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| 00:06.14 | ChannelZ | Is there any other service that uses chan_motif now |
| 00:06.15 | ChannelZ | ? |
| 00:20.06 | file | I don't understand the question |
| 00:26.54 | *** join/#asterisk jpsharp (~jsharp@c-76-101-138-247.hsd1.fl.comcast.net) |
| 00:27.49 | jpsharp | I'm having an issue with a combination of faxing and A2billing. I don't know if it is an A2B issue or Asterisk or a problem with the nut behind the keyboard. |
| 00:29.29 | [TK]D-Fender | What's the issue? |
| 00:29.37 | jpsharp | I can terminate a ulaw-based fax directly onto the A2B server (skipping A2b, just going through extensions) with 100% reliability. |
| 00:30.14 | jpsharp | But if I run the call through a2b and direct it to another machine via an A2B destination, I can't get the call to complete at all. |
| 00:30.18 | [TK]D-Fender | So now there are 2 servers? |
| 00:30.21 | jpsharp | yes |
| 00:30.35 | [TK]D-Fender | this is not an A2B billing issue |
| 00:30.50 | [TK]D-Fender | becasue that is just a dumb AGI & billing system |
| 00:30.54 | [TK]D-Fender | it doesn't make calls fail |
| 00:30.56 | [TK]D-Fender | your path does |
| 00:30.56 | jpsharp | I didn't think so. |
| 00:31.46 | [TK]D-Fender | "I can't get the call to complete at all." <- go look at that call. |
| 00:31.47 | jpsharp | I'm not sure where my path is falling apart, though. |
| 00:32.11 | jpsharp | I should have been more clear...I can't get the fax to connect and receive. |
| 00:32.32 | jpsharp | The call connects, ReceiveFAX answers the line, but the modem portion never gets itself together. |
| 00:43.25 | mjordan | file: I'm guessing a 'google-voice replacement that uses XMPP + Jingle for signaling' |
| 00:44.03 | file | I wish. |
| 00:44.12 | jpsharp | So I'm not sure where my audio path is falling apart. |
| 01:02.04 | ChannelZ | file: chan_motif was primarily for Google but they threw the switch on that. I just wondered who/what else is using the protocol for voice. Custom applications? |
| 01:02.42 | file | It's possible, I don't know of any. |
| 01:03.53 | jpsharp | I'm still making and receiving calls on * via GoogleVoice. |
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| 05:37.54 | qakhan | what is wrong with this CUT exten => _X.,n,Set(foo=${CUT(dial,,1-4)}) |
| 05:38.56 | qakhan | http://pastebin.com/5zsT17vH |
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| 07:00.40 | [TK]D-Fender | [01:37]qakhanwhat is wrong with this CUT exten => _X.,n,Set(foo=${CUT(dial,,1-4)}) <- you diodn't specify a delimiter |
| 07:01.08 | [TK]D-Fender | qakhan: And based on what I see you trying to use it on you aren't even using the right tool for the job. |
| 07:01.25 | [TK]D-Fender | qakhan: Before asking what's wrong you should be clear on what you are trying to DO. |
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| 08:17.24 | r00f | at each reload of sip, asterisk says that nat=yes is deprecated |
| 08:17.38 | r00f | while grepping for nat=yes in sip.conf gives nothing |
| 08:17.46 | r00f | where does asterisk get that from? |
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| 08:21.05 | afournier | r00f: maybe your files contains "nat = yes" instead of "nat=yes" |
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| 08:24.20 | Faustov | or it loads a sample file due to file permissions... |
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| 08:26.23 | meital | hi all, I'm running asterisk with portbind=5060, however it seems that it doesn't bind to it (according to netstat) |
| 08:26.50 | meital | the only thing that is connected is: unix 3 [ ] STREAM CONNECTED 218296 /var/run/asterisk/asterisk.ctl |
| 08:27.10 | meital | I tried googeling it, but I didn't find anything relevant |
| 08:27.52 | Chainsaw | meital: You're looking for UDP sockets? |
| 08:28.50 | Chainsaw | meital: I see UDP & TCP, but then I've configured my system that way. Two results for netstat -a -n | grep 5060 | grep 0.0.0.0 |
| 08:32.06 | meital | Chainsaw: I get an empty output for netstat -a -n | grep 5060 | grep 0.0.0.0 |
| 08:33.02 | r00f | afournier grepped for "nat =", still nothing. the file loaded is correct, for it loads all peers from it |
| 08:33.11 | Chainsaw | meital: That's suspicious. |
| 08:33.32 | meital | Chainsaw: I know, that's why I'm here ;) |
| 08:33.34 | Chainsaw | meital: I'd look into what interface you're binding to and what your firewall setup is. |
| 08:33.51 | Chainsaw | meital: Well, Asterisk is very good at binding to whatever port & interface you tell it to, unless you put roadblocks in its way. |
| 08:34.04 | Chainsaw | would suggest that meital clear the roadblocks |
| 08:35.14 | meital | Chainsaw: firewall is "allow all". where can I check the interface? bindaddr=0.0.0.0 ? what is roadblocks? |
| 08:35.42 | Chainsaw | meital: Well, there could be all sorts of roadblocks. You could be running Red Hat Expensive Linux with some sort of SELinux enforcement. |
| 08:36.29 | meital | <PROTECTED> |
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| 08:37.59 | meital | Chainsaw: the only weird thing that I see is that it has some warnings about loading some modules, but I don't think that it should affect the port binding... |
| 08:38.13 | Chainsaw | meital: It could, without a SIP channel driver not much is going to bind to that 5060 port. |
| 08:38.20 | udp` | hi |
| 08:38.29 | udp` | what is the difference between a channel and a route ? |
| 08:38.32 | Chainsaw | meital: Can you stick those error messages up on a pastebin of your choosing please? |
| 08:38.52 | Chainsaw | meital: It'll probably include the version number by default, but just in case... manual compile or distro-supplied? |
| 08:39.39 | meital | Chainsaw: http://pastebin.com/tjZffWJX |
| 08:40.02 | Chainsaw | [Jun 29 19:18:45] WARNING[7149]: loader.c:423 load_dynamic_module: Error loading module 'chan_sip.so': /usr/lib/asterisk/modules/chan_sip.so: undefined symbol: ast_websocket_write |
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| 08:40.53 | meital | Chainsaw: is this the problem? because it's only a "WARNING" |
| 08:40.54 | Chainsaw | meital: My previous question is becoming increasingly relevant... where did you source this Asterisk from? A native distro-supplied package? Some custom repository? Did you compile it by hand? Or some unholy mixture of more then one of these? |
| 08:41.17 | Chainsaw | meital: Your SIP channel driver is not loading. This is deathly serious, because from the sound of it, you're wanting to use SIP. That's what lives on port 5060. |
| 08:41.53 | Chainsaw | udp`: Going to need to see that in context please. |
| 08:43.04 | meital | Chainsaw: http://www.asterisk.org/sites/asterisk/files/mce_files/documents/asterisk_quick_start_guide.pdf |
| 08:43.48 | Chainsaw | meital: Okay, so your answer is: "some unholy mixture of more then one of these". |
| 08:44.17 | Chainsaw | meital: Did you install Asterisk through apt-get first? |
| 08:44.44 | meital | Chainsaw: lol, ok, so I can create a new vm and install asterisk again, just tell me please how to install it ;) |
| 08:50.50 | Chainsaw | meital: Do a Gentoo VM and emerge asterisk? |
| 08:52.24 | meital | Chainsaw: really? but I hate gentoo! I haven't used it in ages! |
| 08:55.03 | Chainsaw | meital: It's an environment where I can guarantee the distro packaging makes sense. |
| 08:55.35 | meital | Chainsaw: ok, will try |
| 08:55.43 | Chainsaw | meital: In other environments, I cannot. And since you seem to be struggling in this area... I thought I would short-circuit things and get you a quick fix. |
| 08:56.45 | meital | Chainsaw: r u the maintainer or something? :) |
| 08:57.30 | Chainsaw | meital: In Gentoo? Yes. |
| 08:58.07 | meital | Chainsaw: cool :) ok, I'll download and install gentoo, will update later. thanks!!!! |
| 08:58.19 | udp` | meital: good luck :) |
| 08:58.31 | meital | udp`: thanks! |
| 08:58.44 | udp` | meital: ;) |
| 08:59.00 | udp` | Chainsaw: bro, i didnt get you before about the context |
| 08:59.35 | Chainsaw | udp`: There's a sentence out there that is using channel & route. Maybe even a paragraph. |
| 08:59.59 | Chainsaw | udp`: I'd like to see the whole thing, rather then give you all possible meanings of the words. |
| 09:00.05 | udp` | mmmm ok |
| 09:00.19 | Chainsaw | udp`: Not to mention after all that work be told that in your specific sentence/paragraph, none of them make sense. |
| 09:00.40 | udp` | im using PowerPBX v11 |
| 09:00.55 | udp` | the config files are a little bit different than the usual |
| 09:00.58 | Chainsaw | Right. And that has some documentation that uses those words. |
| 09:01.21 | udp` | in the sip.conf |
| 09:01.29 | udp` | there are only headers |
| 09:01.55 | udp` | defined, and to add stuff you need to edit those headers instead of the sip.conf file directly |
| 09:02.24 | udp` | anyway, to use a sip trunk channel |
| 09:02.31 | udp` | i have to define first a route |
| 09:04.19 | Chainsaw | udp`: So you're using the PowerPBX documentation to install a FreePBX, right? |
| 09:04.39 | udp` | what is the purpose of the route, other than handle the dialed number to the correct channel ? |
| 09:06.23 | Chainsaw | You have not answered my question. |
| 09:08.51 | udp` | Chainsaw: im using just powerpbx |
| 09:09.19 | udp` | Chainsaw: or i didnt understand you question |
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| 09:37.10 | udp` | Chainsaw: ? |
| 09:37.34 | Chainsaw | udp`: If you are unable to answer that question I am unable to assist you. |
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| 09:38.35 | udp` | Chainsaw: mmm ok |
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| 09:45.58 | udp` | Chainsaw: another question, i have a sip trunk provider, he gave me an account, under that account i have 10 virtual numbers |
| 09:46.23 | udp` | how do i define them in the system ? |
| 09:47.20 | Chainsaw | udp`: Again, if you will not answer my questions I cannot answer yours. |
| 09:47.39 | udp` | i didnt understood your question amigo |
| 09:48.10 | Chainsaw | udp`: When I look at this "PowerPBX" website you mention, there are guides. Guides to install FreePBX, guides to install FreeSwitch... |
| 09:49.15 | udp` | auhhh ok |
| 09:49.22 | udp` | yes im using it with freepbx |
| 09:55.46 | Chainsaw | Then you need to ask your questions in #freepbx |
| 09:55.51 | Chainsaw | Not #asterisk |
| 09:56.07 | Chainsaw | (This explains why you're asking about terms that are not used in Asterisk) |
| 10:02.35 | udp` | auhhh ok ! :) |
| 10:02.47 | udp` | thanx :) |
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| 10:11.00 | r00f | today i was trying to grab some variables from ${CHANNEL()} and was getting lots of errors. it's so stupid that all those variables were renamed ages ago, but documentation still has old names, even xmldoc in func_chan.c |
| 10:11.42 | r00f | are we all supposed to search inside the source code to get the real var names? |
| 10:12.55 | Chainsaw | r00f: Are you referring to https://issues.asterisk.org/jira/browse/ASTERISK-17185 or is it a different problem? |
| 10:14.43 | r00f | Chainsaw: looks like. my 11.10 still has old names in xmldoc, i was confused |
| 10:15.22 | Chainsaw | r00f: The patch still applies, and I've been carrying it in my downstream patchset for many years. |
| 10:15.41 | Chainsaw | r00f: Perhaps you can help show the importance of it to Digium by adding yourself to the watch list and commenting. |
| 10:16.02 | r00f | okay, i will, thanks for pointing |
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| 10:24.54 | r00f | lol, they call it minor issue. |
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| 10:35.34 | hanuman_ | #django |
| 10:35.54 | eirirs | unchained |
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| 10:43.55 | r00f | eirirs ))) |
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| 12:37.27 | meital | Chainsaw: did I mention how much I hate gentoo? No network for me... =\ |
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| 12:58.07 | Chainsaw | meital: Did you emerge dhcpcd? |
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| 13:10.07 | r00f | i've found the culprit. a2billing was generating config in old format |
| 13:19.09 | meital | Chainsaw: no I did, and it worked! you are my hero! ;) |
| 13:22.36 | meital | *now |
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| 13:41.22 | meital | Chainsaw: even asterisk works now! thanks you so much!!! |
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| 13:41.37 | Chainsaw | meital: Any time :) |
| 13:44.24 | estranger | Filling out a SIP pre-sale questionnaire for asterisk from a provider, got one last question: "Sends INVITESs with SIP URLs (no tel URLs)"? that's verbatim .. I assume that is "Yes"? |
| 13:46.57 | file | yes. |
| 13:47.12 | estranger | thanks :) |
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| 13:51.36 | wdoekes | multiple INVITESses? |
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| 13:55.44 | estranger | I dunno, there are a few things on the form that are quite ambiguous |
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| 14:30.05 | Stefan27 | Hello, im using https://wiki.asterisk.org/wiki to make a new installation of asterisk 12.3. I noticed there are around 300 modules, many of which may not be needed, in which case one should add noload => res_XXX.so in modules.conf ? ... is there a clever way of finding out which modules i really need other than going through each one and reading about what it does? which modules are the most |
| 14:30.06 | Stefan27 | important? how many modules are needed for a minimal asterisk-setup with chan_sip? |
| 14:31.20 | WIMPy | No. Either you enable all, you read through all of them and hope you understand what they do, or you disable aoutoload and add them as you miss functionality. |
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| 14:35.52 | Stefan27 | ok thanks i guess i can just enable all while testing: adding or removing module X does not change the behavior of module Y? |
| 14:36.27 | WIMPy | It can. |
| 14:36.49 | WIMPy | Some modules depend on others. |
| 14:37.48 | Stefan27 | ah, those with common prefix, like res_ari.so and res_ari_sounds.so? |
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| 14:41.50 | Stefan27 | ill just start experimenting to get a grip |
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| 15:37.48 | mjordan | Stefan27: res_ari is the basis of most of the ARI related modules. res_pjsip is the basis of most PJSIP related modules. Disabling both of those will nuke out most of the ARI/PJSIP features. |
| 15:37.54 | mjordan | (I say most, it really is "all") |
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| 16:08.52 | file | grooves |
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| 17:52.31 | c0rnoTa | Hello everyone! |
| 17:52.33 | c0rnoTa | I have Asterisk 1.8.23.0 installed from the source, it is working installation with mysql realtime connectivity. I'm trying to use REALTIME_STORE function, but cannot. Asterisk always return that "ERROR: ast_func_read: Function REALTIME_STORE cannot be read" I'm already found example of usage in source of patch for this function. But it doesn't work too with the same error. Family is configured, table exists. When I change REALTIME_STORE to REALTIME funct |
| 17:52.33 | c0rnoTa | Thanks for Attantion. |
| 17:53.58 | c0rnoTa | Or should I create issue in JIRA? |
| 17:58.31 | newtonr | c0rnoTa, what exact backend are you using? Looking at the documentation of REALTIME_STORE, it sounds like that function is only for *setting* new values in a field |
| 17:58.51 | newtonr | c0rnoTa, did you try reading REALTIME_FIELD ? |
| 17:59.18 | newtonr | c0rnoTa, and also yeah, REALTIME |
| 18:02.12 | newtonr | c0rnoTa, take a look at the "core show function <function name>" help text for each.. |
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| 18:04.07 | c0rnoTa | newtonr, REALTIME works fine. it update and read values from the table. I have not try realtime field but as I see i's like from help text, it's only updating value. I think, it should work, but I can try. newtonr, but it does not solve my task. I need to insert row, not update value. |
| 18:04.20 | c0rnoTa | the backend is mysql |
| 18:04.56 | c0rnoTa | newtonr, res_config_mysql.so |
| 18:06.10 | c0rnoTa | newtonr, don't you think, that this backend could not insert rows? |
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| 18:08.30 | c0rnoTa | newtonr, I wrote an AGI script until I can find solution. Because function is so simple and frequently used that agi is too heavy for this. |
| 18:09.45 | newtonr | c0rnoTa, I don't think there is an issue with the backend. I think you should be able to do what you want with REALTIME_STORE |
| 18:10.23 | newtonr | c0rnoTa, I'm just going by the documentation. I haven't used any of these. :) |
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| 18:19.20 | c0rnoTa | newtonr, yes, I think so too. I read documentation about this function and found source code (where found example of usage ) So, may be it's better tol create an issue in jira. I thought, that this function is not widely used, that's because bug in function is possible.... |
| 18:20.49 | newtonr | c0rnoTa, if you file a bug, make sure to include dialplan showing how you call the function, and an Asterisk debug log showing the action. |
| 18:21.10 | *** join/#asterisk bullium (~will@216.68.250.27) |
| 18:21.32 | bullium | Anyone have a quick way to export a list of all extensions on the system? |
| 18:21.56 | WIMPy | dialplan show |
| 18:22.02 | c0rnoTa | newtonr, sure! Thank you for your suggestions. |
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| 18:25.55 | newtonr | c0rnoTa, np |
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| 18:31.43 | bullium | WIMPy, that's not the kind of output I'm looking for. Basically I'd like to have a list of extensions to print out and use it as a check list |
| 18:34.02 | bullium | We only have SIP peers, so asterisk -rx "sip show peers" almost gives me what I want. I want extension and name |
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| 18:50.44 | WIMPy | Maybe you should define what _you_ mean by "extension". |
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| 19:05.37 | [TK]D-Fender | bullium: parse "sip show peers" and "sip show peer X" for each. And stope calling SIP device definitions "extensions" |
| 19:05.40 | [TK]D-Fender | stope* |
| 19:05.43 | [TK]D-Fender | GAH |
| 19:05.48 | [TK]D-Fender | can't type.... |
| 19:06.56 | mbowie | Typing is overrated on a medium like IRC anyhoo. |
| 19:10.56 | carrar | yeah |
| 19:11.00 | carrar | IRC over SIP |
| 19:11.03 | carrar | FTW |
| 19:16.53 | mbowie | #partyline |
| 19:16.59 | drmessano | IRC <> SIMPLE <> IRC |
| 19:17.02 | drmessano | Make it happen |
| 19:17.09 | drmessano | Why arent we funding this? |
| 19:17.11 | drmessano | YOLO |
| 19:17.51 | drmessano | Asterisk 14 should have an integrated IRC server |
| 19:17.58 | Nivex | SIP <> XMPP <> IRC :) |
| 19:18.01 | drmessano | Why? Because Microsoft had it in Exchange once |
| 19:18.05 | drmessano | and we should be better |
| 19:19.28 | drmessano | The only problem with having IRC integrated into Exchange was that it was FREAKING IRC RUNNING ON AN EXCHANGE SERVER AND CODED BY MS |
| 19:19.29 | bullium | This does it for me awk '/username|fullname/' /etc/asterisk/users.conf | awk '{ if ((NR % 2) == 0) printf("\n"); print; }' |
| 19:19.31 | drmessano | By other than that |
| 19:19.44 | mbowie | I'm not sure that's a yard-stick everyone can all agree on. |
| 19:19.46 | drmessano | But other than that* |
| 19:20.28 | drmessano | I miss the MS CHAT client |
| 19:20.47 | drmessano | Nothing like telling someone to GFY in comic form like an A-HA video |
| 19:21.45 | drmessano | "Johnson, did you just change your MS Chat character to Garfield and tell me to *** last weeks report *** my ****?" |
| 19:21.50 | drmessano | "Yes, sir" |
| 19:21.55 | drmessano | "Oh, well that was much LULZ" |
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| 19:54.28 | anonymouz666 | nice news from atlassian... conquering the world |
| 20:10.38 | TazzNZ | what news that be ? |
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| 20:25.30 | mjordan | drmessano: actually, that would be kind of funny. And doable. |
| 20:25.36 | mjordan | drmessano: I wouldn't write an IRC server |
| 20:25.45 | mjordan | just an IRC client. And tie into the messaging in Asterisk. |
| 20:25.50 | mjordan | So you could SIP MESSAGE requests to IRC and vice versa |
| 20:25.59 | drmessano | That would be awesome |
| 20:26.10 | drmessano | Makes more sense |
| 20:26.13 | drmessano | But yeah |
| 20:26.30 | file | I'd use it for signaling session establishment |
| 20:26.58 | mjordan | file: FUNNY |
| 20:27.13 | drmessano | I'd love to send call notifications to users via IRC |
| 20:27.29 | drmessano | U HAV A CALL FM UR MOM LULZ |
| 20:27.34 | mjordan | heh |
| 20:27.49 | drmessano | U R MOM LEFT ME A VOICEMAIL |
| 20:27.52 | drmessano | ETC |
| 20:28.19 | file | mjordan, hey man - it would work! |
| 20:28.24 | mjordan | every time you go on the phone, it sends a message that "hey, don't bother talking to me" |
| 20:29.16 | drmessano | An Asterisk IRC service would be neat. Set AWAY for users.. let them for messages via a bot |
| 20:29.25 | drmessano | well not a bot, if its an IRC service |
| 20:29.41 | file | mjordan, omg |
| 20:29.47 | file | mjordan, DCC as the data channel |
| 20:29.52 | drmessano | lol |
| 20:30.21 | drmessano | Wait, DCC as the data channel.. so call establishment would now involve genital pics? |
| 20:30.31 | file | drmessano, it could! |
| 20:30.38 | drmessano | Because thats what some.. err... I hear some people used DCC for.... ummm |
| 20:31.35 | mjordan | not that I do. I just ... hear. Things. |
| 20:31.37 | mjordan | OTOH |
| 20:31.50 | mjordan | jpeg of certain size is "Ringing"? |
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| 20:31.57 | drmessano | lol |
| 20:32.14 | drmessano | To hangup a channel you send a goatse |
| 20:32.30 | anonymouz666 | freenode is exception of the rule, IRC was killed by MSN, then Orkut, then Facebook, now Whatsapp. |
| 20:33.37 | drmessano | IRC as a protocol is alive and well. There were new extensions added almost 12 years ago. Thats newer than most RFCs we use with telephony! |
| 20:35.53 | anonymouz666 | ./BitchX |
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| 20:37.42 | TazzNZ | hey hey hey - we can then use IRC for distrubuted states |
| 20:38.41 | *** join/#asterisk sgriepentrog (~sgriepent@nat/digium/x-iyxeqmzmihxyovdg) |
| 20:38.47 | TazzNZ | no more XMPP |
| 20:38.54 | anonymouz666 | no more corosync |
| 20:39.02 | anonymouz666 | no more app_queue |
| 20:39.24 | drmessano | We could use IRC to replace AMI |
| 20:39.28 | anonymouz666 | Build your own queues for fun and profit |
| 20:40.19 | TazzNZ | that would be SO cool |
| 20:41.15 | drmessano | They did that years ago with Trixbox and HUD |
| 20:41.42 | drmessano | Each HUD client was an IRC client.. connected a channel on an IRCD running on the PBX |
| 20:41.58 | drmessano | They had some bot that resided in there that was used for control |
| 20:42.18 | drmessano | I made my own script to interact with it.. Could do the call notifications in channel, place calls, etc |
| 20:42.20 | TazzNZ | you could join the chaneel and see what your users are doing on phones :) |
| 20:42.27 | drmessano | Yep |
| 20:42.36 | WIMPy | Programming for non-programmers? |
| 20:43.25 | drmessano | The neat thing was they actually had params for the server address and port, which defaulted to localhost, so I made their script connect to an IRC I had running on another box |
| 20:43.29 | TazzNZ | mjordan: we all know you have tons of spare time - make it so ! :D hehe |
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| 20:43.48 | drmessano | That was kinda fun for a week |
| 20:44.00 | drmessano | Especially when I put triggers in channel |
| 20:44.03 | drmessano | !call 102 from 103 |
| 20:45.21 | TazzNZ | ROFL - that would even solve the monitoring "issue" - "server_1 ping timeout" <-- oh cr** - server is down !!!! RUN !!!!!!!!!!! |
| 20:45.55 | drmessano | "PBX has left IRC" |
| 20:45.58 | drmessano | Awww crap |
| 20:46.08 | drmessano | yeah |
| 20:46.12 | TazzNZ | LOL |
| 20:46.48 | drmessano | But you know querying the server would be fun |
| 20:46.57 | drmessano | Hey PBX, are you busy? |
| 20:47.08 | drmessano | I have 7 concurrent calls |
| 20:47.24 | drmessano | Is Joe on the phone again |
| 20:47.32 | TazzNZ | LOL |
| 20:47.32 | drmessano | Yep, that idiot is always on the phone |
| 20:48.29 | drmessano | Oh man |
| 20:49.39 | drmessano | MAHMI - Matrix Asterisk Human Manipulation Interface |
| 20:49.49 | drmessano | I even have the name |
| 20:50.04 | TazzNZ | LOL |
| 20:50.38 | drmessano | Nope, need more markety |
| 20:50.54 | drmessano | MAHMI - Multilingual Asterisk Human Manipulation Interface |
| 20:51.05 | drmessano | There, now it speaks N+1 languages |
| 20:51.11 | TazzNZ | AII - Asterisk IRC interface |
| 20:51.38 | rrittgarn | pronounced AYEEEEE ? |
| 20:51.47 | TazzNZ | LOL - yeah - like IAX |
| 20:53.08 | *** join/#asterisk D-Boy (~D-Boy@unaffiliated/cain) |
| 20:56.00 | drmessano | I want to be speak to Asterisk |
| 20:56.06 | drmessano | Need to work on that |
| 20:56.10 | drmessano | "Hangup on Joe" |
| 20:56.13 | drmessano | Done |
| 20:56.20 | drmessano | ugh |
| 20:56.29 | drmessano | I want to be ABLE TO speak to Asterisk |
| 20:58.24 | *** join/#asterisk TazzNZ (~TazzNZ@office-nat1.ytechops.co.nz) |
| 20:58.26 | _0x5eb_ | hi! speaking of XMPP... and gtalk, anyone's been experimenting chan_motif on Asterisk 12? |
| 20:59.49 | _0x5eb_ | (esp. with gtalk users, who unfortunately represents the vast majority of XMPP users) |
| 21:00.15 | WIMPy | Not for long. |
| 21:00.34 | _0x5eb_ | troubles? |
| 21:01.29 | WIMPy | That IRC discussion was evil. Now I have to think about how much sense it would really make to use IRC vor VOIP. I mean it could easily replace SIP. But would you even need a PBX, or could IRC do it? |
| 21:03.34 | _0x5eb_ | well as a matter of fact, IRC is one of the world's most used signalling protocol |
| 21:03.46 | _0x5eb_ | not for the good cause thought |
| 21:04.56 | *** join/#asterisk wonderworld (~ww@ip-62-143-157-238.unitymediagroup.de) |
| 21:05.11 | _0x5eb_ | and Skype could be abused to achieve the same goal btw |
| 21:05.39 | WIMPy | Maybe. Noone knows what it does. So that's disqualified. |
| 21:07.51 | _0x5eb_ | I love to point out EADS' excellent study on it, btw that (could) help promoting regular IPBX instead of Skype as a voice communication platform |
| 21:09.27 | WIMPy | Well, skypes success comes from the fact that it works without haveing a network engeneer next to you to help you get through NAT. |
| 21:09.48 | WIMPy | Off course Asterisk could do the same with IAX, but noone seems to care about that. |
| 21:10.07 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 21:11.02 | _0x5eb_ | too complicated for the average business user |
| 21:11.34 | WIMPy | So why do people think that SIP could do it? |
| 21:12.01 | _0x5eb_ | a softer alternative would be to make them start using Jingle without them knowing it (i.e. gtalk) |
| 21:14.02 | anonymouz666 | "just works" is the "secret" of skype |
| 21:14.06 | anonymouz666 | signalling |
| 21:14.10 | anonymouz666 | audio codec |
| 21:14.11 | anonymouz666 | etc etc. |
| 21:14.17 | _0x5eb_ | and then try to convince them to switch to a (well designed) SIP network with real IP phones |
| 21:15.05 | _0x5eb_ | ... and breaking through most of the firewalls |
| 21:15.21 | anonymouz666 | G729 is not even close to be a good codec for all kinds of network |
| 21:15.39 | WIMPy | Someone sould make IAX phones again. Preferrably better ones. |
| 21:16.22 | anonymouz666 | WIMPy: there are very good techniques you can use with SIP to make it more reliable through NAT |
| 21:16.43 | _0x5eb_ | why are we moving so slowly to IPv6? |
| 21:16.56 | TazzNZ | WIMPy: according to "digium" IAX is a "trunk only protocol" |
| 21:17.03 | WIMPy | Maybe, but chances of it just working are far to small. |
| 21:17.16 | TazzNZ | _0x5eb_: the world is moving too slow :( |
| 21:17.20 | *** join/#asterisk raspberrypifan (~textual@190.131.164.211) |
| 21:17.26 | Nivex | _0x5eb_: because the people who have made all their money off the Internet don't want to spend it to upgrade |
| 21:17.26 | Qwell | TazzNZ: those aren't the words I'd use to describe it |
| 21:17.34 | WIMPy | Well, they don't know what great technology they own. |
| 21:18.12 | WIMPy | Instead they seem to prefer to play buzzwordbingo. |
| 21:18.14 | *** join/#asterisk raspberrypifan (~textual@190.131.164.211) |
| 21:18.15 | WIMPy | :-( |
| 21:18.25 | TazzNZ | Qwell: I can't remember the exact words they used, but that was it in a nutshell :) |
| 21:18.40 | coppice | _0x5eb_: arrogance and incompetence by the IPv6 developers. They stopped, job finished, when there was still much to do.... like a solid migration scheme |
| 21:19.01 | TazzNZ | either way - I can't see the reason behind using IAX vs SIP |
| 21:19.05 | Nivex | I tried to turn on IPv6 for SIP in Asterisk 11 and my v4 SIP broke. I haven't investigated further. |
| 21:19.30 | Qwell | Nivex: Do you run Asterisk on BSD? |
| 21:19.34 | WIMPy | IAX just works. |
| 21:19.40 | Nivex | Qwell: no, Debian. |
| 21:19.40 | *** join/#asterisk raspberrypifan (~textual@190.131.164.211) |
| 21:19.43 | WIMPy | And it's a defines protocoll. |
| 21:19.48 | Qwell | Nivex: then a bindport of ::0 should "just work" |
| 21:20.00 | _0x5eb_ | coppice: this is esp. when I look at Japan: once leader in IPv6 experimentation (kame project/wide) and test deployment, now among the last to actually deploy it to the homes |
| 21:20.11 | Nivex | yes, it bound, but then things coming through the NAT got wonky. |
| 21:20.17 | Qwell | Nivex: err, sorry, bindaddr |
| 21:20.19 | WIMPy | Unlike SIP which is just some useless basics with everyting else being implemented in various incompatible ways. |
| 21:20.55 | Nivex | the only IPv6 SIP client I have to test with is cSIPsimple on Android, which is an abomination on v6 anyway, I just put it back to v4 only |
| 21:21.35 | _0x5eb_ | Nivex: why abomination? |
| 21:21.38 | TazzNZ | hhhmmm v6 - I wonder...... |
| 21:22.30 | Nivex | _0x5eb_: android itself has crap support on wifi and then csipsimple's chosen SIP stack also doesn't handle it |
| 21:22.30 | *** join/#asterisk raspberrypifan (~textual@190.131.164.211) |
| 21:22.34 | coppice | _0x5eb_: much of what is said about IPv6 deployment is rather bogus. Various information from IPv6 proponents says that we here in HK have one of the highest levels of IPv6. In reality there is no option for consumer IPv6, either wired or wireless. |
| 21:24.38 | _0x5eb_ | coppice: here in France, I have native IPv6 at my home on my residential DSL connection for... 10 years |
| 21:25.02 | *** join/#asterisk raspberrypifan (~textual@190.131.164.211) |
| 21:25.05 | _0x5eb_ | coppice: Asia was once leader for IPv6 research; problem is that broadband ISP were never really involved |
| 21:25.50 | anonymouz666 | _0x5eb_: congratulations for the victory today 2x0 |
| 21:26.29 | _0x5eb_ | anonymouz666: thanks ;) |
| 21:26.42 | coppice | _0x5e6_: even our LTE networks, which are all less than 3 years old, have no support for IPv6. |
| 21:27.07 | _0x5eb_ | another way to cope with NAT issues: openvpn tunnel, natively supported by some major brand of IP phones |
| 21:27.39 | WIMPy | Yes. That does work. |
| 21:27.48 | WIMPy | But that's just a work around. |
| 21:28.19 | [TK]D-Fender | adds "King Of Wishful Thinking" to his playlist... |
| 21:28.19 | _0x5eb_ | sure; IPv6 should be (I said should, not that it will be) the right solution |
| 21:29.01 | Nivex | https://www.youtube.com/watch?v=v26BAlfWBm8 |
| 21:29.07 | WIMPy | I'd rather see IPv7 or whatever. |
| 21:29.27 | coppice | _0x5e6_: problems with IPv4 keep showing up for years. I donm't think IPv6 has seen enough use to really flush out how well it works. |
| 21:29.50 | Nivex | coppice: so what are you doing to help find those problems? |
| 21:29.55 | _0x5eb_ | (Snom and Yealink natively support OpenVPN, maybe others?) |
| 21:30.18 | coppice | Nivex: until I have access to native IPv6, nothing |
| 21:30.39 | _0x5eb_ | Nivex: actually using it is the best way to deal with problems :) |
| 21:30.46 | Nivex | _0x5eb_: amen |
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| 21:33.25 | [TK]D-Fender | [17:29]_0x5eb_(Snom and Yealink natively support OpenVPN, maybe others?) <- no "majors" |
| 21:33.54 | coppice | Yealink is the global number 3 |
| 21:34.28 | [TK]D-Fender | Ae they? Guess the volume cheap market is ""something" |
| 21:34.47 | [TK]D-Fender | That was a fast track to #3 |
| 21:35.39 | anonymouz666 | Yealink are very good phones |
| 21:36.48 | WIMPy | I haven't seen one, yet, but when I looked close last week, looking for phones with (the option for) enough buttons and other must haves, I got the impression they might be more like top end. |
| 21:37.27 | coppice | I think Yealink have abandoned all their low end stuff |
| 21:37.41 | anonymouz666 | now people are using phones with android! |
| 21:38.00 | WIMPy | And touchscreens instead of keys. |
| 21:38.41 | Qwell | all the touchscreen deskphones I've used have annoyed the crap out of me. really insensitive/inaccurate |
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| 21:39.06 | coppice | Qwell: there have been lots of resistive touch screens, but they are going capacitive now |
| 21:39.25 | anonymouz666 | why developers thoughts are always incompatible with the customers? heheheh :-) |
| 21:39.41 | Qwell | coppice: that's a start |
| 21:40.00 | WIMPy | It's usually not developers. It's the evil Marketeers. |
| 21:40.10 | coppice | in a year or two I think all but the entry level IP phones will be android machines |
| 21:40.22 | WIMPy | They know better what the customer wants than the customer himselv. |
| 21:40.25 | _0x5eb_ | Polycom's good old hard buttons are nice :) |
| 21:41.10 | coppice | there's nothing like real buttons for people who do a lot of dialing |
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| 21:43.35 | anonymouz666 | even grandstream phones they got much better. From what I saw in 2005... now are totally different products |
| 21:44.14 | coppice | people are very positive about the new grandstream android phones |
| 21:45.00 | anonymouz666 | I just installed about 50 GS android phones in a customer last week |
| 21:45.32 | anonymouz666 | they tested the phone for 1 month and loved it |
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| 21:48.19 | coppice | was the the touch model, or the one with buttons? |
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| 22:21.52 | smd75jr_ | Hi, I finally managed to get my system half working. I can dial out but I cant receive any calls. All of my settings seem to look correct (as far as I can tell that is) but I just can't seem to get it to work. The log of an incoming call is here: http://pastebin.com/Mmk6zEq1 Can anyone help me? |
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| 23:24.08 | syadnom | hi all. anyone know of a decent CDR analysis software? google is no help |
| 23:27.24 | [TK]D-Fender | https://www.google.ca/#q=asterisk%20cdr%20analysis%20tool |
| 23:27.37 | [TK]D-Fender | I see SIX within the first page of my 2-second search alone... |
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