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00:21.31 | WIMPy | res_calendar_ews can be selected in spite of missing requirements. |
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00:43.02 | Micc | how can I setup different users under the same peer name and auth/secret? Like having a customer with a SIP trunk that wants to register each DID as well as the main peer. Or am I smoking the wrong pipe? |
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00:46.17 | WIMPy | You can't. But I'm not sure, what you're trying to acheive. |
00:49.09 | Micc | I just want asterisk to not give a registration error for this customer trying to register a bunch of dids, but with same userauth, realm, and secret. |
00:49.31 | p3nguin | I don't know about smoking the wrong pipe, but you're certainly smoking the wrong terminology. |
00:49.37 | Micc | I'm not sure why it needs to do it in the first place, but thats what this silly UC540 cisco thing is doing. |
00:50.09 | Micc | I know, I'm not quite sure what I'm talking about. |
00:50.10 | p3nguin | DIDs aren't "registered," and "extensions" don't register. |
00:50.19 | Micc | I know that. |
00:50.27 | Micc | But this stupid thing wants to register them all. |
00:51.15 | Micc | I've told the customer to find a way to turn that off, but I'm just trying to see if theres anything I can do on my side to just make it not matter. |
00:51.17 | p3nguin | Register what all? |
00:51.51 | Micc | all DIDs it wants to receive calls on. |
00:51.54 | Micc | I found this http://www.ipcomms.net/product-uc500-didusername.html |
00:52.20 | Micc | so I know it can do just main number, which I don't want it to be the number anyways, just a username. |
00:53.27 | Micc | None of my peers are phone numbers, thats just a headache in my opinion. Except the new panasonic kx-gtp phones don't have a way to use anything but numbers for registration. |
00:53.31 | Micc | I sure hope they fix that. |
00:53.53 | Micc | but in general, I don't use phone numbers for peer names. |
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01:01.00 | Micc | is there any sip.conf option in the peer context to tell asterisk to use the internal nat IP of the registered peer in the invite URI instead of the external IP:port? |
01:01.29 | p3nguin | Yeah. nat=yes |
01:02.25 | Micc | I've tried that, but it still uses external ip, sip show peer shows Reg. Contact : sip:badventures@10.10.2.3:5060 |
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01:02.38 | Micc | I want that to be in the invite, but its not. |
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01:04.40 | p3nguin | Oh, I think I see what you mean. |
01:04.53 | ska | ManxPower: hi |
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01:06.00 | p3nguin | An invite FROM that device should have a Contact field containing the inside address. |
01:06.41 | p3nguin | But if you're inviting that device, then invite seems to have your address and that device's outside address. |
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01:25.21 | WIMPy | has already downgraded all development packages, but is still unable to compile a working Asterisk :-( |
01:25.41 | blaines | Anyone have an opinion on what asterisk system would be best for 3 people? |
01:25.47 | blaines | Just a custom one? |
01:25.59 | blaines | Or maybe the asterisk appliance? |
01:26.28 | blaines | Was also thinking about a hosted asterisk/sip trunk |
01:26.39 | blaines | too many options |
01:27.50 | p3nguin | micc: It looks like the 200 OK is the first spot where the called device reveals its internal IP address. |
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02:56.01 | WIMPy | gives up building a working Asterisk |
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03:05.18 | p3nguin | lol why? |
03:06.04 | WIMPy | I ran out of ideas what else to downgrade. |
03:06.29 | TJNII | WIMPy: Just go ahead and change your name to Rehab. |
03:06.34 | p3nguin | Downgrade? Why not just grab the sources, build, and install? |
03:06.39 | TJNII | Because "You're a Quitter!" |
03:07.15 | WIMPy | Because it generates a non-working version on the current system. |
03:07.42 | p3nguin | I can't imagine how that could be true. |
03:07.51 | WIMPy | It somehow hangs internally, loading a few modules on startup, then quitting. |
03:08.13 | WIMPy | quitting loading modules that is. It keeps running. |
03:08.40 | WIMPy | Then if I try to manually load additional modules, the shell hangs. |
03:09.35 | WIMPy | first suspected the current toolchain, but that't not it (or not all?). |
03:11.23 | WIMPy | Bulding 1.4.21.2 still gives a working copy, 1.6.2.0-rc3, 1.6.2.8-rc8 and trunk from ysterday don't. |
03:11.47 | WIMPy | Unfortunaletly I can't see anything unusual while building. |
03:14.37 | TJNII | Have you nuked /etc/asterisk as a test? |
03:15.06 | WIMPy | Why would I do that? |
03:15.35 | TJNII | To rule out a config based problem, as the binary seems to start and run but hangs when loading configurable bits. |
03:16.06 | WIMPy | I initially didn't cahnge te Asterisk version, just rebuilt it. |
03:17.16 | WIMPy | I just tried the old one after I discovered that such a version still runs on an otherwise similar system. |
03:17.35 | WIMPy | And trunk to rule out it's about something that's already been fixed. |
03:18.15 | TJNII | Well configs are very easy to rule out. |
03:18.21 | TJNII | I would try it. |
03:18.47 | TJNII | (Mostly because I have zero other ideas, but it is stilla good idea.) |
03:20.42 | WIMPy | Now I'm bringing the system back to current software. |
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03:21.06 | WIMPy | Need to do further testing on something faster. |
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03:34.09 | WIMPy | Wow. |
03:34.17 | WIMPy | Installing sample configs seem to help. |
03:34.42 | WIMPy | Pretty interesting, considering the same version was working before. |
03:36.11 | WIMPy | Ok, that's with trunk. Lets try the prviously running version... |
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04:05.11 | WIMPy | It was pp_vad in codecs.conf. |
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04:06.26 | WIMPy | That seems to apply to speex, but that was actually not changed. |
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05:29.39 | Tsolias | Good evening just need some advice regarding Asterisk Voip |
05:30.31 | Tsolias | if anyone can help id really appreciate it as im getting mixed information |
05:30.49 | ruben23 | hi guys i have 10 Mbps leased line connection my callers are only 18 but still getting choppy lines, and other quality issues---> line are shared between voice and data traffic.. |
05:31.36 | Tsolias | 10 Mbps up and Down ruben? |
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05:33.30 | ruben23 | Tsolias: yes |
05:35.59 | WIMPy | ruben23: You nee traffic control (AKA QOS) on your router. |
05:36.31 | Tsolias | i totally agree |
05:37.14 | ruben23 | <PROTECTED> |
05:37.18 | WIMPy | need |
05:37.25 | WIMPy | sure |
05:37.40 | WIMPy | I assume your LAN is faster :-) |
05:37.49 | ruben23 | WIMPy:QoS mean slowing my data traffic , mean slow browsing |
05:37.52 | Tsolias | 1 Gbps Lan network? |
05:38.09 | Tsolias | Ruben QOS : optimising primary traffic for voip |
05:39.09 | WIMPy | ruben23: It does off course mean you won't be able to use more bandwidth for data than is left over by voip, but I'm pretty sure that's what you want. |
05:39.37 | WIMPy | But that does not mean slow browsing. Actually it can make browsing faster as well at the same time. |
05:39.50 | Tsolias | yep |
05:41.03 | ruben23 | WIMPy: wow, what are my chances to implemet QoS on this setup: |
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05:41.36 | ruben23 | internet--->cisco router 1841------->layer 3 cisco switch------>layer 2 cisco switch----->client PC. |
05:42.50 | ruben23 | theres no way i can separte data traffic and voice traffic right..? on my setup.. what you think |
05:43.01 | WIMPy | NFI how good the cisco is at the task. But what's the other side? |
05:43.13 | WIMPy | Why would you want to? |
05:44.40 | ruben23 | <PROTECTED> |
05:45.06 | ruben23 | WIMPy:on my sequence on whihc part i can appy QoS..? |
05:45.28 | WIMPy | The point in voip is that you are able to share bandwidth. |
05:46.11 | WIMPy | If you seperate the traffic you will have unused bandwidth for both. If it's combined you can use all bandwidth. |
05:47.29 | ruben23 | WIMPy: would you think there would be great improvements when QoS is implemented right..? |
05:47.39 | ruben23 | with voice Quality.. |
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05:48.01 | WIMPy | Definitely. |
05:48.23 | Tsolias | yes there would be cause your allowing high priority on your voip |
05:48.36 | ruben23 | i can do separate voice and data traffic also since i used softphones, not IP phone.. |
05:48.40 | ruben23 | i mean |
05:48.45 | ruben23 | i cannot----sorry |
05:49.13 | WIMPy | Yes, you can. |
05:49.47 | WIMPy | But you _might_ have to trust the clients. |
05:50.13 | WIMPy | Where is the voip traffic going? |
05:50.55 | ruben23 | <PROTECTED> |
05:51.56 | WIMPy | Great. So the traffic is both comming from and going to a unique host. |
05:52.28 | WIMPy | Shouldn;t be hard to describe that as a rule. |
05:52.42 | ruben23 | yes, its impossible to separate since i got only one connectivity pipe my 10 mb. |
05:53.02 | ruben23 | and the softphones resides on my PC |
05:55.17 | ruben23 | i got no Idea even s ingle bit for some Cisco QoS for voip. |
05:55.21 | ruben23 | :-( |
05:56.16 | WIMPy | I can't help you with the cisco, but it shouldn't be hard to do. |
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05:57.17 | Tsolias | ill see what i can help him with wimpy as i have done it to a certain point |
05:57.49 | ruben23 | Tsolias:can you share some samples |
05:59.49 | Tsolias | yep looking for you now ruben |
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06:01.46 | Tsolias | ruben is your link 10 mbps frame-atm ? |
06:01.56 | Tsolias | atm = asynchronous transfer mode |
06:07.57 | kruemeltee | hello everybody :-) |
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06:10.48 | kruemeltee | I've got a curious problem here ... Asterisk v1.4 and a couple of gramdstream SIP telephones ... just one telephone has a problem. during a call the other person is not able to hear us anymore. Our agent has to raise his speek-volume. |
06:10.52 | brunner | Please msg me if you sell termination. I do about 70k minutes per day. |
06:11.31 | kruemeltee | I just thought it's the telephone itself, but I've changed it (got a new one) and the headset too ... still the same problem ... |
06:12.53 | kruemeltee | but not during every call ... just some of them ... maybe it's a codec issue ... I've disabled all and allowed first alaw and then ulaw, should be enough, right? |
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06:19.02 | kaldemar | kruemeltee: if the volume is low, it's not a codec issue. in case of codec incompatibility, there would be no audio at all. |
06:19.05 | kaldemar | ~gs |
06:19.16 | infobot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
06:29.13 | kruemeltee | kaldemar, that was my idea too ... if it's a codec problem the whole conversation would not be able ... |
06:30.22 | kruemeltee | infobot, grandstream telephones work fine here, exept this one ... but it seems as if it doen't depend on the phone (because of changing the whole phone doesn't solve the problem) |
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06:31.32 | kaldemar | infobot is a bot, not a human being. :) |
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07:00.40 | DND | hi guys. i have asterisknow 1.5 with asterisk core 1.4, can it handle corei3? |
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07:04.13 | Seb^ | hi |
07:05.07 | Seb^ | can anyone help me with a TRUNK Dial failed due to CHANUNAVAIL (hangupcause: 0) error using a Digium TE420P PIR E1 card? |
07:05.11 | Seb^ | PRI* |
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07:08.31 | Seb^ | this is my error: http://www.trixbox.org/forums/trixbox-forums/open-discussion/trunk-dial-failed-due-chanunavail-hangupcause-0 |
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07:11.31 | kaldemar | Seb^: dahdi-channels.conf and chan_dahdi_additional.conf are also relevant. |
07:11.44 | Seb^ | ok, I didnt know this... |
07:12.20 | Seb^ | so should I make the contents of dahdi-channels.conf the same as my /etc/dahdi/system.conf ? |
07:12.32 | kaldemar | hell no. |
07:12.46 | kaldemar | they are different configuration files. |
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07:12.58 | Seb^ | lol |
07:13.22 | kaldemar | your first choice would be to use freepbx to configure the card. for that, go to #freepbx. |
07:13.48 | Jumpie | h8rs |
07:14.46 | Seb^ | im using trixbox |
07:17.38 | kaldemar | Seb^: i.e. you're using the freepbx GUI. |
07:17.53 | Seb^ | the gui is very limited |
07:18.23 | Seb^ | I am sure that that is all correct |
07:18.32 | Seb^ | you cant really configure the card via the gui |
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07:19.55 | kaldemar | pastebin your dahdi-channels.conf and chan_dahdi_additional.conf and we'll take a look. |
07:19.58 | kaldemar | ~pb |
07:19.59 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
07:21.22 | Seb^ | dahdi-channels.conf - http://pastebin.com/nwVEHiew |
07:22.24 | Seb^ | chan_dahdi_additional.conf is empty |
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07:24.27 | kaldemar | ok, your problem is that the freepbx macro is dialing DAHDI/g1/07739487428. g1 would translate to group=1 in chan_dahdi.conf (or in your case also dahdi-channels.conf or chan_dahdi_additional.conf). you have group=1 in chan_dahdi.conf but no channel definitions _under_ it, so from asterisk's point of view, no channels belong to group 1. |
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07:25.00 | Seb^ | ok |
07:25.08 | Seb^ | so what do I need to enter to rectify that? |
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07:25.27 | kaldemar | the easiest solution is to move the "group=1" line above "#include dahdi-channels.conf" in chan_dahdi.conf and either reload chan_dahdi or restart asterisk. |
07:25.57 | Seb^ | ahhhhhh |
07:25.57 | Seb^ | ok |
07:25.59 | Seb^ | ill try that |
07:26.15 | Seb^ | and should I uncomment the include dahdi-channels.conf |
07:26.16 | kaldemar | erm. dahdi-channels.conf has group definitions too.. they will conflict. |
07:26.42 | kaldemar | it's not commented, #include is actually including dahdi-channels.conf to chan_dahdi.conf. |
07:26.50 | Seb^ | oh ok |
07:26.51 | kaldemar | ; is the comment character |
07:26.57 | Seb^ | ah,ok, so what does # do? |
07:27.01 | Seb^ | nothing? |
07:27.29 | kaldemar | #include includes other files to configuration files |
07:27.39 | Seb^ | ah ok |
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07:28.11 | kaldemar | you also need to remove group=0,11 from dahdi-channels.conf. then your g1 will have "channel => 1-15,17-31" in it. |
07:29.06 | Seb^ | should I also remove the group = 63 ? |
07:29.45 | kaldemar | it doesn't make any difference there. |
07:30.19 | Seb^ | ok |
07:30.25 | Seb^ | in chan_dahdi.conf |
07:30.33 | Seb^ | I have group=1 above the include |
07:31.05 | kaldemar | btw, it might be that freepbx screws up your config again upon a reboot for example, i don't know how it behaves exactly. |
07:31.23 | Seb^ | nah it doesnt seem to touch it |
07:32.34 | *** join/#asterisk BANSAL (~bansal@117.207.82.30) |
07:33.27 | Seb^ | hmm |
07:33.32 | Seb^ | restarted asterisk but get the same issue |
07:34.25 | Seb^ | grrr |
07:40.00 | kaldemar | did you modify dahdi-channels.conf too? |
07:40.32 | Seb^ | yeah |
07:40.33 | Seb^ | 1 sec |
07:41.02 | Seb^ | http://pastebin.com/HgHfK3eX |
07:41.06 | Seb^ | thats what it looks like now |
07:43.00 | kaldemar | what does the cli output look like? |
07:44.45 | ChannelZ | white text, black background |
07:46.30 | Seb^ | lol |
07:46.36 | Seb^ | how do you mean? |
07:46.43 | Seb^ | i can show you a the output from a call |
07:47.48 | Seb^ | http://pastebin.com/tDQncMZc |
07:50.22 | ChannelZ | I haven't been following this whole thing, but based on the previous pastebin, you are dialing DAHDI/g1/.... - and based on the previous paste before that, you don't have any of those channels defined as being in group 1 |
07:50.35 | kaldemar | Seb^: and you restarted asterisk or reloaded chan_dahdi.so? |
07:51.08 | Seb^ | yeah |
07:51.09 | Seb^ | well |
07:51.11 | kaldemar | ChannelZ: he should now have group=1 in chan_dahdi.conf and #include dahdi-channels.conf (which has the channels) under it. |
07:51.17 | Seb^ | i did an asterisk restart |
07:51.22 | Seb^ | and a hadhi restart |
07:51.50 | Seb^ | dahdi* |
07:51.51 | Seb^ | ok |
07:51.53 | ChannelZ | why? Put the group where it belongs by the rest of the channel definitions. |
07:52.19 | Seb^ | so should I have the group in the dahdi-channels.conf ? |
07:52.51 | ChannelZ | well it's not necessarily a requirement but it seems a bit odd to split them up |
07:52.59 | ChannelZ | You're also defining a group 63 twice in your dahdi-channels.conf |
07:53.00 | *** join/#asterisk Tim_Toady (~moi@188.4.4.16.dsl.dyn.forthnet.gr) |
07:53.12 | ChannelZ | so who knows what might be wrong in your chan_dahdi.conf |
07:53.28 | Seb^ | i have removed group 63 |
07:53.33 | Seb^ | as it is not necessary |
07:53.59 | ChannelZ | Also have you tried dialing one of the channels directly, not as a group, just to make sure it's even working? |
07:54.06 | Seb^ | how would I do that? |
07:54.21 | Seb^ | I havent tried, I didnt know I could... |
07:54.22 | ChannelZ | Dial(DAHDI/1/somenumber) |
07:54.29 | Seb^ | from the CLI? |
07:54.52 | ChannelZ | you also have dueling contexts in dahdi-channels.conf |
07:55.50 | ChannelZ | or duplicates rather. It _should_ be doing what you want but is confusing to see none the less |
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07:56.01 | Seb^ | when i try to dial from CLI i get: |
07:56.02 | Seb^ | No such command 'Dial(DAHDI/1/07739487428)' (type 'help Dial(DAHDI/1/07739487428)' for other possible commands) |
07:56.38 | ChannelZ | no put that as an extension in your dialplan and call it |
07:56.47 | ChannelZ | or wait.. you're using FreePBX aren't you |
07:57.03 | Seb^ | well, im using trixbox, which I beleive is freepbx yes |
07:57.09 | Seb^ | but Im doing this all via cli |
07:57.17 | Seb^ | the gui is very limited |
07:57.30 | ChannelZ | cusses |
07:57.39 | Seb^ | lol |
07:58.17 | Seb^ | in freepbx i have setup my extentions, outbound route, and trunk |
07:58.24 | ChannelZ | I don't remember how/if you can dial arbitrary things from the console |
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07:59.10 | ChannelZ | I really despise these asterisk GUIs which everyone seems to run but nobody seems to want. |
08:00.30 | Seb^ | lol |
08:00.39 | Seb^ | well as I said, im in the CLI :) |
08:01.06 | ChannelZ | yes but you are not manually editing extensions and such |
08:01.18 | Seb^ | I am not, no. |
08:02.30 | ChannelZ | try "originate DAHDI/1/number application MusicOnHold" |
08:02.36 | ChannelZ | replacing 'number' with a valid phone number |
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08:02.45 | ChannelZ | it should dial that number and when you answer it you'll hear music |
08:03.01 | Seb^ | hmmm |
08:03.09 | Seb^ | vcsn I paste 4 lines? |
08:03.11 | Seb^ | can* |
08:03.24 | ChannelZ | sure, it's late |
08:03.26 | Seb^ | thpbx1*CLI> originate DAHDI/1/07739487428 application MusicOnHold |
08:03.26 | Seb^ | <PROTECTED> |
08:03.26 | Seb^ | <PROTECTED> |
08:03.26 | Seb^ | <PROTECTED> |
08:03.41 | ChannelZ | uhh |
08:03.57 | Seb^ | ? :) |
08:04.34 | ChannelZ | what does 'dahdi show channels' say to you (pastebin that because for you it should be a lot) |
08:05.17 | Seb^ | http://pastebin.com/4yWZv6Ea |
08:05.22 | Seb^ | 62 channels |
08:05.33 | Seb^ | although we are only using one port on the card so only half will work |
08:05.46 | ChannelZ | ok.. do you have the thing plugged into the right port? |
08:06.11 | Seb^ | yes |
08:06.24 | ChannelZ | try "originate DAHDI/32/number application MusicOnHold" |
08:06.26 | Seb^ | i havent diabled either |
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08:06.55 | Seb^ | hmmm |
08:06.59 | Seb^ | it hasnt errored yet |
08:07.00 | Seb^ | ... |
08:07.14 | Seb^ | doh |
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08:07.18 | ChannelZ | So you do have it plugged into the "wrong" port |
08:07.20 | Seb^ | thpbx1*CLI> originate DAHDI/32/07739487428 application MusicOnHold |
08:07.20 | Seb^ | <PROTECTED> |
08:07.20 | Seb^ | <PROTECTED> |
08:07.32 | Seb^ | same error |
08:07.33 | ChannelZ | wait did the number you dialed ring? |
08:07.36 | Seb^ | no |
08:08.00 | ChannelZ | hmm slightly different |
08:08.14 | Seb^ | oh |
08:08.16 | ChannelZ | well my only thought is the hardware is configured wrong, or the line isn't active, or something |
08:08.33 | Seb^ | well, shall I try swapping the ISDN to the other span? |
08:08.34 | Seb^ | oh |
08:08.44 | Seb^ | I have a loopback cable on the second span |
08:08.48 | Seb^ | that will make a difference |
08:08.49 | ChannelZ | Or you're dialing differently than your telco wants to see |
08:08.54 | Seb^ | one second and ill remove it |
08:10.14 | ChannelZ | wait this is ISDN? |
08:10.22 | ChannelZ | I thought this was T1/E1 |
08:10.37 | Seb^ | it is |
08:10.40 | Seb^ | well ISDN30 |
08:10.43 | Seb^ | which is E1 isnt it? |
08:10.51 | Seb^ | well, I had to set the card to E1 |
08:11.00 | Seb^ | which I beleive is compatible wiht isdn |
08:11.29 | ChannelZ | I'm thinking not. But I know little about PRIs (no personal experience) and even less about ISDN.. and less yet about eurpoean isdn |
08:11.41 | Seb^ | lol |
08:11.57 | Seb^ | why cant these things just be simple! |
08:12.09 | tuxx- | hiya, we have a setup with a duo-bri card, we wanna use 3 lines for incoming traffic, and reserve 1 line for outgoing traffic. Anyone know how we can best handle this? I was thinking about putting the channels of the 4th line in a different context that doesnt have any rules in it, so the call will always fail when its incoming.. is this the correct way to handle a situation like this? |
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08:13.00 | ChannelZ | Anyways, I'm not sure how to help further.. my only guess is the hardware is not configured correctly for your telco |
08:13.24 | Seb^ | well |
08:13.29 | Seb^ | I can receive incoming calls |
08:13.36 | Seb^ | if that makes any difference? |
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08:15.00 | ChannelZ | tuxx-: so the 4th line has a number people can call but you don't want to answer it? Or is it in a hunt group you can't remove? |
08:15.09 | ChannelZ | Seb^: that's strange |
08:15.28 | ChannelZ | Seb^: What does the console say when such a call comes in? What channel does it say? |
08:15.38 | Seb^ | one second |
08:16.19 | tuxx- | ChannelZ: actually, i would prefer if no calls came in on the 4th line, so we can use it as a backup outgoing line. |
08:17.22 | ChannelZ | tuxx-: right but you have no control over that, besides to not tell people what the number is so they don't call it. Done. |
08:18.18 | Seb^ | ChanServ: this is an incoming call log from the CLI: http://pastebin.com/qraJKkdv |
08:18.21 | ChannelZ | tuxx-: as far as dialing out, you want to dial out on any of the 4 lines but prefer to dial out starting with line 4, yes? |
08:18.39 | tuxx- | yep |
08:18.47 | tuxx- | ill just make a group for that i think, so i cann call out via that group? |
08:19.58 | ChannelZ | yes, and you can use a big-G to make it go in reverse |
08:20.02 | Seb^ | that looks to me like its using a group called 1-1 :\ |
08:20.23 | tuxx- | mkay, thanks! :) |
08:20.27 | ChannelZ | so say lines 1-4 are all in group 1, you can Dial DAHDI/G1 instead of DAHDI/g1 |
08:23.37 | ChannelZ | Seb^: So maybe you're dialing an incorrect number format for your telco.. I dunno how the wacky european numbers go |
08:23.53 | Seb^ | lol |
08:23.59 | Seb^ | he number I am dialling should be fine |
08:24.03 | Seb^ | its a standard mobile phone number |
08:25.30 | Seb^ | I tried this |
08:25.30 | Seb^ | thpbx1*CLI> originate DAHDI/1-1/07739487428 application MusicOnHold |
08:25.30 | Seb^ | <PROTECTED> |
08:25.31 | Seb^ | <PROTECTED> |
08:25.31 | Seb^ | <PROTECTED> |
08:25.42 | Seb^ | because the incoming call was on DAHDI/1-1 |
08:25.46 | Seb^ | but got the same issue |
08:26.10 | ChannelZ | yeah you don't want 1-1 |
08:26.36 | Seb^ | ok |
08:26.55 | Seb^ | will the contents of /etc/dahdi/system.conf make any difference? |
08:27.31 | ChannelZ | probably not to me. |
08:28.06 | ChannelZ | 'cause 1' is, AFAIK, is something along the lines of 'number cannot be reached as dialed' |
08:30.58 | ChannelZ | Are you sure your carrier wants the leading 0? (or is it 07 a country code? is it in the same country as you?) |
08:31.26 | Seb^ | yes |
08:31.33 | Seb^ | 07 for a mobile number is normal |
08:31.49 | Seb^ | every number in the UK starts with a 0 |
08:32.04 | ChannelZ | is there a handset device (an isdn phone?) you can plug in and dial that exact string and it works? |
08:32.12 | Seb^ | we dont have one :( |
08:33.28 | ChannelZ | well I'm still feeling like whatever your dialing, your provider doesn't like. * seems to be communicating on the channel correctly based on your incoming calls working, and the error you're getting trying to dial out. They're rejecting you for whatever reason |
08:33.46 | Seb^ | hmmm |
08:33.51 | Seb^ | youre the second person to have said this |
08:33.58 | ChannelZ | Just because you dial a certain way picking up a phone doesn't necessarily mean your carrier wants digits the same way. |
08:34.09 | Seb^ | but we took the trixbox to another building last night with an isdn30 line, and it didnt work there either |
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08:34.29 | ChannelZ | For instance the ITSP I just signed up to, wants 1 before the number, even if it's local (in the US 1 usually signifies long distance) which is NOT how I'd normally dial a local number. |
08:34.30 | Seb^ | and we know that that line works becaue it has a production box on it which is working fine |
08:34.44 | ChannelZ | wait, a production box of what? |
08:34.49 | Seb^ | trixbox |
08:34.56 | Seb^ | by production i mean its live |
08:34.59 | Seb^ | and theyre using it |
08:35.17 | ChannelZ | wait.. so this same line is normally serviced by another system? |
08:35.42 | Seb^ | the line here, is a brand new line |
08:35.52 | Seb^ | last night I took this trixbox to another building |
08:36.04 | Seb^ | which has another ISDN30, which is serviced by another system |
08:36.14 | Seb^ | and it wouldnt work their either |
08:36.23 | Seb^ | so it must be the trixbox that is the issue... |
08:36.53 | ChannelZ | What services the other line normally? nothing? |
08:37.20 | ChannelZ | (you said you took it to another building with a different line and it didn't work -- what was that building running before) |
08:38.21 | Seb^ | that other building is running a very old version of trixbox. |
08:38.34 | Seb^ | and freepbx |
08:38.35 | ChannelZ | but it works |
08:38.37 | Seb^ | yes |
08:38.56 | ChannelZ | so look at its console and see what it's dialing |
08:39.25 | Seb^ | I have |
08:39.34 | Seb^ | I can show you, one second |
08:44.45 | Seb^ | this is a pastebin from the working phone system |
08:44.46 | ChannelZ | ...? I really need to get to bed.. |
08:44.46 | Seb^ | http://pastebin.com/A8L7bufN |
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08:46.37 | ChannelZ | hmm. Well I think you have some ISDN voodoo going on which is over my head. |
08:46.43 | Seb^ | :( |
08:46.44 | Seb^ | doh |
08:46.46 | ChannelZ | Humor me and try the originate command again but remove the leading 0 |
08:46.51 | Seb^ | ok |
08:47.06 | Seb^ | OMG |
08:47.13 | Seb^ | JESUS CHRISTING HELL |
08:47.15 | Seb^ | :| |
08:47.22 | ChannelZ | ok... so now do this |
08:47.36 | ChannelZ | in your /etc/asterisk/chan_dahdi.conf add: |
08:47.47 | Seb^ | it seemed to work lol |
08:47.50 | ChannelZ | pridialplan=unknown |
08:47.51 | ChannelZ | prilocaldialplan=unknown |
08:48.07 | ChannelZ | and then restart asterisk completely (nor just reload) |
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08:48.11 | ChannelZ | s/nor/not/ |
08:48.17 | Seb^ | ok |
08:48.18 | Seb^ | well |
08:48.21 | Seb^ | i have that in there |
08:48.27 | ChannelZ | already? |
08:48.29 | Seb^ | yes |
08:48.30 | Seb^ | 1 sec |
08:48.44 | ChannelZ | ok.. then one of them is probably wrong for you |
08:48.51 | Seb^ | http://pastebin.com/VUzVKCfe |
08:48.53 | Seb^ | this is it |
08:50.04 | ChannelZ | ok so I don't know how these settings all interact but they're wrong in some fashion |
08:50.09 | Seb^ | well |
08:50.20 | Seb^ | i can juts strip the leading 0 off every number on my trunk |
08:50.23 | Seb^ | thatll work.... |
08:50.47 | ChannelZ | I think maybe your 'nationalprefix' is adding an additional 0 onto the front of your string |
08:51.05 | Seb^ | ahhhhhhhhh |
08:51.05 | Seb^ | ok |
08:51.12 | Seb^ | ill remov ethat |
08:51.32 | ChannelZ | I'm only guessing her |
08:51.44 | ChannelZ | as I've never had occasion to use all this stuff |
08:52.32 | ChannelZ | and they don't seem to be documented that well in my sample config so I have no idea what they do |
08:52.51 | Seb^ | this is very odd |
08:53.03 | Seb^ | if I dial from the CLI |
08:53.11 | Seb^ | if I dial from the CLI originate DAHDI/1/7796955619 application MusicOnHold |
08:53.13 | Seb^ | this works |
08:53.26 | ChannelZ | but in any case now you know what to jack around with, it is because you're sending a dial string to your provider that it doesn't like. best way to fix it, I dunno. Maybe someone else here will wake up at some point who is more familiar with european ISDN in particular |
08:53.39 | Seb^ | ok :) |
08:53.42 | Seb^ | wel thanks for your help |
08:54.03 | ChannelZ | sure, sorry it wasn't definitive |
08:54.20 | ChannelZ | what were you saying above |
08:54.23 | Seb^ | if i was to try originate DAHDI/g1/7796955619 application MusicOnHold |
08:54.27 | Seb^ | should that wotk? |
08:54.43 | ChannelZ | assuming your channels are grouped correctly yes |
08:55.19 | ChannelZ | although I think you have all 62 in group 1 which means after 31 it will start trying to dial on channel 32 which you said isn't hooked up to anything |
08:55.29 | Seb^ | ok |
08:55.41 | ChannelZ | no your chans aren't grouped right I see |
08:55.50 | Seb^ | is there an easy way to diable the 2nd port on the card? |
08:55.50 | ChannelZ | you have the group after your #include that defines the channels. Do this |
08:55.59 | ChannelZ | Remove group=1 from your chan_dahdi.conf |
08:56.06 | Seb^ | ok |
08:56.22 | ChannelZ | then in your dahdi-channels.conf put group=1 in right before the first channels => 1-31 etc line |
08:56.49 | ChannelZ | as for disabling the second span, just comment the whole thing out in dahdi-chanels.conf |
08:57.18 | Seb^ | ok |
08:57.24 | Seb^ | then restart asterisk? |
08:57.27 | ChannelZ | yeah |
08:57.39 | ChannelZ | well for this you can probably just reload |
08:57.43 | ChannelZ | but either wya |
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08:58.06 | Seb^ | ahar |
08:58.11 | Seb^ | group1 seems to work now |
08:58.58 | ChannelZ | yay |
09:03.33 | Seb^ | thanks for all your help |
09:05.18 | Seb^ | hmm |
09:05.22 | Seb^ | international calls fail though |
09:05.24 | Seb^ | thats annotinbh |
09:05.29 | Seb^ | thats annoying |
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09:22.45 | arturio | Hello. I have trixboxCE server. When I'm connecting to users via phones there is a text on phone display "Unknown". I'm using php-api and connect throw $socket = fsockopen(....); |
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09:23.30 | arturio | I know that I need to write something in: fputs($socket, "CallerID: What Ever Appropriate\r\n"); |
09:23.53 | arturio | But how to get CallerID and CallerName |
09:23.56 | arturio | ? |
09:37.27 | arturio | no matter. i've done it :) |
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09:49.27 | Gopal | Is it possible to generate a simulator for Asterisk with TDM signals? |
09:50.00 | *** join/#asterisk Moz (~me@81.179.238.144) |
09:50.22 | Moz | Hi All. Has anyone here got Asterisk up and running with SS7 (over SIGTRAN)? |
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10:07.32 | *** join/#asterisk Dovid (~annon@213.8.118.62) |
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10:12.38 | tzafrir_laptop | Gopal, what would you expect of such a simulator? |
10:14.03 | Gopal | tzafrir_laptop: I need to simulate PSTN calls using T1/E1 line within two asterisk server. |
10:14.22 | Gopal | tzafrir_laptop: I suspect it is possible with SIPp |
10:14.42 | tzafrir_laptop | Gopal, you have two Asterisk servers with E1 adapters? |
10:15.09 | Gopal | tzafrir_laptop: yes |
10:15.10 | tzafrir_laptop | I personally find originating calls directly with 'originate' simple and often powerful enough |
10:15.36 | tzafrir_laptop | help originate |
10:15.58 | Gopal | tzafrir_laptop: are you saying to use originate action to dial a call? |
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10:17.13 | tzafrir_laptop | originate from the asterisk CLI is simpler |
10:17.29 | tzafrir_laptop | Does not provide all the options that the manager action provides |
10:17.35 | tzafrir_laptop | But often it will do |
10:17.48 | tzafrir_laptop | sipp is naturally an option as well |
10:18.02 | tzafrir_laptop | (Debian package: sip-tester) |
10:19.10 | Gopal | tzafrir_laptop: sip-tester is for testing sip calls rite? |
10:19.43 | tzafrir_laptop | Depends on what do you mean by that |
10:19.58 | tzafrir_laptop | Do you mean: test if the system can accept a certain call rate? |
10:20.00 | Gopal | tzafrir_laptop: what i need to test is tdm calls |
10:20.27 | tzafrir_laptop | sipp can generate those calls. You can redirect them (in the dialplan) to asterisk |
10:20.42 | Gopal | tzafrir_laptop: ok thanks |
10:20.43 | tzafrir_laptop | Do you generally plan to have sip<->PSTN calls? |
10:20.54 | Gopal | tzafrir_laptop: yes |
10:20.58 | tzafrir_laptop | If so, that is a close-to-reality way to test |
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10:44.26 | Faustov | any idea what can cause a "603 declined" response from a SIP provider whenever more than 1 sip call is being established? |
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10:47.20 | stodorovic | Hi. Got a client that has Asterisk, and most of the phones generally work, but one particular person has trouble where if they call landlines, they have fluctuating volume problems. For about a second, it's loud, then the next second, the other person sounds very quiet and muffled, then loud, then muffled etc. He's switched handsets, with no difference in this problem. What could this be, please? |
10:48.57 | frk2 | stodorovic, ip phones? |
10:49.17 | stodorovic | yes, they are VOIP phones |
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10:54.30 | frk2 | do ip-ip calls work fine? |
10:55.12 | stodorovic | Not sure. I will ask next time I speak to the person. However, I dialed the client as an ip-ip call and that seemed fine. |
10:55.30 | stodorovic | I'm just going to do a 3-way diff on the users.conf extension configs |
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10:58.19 | stodorovic | codec and signalling differs |
10:58.23 | stodorovic | could this be a problem? |
11:01.14 | *** join/#asterisk Tim_Toady (~moi@188.4.4.16.dsl.dyn.forthnet.gr) |
11:03.16 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
11:04.27 | *** join/#asterisk lantizia (~lantizia@217.154.146.153) |
11:04.45 | lantizia | Hey... I just re-installed DAHDI... but on the asterisk CLI I no longer have the dahdi command |
11:04.58 | lantizia | and ideas why? (dahdi is loading my card correctly OK it seems) |
11:05.57 | kaldemar | chan_dahdi.so is not loaded. |
11:05.59 | Chainsaw | lantizia: Sounds like you don't have the dahdi channel driver. |
11:06.03 | *** join/#asterisk kombi (~kombi@port-92-198-15-96.static.qsc.de) |
11:06.05 | Chainsaw | lantizia: So module load chan_dahdi.so |
11:06.12 | Chainsaw | lantizia: If it complains about anything, address it. |
11:07.05 | kombi | where would I find info on how to set up dahdi to route an S0 bus as output to a B410P? |
11:07.21 | lantizia | Chainsaw: how can I address it? |
11:07.37 | Chainsaw | lantizia: The warning? Can't tell you until you share it with me. |
11:07.41 | lantizia | it just says "Unable to load module chan_dahdi.so" "Command 'load chan_dahdi.so' failed" |
11:07.49 | Chainsaw | lantizia: core set verbose 10 |
11:07.51 | Chainsaw | lantizia: core set debug 10 |
11:07.59 | Chainsaw | lantizia: And then load again. |
11:08.05 | lantizia | same message |
11:08.13 | Chainsaw | Obviously, but with more information surrounding it. |
11:08.23 | lantizia | it no it is identical |
11:08.32 | Chainsaw | You win. No idea. |
11:08.35 | Chainsaw | Good luck. |
11:09.02 | Tim_Toady | lantizia its 'module load chan_dahdi.so' not 'load chan_dahdi.so' |
11:09.24 | tzafrir_laptop | lantizia, maybe chan_dahdi.so failed to load |
11:09.26 | lantizia | same result |
11:09.36 | tzafrir_laptop | module show like chan_dahdi |
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11:09.54 | lantizia | it is already loaded |
11:10.08 | lantizia | ok.... module show |
11:10.13 | lantizia | tells me it is already loaded |
11:10.17 | lantizia | but I have no dahdi command |
11:11.16 | *** join/#asterisk Seb^ (~sebspiers@188.39.20.226) |
11:13.27 | tzafrir_laptop | lantizia, module unload chan_dahdi.so |
11:13.35 | tzafrir_laptop | module load chan_dahdi.so |
11:13.43 | tzafrir_laptop | this will give you the acctual error |
11:14.29 | lantizia | tzafrir_laptop: unable to load module chan_dahdi.so .... command mailed ... registred application dahdisendkeypadfacility then it parses 4 files |
11:14.52 | lantizia | tzafrir_laptop: basically it doesn't give me any more information than it normally does... and if you do a modules show then it still says it is loaded |
11:15.00 | tzafrir_laptop | lantizia, what error did you see before "unable to load module chan_dahdi.so" ? |
11:15.12 | lantizia | tzafrir_laptop: none |
11:15.16 | tzafrir_laptop | again: first unload, then load |
11:16.29 | lantizia | tzafrir_laptop: I've already done that |
11:18.01 | tzafrir_laptop | lantizia, do you have the line '[channels]' in /etc/asterisk/chan_dahdi.conf ? |
11:18.27 | tzafrir_laptop | Also: what is the output of: logger show channels |
11:18.27 | lantizia | tzafrir_laptop: I've been using zapata.conf not that file |
11:18.34 | tzafrir_laptop | Any 'Console' line? |
11:18.50 | tzafrir_laptop | lantizia, so in zapata.conf |
11:19.16 | lantizia | yes I have [channels] in there |
11:20.06 | lantizia | tzafrir_laptop: http://pastebin.com/FB4FxJML |
11:20.20 | lantizia | it's using qozap (fully installed and working - dahdi loves it) |
11:20.29 | lantizia | so it's a Junghanns.NET quadBRI card |
11:20.51 | lantizia | they told me how the system.conf should look but now how chan_dahdi.conf / zapata.conf should look |
11:21.18 | lantizia | tzafrir_laptop: should I use chan_dahdi.conf INSTEAD of zapata.conf (i.e. they both shouldn't exist at the same time?_) |
11:21.23 | tzafrir_laptop | lantizia, do you use dahdi or zaptel? |
11:21.30 | lantizia | dahdi... there is no zaptel |
11:21.31 | tzafrir_laptop | (at the kernel level) |
11:21.40 | tzafrir_laptop | what's the output of: lsdahdi |
11:21.54 | lantizia | tells me about my spans |
11:22.10 | tzafrir_laptop | Can you pastebin the output? |
11:22.39 | lantizia | http://pastebin.com/xAeFmzQJ |
11:22.40 | Gopal | If I dial from soft phone to FXS connected in Asterisk there is only one way audio |
11:22.50 | Gopal | the softphone is at remote end |
11:23.26 | tzafrir_laptop | Gopal, what happens if you try voicemail? Playback? Echo (echo test)? |
11:23.42 | lantizia | tzafrir_laptop: did you get it? |
11:24.12 | tzafrir_laptop | lantizia, right. So channels are configured. None is used by Asterisk |
11:24.22 | tzafrir_laptop | What version of asterisk is it? |
11:24.30 | lantizia | tzafrir_laptop: ok should I be using zapata.conf or not? it is asterisk 1.6 |
11:24.58 | tzafrir_laptop | It ignores zapata.conf and only uses chan_dahdi.conf |
11:25.08 | tzafrir_laptop | Start with renaming zapata.conf to chan_dahdi.conf |
11:25.16 | lantizia | right ok - let me configure chan_dahdi.conf then with the BRI info and also channels |
11:25.17 | tzafrir_laptop | (symlink it. whatever) |
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11:27.45 | Gopal | tzafrir_laptop: echo test let me check, i have not enabled it |
11:28.25 | tzafrir_laptop | Gopal, the point is: to check the connection from the SIP device to Asterisk separately |
11:28.52 | Gopal | ok |
11:29.11 | lantizia | tzafrir_laptop: ok I've got the dahdi command back now :) |
11:29.27 | lantizia | tzafrir_laptop: getting alot of messages saying.... dahdi: Master changed to ztqoz/1/1 |
11:29.36 | lantizia | flying past the screen, good/bad? |
11:30.31 | Wimme | i have an issue with a digium bri card, it seems to have stopped working. |
11:30.47 | tzafrir_laptop | lantizia, one span keeps going up and down? |
11:31.04 | Wimme | zaptel_hardware says: pci:0000:0f:03.0 wcb4xxp- d161:b410 Digium Wildcard B410P |
11:31.08 | lantizia | tzafrir_laptop: it's that same message but it either ends in 1, 2, 3 or 4 |
11:31.29 | Wimme | if i run genzaptelconf it says 0 channels to configure. |
11:31.46 | tzafrir_laptop | lantizia, are most ports (spans) in RED alarm most of the time? |
11:31.50 | Wimme | it has worked for ages untill now |
11:32.12 | lantizia | tzafrir_laptop: no isdn2 is plugged in currently - however I've only enabled 1-2 and 4-5 in the chan_dahdi.conf file |
11:32.12 | tzafrir_laptop | Wimme, genzaptelconf doesn't really work for dahdi |
11:32.22 | Wimme | its a 1.4 box |
11:32.34 | tzafrir_laptop | Wimme, it appears that the module is not loaded |
11:33.00 | tzafrir_laptop | try: dahdi_genconf modules; /etc/init.d/dahdi start; dahdi_genconf; /etc/init.d/dahdi start |
11:34.29 | Wimme | tzafrir_laptop, its an asterisk 1.4.22, iirc dahdi is from asterisk 1.6 |
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11:35.55 | tzafrir_laptop | dahdi works with 1.4.x for x >= 22. But the BRI support is not included up until 1.6.0, right |
11:36.31 | tzafrir_laptop | That said, your issue is at the DAHDI level, even before getting to Asterisk |
11:39.32 | Faustov | any idea what can cause a "603 declined" response from a SIP provider whenever more than 1 sip call is being established? |
11:44.36 | kaldemar | Faustov: the provider offering you only one call at a time. |
11:45.03 | Faustov | kaldemar: I have multiple accounts with that provider and all of them allow more than one |
11:45.08 | Faustov | could this be some kernel setting? |
11:45.18 | Seb^ | Hi, can anyone help me with this issue??? http://trixbox.org/forums/trixbox-forums/open-discussion/help-trixbox-outgoing-calls-driving-me-nuts |
11:46.07 | kaldemar | Faustov: ask your provider. |
11:46.25 | Faustov | kaldemar: done, they claim they allow more |
11:48.28 | lantizia | tzafrir_laptop: do you know where I can get a sample chan_dahdi.conf for BRI cards (if Junghanns.NET or Digium) ? |
11:48.44 | tzafrir_laptop | dahdi_genconf |
11:48.54 | kaldemar | Faustov: tell them that they're responding with 603. |
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11:53.23 | lantizia | tzafrir_laptop: no that doesn't generate that file |
11:53.57 | tzafrir_laptop | It generates /etc/asterisk/dahdi-channels.conf |
11:54.30 | lantizia | tzafrir_laptop: so just rename it? |
11:54.57 | tzafrir_laptop | generally: add it at the end of the existing chan_dahdi.conf |
11:55.07 | tzafrir_laptop | It does not include the [channels] line |
11:55.16 | tzafrir_laptop | It is intended to be the "generated" part of it |
11:55.20 | lantizia | tzafrir_laptop: and I can use chan_dahdi.conf.template if i've mucked it up? |
11:56.23 | tzafrir_laptop | I guess so |
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12:02.04 | ijpalmer | good afternoon, how would I let a person making an outbound call know they've dialled an invalid number |
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12:30.13 | stodorovic | which audio codec to use in allow= line? |
12:30.28 | stodorovic | got alaw on atm, but it seems to have choppy quality |
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12:39.14 | tzafrir_laptop | stodorovic, alaw is not (much) compressed. Thus it gives you good quality if there are not network/bandwidth issue |
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13:03.01 | stodorovic | tzafrir_laptop: what about gsm? |
13:03.35 | stodorovic | [May 27 13:41:47] WARNING[733]: chan_sip.c:1949 retrans_pkt: Maximum retries exceeded on transmission |
13:03.38 | stodorovic | that's bad? |
13:04.07 | tzafrir_laptop | stodorovic, it's generally a better option if you have a worse network connection |
13:04.29 | tzafrir_laptop | it: gsm |
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13:04.54 | stodorovic | hmm well the one user that seems to use gsm seems to have worse conversation clarity when the call is in progress |
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13:20.06 | stodorovic | [May 27 14:19:43] WARNING[732]: chan_zap.c:6685 handle_init_event: Detected alarm on channel 1: No Alarm |
13:20.11 | stodorovic | what does that mean? |
13:20.22 | stodorovic | why is there an alarm but no alarm? |
13:23.25 | Baylink-work|afk | stodorovic: Could be a race condition in the notify code. |
13:23.59 | stodorovic | :/ |
13:24.20 | Katty | hello my asterisk does not work at all how to fix pls |
13:24.34 | Baylink-work|afk | Buy new shades. :-) |
13:25.44 | stodorovic | Baylink-work|afk: this might even be an old version of asterisk. not sure |
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13:29.55 | shader | I keep getting noticies about incomplete support for comfort noise in asterisk. Is there anything I need to do about this? |
13:30.00 | anny__ | hey all |
13:30.50 | anny__ | i want to connect 2 asterisk servers using SIP, i followed a lot of links on the net but with no luck |
13:31.04 | anny__ | does anyone have a guide i can use |
13:31.50 | [TK]D-Fender | shader: Yes... tell your client to STOP USING IT |
13:32.04 | shader | [TK]D-Fender: how important is that? |
13:32.24 | [TK]D-Fender | shader: How important is that warning being spewed into your console? |
13:32.29 | shader | i.e. what can go wrong if I don't? |
13:33.05 | [TK]D-Fender | anny__: You've already tried following guides and something isn't working out. Maybe you should show us the problem so we can help you with it. |
13:33.11 | [TK]D-Fender | anny__: PASTEBIN is your friend |
13:33.12 | [TK]D-Fender | ~pb |
13:33.13 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
13:33.24 | tzafrir_laptop | stodorovic, hmm... IIRC it means that the alarm was cleared |
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13:34.09 | anny__ | ok will do |
13:34.49 | ijpalmer | does anyone know how I can get the BT (uk) unobtainable message to one of my users when they dial an invalid number |
13:35.04 | stodorovic | tzafrir_laptop: yeah it was raised and cleared in the same second. Still, what could be the cause of it? |
13:35.47 | tzafrir_laptop | if the alarm was cleared pretty fast: yes |
13:36.11 | stodorovic | allegedly, within the same second |
13:36.28 | tzafrir_laptop | The "alarm" event does not tell which type of alarm there is. DAHDI need to check for it. Or maybe.. |
13:36.33 | stodorovic | which to a computer, is an eternity :) |
13:36.51 | tzafrir_laptop | I recall that this test was buggy at that time. Not sure if it actually applies there |
13:39.34 | *** part/#asterisk MhaddogM1 (~MhaddogM1@z65-50-118-232.ips.direcpath.com) |
13:39.50 | anny__ | i have created an entry for my problem here http://asterisk.pastey.net/136972, any help is apprectiated |
13:39.59 | anny__ | *appreciated |
13:43.37 | *** part/#asterisk muiro (~muiro@unaffiliated/muiro) |
13:44.24 | shader | anny__: could you paste a copy of the console log of the failed call attempt? |
13:45.19 | anny__ | failed to authenticate on invite for '444' |
13:45.35 | anny__ | user 444 is defined in sip.conf for server A |
13:45.58 | stodorovic | allow = gsm,adpcm,ulaw,alaw,g726 <-- is this good, or is adpcm or ulaw pretty rubbish? |
13:46.39 | shader | I think ulaw > alaw > gsm |
13:46.55 | shader | but I don't know about the others |
13:47.06 | stodorovic | hmm |
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13:47.29 | shader | I used to allow alaw and gsm on my system, but switching to only ulaw has dramatically improved quality |
13:48.02 | stodorovic | it's just the extension with the gsm,adpcm,ulaw,alaw,g726 is having sound quality and fluctuating volume problems, but the extension with allow = alaw,gsm,ulaw,adpcm has been fine |
13:48.03 | shader | anny__: is that failed to authenticate message from A or B? |
13:48.47 | anny__ | i'll rerun the test and paste the error in the pastebin |
13:48.53 | shader | ok |
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13:53.46 | anny__ | i updated my pastebin entry |
13:53.50 | anny__ | with the problem |
13:53.52 | [TK]D-Fender | anny__: and I see this listed as "555" You need to set up a peer on "A"'s side |
13:54.21 | [TK]D-Fender | anny__: and do Dial(SIP/thepeer/1234567890) |
13:54.45 | [TK]D-Fender | anny__: Set that up accordingly. in your peer do : fromuser=thenameisbracketsontheotherside |
13:55.01 | [TK]D-Fender | anny__: sendrpid=yes |
13:55.09 | [TK]D-Fender | anny__: trustrpid=yes |
13:56.08 | anny__ | host=dynamic? |
13:56.26 | shader | if its ip might change |
13:56.45 | shader | otherwise you can just use host=<ip address> |
13:56.50 | [TK]D-Fender | anny__: No, you already know the IP of the remote host... |
13:56.51 | anny__ | this peer should be server B |
13:57.33 | anny__ | and should i create a user on B's side |
13:57.41 | anny__ | identifying server A |
13:59.08 | [TK]D-Fender | anny__: http://asterisk.pastey.net/136975 |
14:00.15 | anny__ | thx D-Fender, i'll try this out |
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14:07.19 | anny__ | D-Fender: i got the same authentication problem, i pasted the error the pastebin |
14:07.30 | anny__ | *in the pastebin |
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14:13.37 | [TK]D-Fender | anny__: pastebin again with the configs on both sides, and with SIP DEBUG enabeld |
14:13.46 | anny__ | ok |
14:15.37 | anny__ | did u add an entry to pastebin? |
14:16.10 | [TK]D-Fender | anny__: not since your last |
14:16.43 | anny__ | so, i will re-run the tests with debug enabled |
14:16.44 | *** join/#asterisk pigpen (~mark@fw.seamans.cc) |
14:16.49 | anny__ | same tests as before |
14:17.49 | [TK]D-Fender | anny__: Oh, add "username=555" on your outgoing peer |
14:18.15 | Baylink-work|afk | ulaw and alaw should be roughly equivalent; the only different is the PCM curve. |
14:18.34 | Baylink-work|afk | And they should both be better than anything except maybe the 722 family. |
14:19.02 | anny__ | D-Fender:my outgoint peer aka otherserver has the fromuser=555 |
14:19.14 | [TK]D-Fender | addadd the username as well |
14:19.15 | anny__ | should i add useranme=555 also |
14:19.19 | [TK]D-Fender | yes |
14:19.20 | anny__ | ok |
14:19.31 | Chainsaw | Baylink-work|afk: Then again, last time I allowed 722 I had a call destination in Italia that ended up at half speed. Like the sample rate was wrong. |
14:20.03 | Baylink-work|afk | There's a documented sample rate problem in the 722 def; it's *supposed* to be ignored by proper drivers. |
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14:20.47 | Chainsaw | Baylink-work|afk: It was Telecom Italia or my SoundPoint IP 670 at fault. |
14:21.32 | Baylink-work|afk | Yeah; 722 appears a work in progress. :-) |
14:21.54 | Chainsaw | Baylink-work|afk: I shall just leave it on ulaw/alaw for the time being then :) |
14:23.26 | anny__ | D-Fender: i added username = 555, no luck, i pasted the sip packets in the pastebin |
14:23.46 | anny__ | D-Fender: i noticed that the error packet is proxy authentication failure |
14:23.49 | *** join/#asterisk Polysics (~luca@host113-41-static.25-87-b.business.telecomitalia.it) |
14:23.58 | Polysics | hello |
14:24.25 | Polysics | i still haven't figured out how/if i can get a client that has NO open outbound ports to connect to an Asterisk server |
14:24.43 | Polysics | doesn't Skype do something like that using a middleman server? |
14:24.58 | Polysics | i can communicate with the world only through an HTTP proxy |
14:25.06 | Deeewayne | has anyone ever heard of anyone successfully using chan_dahdi configured for CAMA in The Real World(TM) ? |
14:25.17 | [TK]D-Fender | anny__: Link please |
14:25.30 | anny__ | http://asterisk.pastey.net/136977 |
14:25.50 | Polysics | apparently STUN would be one half of the solution, would the other half be a client capable of firewall punching? |
14:26.31 | *** join/#asterisk Slugs_ (~Slugs@unaffiliated/slugs) |
14:27.04 | [TK]D-Fender | anny__: also debug from the other side, along with current configs |
14:27.25 | Polysics | am i delusional or is that some sort of solution? |
14:27.41 | angryuser | Polysics, yes, you are |
14:28.06 | Polysics | angryuser, as in "no outbound ports = dead in the water"? |
14:28.16 | Gnarfy | Polysics: you can always try something like cisco anyconnect to a cisco router, it makes a tunnel over http and just route your calls over that |
14:28.20 | Polysics | i don't want to reiterate, but Skype does it :-) |
14:28.40 | file | define an 'outbound port' |
14:28.42 | *** join/#asterisk Slugs_ (~Slugs@unaffiliated/slugs) |
14:28.48 | angryuser | Gnarfy, no outbound ports........ |
14:28.53 | file | as in no outbound connections? |
14:29.05 | Gnarfy | no outbound port "except http proxy" he said :) |
14:29.38 | file | then unless you tunnel through that http proxy using something else you can't establish a connection |
14:29.46 | angryuser | well, a tunnel |
14:29.53 | file | and Skype works because Skype controls the protocol and they wrote it so that it'll try every possible way it can to get out |
14:30.07 | file | direct connections, connecting through port 80, port 443, through an http proxy |
14:30.08 | Polysics | doesn't firewall punching allow for a temporary UDP connection, which I could use for IAX? |
14:30.36 | Polysics | file, so Skype routes voice itself through port 80? |
14:30.42 | file | if it has to |
14:31.24 | [TK]D-Fender | Skype tries every dirty trick in the book |
14:31.33 | [TK]D-Fender | And then writes a few more |
14:32.02 | *** join/#asterisk n3hxs (~HAMming@173-162-253-153-NewEngland.hfc.comcastbusiness.net) |
14:33.17 | Polysics | so, what could be a solution, at least a partial one? switch to IAX so everything goes through on a single UDP port and ask for that port to be allowed? |
14:34.31 | *** join/#asterisk mace (~mace@debian/developer/mace) |
14:34.55 | [TK]D-Fender | Polysics: Not the worst you could do... sounds like you're in a very hostile series of networks |
14:35.06 | *** join/#asterisk hugorebelo (~hugo@200-171-132-124.completo.com.br) |
14:35.15 | *** join/#asterisk bent_screwdriver (~socain00@74.255.249.66) |
14:35.39 | Polysics | [TK]D-Fender, government networks |
14:35.41 | *** join/#asterisk moy (~moy@bas1-unionville55-1177733627.dsl.bell.ca) |
14:35.59 | Polysics | i do have access to a DMZ though, does that allow for anything else? |
14:36.05 | *** part/#asterisk mace (~mace@debian/developer/mace) |
14:36.05 | Polysics | like, say, a SIP proxy? |
14:37.10 | Polysics | if such a thing exists, but it looks like it does |
14:37.22 | file | blinks |
14:38.03 | Gnarfy | Polysics, if you have access to place a proxy in the DMZ, surely you can open a few ports... like say: 5060 and 10k to 20k and forward them :p |
14:38.19 | Polysics | which proxy do you recommend? |
14:41.37 | *** join/#asterisk retentiveboy (~pdugas@74-95-28-37-Atlanta.hfc.comcastbusiness.net) |
14:42.07 | Gnarfy | btw, didn't asterisk get skype support? maybe relay your calls through skype... |
14:42.57 | *** join/#asterisk kartik (~koolkarti@117.199.113.102) |
14:46.26 | Gnarfy | I've got a question about rtpkeepalive, how do i go about testing/seeying if this works? incoming calls (ending in a silent asterisk goto loop), are terminated after 60 secs of silence. but logging: sip set debug, never shows any keep alive signals being sent... or am i looking in the wrong place? |
14:46.32 | *** join/#asterisk Chinorro (~Chino@244.213.117.91.dynamic.mundo-r.com) |
14:46.44 | pabelanger | ~sfa |
14:47.01 | *** join/#asterisk DrDamnit (~michael@173-165-161-161-atlanta.hfc.comcastbusiness.net) |
14:47.02 | pabelanger | infobot :) |
14:47.03 | infobot | (: |
14:47.16 | [TK]D-Fender | ~skypeforasterisk |
14:47.16 | infobot | from memory, skypeforasterisk is a Digium-made channel driver that allows Asterisk to connect to the Skype network using a Skype-supported connection method. Unlike all other solutions it does not require X11, kernel modules, or a Windows machine. See http://www.digium.com/skype for details |
14:47.18 | *** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
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15:01.00 | anny__ | D-Fender: thx for your help |
15:02.16 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
15:02.20 | [TK]D-Fender | anny__: Functional now? |
15:02.50 | anny__ | yes |
15:02.50 | Polysics | ok, no, proxy is not an option |
15:03.01 | anny__ | i missed creating the extension in the context |
15:03.11 | anny__ | i saw it in the debug messages |
15:03.13 | anny__ | on server B |
15:03.21 | anny__ | thx for ur help |
15:03.23 | [TK]D-Fender | anny__: Probably became evident when you looked at the debug on the other side :) |
15:03.30 | anny__ | true :) |
15:03.41 | [TK]D-Fender | anny__: Gotta pay attention to what the 404 is in response to. |
15:03.46 | Polysics | anyone care to explain why firewall punching will not work in the above situation? |
15:03.54 | Polysics | assuming i do have a client capable of doing it |
15:04.39 | [TK]D-Fender | Polysics: firewall punching requires an intermediary server to be always contactable |
15:04.46 | [TK]D-Fender | Polysics: Its not just a client |
15:04.47 | anny__ | D-Fender: yes |
15:05.02 | [TK]D-Fender | anny__: Ok, well keep at it... sounds like you're on your way |
15:05.07 | Polysics | [TK]D-Fender, i can have that through theHTTP proxy, no? |
15:05.34 | Polysics | i can put together my own server |
15:05.44 | [TK]D-Fender | Polysics: And what protocl is going to let you do that? |
15:06.13 | Polysics | what is the firewall punching supposed to do? some sort of STUN registration, right? |
15:06.40 | Polysics | as long as i get the relevant information to the clients, i can trasmit it over HTTP web services, or not? |
15:06.57 | Polysics | problem is i would then have to code a client to support this |
15:07.35 | *** join/#asterisk rare1980_ (~as@109.169.28.42) |
15:07.39 | kaldemar | Gnarfy: sip debug shows SIP messages, not RTP. try "rtp set debug on". |
15:08.27 | Gnarfy | kaldemar: doh, will try |
15:09.55 | Gnarfy | haha, such spam :x |
15:10.34 | [TK]D-Fender | Polysics: Both sides need to eb doing this. You can't punch your way in... you need an outbound attempt to leaev the door a little open first. |
15:10.44 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
15:10.44 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
15:12.09 | Polysics | [TK]D-Fender, of course, but i am more worried about the client |
15:12.32 | Polysics | it is starting to look like this simply cannot be done with *, not asterisk's fault, fo course |
15:12.46 | leifmadsen | damn, I missed something possibly cool! :) |
15:12.51 | [TK]D-Fender | Polysics: You're WAY into ##networking land... and having to invent crazy clients... |
15:12.51 | Polysics | more to do with donkey-brained government sysadmins |
15:13.30 | Polysics | whose idea of security is building houses without doors or windows :-) |
15:13.55 | leifmadsen | takes the tunnel |
15:14.16 | [TK]D-Fender | fires up his Heisenburg Compensators |
15:14.22 | Polysics | leifmadsen, if you were referring to our conversation, still the old "client is inside a network without ANY outbound ports, only an HTTP proxy" |
15:14.34 | leifmadsen | yuck |
15:14.59 | Polysics | i was wondering about possibilities like firewall punching ala Skype |
15:15.33 | Polysics | but apparently it would require to rewrite both clients and probably the server to support getting the punching information over HTTP from a central server |
15:15.40 | [TK]D-Fender | Polysics: at that point it'd be like encapsulating an SSL VPN into HTTP because if you're raw-proxy'd then it needs to route via HTTP as well... not just sit on port 80 |
15:15.46 | [TK]D-Fender | Polysics: Such a horrific mess |
15:15.47 | Polysics | it is doable in theory, not something i would actually do |
15:16.26 | Polysics | tbh it looks like i would just use IAX and get ONE port opened |
15:18.36 | Polysics | am i right in assuming that simply having some sort of STUN server wouldn't help anyway? |
15:18.51 | Polysics | because no open ports = no way to "call" the STUN server |
15:19.05 | leifmadsen | ya, STUN won't help in that situation |
15:19.26 | leifmadsen | it just tells the end point what their external IP is and some port information, but if all are blocked (especially for the request) it won't even work |
15:19.36 | leifmadsen | STUN is not a magic bullet :) |
15:21.53 | Polysics | leifmadsen, do you know of any SIP softphone clients that have better than normal firewall traversal support? |
15:22.09 | Polysics | i am probably hoping to find something that just doesn't exist though |
15:22.48 | *** part/#asterisk JayT (unstable@tor/regular/sid) |
15:23.35 | leifmadsen | Polysics: sorry, no idea |
15:23.47 | leifmadsen | skype is really probably the best app of getting through crazy networks |
15:24.02 | file | I'm afraid you can't change reality. |
15:24.09 | leifmadsen | file: lies! |
15:24.18 | shader | I keep getting Error: Couldn't find mailbox 202 in context default. What determines which mailboxes the system is looking for? How do I delete a mailbox and prevent * from continuing to poll it? |
15:24.30 | leifmadsen | shader: voicemail.conf |
15:24.36 | *** join/#asterisk fofware (fabian@190.7.25.160) |
15:24.53 | leifmadsen | shader: also if the phone is asking for mailbox 202, tell the phone to stop subscribing to it |
15:25.33 | shader | leifmadsen: I don't have any phones asking for it; what I'm trying to do is delete the mailbox, so it is no longer in voicemail.conf but asterisk keeps looking for it |
15:25.48 | leifmadsen | then you haven't deleted it or reloaded app_voicemail.so |
15:26.27 | shader | deleted which? the actual directory or the entry in voicemail.conf |
15:26.35 | leifmadsen | the entry in voicemail.conf |
15:26.48 | shader | the entry is certainly no longer there, and I've done a voicemail reload |
15:26.57 | shader | so I have to reload the module? |
15:27.04 | leifmadsen | you just said you did |
15:27.07 | shader | oh |
15:27.17 | leifmadsen | then you have a dialplan line that is trying to get to voicemail 202 |
15:27.25 | leifmadsen | Voicemail(202@default) would now give you that error |
15:27.51 | shader | I don't have any dialplan looking for 202 |
15:28.02 | leifmadsen | well asterisk isn't just randomly looking for mailbox 202 |
15:28.04 | shader | I do have voicemail set to poll the mailboxes |
15:28.07 | leifmadsen | more information required |
15:28.12 | shader | ok |
15:29.19 | Gnarfy | My DID provider hangs up calls incoming to asterisk after 1 minute of silence, i've been trying rtpkeepalive, but this is not helping. What other feature do i need to look at, session timers? |
15:29.36 | leifmadsen | session timers are for SIP not RTP |
15:29.38 | [TK]D-Fender | shader: Where is the DEBUG and CONFIGS for us to look at? |
15:30.17 | *** join/#asterisk cusco (~trilili@213.63.137.210) |
15:30.20 | cusco | hi |
15:30.31 | shader | http://pastebin.com/CkfMpEfZ |
15:30.35 | cusco | dahdi_scan returns alarms=UNCONFIGURED |
15:30.36 | shader | config for voicemail.conf |
15:30.41 | cusco | meaning problem with the PRI line? |
15:30.54 | [TK]D-Fender | shader: DEBUG |
15:31.10 | cusco | http://paste.debian.net/75068/ |
15:31.10 | Gnarfy | leifmadsen, i'm not sure why they hangup... can this not be done with sip keepalive signals ? |
15:31.19 | shader | [TK]D-Fender: any particular debug? |
15:31.23 | leifmadsen | Gnarfy: I also do not know why they hangup |
15:31.30 | [TK]D-Fender | shader: Show us the ERROR. |
15:31.40 | shader | all I'm getting is ERROR[31749]: app_voicemail.c:1840 __messagecount: Couldn't find mailbox 202 in context default |
15:32.03 | shader | I mentioned that earlier though |
15:32.13 | [TK]D-Fender | shader: ENTIRE output with debug 10 |
15:32.25 | Trixboxer | cusco: Have you configured the span in /etc/dahdi/system.conf |
15:32.46 | *** join/#asterisk grEvenX (~even@cC0FD00C3.dhcp.bluecom.no) |
15:32.56 | cusco | Trixboxer: dahdi_gen_conf did it for me, and I checked it |
15:32.58 | cusco | seems OK |
15:33.06 | shader | [TK]D-Fender: what do you mean debug 10? |
15:33.17 | [TK]D-Fender | shader: core debug 10 |
15:33.21 | Trixboxer | which country ? |
15:33.32 | Trixboxer | and what does that file contains now ? |
15:33.40 | cusco | Trixboxer: us but I changed it to pt |
15:33.48 | cusco | though it makes no difference |
15:33.50 | cusco | hold |
15:33.53 | drmessano | Wasn't "trixboxer" a Fiona Apple song? |
15:33.53 | shader | no such command 'core debug' |
15:34.17 | [TK]D-Fender | shader: core set debug 10 |
15:34.19 | cusco | Trixboxer: http://paste.debian.net/75070/ |
15:34.20 | Trixboxer | I never heard that :) |
15:34.33 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
15:34.54 | shader | [TK]D-Fender: still gives the same exact error |
15:34.57 | shader | and nothing else |
15:35.22 | Trixboxer | cusco: that is identical to mine : |
15:35.30 | [TK]D-Fender | shader: go look in the help for the syntax |
15:35.52 | cusco | Trixboxer: yes should be fine. I am doubtibng of telco equipment right now |
15:36.02 | shader | no, I mean that I got debug to go from 0 to 10, and it's not outputting any more information |
15:36.24 | Trixboxer | cusco: Check the LED at the back of card.. what color does it show ? |
15:36.39 | [TK]D-Fender | shader: When do you get that message? |
15:36.48 | *** join/#asterisk Carlos_PHX (~Carlos@ip68-99-199-10.ph.ph.cox.net) |
15:36.56 | cusco | Trixboxer: can't. Machine is not here |
15:37.46 | shader | in groups of 3 every 30 seconds, like the poll mailbox setting seems to imply it should |
15:38.32 | [TK]D-Fender | shader: Pastbin your configs, and the CLI output. SIP peers, everything |
15:38.50 | shader | nothing else mentions mailbox 202, only voicemail.conf |
15:38.59 | cusco | Trixboxer: ouch, dahdi_scan suddently changed to alarm=OK |
15:39.04 | Trixboxer | wow |
15:39.07 | cusco | where as before was = UNCONFIGURED |
15:39.09 | cusco | dunno why |
15:39.09 | cusco | :) |
15:39.12 | Trixboxer | :) |
15:39.16 | cusco | but asterisk shows pri as DOWN |
15:39.17 | shader | and even voicemail.conf doesn't mention it anymore |
15:39.33 | Trixboxer | cusco: U just need an inbound route n start taking calls :) |
15:39.37 | cusco | (maybe because I just installed dahdi and restaerted asterik, it needed a moment to bring the interface up??) |
15:39.59 | shader | so I've already posted all of the relevent config, and you've seen the CLI error, but if you want I can add it to the pastebin |
15:40.17 | [TK]D-Fender | shader: And have you restarted * completely? |
15:40.33 | shader | no, I was hoping that wasn't necessary to remove a mailbox |
15:40.45 | [TK]D-Fender | shader: You have to reload all of the VM modules. |
15:40.50 | cusco | Trixboxer: I don't know the DDI |
15:40.52 | cusco | ! |
15:40.58 | shader | ok |
15:41.26 | [TK]D-Fender | cusco: Call out and see what you get for CID |
15:41.32 | Trixboxer | it may be the last 3 or 4 digits of the telephone no |
15:44.40 | Gnarfy | hmm, bah... rtpkeepalive does not show up with rtp debug enabled. |
15:44.55 | *** join/#asterisk bmoraca_work (~bmoraca@66-242-174-254.ceres.bvn.net) |
15:45.01 | stodorovic | eeeeesh! this Asterisk version is 1.4.18.1 :( |
15:45.10 | cusco | [TK]D-Fender: can't pri show spans is down |
15:45.13 | cusco | dunno why |
15:45.14 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
15:47.16 | shader | [TK]D-Fender: so, reloading app_voicemail.so didn't help; I had to restart asterisk |
15:47.19 | shader | oh well |
15:50.35 | cusco | [TK]D-Fender: why does pri show spans show span as DOWN |
15:50.36 | cusco | ? |
15:50.42 | cusco | dahdi_sacn shows alarm=OK |
15:51.03 | *** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com) |
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15:58.25 | *** part/#asterisk c0rnoTa (~c0rnoTa@nas-8.emserv.ru) |
16:02.29 | *** join/#asterisk Raden (~Raden@71.89.121.119) |
16:04.18 | *** join/#asterisk manatails (~admin@reactos/tester/manatails) |
16:04.26 | manatails | hii |
16:04.58 | manatails | what should I do to make calls with external sip server? |
16:05.46 | p3nguin | Pay your bill. |
16:06.06 | *** join/#asterisk grEvenX (~even@cC0FD00C3.dhcp.bluecom.no) |
16:07.51 | shader | does reloading configs drop calls? |
16:08.24 | *** join/#asterisk ChrisWi (~admin@mx2.wwserver.net) |
16:09.31 | p3nguin | Not typically. |
16:11.10 | shader | ok |
16:18.30 | *** join/#asterisk diegomad (~mad@190.146.200.120) |
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16:27.01 | cusco | what can I do when pri show spans returns "PRI span 1/0: Provisioned, Down, Active" |
16:28.18 | Baylink-work|afk | Down generally means the switch at the other end isn't syncing up; if unplugging it, and/or rebooting the switch doesn't clear it, you'll have to call the carrier. |
16:29.15 | cusco | pri intense debug span 1 keeps showing: http://paste.debian.net/75080/ |
16:29.35 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
16:32.31 | TSM | is it normal to see a PRI span restart itself every now and again 'B-channel 0/1 successfully restarted on span 1' |
16:32.56 | Baylink-work|afk | Yeah; they do that about hourly here |
16:33.50 | TSM | oks |
16:39.37 | *** join/#asterisk g_r_eek (~g_r_eek@dslb-094-218-206-184.pools.arcor-ip.net) |
16:42.25 | tzafrir_laptop | TSM, what version of asterisk? |
16:42.49 | *** join/#asterisk centoslinux (~centoslin@s0021-0018.dsl.start.no) |
16:50.08 | *** join/#asterisk Wolfeyes (~Wolfeyes@41.124.132.70) |
16:50.34 | *** join/#asterisk saftsack (~oliver@p5DDCF965.dip.t-dialin.net) |
16:50.45 | *** join/#asterisk mrgabu (~gbdurante@187.38.158.209) |
16:50.55 | saftsack | hey, i have a problem while negotiating t.38. here is a log: http://nopaste.info/b1c845cbd6.html |
16:51.00 | saftsack | are there obvious errors? |
16:51.19 | saftsack | the patton says "# SIP/2.0 415 Unsupported Media Type " but i don't know why |
16:52.15 | *** part/#asterisk mrgabu (~gbdurante@187.38.158.209) |
16:52.32 | *** join/#asterisk mrgabu (~gbdurante@187.38.158.209) |
16:52.51 | Wolfeyes | anyone here do asterisk in south africa? |
16:53.12 | chazzam | saftsack: looks like the patton doesn't support T.38 |
16:53.56 | saftsack | the patton supports t.38. that's the issue ... |
16:54.12 | chazzam | is it enabled? |
16:54.35 | saftsack | scenario as follows: fax from an external station comes over the patton and goes then to the asterisk where it should terminate on receivefax() |
16:54.42 | saftsack | chazzam, yes it is enabled |
16:55.00 | chazzam | asterisk is trying to switch to T.38 and we are getting back not supported from the other end |
16:56.11 | *** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com) |
16:57.50 | saftsack | hmm it is on. with the latest non svn asterisk i got another errormsg yesterday |
16:58.02 | saftsack | there the patton said: not acceptable proposal |
16:58.17 | TSM | just wandering does the upgrade of freepbx to 2.7 change the extensions.conf file at all or even replace it? |
16:58.44 | KavanS | Wolfeyes, probably! |
17:00.46 | *** join/#asterisk pabelanger (~pabelange@nat/digium/x-vkbdiqgnneqeyegf) |
17:01.04 | *** join/#asterisk centoslinux (~centoslin@s0021-0018.dsl.start.no) |
17:01.48 | *** join/#asterisk emora (~emora@213.236.9.114) |
17:02.24 | emora | hello |
17:02.34 | TSM | woops wrong room |
17:09.41 | emora | Any opinions on using Cisco Unified SIP Phone 3911 with Asterisk ? |
17:09.50 | *** join/#asterisk jpeeler (~jpeeler@asterisk/digium-software-dev/jpeeler) |
17:10.45 | emora | We can get a really good deal on these at the moment but since we've never used them before, I know nothing about them |
17:11.19 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
17:13.14 | Wolfeyes | KavanS, thank you but I am trying to contact anyone that does for help and well noone is replying. |
17:14.49 | Wolfeyes | I need the per minute billing for my country if anyone knows that? (South Africa) |
17:15.11 | Nugget | emora: avoid cisco phones. |
17:15.35 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
17:16.42 | emora | Nugget: why do you say to avoid them ? We've had good experiences with Cisco Linksys phones so far. |
17:17.41 | Nugget | they're undocumented, unreliable, and it's a pain in the ass to legally acquire the firmware updates. |
17:18.01 | Nugget | and they're expensive (relatively) compard to better performing options |
17:18.01 | Katty | hi |
17:18.05 | Katty | hello my asterisk does not work at all how to fix pls |
17:18.06 | Nugget | huggles katty |
17:18.25 | Katty | huggles on Nugget |
17:18.50 | emora | Firmware: yep, that's true. Documentation: I've found them to be very well documented. |
17:18.52 | cusco | hi.... |
17:18.54 | cusco | [May 27 18:17:42] WARNING[3072]: loader.c:386 load_dynamic_module: Error loading module 'app_fax.so': libspandsp.so.2: cannot open shared object file: No such file or directory |
17:19.08 | cusco | there is a /usr/local/lib/libspandsp.so.2 |
17:19.08 | emora | What would you consider a "better performing option"? |
17:19.19 | Nugget | emora: the linksys phones must not use the newer-style xml configuration then |
17:19.22 | KavanS | I've not used cisco phones |
17:19.25 | *** join/#asterisk verywiseman (~khaled@unaffiliated/verywiseman) |
17:19.27 | KavanS | these polycoms are epic |
17:19.36 | Nugget | Polycom seems to be a popular choice here among people whose opinion I trust |
17:19.43 | Qwell | cusco: and is /usr/local/lib/ in your LD search path? |
17:19.45 | Nugget | but I run all ciscos |
17:20.02 | Qwell | (hint: it's not) |
17:20.03 | KavanS | doh, not your decision I take it? |
17:20.03 | emora | I have a problem sourcing Polycom in Spain. |
17:20.18 | Nugget | I'm just stubborn. :) |
17:20.29 | Nugget | I'd rather run ciscos and whinge about it than switch |
17:20.51 | emora | Yes, up til now we've used the SPA-922 and SPA-942 models mostly. They are not configured via XML |
17:20.52 | cusco | Qwell: how do I check? |
17:21.08 | cusco | $LD_LIBRARY_PATH is empty |
17:21.20 | Qwell | cusco: /etc/ld.so.conf |
17:21.31 | emora | KavanS: I have the final word |
17:21.37 | KavanS | lol |
17:22.55 | cusco | Qwell: yes it is! # libc default configuration |
17:22.55 | cusco | /usr/local/lib |
17:23.39 | *** join/#asterisk wonderworld (~ww@ip-62-143-22-226.unitymediagroup.de) |
17:23.39 | Qwell | cusco: run ldconfig |
17:24.27 | cusco | ok taht did it |
17:24.28 | cusco | :/ |
17:24.39 | Qwell | that'll be $199.99 |
17:24.46 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
17:24.54 | leifmadsen | I'll discount by $0.01! |
17:24.59 | cusco | heh |
17:25.07 | emora | Does anyone now when the Cisco Linksys SPA-942 and SPA-922 go EOL? |
17:25.07 | Qwell | leifmadsen: sorry, that coupon code has expired. |
17:25.13 | leifmadsen | Qwell: darn! |
17:25.17 | Qwell | it now adds $47.83 |
17:25.21 | leifmadsen | eep! |
17:25.24 | leifmadsen | good thing he didn't use that code |
17:25.41 | Qwell | but you did! |
17:25.41 | cusco | thanks :) |
17:26.22 | cusco | I am available to collect those $47.83 (now that somebody actually spent them) |
17:27.24 | Faustov | any idea what can cause a "603 declined" response from a SIP provider whenever more than 1 sip call is being established? I have multiple accounts with this provider and just one of them is having this problem, so I think it is something local... |
17:27.49 | Qwell | they probably don't allow multiple calls on that account |
17:28.31 | Katty | Qwell |
17:28.34 | cusco | that seemed rectoric |
17:28.36 | Qwell | Katty |
17:30.53 | Katty | shall we hug |
17:31.01 | *** join/#asterisk wcselby (~dubba@99-146-243-194.lightspeed.hstntx.sbcglobal.net) |
17:31.04 | Katty | :> |
17:31.05 | Qwell | we shan't |
17:31.06 | wcselby | o/ |
17:31.10 | wcselby | ~pb |
17:31.11 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
17:31.11 | shader | group hug! |
17:31.11 | Katty | why? do you have germs? |
17:31.17 | Katty | hugs wcselby |
17:31.25 | wcselby | hey Katty |
17:31.27 | TSM | is there a way to detect digits when playingback a audio file? |
17:31.35 | Qwell | TSM: Background() |
17:31.43 | wcselby | ugh.....where are my timestamps |
17:31.46 | shader | instead of Playback(), that is |
17:31.47 | Katty | i ate them. |
17:31.56 | Katty | they were delicious |
17:31.58 | Katty | and with good timing too |
17:32.08 | wcselby | ahhh, there they are |
17:32.17 | wcselby | lol Katty |
17:32.19 | TSM | Qwell: this i thought but the problem being, what happenens when the audio finishes |
17:32.28 | wcselby | little dash of thyme with those timestamps? |
17:32.28 | Katty | anyone watch a good movie lately? |
17:32.31 | Qwell | Katty: I love fudging me some timestamps. so much better |
17:32.47 | TSM | Qwell: I then want to jump to another macro |
17:32.49 | wcselby | Katty - I saw shrek final chapter yesterday with the kids. |
17:32.51 | wcselby | was pretty good |
17:33.02 | wcselby | watched zombieland the other night by myself, it was funny, in a gross sort of way |
17:33.03 | Qwell | ~book |
17:33.04 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
17:33.08 | Qwell | TSM: read the chapter on dialplan basics |
17:33.09 | Katty | i was hoping for something i could rent, or get off amazon or something |
17:33.24 | *** join/#asterisk moy (~moy@bas1-unionville55-1177733627.dsl.bell.ca) |
17:33.58 | wcselby | Katty - what do you consider a good movie? chick flick? action? gore / horror? |
17:34.18 | wcselby | also, can anyone help me with this dahdi compile issue I'm having ---> http://pastebin.com/WHbpx56M |
17:34.39 | Qwell | wcselby: upgrade |
17:35.03 | wcselby | Qwell - upgrade what, exactly? this is 2.2.1.2+2.2.1.1 released 2 days ago |
17:35.14 | Katty | wcselby: anything but horror or...just really really stupid comedy |
17:35.24 | Katty | wcselby: i don't handle horror very well :< |
17:36.00 | Qwell | hrm |
17:36.13 | wcselby | Katty - the wife and I were watching Bourne Identity last night, it's a good one. |
17:36.21 | Katty | that is a good one. |
17:36.28 | Katty | i enjoyed it, well the latest one |
17:36.29 | *** join/#asterisk sysreq (~sysreq@unaffiliated/sysreq) |
17:36.38 | Katty | idk if there were other ones or not |
17:36.41 | Katty | hi sysreq |
17:38.04 | Katty | i watched a really really nice movie last night called Young Victoria |
17:38.09 | Katty | it was sappy |
17:38.12 | *** join/#asterisk p3nguin (gpz5GvdFkf@cpe-0210-4bff-fe2b-9074.dhcp.a2infotech.com) |
17:38.17 | Katty | hi p3nguin |
17:38.32 | Qwell | tzafrir_laptop: I thought I saw something fixed recently with spinlock stuff.. having trouble finding it now. do you know? |
17:39.45 | Katty | wcselby: it was basically about queen victoria growing up, becoming queen, and getting married to prince albert |
17:40.17 | wcselby | yeah i think my wife watched it |
17:40.21 | tzafrir_laptop | Qwell, there was a recent addtion of an explicit '#include <linux/sched.h>', but this is not it |
17:40.30 | Katty | wcselby: http://thoughtsonfilms.files.wordpress.com/2010/03/theyoungvictoria-2.jpg <- random shot from movie |
17:40.46 | Katty | wcselby: reminded me of Pride and Prejudice a bit |
17:40.57 | Katty | wcselby: with keira knightley and whatshisface |
17:41.13 | wcselby | Katty - yeah I know which one you mean |
17:41.32 | Katty | wcselby: did you watch Legion? |
17:41.50 | wcselby | let's see, if you're into that kind of stuff, there's Shakespere in Love, Elizabeth, the second Elizabeth movie, hmmm.... |
17:41.53 | wcselby | no I didn't watch Legion |
17:42.01 | *** part/#asterisk Wolfeyes (~Wolfeyes@41.124.132.70) |
17:42.09 | Katty | wcselby: it's one of those apocalypse movies |
17:42.25 | wcselby | Katty - yeah I saw the preview, the crazy old demon lady |
17:42.30 | Katty | ya that's the one |
17:43.07 | Katty | guess i could always watch the princess and the frog. |
17:48.30 | Katty | Qwell: are you goin to blizzcon this year |
17:48.49 | Katty | Qwell: tickets go on sale june 2nd |
17:49.20 | Qwell | Katty: no |
17:49.45 | wcselby | Katty - a good friend of mine is going to the SOE equivalent of that, he's a major EQ2 buff |
17:49.57 | Qwell | tzafrir_laptop: check out the pastebin above by wcselby. kinda funky. |
17:50.00 | *** join/#asterisk Slugs_ (~Slugs@unaffiliated/slugs) |
17:50.05 | Qwell | wcselby: also you should probably report a bug. issues.asterisk.org |
17:50.10 | wcselby | http://pastebin.com/WHbpx56M |
17:50.26 | tzafrir_laptop | saw it. Looks familiar |
17:50.40 | wcselby | tzafrir_laptop - I posted it to the list a couple nights ago |
17:50.47 | wcselby | you may have seen it there |
17:50.50 | Qwell | tzafrir_laptop: yeah, to me too.. can't remember why though |
17:50.53 | Qwell | wcselby: that might be it |
17:51.13 | wcselby | I got no response for a couple days, so I came in here, to see if anyone had any ideas... |
17:51.35 | Qwell | ahh, there it is |
17:52.03 | Qwell | wcselby: can you try upgrading kernels? I think there's newer in the 5.4 line than .11.1.el5 |
17:52.34 | Katty | Qwell: well bummer. |
17:52.47 | Katty | wcselby: that sounds like a lot of fun too |
17:53.00 | Katty | wcselby: i ain't got time for EQ tho :P |
17:53.11 | Katty | Qwell: i started a new pally, alliance side, on vashj |
17:53.16 | Qwell | Katty: tsk tsk |
17:53.28 | Katty | Qwell: i know, but i got a friend who plays on that server |
17:53.37 | Katty | Qwell: it's pvp :< |
17:53.38 | wcselby | Qwell - yeah, I think I may get access to the box again tonight. It's at a client site, and they don't want any downtime during business hours. I inherited this from a tech who tried to fix it but couldn't. He said the only things he'd done was add some new kernel modules for the NIC drivers he had, but he wasn't very clear or sure what he had done. :( |
17:54.03 | wcselby | Katty - pally....ewww. I got to level 40 with a pally and gave up, it was soooooo slow to do anything |
17:54.19 | wcselby | but then again, I was bored of the game by then. |
17:54.33 | Qwell | tzafrir_laptop: you sure the sched.h addition wouldn't cause this? looks like module.h is including it directly. things are being redefined. |
17:55.20 | tzafrir_laptop | nothing wrong with including it twice |
17:56.01 | *** join/#asterisk m_tadeu (~quassel@89.180.25.208) |
17:56.22 | m_tadeu | hi everyone |
17:56.56 | Katty | wcselby: my main is a healadin |
17:57.59 | m_tadeu | in order to have asterisk receiving several calls to several users, what caracteristics should the phone line have? |
17:58.20 | Qwell | m_tadeu: none. not possible. |
17:58.26 | Qwell | 1 line = 1 call |
17:58.45 | Qwell | you'd need a PRI, or multiple analog lines. |
17:59.21 | m_tadeu | what's a PRI? |
17:59.27 | Qwell | ~book |
17:59.28 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
17:59.30 | Qwell | start there |
17:59.32 | leifmadsen | or Google |
17:59.35 | Qwell | ~telephony 101 |
17:59.36 | Qwell | or there |
17:59.42 | Qwell | glares at infobot |
17:59.44 | Qwell | ~telephony101 |
17:59.49 | Qwell | stupid bot |
17:59.53 | m_tadeu | lolol |
17:59.58 | leifmadsen | stupid Qwell |
18:00.01 | Qwell | ~101 |
18:00.02 | infobot | rumour has it, 101 is Telephony 101, which is a good read if you're unfamiliar with traditional TDM telephony. You can download it at http://www.stromcarlson.com/docs/basics/NTtelephony101.pdf |
18:00.02 | m_tadeu | thx |
18:00.07 | wcselby | m_tadeu - PRI is otherwise known as a T1 or an E1 depending on what part of the world you're in, but not always. read the book. :) |
18:00.09 | Qwell | infobot: God I hate you |
18:00.26 | leifmadsen | or J1 |
18:00.36 | Qwell | leifmadsen: silly J1ers ;) |
18:00.40 | wcselby | leifmadsen - hence "but not always". :) |
18:00.44 | m_tadeu | because I need to have the same phone number for multiple connections |
18:00.45 | leifmadsen | but those can really carry any signalling -- PRI is just one method |
18:01.00 | Qwell | m_tadeu: then you'll need a PRI, or some rollover features on the multiple analog lines |
18:01.24 | leifmadsen | or an ITSP that allows multiple simultaneous calls |
18:01.25 | tzafrir_laptop | wcselby, what happens if you remove the explicit '#include <linux/sched.h>' from drivers/dahdi/dahdi-base.c ? |
18:01.33 | m_tadeu | ok...I'm gonna check that stuff that Qwell posted |
18:01.37 | *** join/#asterisk emora (~emora@213.236.9.114) |
18:01.52 | Qwell | listen to leifmadsen. he practically wrote the book on telephony! |
18:01.59 | leifmadsen | just asterisk |
18:02.18 | Qwell | well then. listen to leifmadsen. he practically wrote the book on Asterisk! |
18:02.34 | leifmadsen | I just faked my way through it |
18:03.08 | leifmadsen | Qwell: you should see the dialplan I just wrote for the next edition just for allowing calling between extensions and outbound |
18:03.09 | wcselby | Qwell, tzafrir_laptop - as requested, the bug has been submitted on issues.asterisk.org - https://issues.asterisk.org/view.php?id=17411 |
18:03.11 | leifmadsen | it's like 3 pages long |
18:03.36 | m_tadeu | cool leifmadsen...I'll be glad to listen...but I'm still missing some basic knowledge |
18:04.01 | wcselby | tzafrir_laptop - I'll have to try that tonight. production system, users don't want to be offline. |
18:04.12 | Qwell | wcselby: wont have to install it or anything |
18:04.16 | Qwell | just do a make |
18:04.23 | Qwell | wont hurt anything |
18:04.43 | wcselby | Qwell - good point, I'll see if I can get access to the box remotely (i'm at my home office at the moment). |
18:05.54 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
18:06.13 | Qwell | wcselby: FWIW, there is a newer kernel. .15.1.el5 |
18:06.35 | Qwell | it's possible that it's a bug specific to that kernel you've got. maybe they didn't include protection in sched.h, I dunno |
18:08.12 | *** join/#asterisk DennisG (~DennisG@84.30.136.208) |
18:10.58 | *** join/#asterisk cosmicwombat (~cosmicwom@69.7.44.68) |
18:14.22 | wcselby | Qwell, tzafrir_laptop - I'm waiting to hear back about the remote access information from the original tech. I'm going to step away for a little while. I'll update next I come back if he's responded or not and if I'm able to make the requested change. Thanks again for your help! |
18:19.39 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
18:29.12 | *** join/#asterisk verywiseman (~khaled@unaffiliated/verywiseman) |
18:30.06 | *** join/#asterisk lordvadr (~something@jose-tc.ctc.biz) |
18:31.10 | lordvadr | Hi everybody. Can I use boolean data types in realtime config tables for options that are boolean in nature? Basically, does 't' and 'f' mean 'yes' and 'no' to asterisk? |
18:33.10 | Corydon76-dig | lordvadr: yes |
18:33.30 | lordvadr | thank you |
18:33.51 | Corydon76-dig | You can also spell them out, and they are not case-sensitive |
18:34.17 | Corydon76-dig | Also, "on" and "off" |
18:34.48 | Corydon76-dig | Also, "1" and "0" for options which are truly boolean (NOT qualify) |
18:36.58 | *** join/#asterisk rubbs (~rubbs@cpe-71-72-56-140.neo.res.rr.com) |
18:38.03 | m_tadeu | what sound server does asterisk use? |
18:38.03 | lordvadr | yeah, the next think I have to do figure out which ones are truly boolean |
18:38.54 | lordvadr | can one pull off templates with the realtime config? |
18:38.55 | m_tadeu | I'm asking because it shouldn't be mixed with desktop environments, because may cause sound quality problems |
18:39.29 | lordvadr | m_tadeu: I don't believe it uses a sound server. I think all mixing is done internally. |
18:39.50 | Corydon76-dig | lordvadr: no, templates cannot be done with realtime |
18:39.51 | *** part/#asterisk mrgabu (~gbdurante@187.38.158.209) |
18:39.54 | m_tadeu | lordvadr: ah ok |
18:39.59 | *** join/#asterisk TimeRider (steve@5ac7b311.bb.sky.com) |
18:40.52 | lordvadr | Corydon76-dig: Thanks. I was planning on pulling something similar off with views anyway but templates would be nice |
18:41.44 | lordvadr | m_tadeu: In fact, I'm almost 100% certain it doesn't since all my asterisk installations are on headless machines with no sound server running. |
18:42.43 | *** join/#asterisk mrgabu (~gbdurante@187.38.158.209) |
18:43.41 | m_tadeu | so it won't take advantage of a sound card dsp? |
18:44.11 | Qwell | m_tadeu: it won't use your sound card |
18:46.16 | m_tadeu | oki |
18:49.55 | *** join/#asterisk niekvlessert (~niek@82-171-252-6.ip.telfort.nl) |
18:50.23 | niekvlessert | if I reload asterisk config with reload pbx_config, will my queue stats still be there? |
18:51.15 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
18:51.52 | *** join/#asterisk IsUp (~nocturne@unaffiliated/isup) |
18:51.54 | IsUp | hello |
18:52.27 | IsUp | I need a VoIP provider which supports any CLI |
18:52.42 | IsUp | i want to call myself from +01234567 for example |
18:52.48 | Baylink-away | "CNID". |
18:52.57 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-50.hsd1.nj.comcast.net) |
18:53.04 | IsUp | can anybody test with me? |
18:53.14 | Baylink-away | I gather an impression that SIP providers that will let you set a random CNDI are fairly rare, |
18:53.15 | IsUp | if i can find any provider, i'll buy credits |
18:53.40 | IsUp | oh, SonoVoip is allowing that but i am looking for another provider |
18:53.50 | *** join/#asterisk verywiseman (~khaled@unaffiliated/verywiseman) |
18:53.53 | IsUp | also Teliax, but its closed i think |
18:54.24 | IsUp | can anybody send me a call with a random CLI? |
18:54.36 | IsUp | i want to see if it works or not |
18:55.22 | lordvadr | IsUp: you could do that in your dialplan yourself |
18:55.29 | Baylink-away | Oh. Yeah, sure; but whether it works inbound says nothing about whether you can *send* it. |
18:55.49 | IsUp | lordvadr, i dont have any voip provider |
18:56.17 | *** join/#asterisk emora (~emora@213.37.33.105.dyn.user.ono.com) |
18:56.21 | IsUp | i just need someone to call me with a random CLI |
18:56.24 | Baylink-away | We're confused, then. How did you want us to send you a call with random CNID, and what did you expect that to prove? |
18:56.33 | IsUp | for example, +0123456789, +05555333 |
18:56.36 | Baylink-away | CLI == Command Line Interface. |
18:56.42 | IsUp | i am looking for a provider |
18:56.51 | IsUp | oh, i mean CID, sorry :) |
18:57.01 | Baylink-away | We're confused, then. How did you want us to send you a call with random CNID, and what did you expect that to prove? |
18:57.09 | Baylink-away | ...if you have no provider. |
18:57.20 | lordvadr | IsUp: In your dial plan, set the CID to something and then send the call to whatever your incoming context is. |
18:57.33 | IsUp | I am looking for a provider Baylink-away, i want to see if anybody able to send me a call with a random Caller ID |
18:57.42 | IsUp | and if it works, i'll use that provider |
18:57.46 | IsUp | do you understand? |
18:58.01 | Baylink-work | Oh. Now I do, finally, yes. All my lines are PRI; sorry, I can't help you. |
18:58.28 | lordvadr | I can set arbitrary CID on my pri's, but I don't understand what you're trying to accomplish |
18:58.47 | Baylink-work | And you can't necessarily assume that even on a specific carrier, that that will be the same from country to country. |
18:58.47 | IsUp | i am looking for a SIP or IAX provider |
18:58.50 | IsUp | and i want to test it |
18:58.57 | IsUp | because i am living in Turkey |
18:59.05 | Baylink-work | He wants someone with a SIP provider to attempt to call him and see if he sees the random CNID. |
18:59.06 | IsUp | and some providers are not supported |
18:59.10 | *** join/#asterisk Z_God (~julius@wlan234013.mobiel.utwente.nl) |
18:59.15 | IsUp | exactly Baylink-work |
18:59.17 | Baylink-work | He will then believe that carrier can do that for him, which may or may not be tru. |
18:59.24 | Baylink-work | true, even. |
18:59.34 | Baylink-work | I've been doing tech support for a living for 25 years. :-) |
19:00.06 | lordvadr | any provider can set any cid they want in the sip or iax2 header. They're going to set if off of the cid they get from wherever they get the call. Are you trying to route the call based on caller id? |
19:00.32 | IsUp | lordvadr, i know what you talking about |
19:00.33 | Baylink-work | lordvadr: He's looking for an *outbound* provider, that will let *him* provide CNID arbitrarily. |
19:00.35 | IsUp | just basicly |
19:00.40 | Baylink-work | As I noted to him, those are pretty rare. |
19:00.42 | IsUp | i am looking for a outbound provider, yes |
19:00.53 | *** join/#asterisk Jumpie (n3rdz@ip68-230-28-186.ph.ph.cox.net) |
19:00.54 | IsUp | i want to call outside with any CLI |
19:00.58 | IsUp | Caller ID i mean |
19:01.03 | Jumpie | why is it i manage to be disconnected from freenode every few hours |
19:01.06 | Jumpie | unstable shit |
19:01.19 | lordvadr | I use junction networks and voip jet. Both will let me set an arbitrary caller id. |
19:01.37 | lordvadr | Junction Networks is in California. Voip Jet is somewhere in canada. |
19:01.40 | Qwell | Jumpie: yell at Cox. |
19:01.57 | Baylink-work | The real reason, IsUp, that most SIP carriers don't let you do that, is that carriers who will generally want you to be a contract customer, so they have some control over you if you start spoofing for no justifiable reason... and most SIP accounts are pre-paid, not post-paid. |
19:02.20 | lordvadr | Voip Jet will give you $0.25 in free credit |
19:02.33 | lordvadr | just for signing up... or at least they used ot |
19:02.33 | lordvadr | to |
19:03.00 | *** join/#asterisk AAA (spencer@beer.tclug.org) |
19:03.08 | Gnarfy | IsUp: flowroute.com claims to be able to do that for USA/Canada and some other countries... its the provider mr. mitnick used with the hidden caller id "hack" |
19:03.43 | IsUp | most of providers are using "gsm gateways" for calling Turkey |
19:03.52 | IsUp | so most of them not working |
19:04.10 | IsUp | they are using gsm gateways for cheap termination |
19:04.43 | IsUp | so anybody can drop me a test call? |
19:04.48 | IsUp | with a random caller id |
19:05.10 | IsUp | then i'll see if it works or not |
19:05.54 | Baylink-work | then you'll see "whether incoming calls *to* turkey by a provider will carry random CNID". FTFY. You won't see whether that carrier can deliver arbitrary CNID intra- or out of Turkey. |
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19:07.41 | lordvadr | Which brings me back to my point that yes, a udp packet from anywhere to turkey can contain any sip header. |
19:08.21 | lordvadr | Unless he's trying to get a call to some sort of landline or mobile phone, at which point I don't see where the provider comes in. |
19:10.58 | IsUp | argh |
19:11.10 | IsUp | nevermind |
19:11.12 | IsUp | thank you folks |
19:11.22 | wcselby | tzafrir_laptop, Qwell - I was able to get remote access to that box with the dahdi compile issues. removing the explicit '#include <linux/sched.h>' from drivers/dahdi/dahdi-base.c' doesn't make any difference in the make output that I can tell. |
19:13.23 | wcselby | updated pastebin: http://pastebin.com/mhGeJXhQ |
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19:43.20 | Jumpie | if i'm wanting to run asterisk on a VM, are there some recommended resources i should dedicate per vm? lets say, 5 concurrent call volume estimates |
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19:43.32 | Jumpie | 4gb ram and 200gb hd be ok? |
19:43.40 | Jumpie | cpu on main server is 8 quad cores |
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19:46.35 | pabelanger | Jumpie: Should be good, give it a try |
19:46.37 | Gnarfy | Performance required for only sip isn't much, biggest issue was the virtualization of the network causing delays for me... i ran asterisk on openbsd as guest in vmware esxi, but it worked fine |
19:47.02 | pabelanger | Jumpie: it also depends if you are transcoding audio for example |
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20:19.53 | jblack | Ohhh, there's a gpl c compiler for the parallax propeller |
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20:33.18 | seanjohn | anyone use freepbx for the basic dial plan? |
20:34.31 | pabelanger | what replaced AgentCallBackLogin in 1.6? |
20:35.17 | pabelanger | nm, found documents |
20:37.30 | wcselby | tzafrir_laptop, Qwell - I was able to get remote access to that box with the dahdi compile issues. removing the explicit '#include <linux/sched.h>' from drivers/dahdi/dahdi-base.c' doesn't make any difference in the make output that I can tell. |
20:40.33 | [TK]D-Fender | seanjohn: As opposed to? And there is NOTHING "basic" about FreePBX's dialplan |
20:47.38 | Baylink-afk|nigh | What he said. :-) |
20:48.00 | Baylink-afk|nigh | I've trawled through FPs dialplanning. You could invade *Normandy* with a plan that big. |
20:48.44 | wcselby | anyone know of a polycom wireless conference phone, preferably voip? |
20:48.50 | wcselby | i know they make an analog version |
20:49.08 | Sweeper | wireless conference phone? |
20:49.16 | Sweeper | I don't see much point in that.... |
20:49.30 | roe | Anyone know what happened to 'snap a number'? |
20:49.54 | Sweeper | generally conference phones sit in one place, like the middle of a table, and can be used for hours on end. not much point in being wireless |
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20:50.45 | Baylink-afk|nigh | Sure there is: running cables to a 500 pound marble table in the middle of a room's a pain. |
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20:51.14 | Baylink-afk|nigh | The Soundstation 2W is the analog one, wcselby. We like ours a lot. I don't think they ever quite made a wireless SIP one. I hook mine to an ATA. |
20:51.32 | wcselby | Baylink-afk|nigh - yeah I just found the 2w. |
20:51.37 | Baylink-afk|nigh | Most professional wireless mics only go 20-30 ft... |
20:51.40 | wcselby | i've got one client that uses it as well, they hook to an ATA |
20:51.48 | Baylink-afk|nigh | Factory refurb price $250 |
20:52.14 | Baylink-afk|nigh | Aw, c'mon; I worked hard on that Normandy joke... |
20:53.00 | Sweeper | so what happens when the battery dies? |
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20:55.01 | wcselby | Sweeper - I think these have like 24 hours of talk time capabilities |
20:55.28 | wcselby | plus I think they have ac ports...can't remember off the top of my head |
20:57.48 | wcselby | i've gotta reboot, brb |
20:57.51 | Baylink-afk|nigh | They run long, but probably not 24 hrs of talk. They have a charge port on the side, they'll run off it. |
20:58.10 | Baylink-afk|nigh | I'm pretty sure we've gotten 4 hour talk off a fully charged unit. |
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21:07.56 | seanjohn | http://pastebin.ca/1873105 why can't I make menuselect? |
21:08.44 | alexx1523 | Hi everyone: I'd like to change my cdr output from csv to json. I can't seem to find any information on the web as to the best way to do this... any hints? |
21:08.57 | seanjohn | what is json? |
21:09.39 | alexx1523 | javascript object notation... |
21:11.08 | niekvlessert | we've been running our asterisk on xen for a few months now... 40 people business |
21:11.11 | niekvlessert | no problems at all |
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21:16.20 | seanjohn | http://pastebin.ca/1873105 why can't I make menuselect? |
21:16.50 | Sweeper | wcselby: well if they have AC ports...not wireless anymore :P |
21:17.07 | Qwell | seanjohn: make nmenuselect |
21:17.16 | Qwell | you have gtk-devel stuff installed, so it's trying to use that. |
21:17.22 | Qwell | russellb: thoughts about removing that? :p |
21:17.41 | Qwell | actually, looks like png-devel just isn't installed, so it's causing it to fail |
21:17.49 | russellb | shouldn't try to use that unless you run gmenuselect directly |
21:18.10 | Qwell | it compiles whatever it can |
21:18.18 | russellb | ah, i see |
21:18.25 | russellb | package bug :-p |
21:18.28 | Qwell | maybe it shouldn't |
21:18.39 | russellb | yeah, just compile whatever it's going to run *shrugs* |
21:18.48 | russellb | in any case, redhat package bug :-p |
21:18.59 | Qwell | nah, there's no reason cairo-devel needs libpng-devel |
21:19.04 | Qwell | maybe. |
21:19.17 | russellb | it's not even the devel stuff, libcairo depends on libpng apparently |
21:19.21 | russellb | or something like that |
21:22.14 | alexx1523 | I could heavily modify custom_cdr.conf to resemble json, and then write a function to escape the json... |
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21:32.07 | mrfaison | hi guys!, how can i connect an aastra matra i760 phone on asterisk? |
21:32.34 | Qwell | mrfaison: with an Ethernet cable |
21:33.38 | mrfaison | Qwell: do you know how to configure them? |
21:33.56 | Qwell | nope. shouldnt be hard to find the manual |
21:33.56 | wcselby | hey Qwell, i was able to get remote access to that system. running make with the "include" removed didn't make a difference |
21:34.28 | wcselby | Qwell - updated pastebin - http://pastebin.com/mhGeJXhQ |
21:34.44 | Qwell | wcselby: I saw. I'm not a kernel guy. Just threw out a suggestion based on recent commits |
21:34.54 | wcselby | Qwell - gotcha. thanks :) |
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21:37.10 | wcselby | Sweeper - the ac port is to charge when the battery gets low...i.e between conference calls. like i said, has a pretty good battery in it for long talk times. |
21:37.29 | wcselby | Sweeper - and that's if it even has one, I don't really recall off the top of my head. |
21:39.46 | Sweeper | yea, I guess :) |
21:40.08 | wcselby | http://www.dailymotion.com/video/x2767r_biz-markie-just-a-friend_music |
21:40.10 | Sweeper | just not something I've ever seen a user for |
21:40.15 | wcselby | bah wrong window, sorry |
21:43.20 | mrfaison | Qwell: aastra matra 760i is not a regular sip phone |
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22:03.41 | alexx1523 | Hmm, is there an asterisk command for grabbing all current cdr values? As opposed to reading one at a time? |
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22:14.56 | KavanS | thoughts on asterisk 1.4.31 vs. any of the other 1.4's? |
22:15.22 | [TK]D-Fender | KavanS: The number is bigger... |
22:16.17 | KavanS | lol |
22:16.32 | KavanS | k .. running 1.4.28 and have this nasty debug msg showing up from console |
22:16.54 | KavanS | <PROTECTED> |
22:17.09 | KavanS | assuming 1.4.31 would fix this - I need to read the changelog a bit closer I think |
22:17.16 | KavanS | I cannot seem to locate the official bug |
22:17.28 | russellb | i don't remember if it has been fixed, but it's a harmless race condition |
22:17.37 | [TK]D-Fender | KavanS: How much does this situation affect you? |
22:18.18 | KavanS | heh it doesn't effect it at all - just makes console messy |
22:18.25 | KavanS | sometimes it is nice to "monitor" the console for calls taking place |
22:18.35 | KavanS | and with all this "junk" it is hard(er) to read |
22:18.48 | KavanS | so in the big picture "not much" but on niche scenarios - it is causing headaches |
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22:34.27 | QN | hi all. just to check.. is the iax2 packaged with asterisk? |
22:34.50 | russellb | yes. |
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22:36.58 | QN | hmm.. if this is the case, does the iax2 that comes with 1.6.2.6 has any known problems? |
22:37.16 | [TK]D-Fender | Packaged.. lol |
22:37.47 | [TK]D-Fender | QN: Well 1.6.2.7. is out.. don't know why you'd be aiming short |
22:52.54 | QN | [TK]D-Fender: this is because the distro i'm using now is not plain asterisk.. that'll likely cause things like freepbx to break i assume.. so if i can identify the actually problem for iax2 (if there's any), i can probably patch that portion? |
22:53.03 | *** join/#asterisk cusco (~trilili@213.63.137.210) |
22:53.05 | cusco | hello |
22:53.21 | cusco | how do I configure a dummy dahdi interface so I can use meetme? |
22:53.23 | cusco | dahdi is installed |
22:53.26 | cusco | but no dummy |
22:53.49 | [TK]D-Fender | QN: No, a fractional upgrade in the same branch doesnt' change anything like that |
22:54.35 | [TK]D-Fender | QN: You've also failed to give any information on the problem you seem to claim you have |
22:55.01 | [TK]D-Fender | cusco: Did you completely recompile & install * AFTERWARDS? |
22:55.18 | cusco | asterisk? |
22:55.47 | [TK]D-Fender | yes |
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22:55.57 | cusco | ah, I needed to modprobe dahdi_dummy |
22:58.58 | QN | [TK]D-Fender: sorry for that. the issue i encountered was between iax2 & iaxmodem. it seems like the iaxmodem are having difficulties in auth and stay connected with the iax2 at times which results iaxmodem is not connected with the iax2 account which was created for it.. |
22:59.10 | cusco | ok bye |
22:59.13 | cusco | thanks |
23:01.10 | [TK]D-Fender | QN: Where is the IAXmodem running on? |
23:01.52 | *** join/#asterisk talntid (swarm@c-67-185-219-139.hsd1.wa.comcast.net) |
23:02.05 | QN | its running within the same box.. |
23:02.25 | [TK]D-Fender | QN: pastebin actual CLI output of the errors |
23:02.28 | [TK]D-Fender | ~pb |
23:02.29 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
23:02.39 | talntid | I have a polycom that is connected to a remote asterisk server. When it makes calls, I can hear the person I am calling 100% fine, but they get choppy sound from me. Ideas? |
23:03.36 | wcselby | QN - is it becoming UNREACHABLE for a few seconds and then coming back? |
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23:06.58 | QN | wcselby: if i'm not wrong, yes it is. |
23:10.16 | [TK]D-Fender | talntid: You have an upstream bandwidth issue |
23:10.46 | wcselby | QN - I've seen that before, it always comes back quickly enough to have not been an issue for me. Is it not coming back for you? |
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23:13.12 | talntid | [TK]D-Fender Hmm :( |
23:13.16 | talntid | Ok. |
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23:30.54 | thansen | how do I get asterisk to answer a call as a sip client? I keep getting extension not found notices |
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23:34.26 | TedNJ40 | Hi guys. Can someone help me please? Somehow I have managed to get rid of the WEB Interface and I need to re-install it again (tbm-guicore) with all of its dependencies. Does anyone know how I can do that? |
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23:58.34 | ChannelZ | Getting rid of the web interface seems desireable. |