| 00:17.14 | *** join/#asterisk EPBX29 (18c1698d@gateway/web/freenode/ip.24.193.105.141) |
| 00:18.11 | EPBX29 | need help with an issue |
| 00:18.13 | EPBX29 | anyone here? |
| 00:18.52 | EPBX29 | having an issue adding a new extension |
| 00:20.00 | EPBX29 | hello room |
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| 00:20.36 | jameswf | EPBX29: post your issue and someone will respond at their convenience |
| 00:20.42 | EPBX29 | thanks |
| 00:23.57 | [TK]D-Fender | I highly recommend Chapter 6.... |
| 00:24.54 | EPBX29 | we just created a new extension for internal dialing. When you pick up the phone, you get a dial tone. But when you try to dial any internal/external numbers you immediately get a disconnect tone. Also, see attached. I use cisco 7960 phone and towards the top right corner, next to the 247 extension, the phone icon has an X next to it. It's not the phone because I assigned an existing working extension and the phone works. Is ther |
| 00:26.32 | [TK]D-Fender | "But when you try to dial any internal/external numbers you immediately get a disconnect tone." <- and what do you see in * CLI with "sip set debug on" when you make that call? |
| 00:26.36 | EPBX29 | http://i1092.photobucket.com/albums/i410/TechBeam/Error_zps9f793155.png |
| 00:26.41 | EPBX29 | that's what I get on the phone |
| 00:27.09 | EPBX29 | so the phone picks up the extension, so there is not issue with SIP |
| 00:27.33 | EPBX29 | no* |
| 00:28.23 | EPBX29 | I thought it was a phone issue, so I added a working extension to the original phone with the issue and the call worked |
| 00:28.24 | [TK]D-Fender | [20:27]EPBX29so the phone picks up the extension, so there is not issue with SIP <- this really doesn't mean anything.... |
| 00:28.36 | EPBX29 | just putting it out there |
| 00:28.52 | EPBX29 | but your right it doesn't :) |
| 00:28.58 | [TK]D-Fender | Show us the failed call attempt |
| 00:29.47 | EPBX29 | one moment |
| 00:29.49 | EPBX29 | i'll paste the logs |
| 00:30.58 | *** join/#asterisk u0m3 (~u0m3@92.80.126.165) |
| 00:31.02 | [TK]D-Fender | * CLI with SIP DEBUG enabled as I described, not "logs" |
| 00:37.17 | EPBX29 | damn it, not in the office where i'm having the phone issue. Will have to return back to the forums to get an output via CLI |
| 00:37.37 | EPBX29 | unless there is a way to do it via command line.... |
| 00:38.23 | EPBX29 | thanks |
| 00:38.24 | [TK]D-Fender | ... * CLI ... IS "command-line" |
| 00:38.55 | [TK]D-Fender | And if you can get a shell... you can get "whatever". Not that we should need more. |
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| 08:32.57 | dan_j | Hi, I log the cli verbose to a logfile so I can diagnose issues. However when I issue a dialplan reload using the manager api, it also gets logged. With older versions of asterisk, i used to set verbose to 0 before issuing the reload. However with newer versions, this doesnt help since each connection has its own verbose level. |
| 08:33.24 | dan_j | Is there any way to alter this without changing source code, so that the logger does not include the 'dialplan reload' output? |
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| 10:23.43 | robscow | Is it possible to see a detailed log of calls? ie, to see each request and response, in the CLI it looks like quite a high-level report, even running with -cvvvvv for example |
| 10:25.02 | kaldemar | enable debug for the protocol you're using. |
| 10:25.20 | robscow | I'm after a breakdown showing things like INVITE, 100 Trying, 407 Proxy Auth Required, ACK, INVITE, etc. |
| 10:25.29 | kaldemar | sip set debug on |
| 10:27.46 | robscow | thanks, but my god that's detailed! :) can i now get a higher level view, just showing the main requests and responses? |
| 10:28.20 | robscow | basically the first line of each request/response |
| 10:32.00 | kaldemar | nope. it's all or nothing. but you can of course grep it, e.g.: asterisk -vvvr | egrep -A 1 "(<--- SIP read|<--- Transmitting)" |
| 10:33.25 | robscow | aye, will go down that road, thanks again |
| 10:50.40 | robscow | When making a SIP call to my provider, there is a silent delay of 1-2 seconds between SIP/2.0 100 Trying and SIP/2.0 180 Ringing - How do I set the dialtone for this delay, or play another sound? Perhaps of the numbers being dialled? Tried playing with Playtones, but that's not doing anything. :-/ |
| 10:56.22 | robscow | ignore me, sorted-ish |
| 10:57.32 | Chainsaw | That makes us happy-ish. |
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| 11:46.27 | hebber | Anyone know what this is: [Apr 11 18:32:10] WARNING[18981]: frame.c:936 speex_get_wb_sz_at: Encountered corrupt speex frame; too many wideband frames in a row. |
| 11:46.28 | hebber | [Apr 11 18:32:10] WARNING[18981]: frame.c:966 speex_samples: Had error while reading wideband frames for speex samples |
| 11:46.28 | hebber | [Apr 11 18:32:10] WARNING[18981]: frame.c:994 speex_samples: Unknown speex control frame 9 |
| 11:46.51 | hebber | getting tons of them on Asterisk 11.7 don't even use the speex codecs |
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| 11:50.33 | hebber | it happen on a channel with ChanSpy |
| 11:54.20 | davlefou | hi, what is dbsecret in realtime database iax? |
| 11:56.54 | Chainsaw | davlefou: We're not allowed to tell you. |
| 11:58.01 | tparcina | Chainsaw: Because it's a secret? :) |
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| 11:58.13 | Chainsaw | tparcina: Precisely. |
| 11:59.15 | davlefou | there are secret, md5secret and dbsecret, so what is the last? |
| 11:59.53 | robscow | I'm trying to play a wav file while waiting between |
| 12:00.07 | robscow | ... |
| 12:01.27 | robscow | between 100 Trying, and 180 Ringing. I can do this with Playtones, but not with Playback or Background. is this possible? I can play the sound file, but it waits until it's finished playing before trying the call, I'm trying to fill the 2 second delay when calling out, but i just get the 2 second delay after playing the file |
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| 13:31.33 | seik0 | hi. Can somebody help to figure out why CEL logging is not logging to ODBC & |
| 13:31.35 | seik0 | *? |
| 13:35.03 | seik0 | what I have is: asterisk 1.8.26.1. Configured cel_custom, and cel_odbc. cel_custome works fine, cel_odbc as of debug in CLI even not trying to make insert in DB |
| 13:36.32 | aurs | seik0, module load cel_odbc.so perhaps? |
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| 13:37.21 | seik0 | cel show status shows: CEL Event Subscriber: ODBC CEL backend |
| 13:39.03 | aurs | and the connection => name in cel_odbc.conf matches the name from res_odbc.conf? |
| 13:39.41 | seik0 | yes |
| 13:39.57 | seik0 | btw, cdr logging to odbc works fine too |
| 13:40.05 | aurs | events=ALL in cel.conf? |
| 13:40.12 | seik0 | yes |
| 13:40.25 | aurs | do you have cel csv loaded as well? |
| 13:41.15 | seik0 | only cel_custom.so |
| 13:41.22 | seik0 | or what? |
| 13:41.41 | aurs | I have that one too |
| 13:41.56 | seik0 | i don't have cel_csv.so at all |
| 13:42.05 | marceloamorim | guys, whats the problem with this context: http://pastebin.com/Weib4y8F |
| 13:42.15 | aurs | was takling about this one: CEL Event Subscriber: CEL Custom CSV Logging |
| 13:42.26 | aurs | talking* |
| 13:42.28 | marceloamorim | I was reading about ''Time-conditional include statements'' |
| 13:42.59 | seik0 | aurs, yes, it shows Custom CSV |
| 13:43.00 | marceloamorim | but I use ael and didn fit with that context |
| 13:43.16 | seik0 | aurs, and csv cel logging is working ok |
| 13:43.33 | aurs | seik0, ok |
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| 13:46.21 | aurs | seik0, I was thinking that you might have the dsn name configured in cel_odbc.conf and not the [name] from res_odbc.conf, but I guess not. Then I have no more suggestions I'm afraid :) |
| 13:46.22 | seik0 | Debug shows no error, so i have not clues at all ( |
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| 13:48.00 | marceloamorim | add this information to my problem, sorry write this while you guys talking about this other situation, => Error: syntax error, unexpected KW_INCLUDES |
| 13:48.04 | seik0 | aurs, if name will be wrong i must see errors |
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| 13:52.18 | [TK]D-Fender | marceloamorim: You appear to be trying to do a conditional "include". that is not how this works. |
| 13:52.38 | FlipZTechGuy | Anyone have a preference on a store they buy there equipment from? Looking to buy som FXS cards and want to find a good place to get them. |
| 13:52.45 | [TK]D-Fender | marceloamorim: includes exist in the context from before you enter. they are not executed |
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| 13:53.40 | seik0 | aurs, now I'm thinking about one more thing: firstly asterisk was compiled with iODBC, but we need unixODBC, so I recompiled only res_odbc.so and moved it. For now I tried to recompiled cel_odbc.so too, but it didn't help. Maybe some others modules depends on it, (though shouldn't be any except res_odbc.so) |
| 13:54.43 | aurs | seik0, I'm using unixODBC. the cel_odbc config has connection=blabla and table=cel in the [first] section |
| 13:55.36 | file | prefers buying from a place selling Digium equipment, but he isn't biased AT ALL |
| 13:55.37 | seik0 | aurs, I have exactly the same |
| 13:55.56 | marceloamorim | so this ¨Time-conditional include statements¨ isn´t work anymore? |
| 13:56.09 | aurs | and the table has got columns like eventtype, eventtime, userdeftype, cid_name and so on |
| 13:56.41 | marceloamorim | I'm reading the-asterisk-book/version-1.6 |
| 13:57.12 | [TK]D-Fender | marceloamorim: You tried using DIALPLAN LOGIC to do this. That is entirely wrong |
| 13:57.42 | [TK]D-Fender | You can't execute an extension.. to INCLUDE an extension into a context. |
| 13:57.47 | [TK]D-Fender | It's either there, or it isn't |
| 13:58.27 | seik0 | aurs, if I try "module reload cel_odbc.so" it amongs other thing shows: "-- Found CEL table cel@conname" |
| 13:58.27 | marceloamorim | [TK]D-Fender: I want to do this example include => context|<time>|<day>|<day-of-month>|<month> |
| 13:58.37 | marceloamorim | http://the-asterisk-book.com/1.6/includes-im-dialplan.html |
| 13:59.53 | [TK]D-Fender | And that is not an AEL syntax guide |
| 14:00.05 | marceloamorim | but I change for ael, because I use my main context at ael. |
| 14:00.10 | FlipZTechGuy | I was looking at the A200 from sangoma |
| 14:00.11 | [TK]D-Fender | You are inventing it out of thin air on a basis that certainly doesn't make sense |
| 14:00.26 | [TK]D-Fender | marceloamorim: http://www.voip-info.org/wiki/view/Asterisk+AEL2 |
| 14:02.07 | [TK]D-Fender | marceloamorim: https://wiki.asterisk.org/wiki/display/AST/AEL+Includes |
| 14:05.07 | FlipZTechGuy | Digium 1A8A03F recomended if adding 8 PSTN lines to my asterisk box? |
| 14:05.54 | marceloamorim | I did that, because when i tried : context teste_e1 { IFTIME(10:00-10:30|fri|11|apr) includes {teste_e2}; }; and didn work |
| 14:06.28 | marceloamorim | oh, I see where is the problem. Ty tk |
| 14:06.33 | marceloamorim | Il try |
| 14:10.49 | aurs | seik0, I actually can't see the cel stuff in verbose/debug either, but it gets stored in the db |
| 14:11.33 | WIMPy | FlipZTechGuy: Can't you get (a) better line(s) in your area? |
| 14:11.58 | seik0 | aurs, if enable debug in logger.conf (add debug to console), then there are lots of "DEBUG[53618]: pbx.c:4322 ast_str_substitute_variables_full: Evaluating 'CSV_QUOTE(${CHANNEL(context)})' (from 'CS...." |
| 14:12.30 | seik0 | aurs, which is around cel and csv and not here if unload cel_custom.so |
| 14:12.54 | seik0 | aurs, but mainly seems cel is not very talkative ) |
| 14:13.00 | seik0 | aurs, at all |
| 14:13.08 | FlipZTechGuy | Wimpy: I wish. Middle of nowhere |
| 14:13.50 | FlipZTechGuy | well, without going up to PRI and the cost involved. |
| 14:14.50 | WIMPy | The PRI is too expensive? No BRIs available? |
| 14:14.54 | FlipZTechGuy | I am just trying to decide bwtweek an 8 port digium card and a a200 with a remora card from sangoma |
| 14:15.13 | aurs | well good luck seik0, gotta go :) |
| 14:15.19 | FlipZTechGuy | If we hit around 12 lines needed , the PRI would match the monthly cost but we don't need that many. |
| 14:15.39 | seik0 | aurs, bye! =) |
| 14:15.46 | FlipZTechGuy | and BRI, I dont *think* so. think is the big word. I will check and see on that though |
| 14:15.59 | WIMPy | FlipZTechGuy: But it would be much easier and much more usable. I'd strongly recommend taking it anyway. |
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| 14:16.25 | FlipZTechGuy | i would completly agree....but the people who run oney do not agree much |
| 14:16.37 | FlipZTechGuy | Dealing with PRi was so much easier in the past that this |
| 14:17.53 | WIMPy | Well, they wouldn't have to change anythign to expand in the near future. |
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| 14:18.17 | FlipZTechGuy | i agree. and woudl make my life sooo much easier |
| 14:18.47 | FlipZTechGuy | but i doubt they will go for it. they were thinking of dropping down to 6 lines already |
| 14:18.58 | WIMPy | Maybe you should try to estimate the extra anout of work/time/cash. |
| 14:18.58 | FlipZTechGuy | digium vs sangoma |
| 14:19.24 | FlipZTechGuy | yep. tried that one. told them what they pay me and how much in the end....but no luck on that one |
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| 14:31.14 | Groenleer | What is the most popular php agi class today ? |
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| 14:47.00 | seik0 | Maybe now there is somebody another who can help to figure out why CEL is not logging events to ODBC? |
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| 15:03.07 | [TK]D-Fender | [10:31]GroenleerWhat is the most popular php agi class today ? <- I've only ever seen one published out there. |
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| 15:14.54 | gusto | hi WIMPy |
| 15:14.59 | gusto | how are you? |
| 15:15.04 | gusto | Penguin: are you here? |
| 15:15.12 | WIMPy | ok |
| 15:15.19 | gusto | thats good |
| 15:15.27 | gusto | have you finally fixed your ISDN problems? |
| 15:15.47 | WIMPy | My ISDn problems? |
| 15:15.57 | gusto | that adapter you were trying to get to work |
| 15:16.11 | WIMPy | No. I haven't tried. |
| 15:16.34 | gusto | so you are refusing to continue now |
| 15:16.46 | WIMPy | I just use it with DAHDI, as I need to have that installed for the FXS card anyway. |
| 15:17.02 | gusto | did you hear the news? heartbleed is out! lol, for those ppl who relied on ssl |
| 15:17.05 | WIMPy | Yes. They will have to live with how it works that way. |
| 15:17.24 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 15:17.24 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 15:17.59 | WIMPy | But the question remains, where to buy cards these days. |
| 15:18.21 | gusto | we have to make them by ourselves |
| 15:18.25 | WIMPy | I guess OpenVox is the best bet. |
| 15:18.59 | WIMPy | I know someone who did as he couldn't find a HFC-E1 card, but could find the chips. |
| 15:19.00 | gusto | or use IP only ... but then the question is how to reach PSTN |
| 15:19.28 | WIMPy | Who wants to support that? |
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| 15:20.35 | gusto | are the propraitary solution too expensive, like cisco or such? |
| 15:22.20 | WIMPy | Depends on how much your time is worth, I guess. |
| 15:22.48 | gusto | but there are some around, right? |
| 15:23.24 | WIMPy | A lot. |
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| 17:08.40 | EPBX21 | Got an issue here. we just created a new extension for internal dialing. When you pick up the phone, you get a dial tone. But when you try to dial any internal/external numbers you immediately get a disconnect tone. |
| 17:08.41 | EPBX21 | I use cisco 7960 phone and towards the top right corner, next to the 247 extension, the phone icon has an X next to it. It's not the phone because I assigned an existing working extension and the phone works. http://tinyurl.com/l793rl4 |
| 17:08.41 | EPBX21 | Is there something i'm missing when adding a new extension? is there somewhere I need to tell Asterisk it should allow more extensions? not an Asterisk expert but I can defend my self. However, this I cannot figure out. |
| 17:08.49 | EPBX21 | This is the CLI debut output |
| 17:08.49 | EPBX21 | [Apr 11 10:31:34] NOTICE[31004]: chan_sip.c:15157 handle_request_invite: Failed to authenticate user "247" <sip:247@192.168.55.10>;tag=00152ba20c3900097b45e980-2a473234 |
| 17:09.30 | EPBX21 | I copied the configurations of an existing working extension so I know what we have configure works with existing phones/extensions |
| 17:09.36 | EPBX21 | thank for your help |
| 17:11.30 | EPBX21 | thanks* |
| 17:11.54 | kaldemar | are you using freepbx? |
| 17:12.33 | EPBX21 | Asterisk 1.4.36 |
| 17:12.37 | EPBX21 | I know it's an old version |
| 17:13.33 | drmessano | The log says it all |
| 17:14.26 | kaldemar | i didn't ask for your asterisk version but if you're using freepbx to configure asterisk. |
| 17:16.21 | *** part/#asterisk aruntomar (~arun@49.248.152.227) |
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| 17:17.48 | EPBX21 | using Digium |
| 17:21.39 | kaldemar | so asterisk gui. its support is close to none nowadays. you might have luck in #asteriskgui. |
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| 17:30.38 | drmessano | Most of the asterisk-gui devs are dead, moved on, or were deported |
| 17:30.40 | drmessano | :( |
| 17:30.56 | file | dead? that's rather harsh |
| 17:31.11 | file | passes out at this table |
| 17:31.16 | drmessano | lol |
| 17:31.50 | drmessano | points to the deported one |
| 17:32.11 | drmessano | Funny how that just worked out |
| 17:32.42 | file | I bet this table isn't very comfy |
| 17:32.59 | file | has taken over part of mjordan's office, specifically a corner of it and claimed it as CANADA |
| 17:35.15 | drmessano | ha |
| 17:35.48 | drmessano | Is there a "blamecanada" resolution in Jira yet? |
| 17:42.23 | file | there is not |
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| 17:47.21 | *** join/#asterisk Anvic (mihail@ip-46-73-244-154.bb.netbynet.ru) |
| 17:47.35 | Anvic | Всем привет! |
| 17:50.03 | drmessano | Motherland! |
| 17:51.12 | Anvic | тут русские есть? |
| 17:52.00 | Anvic | hi all |
| 17:52.52 | navaismo | finally something i can barely understand |
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| 17:54.26 | navaismo | o/ Anvic |
| 17:57.24 | *** join/#asterisk Anvic (Anvic@ip-46-73-244-154.bb.netbynet.ru) |
| 17:57.30 | Anvic | There is Russian? |
| 18:01.20 | *** join/#asterisk rmudgett (~rmudgett@nat/digium/x-pkeudljeqstubjod) |
| 18:02.58 | *** join/#asterisk Anvic (Anvic@ip-46-73-244-154.bb.netbynet.ru) |
| 18:03.03 | Anvic | Or can tell me where can assist with setting up asterisk? |
| 18:03.52 | Anvic | please help me |
| 18:04.18 | Anvic | i nid setting asterisk help |
| 18:05.41 | sgriepentrog | Have you seen http://www.asteriskdocs.org/ ? |
| 18:06.09 | Anvic | no need inform |
| 18:06.51 | Anvic | need Russian help |
| 18:08.16 | navaismo | there is a mailing list or forum asterisk-ru |
| 18:08.21 | Anvic | ok thanks |
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| 18:25.52 | Nugget | http://www.mozilla.org/keymaster/gatekeeper/there.is.only.xul |
| 18:34.15 | WIMPy | Those were the days when you could change universe. |
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| 18:59.33 | hexanol | someone knows if it's possible to use |
| 18:59.39 | hexanol | the Atxfer AMI command |
| 18:59.48 | hexanol | on a channel that is bridge with another channel via the Dial application |
| 18:59.53 | navaismo | MaliutaLap, drmessano ubuntu with Mac look&feel-->http://elementaryos.org/ |
| 18:59.54 | hexanol | but without the tT Dial options |
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| 19:02.08 | Katty | packs her bags. |
| 19:02.13 | Katty | hides in leifmadsen's luggage |
| 19:03.47 | hexanol | my problem is |
| 19:04.14 | hexanol | using asterisk 11, I need to be able to have direct media activated |
| 19:04.24 | hexanol | between two SIP peer |
| 19:04.27 | Katty | and tylenol |
| 19:04.32 | Katty | don't forget the tylenol. you'll also need that. |
| 19:04.34 | hexanol | and I need to be able to use the attended transfer feature from the manager |
| 19:04.38 | hexanol | i.e. the Atfxfer command |
| 19:04.57 | hexanol | but I can't put the 'tT' options in the Dial app since this breaks the direct media |
| 19:04.59 | navaismo | sounds like incompatible directmedia and transfer |
| 19:05.01 | hexanol | I'm sad, even with tylenol |
| 19:05.34 | hexanol | but then, I feel like the fact the Atxfer AMI command is implemented by queuing DTMF frame is kinda a hack |
| 19:06.28 | hexanol | someone has any idea on how I could solve my problem ? |
| 19:06.43 | Katty | yes. |
| 19:06.45 | Katty | vodka. |
| 19:06.50 | hexanol | that's what I was thinking |
| 19:06.58 | hexanol | but it only solves it for a short duration |
| 19:07.03 | Katty | let's go. first rounds on me |
| 19:07.11 | hexanol | ;) |
| 19:07.17 | Katty | navaismo can drive. |
| 19:07.55 | navaismo | naa i want to puke over all |
| 19:08.45 | Katty | that's not fun tho. |
| 19:09.11 | navaismo | thats the end of the previous fun |
| 19:11.15 | Katty | you can have previous fun with no puking afterwards, you know |
| 19:12.06 | navaismo | na na I want to be a "mala copa" guy |
| 19:12.32 | navaismo | molesting all the girls puking on the food with my vodka bottle in my hand |
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| 19:14.54 | navaismo | I guess i will never be invited to an astricon party after wrote that |
| 19:15.17 | Katty | i think that sounds like the definition of a cluecon party |
| 19:15.29 | Katty | except you end up going to Diamonds later. |
| 19:15.39 | Katty | and then forget your visa in someone else's hotel room |
| 19:16.05 | navaismo | but uSA can deport me so free return to my country YAY |
| 19:16.25 | Katty | i thought they were going to deport Junky |
| 19:16.36 | Katty | not really, but he was very unhappy about them taking his passport away |
| 19:17.07 | Katty | navaismo: i can't imagine being deported by the USA goverment is a very fun process :/ |
| 19:17.28 | Katty | navaismo: probably not a....oh let's call you a cab and get you on a nice flight out in the morning, would you like some coffee? that sounds like sweden |
| 19:19.36 | navaismo | i hope the sadism histories about Immigration US office are just a myth but based on this i guess not--->https://www.youtube.com/watch?feature=player_detailpage&v=lTdtgOlVtsU#t=89 |
| 19:19.59 | Katty | looks |
| 19:21.11 | Katty | sighs |
| 19:21.33 | Katty | and that's probably not the only incident either :/ |
| 19:22.05 | navaismo | nope the worst was Police shooting a kid from the other side of the border |
| 19:22.24 | navaismo | :'( |
| 19:22.57 | Katty | just stay home, don't come visit :< |
| 19:24.13 | navaismo | if I go to visit USA im going to do taking the legal path, not trying to swim across the river |
| 19:24.41 | Katty | that'd be rather difficult without arm floaties, me thinks |
| 19:25.12 | navaismo | or crossing the dessert |
| 19:25.21 | Katty | mmm, dessert. |
| 19:25.35 | navaismo | strwaberry icecream |
| 19:25.41 | navaismo | derp |
| 19:25.44 | Katty | the usa does dessert well, i think |
| 19:26.10 | navaismo | dont know |
| 19:26.18 | navaismo | i want a cake from the "cake boss" |
| 19:27.32 | Katty | mmm cake. |
| 19:28.36 | navaismo | http://humorgeeky.com/wp-content/uploads/2012/03/pastel-lego.jpg |
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| 20:32.06 | drmessano | <PROTECTED> |
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| 20:47.09 | ziz212 | Hi |
| 20:47.14 | ziz212 | I am with some issue |
| 20:47.43 | ziz212 | I have given a sip trunk to mera softswtich and they say it is standard one. |
| 20:48.45 | ziz212 | but I also see the sip registrations but no debug or any message seen in the asterisk cli |
| 20:49.09 | ziz212 | this is the tcpdump .. pls help me |
| 20:49.09 | fireman_biff | When I do "core show hints" I see the number of "watchers". Is there anyway to see who the watchers are? |
| 20:49.09 | ziz212 | http://pastebin.com/eyRpqeXp |
| 20:49.34 | ziz212 | this is the tcpdump .. pls help me http://pastebin.com/eyRpqeXp |
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| 22:17.30 | SuperNull | anyone able to clarify dtmfmode things to me ? for example i have an ATA that seems to ONLY work doing inband, even when asterisk is explicitly set to info and the ata is set to info. Info messages have been verified. |
| 22:21.42 | SuperNull | i guess the question is.. why would things work via inband when nothing is configured for it (that i know of) |
| 22:22.16 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
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| 22:22.33 | SuperNull | Whats up TK |
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| 23:28.30 | kemmler | Hey, I'm using x-lite to make a call to my cell. Asterisk is on a linode box so no nat necessary that I know of. I hear audio coming from x-lite -> asterisk -> phone, but no audio from phone -> asterisk -> x-lite |
| 23:29.24 | BludSuckingFiend | Might be that you are behind a NAT. Most home connections are |
| 23:29.59 | BludSuckingFiend | You would likely need to configure X-lite to use a specific UDP port for receiving RTP, and forward that port in your router to your PC |
| 23:30.33 | kemmler | I am behind a router here at home. I'll try that, thanks |
| 23:30.42 | BludSuckingFiend | np |
| 23:31.02 | *** join/#asterisk jmmills (~jmmills@204.16.200.135) |
| 23:31.48 | jmmills | So I'm having a similar issue to this http://forums.digium.com/viewtopic.php?f=1&t=86684&start=0 |
| 23:31.54 | jmmills | but, I don't see a clear resolution there |
| 23:32.14 | jmmills | basically I'm trying to get a nagios sip check going thus asterisk needs to reply with a 200 to an OPTIONS packet |
| 23:33.14 | [TK]D-Fender | Why exactly? |
| 23:33.28 | jmmills | Why exactly am I setting up a nagios check? |
| 23:33.46 | [TK]D-Fender | perhaps you could tell it not to care about options..... |
| 23:33.59 | BludSuckingFiend | I would think any response is good. |
| 23:34.02 | BludSuckingFiend | 404 or 200 |
| 23:34.09 | [TK]D-Fender | For what * cares about.... yes |
| 23:34.11 | jmmills | I don't have control over that, and I didn't write the check |
| 23:34.26 | [TK]D-Fender | And you can't stop it? |
| 23:34.52 | jmmills | I can stop the check, but then the monitoring system won't be checking that asterisk is actually responding with sip |
| 23:34.59 | jmmills | I don't own the monitor host |
| 23:35.02 | [TK]D-Fender | So it's all or nothing? |
| 23:36.14 | jmmills | yeah, the situation is poop |
| 23:36.34 | [TK]D-Fender | jmmills: Pastebin your complete CLI output with "sip det debug on", "core set verbose 10" |
| 23:36.36 | [TK]D-Fender | ^^^ |
| 23:36.39 | jmmills | really the problem is in the sip check plugin, it should have a flag that says any response is a good response |
| 23:37.17 | BludSuckingFiend | would be handy if the SIP check plugin was written in an interpreted language that's easy to edit |
| 23:38.05 | jmmills | eck, I doubt that would be much help, I just get 404's to option packets... when should asterisk reply with a 200 |
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| 23:38.28 | jmmills | BludSuckingFiend: it is, but I don't own the monitoring box - I suppose I could ask them to accept a modified check |
| 23:38.33 | jmmills | that's actually what I'm going to do |
| 23:38.44 | jmmills | easiest path |
| 23:38.54 | [TK]D-Fender | ~pb |
| 23:38.54 | infobot | 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/; or install pastebinit with yum or aptitude. |
| 23:38.56 | [TK]D-Fender | ^^^ |
| 23:44.39 | BludSuckingFiend | hey [TK]D-Fender ... do you know what happens in asterisk when a jitterbuffer goes empty mid-stream? Will it halt audio momentarily to refill the buffer, or just continue unbuffered? |
| 23:45.16 | [TK]D-Fender | Dunno.... |
| 23:47.25 | BludSuckingFiend | I have this bad app that streams RTP, and is consistently .5ms late with its 20ms g.711 chunks. Had to enable a 500ms jitter buffer for the audio to sound decent |
| 23:48.45 | [TK]D-Fender | late != jitter |
| 23:48.58 | [TK]D-Fender | mis-ordered = jitter |
| 23:49.12 | *** join/#asterisk Bryanstein (~Bryanstei@shellium/admin/bryanstein) |
| 23:49.16 | [TK]D-Fender | but I suppose the same buffer can cover both. |
| 23:50.34 | BludSuckingFiend | Yeah. It helps. But the real bit rate is about 78.5kbps coming into the buffer thanks to the misbehaving RTP source... and it's going out at the proper 80kbps. Something bad has to happen when that buffer goes away. |
| 23:51.55 | [TK]D-Fender | I would think that * would keep a certain amount buffered constantly... |
| 23:52.07 | [TK]D-Fender | it can't be a buffer if it's deliberately kept empty, right? |
| 23:53.07 | BludSuckingFiend | Well, asterisk can fill it by waiting 500ms before starting output... but if you're only getting 78.5kbps into the buffer, and you're needing to output 80kbps audio rate for the sound to work |
| 23:53.23 | BludSuckingFiend | the buffer will drain |
| 23:54.06 | BludSuckingFiend | I suppose it could insert silence |
| 23:54.32 | BludSuckingFiend | to make sure the incoming rate is the same. The other app is definitely just broken though |
| 23:54.40 | BludSuckingFiend | I am trying to convince the devs how broken it is |
