36.hsd1.de.comcast.net) |
| 21:28.48 | carloimperia | But I have a patton with support T38: is possible to use a connector with T38 ? Have you used this conf ? |
| 21:29.18 | carloimperia | Excuse for my trivial quest |
| 21:30.03 | _Corey_ | It will likely work better with 1.6+ but may also work with 1.4. You will have to try. (I think Digium gives you 1 free channel license for this sort of thing) |
| 21:34.39 | carloimperia | tnx |
| 21:53.04 | *** part/#asterisk carloimperia (~gpt@109.112.68.226) |
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| 23:26.15 | cVsup | can i configure fxo to not answered call? |
| 23:27.31 | cVsup | i need this fxo only call not answer. |
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