| 00:03.07 | slav3_kitten | dijib, want a laptop |
| 00:13.21 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
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| 00:30.11 | epaphus | COuld anybody please explain to me if ChanSpy is meant to listen on specific calls or can I also listen on all calls that an extension (device) is handling? |
| 00:37.13 | slav3_kitten | epaphus, what's the command |
| 00:39.48 | epaphus | ChanSpy() |
| 00:43.50 | slav3_kitten | a quick google later https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_ChanSpy |
| 00:44.49 | slav3_kitten | i like google |
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| 01:04.09 | *** join/#asterisk Sean-Der (~sean@NW1-DSL-74-215-64-154.fuse.net) |
| 01:04.48 | Sean-Der | Does anyone know the definitive source to get either the UTC offsets OR timezones of each area code in the US? |
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| 01:07.14 | slav3_kitten | with a few exceptions most states are 1 timezone |
| 01:07.51 | slav3_kitten | 14 of 50 are split |
| 01:10.15 | Sean-Der | Thats a decent amount of split :/ I found a decent source that I scraped |
| 01:10.43 | slav3_kitten | majority of the states are not split |
| 01:17.13 | dijib | anybody available to debug a dialplan... when i dial from any of the sip phones it does 15555555555@voipms-inbound when it should be using the voipms-outbound context... did i mess up the includes???? |
| 01:17.17 | dijib | http://pastebin.com/Y1yCiuAe |
| 01:27.33 | *** join/#asterisk serafie (~erin@76.73.167.231) |
| 01:33.25 | slav3_kitten | wait what? |
| 01:34.43 | slav3_kitten | why do you have 555-555-5555 twice on your inbound |
| 01:36.31 | slav3_kitten | dijib, |
| 01:37.43 | slav3_kitten | also dijib http://wiki.voip.ms/article/PBXs#Asterisk_.28IAX2.29 |
| 01:38.19 | slav3_kitten | ; Make sure to include inbound prior to outbound because the _NXXNXXXXXX handler will match the incoming call and create a loop |
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| 02:23.13 | dijib | slav3_kitten: its an example |
| 02:23.20 | dijib | sorry ive been dialplan debugging |
| 02:23.47 | slav3_kitten | dijib, it also tells you that you have your includes in the wrong order |
| 02:24.34 | p3nguin | You shouldn't have the inbound context available to the phones. |
| 02:24.48 | dijib | your right. |
| 02:24.54 | p3nguin | My right. |
| 02:26.19 | p3nguin | Apparently it is my right. |
| 02:26.26 | p3nguin | Why isn't it your right? |
| 02:26.35 | p3nguin | You have less rights than I have? |
| 02:27.25 | jpsharp | I only have one right. |
| 02:27.37 | jpsharp | Unless I face the other way. |
| 02:27.59 | dijib | in the phones context p3nguin ? |
| 02:28.04 | dijib | i shouldnt have inbound? |
| 02:28.11 | p3nguin | I would have expected you to have all the rights that everyone else in this country has. |
| 02:28.19 | p3nguin | (2024.34) <p3nguin> You shouldn't have the inbound context available to the phones. |
| 02:28.26 | p3nguin | Answer ^^^^^^^^^^^^^^^^^^^^^^^^^^^ |
| 02:29.04 | p3nguin | Phones should have access to internals and outbound, as well as any special extensions for PBX featurs, etc. |
| 02:29.12 | dijib | whats this??? http://pastebin.com/SQU2YBuN |
| 02:29.47 | dijib | no inbound in phones |
| 02:31.01 | p3nguin | Also, ALL_OUT isn't a very good name for an account for a phone. |
| 02:31.15 | p3nguin | But neither is 11 for the phone's name. |
| 02:31.47 | dijib | also this http://pastebin.com/91Zt17dK |
| 02:32.08 | dijib | thats my cdr |
| 02:32.39 | dijib | ALL_OUT, ALL_IN, and MASTER.csv |
| 02:34.15 | p3nguin | You should really go do some reading. The ACL notice is self-explanatory. |
| 02:35.22 | dijib | i like pasta. |
| 02:35.38 | dfgas-cr48 | me too |
| 02:35.43 | p3nguin | And I like to help people who read. |
| 02:35.46 | dijib | so its unable to authenticate |
| 02:35.54 | dfgas-cr48 | :P |
| 02:36.09 | dijib | but its not a password error its an access control list |
| 02:36.19 | p3nguin | Go fix it. |
| 02:36.21 | dijib | so the device is the issue? |
| 02:36.26 | dijib | fix what? |
| 02:36.31 | dijib | i dont even knw where to look |
| 02:36.37 | p3nguin | ACL <-------- |
| 02:37.00 | p3nguin | Is that IP address one of a phone? |
| 02:37.34 | dijib | acl.conf? |
| 02:37.45 | dijib | yes it seems to be connecting through 4g |
| 02:37.46 | dijib | 3g |
| 02:37.48 | p3nguin | ACL should be well-documented in the book. Read it instead of guessing. |
| 02:38.02 | dijib | https://wiki.asterisk.org/wiki/display/AST/Named+ACLs |
| 02:40.27 | dijib | ok i dont even have acl.conf in my /etc/asterisk dir so i killed his and core restart'ed |
| 02:40.44 | slav3_kitten | so just like an acl on a router |
| 02:41.16 | p3nguin | I'm pretty sure acl.conf isn't in 1.8, so if you are using that, don't expect to see it. |
| 02:41.26 | dijib | 11.0.1 |
| 02:41.42 | p3nguin | It is, however, in 11, and that is what the notice is complaining about. |
| 02:41.49 | slav3_kitten | so acl's aren't in the book, the printed book that is |
| 02:42.20 | p3nguin | I guess that's true. I should retract my remark about it being documented in the book. |
| 02:42.38 | p3nguin | But it is in the wiki, which is nearly as useful. |
| 02:43.01 | dijib | dfgas-cr48: wanna give that a try |
| 02:43.04 | slav3_kitten | i have a hard time reading wiki's |
| 02:43.13 | slav3_kitten | and seeing that my damn ebook died... |
| 02:43.19 | p3nguin | It even TELLS YOU which ACL is responsible for the blockage. |
| 02:43.33 | slav3_kitten | i now gotta figure out what ebook i want to get. thinking a nook simpletouch glowlight |
| 02:43.48 | p3nguin | Not a kindle fire? |
| 02:44.13 | slav3_kitten | i wanted the new kindle e-ink book but 119 bucks, no AC adapter, and 20 to not have advertisements on it |
| 02:44.15 | dfgas-cr48 | dijib, same |
| 02:44.30 | slav3_kitten | vs 100 for the nook that is ad free, and comes with an AC adapter |
| 02:44.41 | p3nguin | Sounds like you've made your decision. |
| 02:44.48 | slav3_kitten | kindle has a nigher resolution though |
| 02:45.09 | slav3_kitten | so still unsure |
| 02:45.48 | dijib | dfgas-cr48: i see that |
| 02:45.58 | dijib | this thing is driving me batty |
| 02:48.17 | dfgas-cr48 | :( |
| 02:48.20 | dijib | ok ive overwritten the outbound dialplan with the voip-ms example |
| 02:49.21 | slav3_kitten | link? |
| 02:49.22 | p3nguin | The one from voip.ms? |
| 02:50.05 | p3nguin | They have a horrible example that shouldn't be used when you can create something much nicer. |
| 02:50.38 | slav3_kitten | p3nguin, what's wrong with their example? |
| 02:50.57 | p3nguin | http://pastebin.com/Piqv4Egj I have put a much better one here. |
| 02:51.12 | slav3_kitten | other then being way out of date |
| 02:51.19 | SeRi | waz up guys |
| 02:51.22 | dfgas-cr48 | dijib, can hear it? |
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| 02:52.29 | dijib | core dfgas-cr48 yes i could hear monkeys |
| 02:52.35 | dijib | your back SeRi |
| 02:52.49 | p3nguin | His back? |
| 02:52.58 | p3nguin | Why are you concerned with seri's back? |
| 02:53.03 | p3nguin | Sounds a little... you know. |
| 02:53.15 | SeRi | my back is nobodys back |
| 02:53.17 | slav3_kitten | p3nguin, :| |
| 02:53.24 | SeRi | my fronts is my wifes |
| 02:53.29 | SeRi | LOL |
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| 02:55.31 | SeRi | p3nguin!!! |
| 02:59.15 | SeRi | p3nguin: jump in |
| 03:04.50 | SeRi | SIP ALG!!! |
| 03:04.54 | slav3_kitten | p3nguin, i read the book's bit about dial plans, never saw a thing about the whole top section |
| 03:05.24 | p3nguin | What top section? |
| 03:05.35 | slav3_kitten | [general] |
| 03:05.59 | p3nguin | Like you'd make a config without a general section or something? |
| 03:07.05 | slav3_kitten | i would and did |
| 03:11.26 | infinity1 | shoudl a pbx require "1" to dial long distance or should it just add it? |
| 03:11.40 | p3nguin | Depends. |
| 03:12.15 | p3nguin | I support both 10- and 11-digit dialing for North American numbers. |
| 03:12.42 | slav3_kitten | infinity1, mine supports 11, 10, and 7 digit dialing for North America |
| 03:12.59 | infinity1 | the way i've done it in the past is 11 digits fro long distance. the problem though is when caller ID comes in its 10 digits and people complain they can't scroll through the last call list on their phone and select dial. |
| 03:13.26 | infinity1 | because caller ID is 10 digits. |
| 03:13.36 | infinity1 | so i added a 1 to caller ID ..but i'm thinking thats lame |
| 03:13.39 | infinity1 | heh |
| 03:20.58 | slav3_kitten | p3nguin, i like your outbound calling |
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| 03:41.59 | p3nguin | slav3_kitten: Me too. |
| 03:45.39 | slav3_kitten | so how long you been writing dialplans an such? |
| 03:46.39 | p3nguin | years |
| 03:47.08 | slav3_kitten | ah, well you sure picked up tricks |
| 03:47.19 | infinity1 | k. i allow 10 and 11 digits for NA dialing now :) |
| 03:48.00 | p3nguin | I also allow 7-digit since we have only one area code encompassing our geography. |
| 03:48.37 | infinity1 | is there microbrowser php code for polycom around? i just want something to try like weather or whatever. |
| 04:00.15 | p3nguin | -= 722 extensions (2707 priorities) in 183 contexts. =- |
| 04:01.14 | dijib | http://pastebin.com/FfCXnTHp |
| 04:02.41 | slav3_kitten | stupid question but "== Spawn extension (Wake-Up, 23, 4) exited non-zero on 'SIP/testphone-0000004d'" is normal right? |
| 04:03.00 | p3nguin | Could be. |
| 04:03.18 | p3nguin | Did it do something that you did not expect? |
| 04:04.15 | slav3_kitten | no it did as expected, just wish i could make it more verbose like giving the exit cause |
| 04:04.47 | p3nguin | You want it to say, "Ran out of things to do, exiting." |
| 04:05.22 | slav3_kitten | also p3nguin in the book they have you make a [LocalSets] for your phones. why do you have a [phones] |
| 04:05.36 | slav3_kitten | nah i'd like it to say like Hangup, Failed, etc |
| 04:06.08 | p3nguin | I had a context called 'phones' before the book was written. |
| 04:06.56 | slav3_kitten | just curious why you had a context phones, as if maybe it's some default context i should change mine to reflect |
| 04:07.15 | p3nguin | While the names of contexts can be arbitrary, they should reflect what the contexts are to be used for. In my case, the phones context is for phones, which Leif's phones' context is called LocalSets. |
| 04:08.15 | p3nguin | The only default contexts that I know of are general, default, globals. |
| 04:08.35 | slav3_kitten | *nods* |
| 04:10.57 | slav3_kitten | p3nguin, earlier. is it supposed to say "ran out of things to do, exiting" ? |
| 04:11.21 | p3nguin | It seemed like you wanted to SEE it tell you that every time a call ended. |
| 04:11.33 | p3nguin | When the extension runs out of things to do, it exits. |
| 04:13.35 | slav3_kitten | right, i was just unsure what the non zero bit meant an if it was supposed to exit 0 or non zero if everything went well |
| 04:13.58 | slav3_kitten | dijib, what' up with line 129. i didn't see that exten anyplace |
| 04:14.06 | slav3_kitten | or label |
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| 04:41.41 | dijib | slav3_kitten: ignore that |
| 04:44.20 | SeRi | http://tinyurl.com/7wllmrc |
| 04:44.43 | slav3_kitten | dijib, why? |
| 04:44.43 | SeRi | Thats my dial plan for those interested ^^ |
| 04:46.42 | SeRi | dijib: Here is my dial plan http://tinyurl.com/7wllmrc |
| 04:47.26 | dijib | http://tinyurl.com/7wllmrc |
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| 04:51.21 | SeRi | google.com |
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| 05:39.32 | slav3_kitten | i /think/ i have the damn dialplan fixed up a bit |
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| 06:05.02 | slav3_kitten | i really want to get cdr shit working next |
| 06:06.29 | WIMPy | exten => youwannashit,1,Goto(the,loo,1) |
| 06:07.16 | slav3_kitten | lol |
| 06:09.44 | ChrisInSydney | g'day all |
| 06:10.27 | WIMPy | Good morning. |
| 06:10.36 | ChrisInSydney | afternooon |
| 06:10.41 | WIMPy | Or maybe rather good night. |
| 06:10.46 | ChrisInSydney | :-) |
| 06:10.52 | ChrisInSydney | 17:10 here |
| 06:10.56 | ChrisInSydney | so arvo it is |
| 06:11.29 | WIMPy | is missing the 1. |
| 06:12.33 | ChrisInSydney | has got a lamb shoulder in the oven on 160C or so. Will be in there for another few hours. Marinaded overnight in Tandori spices and yoghurt |
| 06:12.42 | ChrisInSydney | :P |
| 06:12.59 | ChrisInSydney | Sunday roast |
| 06:13.05 | WIMPy | How many more hours? |
| 06:13.40 | ChrisInSydney | went on at 15:00 or so 19:30 I recon is should be right |
| 06:13.44 | ChrisInSydney | it |
| 06:14.08 | WIMPy | No, I can't make it in time :-( |
| 06:14.21 | ChrisInSydney | it was too windy to use the BBQ with the hood |
| 06:14.40 | ChrisInSydney | sorry, ill send you tasting notes :P |
| 06:15.15 | WIMPy | waits for the invention of real multimedia. |
| 06:15.25 | ChrisInSydney | Anyways,I'd better get onto my MySQL GUI thingy for the extension settings / presence / provisioning |
| 06:15.25 | jpsharp | FedEx a plate of that here. |
| 06:15.45 | ChrisInSydney | jsharp: done ;-) |
| 06:16.17 | ChrisInSydney | Think I'll crack a bottle of red too. |
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| 06:26.09 | sanman | NOTICE[1436]: chan_iax2.c:8883 update_registry: Restricting registration for peer 'XXX' to 60 seconds (requested 120) |
| 06:26.33 | sanman | i thought i fixed it but nope |
| 06:27.49 | slav3_kitten | ok does anyone know how to generate an asterisk notice so you can log events to disk? |
| 06:28.21 | sanman | how i make this go away |
| 06:28.27 | sanman | NOTICE[1436]: chan_iax2.c:8883 update_registry: Restricting registration for peer 'MSI' to 60 seconds (requested 120) |
| 06:28.51 | slav3_kitten | sanman, change the bit in your iax where you have a 120 second registration time |
| 06:29.07 | slav3_kitten | it's saying MSI want's 60 seconds nd you asked for 120 |
| 06:29.44 | WIMPy | slav3_kitten: notice? |
| 06:30.03 | slav3_kitten | yea in the logger.conf you can log warnings, notices, etc |
| 06:30.18 | p3nguin | Log() |
| 06:30.27 | WIMPy | Log(NOTICE,blah) |
| 06:30.29 | slav3_kitten | :/ well fuck |
| 06:30.47 | slav3_kitten | why would it be something logical like that |
| 06:30.49 | sanman | i dont have anything in my iax with those value |
| 06:31.35 | slav3_kitten | sanman, defaults |
| 06:32.08 | sanman | i've check everything |
| 06:32.18 | sanman | I went over the sample |
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| 06:33.51 | slav3_kitten | http://www.voip-info.org/wiki/view/IAX < read the bit about registration |
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| 06:38.40 | dijib | well that didnt work |
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| 06:39.17 | dijib | the s option broke it... i have reverted |
| 06:39.30 | dfgas-cr48 | hmmmm |
| 06:39.32 | dfgas-cr48 | fired |
| 06:39.35 | dfgas-cr48 | lol |
| 06:39.48 | dijib | meh wasnt getting paid anyways |
| 06:40.13 | dfgas-cr48 | lol |
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| 06:44.02 | sanman | minregexpire = 120 maxregexpire = 120 didn't fix it |
| 06:44.13 | sanman | minregexpire = 60 maxregexpire = 60 |
| 06:44.18 | sanman | didn't fix it |
| 06:46.25 | dijib | The syntax for the new ConfBridge application is as follows: |
| 06:46.25 | dijib | ConfBridge([confno][,bridge_profile[,user_profile[,menu]]]) |
| 06:47.22 | WIMPy | s/no/na/ |
| 06:48.23 | p3nguin | So you have to use ConfBridge(1234,,,s) ? |
| 06:49.41 | dijib | no. |
| 06:50.29 | dijib | i have to use ConfBridge(${EXTEN},default_bridge,default_user,user_menu) its all confirgured in confbridge.conf |
| 06:50.32 | dijib | now |
| 06:50.42 | p3nguin | On my ConfBridge, it's #1 to mute/unmute. |
| 06:50.57 | dijib | they rewrote confbridge |
| 06:51.04 | p3nguin | Oh. I still use 1.8, so ConfBridge is primitive. |
| 06:51.13 | dijib | yes now its advanced |
| 06:51.40 | p3nguin | You guys should conf on my box. |
| 06:51.46 | WIMPy | The one in 1.8 was more of a POC I think. |
| 06:52.05 | dijib | mines almost fixed |
| 06:52.10 | dijib | maybe not |
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| 06:53.01 | dijib | how do i reload to conference config |
| 06:53.27 | WIMPy | You have to reload the module. |
| 06:54.04 | p3nguin | The first thing I would do is try conf<TABKEY> on the CLI to see if there is a reload command for it. |
| 06:54.11 | p3nguin | If it isn't there, then module show like conf |
| 06:54.20 | p3nguin | And then module reload whatever. |
| 06:56.03 | dijib | there we go fixed it |
| 06:59.27 | dijib | that isnt picking up p3nguin |
| 06:59.51 | p3nguin | My conf isn't? |
| 07:00.54 | dijib | nope |
| 07:01.44 | p3nguin | I'm sitting on it, so I know it's available. |
| 07:02.29 | dijib | ive taken out eveything after the ,30) |
| 07:02.37 | dijib | looks like that |
| 07:02.43 | dijib | and died when called |
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| 10:59.43 | *** join/#asterisk ghost75 (~trechber@dslb-088-064-220-074.pools.arcor-ip.net) |
| 11:00.37 | ghost75 | if i assign arguments in agi script, am i supposed to use same variables as asterisk uses or should it be a different name? |
| 11:01.11 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
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| 11:42.51 | ghost75 | how i get variable from agi perl script, someone know? |
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| 12:04.01 | bratner | Hi! A customer asks for "Dual DTMF" support. What is the meaning of "dual" in this context? |
| 12:07.06 | unicron | it is redundant; DTFM = dual tone, multi frequency |
| 12:07.25 | unicron | MF* |
| 12:07.25 | WIMPy | qtmf? |
| 12:08.45 | unicron | my guess is that your customer is ether asking whether she or he can use DTMF to communicate with your system, or whether your system will allow him or her to send DTMF to other systems, through yours, after the call is placed |
| 12:09.16 | unicron | in other words, can your customer dial using the phone's touchpad? |
| 12:40.22 | bratner | unicron, sure my system supports DTMF , both rfc2833 and SIP INFO, the thing is i have no idea what "dual" means. Maybe the customer is refering to the ability to dial an extension and the receive DTMF from the callee... that is still a guess.. |
| 12:48.28 | unicron | or maybe the user wants to dial a bank or a conference line |
| 12:48.50 | unicron | for that to be reliable then you would need to use dtmf inband, and that would require an uncompressed codec like ulaw |
| 13:13.21 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 13:28.19 | *** join/#asterisk ghost75 (~trechber@dslb-088-064-220-074.pools.arcor-ip.net) |
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| 13:53.26 | slav3_kitten | so question. if i'm doing north american and international, should i also allow alaw codec? |
| 14:02.12 | ghost75 | alaw is europe |
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| 14:10.59 | slav3_kitten | so in short, yes? |
| 14:14.55 | ghost75 | i would |
| 14:15.09 | ghost75 | myself i use g729 now |
| 14:15.20 | slav3_kitten | bandwidth concerns? |
| 14:15.36 | ghost75 | yes, 384kbit speed :< |
| 14:15.42 | slav3_kitten | ouch |
| 14:15.50 | ghost75 | shared with torrent and apache |
| 14:16.14 | slav3_kitten | i got 1.5 meg up and there are 4 people in the house so the worst it could be is 8 channels of ulaw or alaw |
| 14:16.24 | ghost75 | i already use agi scripts to throttle down during calls |
| 14:19.33 | slav3_kitten | *nods* |
| 14:19.57 | ghost75 | and apache gets into maintenance mode oO |
| 14:20.55 | ghost75 | nobody uses perl in agi ? |
| 14:21.10 | slav3_kitten | shrugs |
| 14:21.19 | slav3_kitten | agi's are beyond my scope of understanding atm |
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| 14:21.49 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 14:22.16 | ghost75 | is just a script with returning values |
| 14:25.42 | kaldemar | not quite. more like anything that reads stdin and writes to stdout. |
| 14:25.59 | slav3_kitten | ghost75, i know that, but it's past what i know how to use or what to use it for |
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| 14:29.02 | ghost75 | kaldemar: how i get variable from it? |
| 14:29.33 | ghost75 | do i need to print variables to console then? |
| 14:30.59 | WIMPy | setvar |
| 14:34.03 | ghost75 | perl or bash? |
| 14:35.07 | ghost75 | $session->setvar("bla", $bla) |
| 14:36.33 | ghost75 | but i already have some value in $bla |
| 14:37.40 | WIMPy | However you have to do it with whatever you use. |
| 14:37.49 | WIMPy | I can only tell about AGI itself. |
| 14:38.02 | mathis_ | boo fucking ya. wideband modification of app_jack nearly done |
| 14:41.46 | kaldemar | ghost75: what variable? |
| 14:42.07 | ghost75 | the variable which is stored in perl |
| 14:42.15 | kaldemar | ghost75: get in where? |
| 14:42.19 | ghost75 | in asterisk |
| 14:42.27 | kaldemar | with setvar. |
| 14:42.34 | ghost75 | hmm how |
| 14:43.34 | kaldemar | no idea what your setvar($$) does. in plain AGI it would be to write "set variable var value" into stdout. |
| 14:44.18 | kaldemar | and then read the response from stdin. |
| 14:44.45 | ghost75 | why set when the perl variable already contains what i need |
| 14:45.13 | kaldemar | you just said you wanted the value in dialplan. make up your mind. |
| 14:45.40 | ghost75 | the syntax would be $session->setvar("bla", $bla) |
| 14:46.12 | kaldemar | if you mean just printing stuff in CLI, you can use "exec" and then print what you want with a NoOp or Verbose. |
| 14:46.31 | *** join/#asterisk elico (~Thunderbi@bzq-79-181-215-110.red.bezeqint.net) |
| 14:46.34 | kaldemar | ghost75: still, i have no idea what your setvar function is or does, so i can't say. |
| 14:47.43 | kaldemar | if you want to print into CLI while the agi is still executing use exec. if you want to set a variable that your dialplan prints after the agi, use set variable. |
| 14:48.26 | ghost75 | http://pastebin.com/1fcP2YzK |
| 14:48.27 | ghost75 | http://pastebin.com/08U0xzQd |
| 14:49.02 | ghost75 | reversecheck.agi is the perl above |
| 14:50.23 | kaldemar | that's not a proper agi in any way. |
| 14:50.52 | ghost75 | its normal perl :) |
| 14:51.08 | kaldemar | yes, but not agi. |
| 14:51.20 | SeRi | good morning all |
| 14:51.26 | ghost75 | what i need to change there |
| 14:51.31 | kaldemar | everything. |
| 14:51.43 | ghost75 | u are kidding |
| 14:51.49 | kaldemar | it doesn't do anything that an agi script would be required to do. |
| 14:52.40 | kaldemar | you should first read the arguments given to app AGI from stdin, they are not set to ARGV. |
| 14:53.37 | kaldemar | and you can't just print what ever crap to asterisk like that, you need to conform to the interface. "agi show commands" |
| 14:54.16 | *** join/#asterisk fritz09 (~Adium@pop1-224.catv.wtnet.de) |
| 14:55.12 | ghost75 | set variable |
| 14:55.21 | ghost75 | on stdout |
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| 14:56.58 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 14:57.57 | slav3_kitten | ok question: outbound calling. if the caller hangs up the callee continues to ring until they answer which then it's disconnected |
| 14:58.12 | kaldemar | ghost75: print "set variable varname" . $perlvarwithvalue; |
| 14:58.29 | slav3_kitten | my console shows a warning about sip retransmission timeout. what am i doing wrong or is that just how pstn terminated sip works? |
| 14:58.30 | ghost75 | ok |
| 14:58.31 | kaldemar | and then read stdin for what asterisk has to say. |
| 14:58.46 | kaldemar | anyway, why are you even using AGI for this? you could use func SHELL as well. |
| 14:59.52 | ghost75 | shell from where |
| 15:00.39 | kaldemar | asterisk has functions in dialplan. one of them is called SHELL. "core show function SHELL" |
| 15:01.29 | kaldemar | doesn't seem like you're doing anything that would require the use of AGI. |
| 15:01.55 | ghost75 | how i get the output with SHELL |
| 15:03.05 | kaldemar | did you read what "core show function SHELL" says? |
| 15:03.18 | ghost75 | it delivers only "result" |
| 15:03.33 | kaldemar | Example: 'Set(foo=${SHELL(echo \bar\)})' |
| 15:05.02 | SeRi | slav3_kitten: something is wrong |
| 15:05.03 | ghost75 | if i store everything in one it could work |
| 15:05.46 | slav3_kitten | SeRi, what could be wrong an how do i fix it |
| 15:06.13 | SeRi | slav3_kitten: most of the time sip retrasmission is due to your itsp going off line or your system not been able to reach your itsp. |
| 15:07.02 | SeRi | and when I mean going off line I mean you not been able to reach them for what ever reason |
| 15:07.10 | slav3_kitten | SeRi, i've got a Hangup after the Dial in my outbound |
| 15:08.56 | SeRi | slav3_kitten: core set verbose 3/sip set debug on |
| 15:09.06 | SeRi | pastebin the output |
| 15:09.21 | slav3_kitten | http://pastie.org/5396769 |
| 15:09.23 | SeRi | and of course replicate the issue |
| 15:09.32 | slav3_kitten | that's my outbound |
| 15:09.45 | SeRi | I dont need that |
| 15:09.50 | SeRi | I need to see your error |
| 15:10.01 | slav3_kitten | i'm workign on that figured i'd post it all |
| 15:10.08 | SeRi | ok |
| 15:10.55 | SeRi | slav3_kitten: who is your ITSP? |
| 15:11.11 | slav3_kitten | flowroute |
| 15:11.16 | SeRi | setting your cid to block can be an issue depending on your itsp |
| 15:12.43 | slav3_kitten | one second, looking for where that's all logged to |
| 15:14.05 | slav3_kitten | my asterisk debug skills are kinda non existent |
| 15:17.01 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 15:18.21 | SeRi | slav3_kitten: slav3_kitten is logged to the cli |
| 15:18.25 | SeRi | not a file |
| 15:18.34 | *** join/#asterisk gusto (~gusto@2001:a60:11ff:1200::42:4) |
| 15:18.56 | SeRi | you have to copy and paste on cli |
| 15:19.17 | slav3_kitten | SeRi, which i can't seem to pageup and it's floods cli, thus i made a verbose logger file |
| 15:20.34 | slav3_kitten | it looks like i have something screwed up with nat |
| 15:22.27 | SeRi | show me |
| 15:23.40 | slav3_kitten | give me a couple minutes to try an fix it on my own? |
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| 15:23.50 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 15:24.13 | slav3_kitten | it's got my isp's natted wan ip in the sip packets |
| 15:26.51 | SeRi | slav3_kitten: You ask for help not me. so dont ask for help if you dont need it ;) |
| 15:27.01 | SeRi | and yeas that would cause an issue. |
| 15:27.35 | slav3_kitten | SeRi, i did ask for help, you said "hey dumbass turn on sip debug" |
| 15:27.39 | slav3_kitten | and i saw a blatant problem |
| 15:28.40 | *** part/#asterisk mjordan (~mjordan@user-24-214-136-35.knology.net) |
| 15:28.54 | SeRi | slav3_kitten: my console shows a warning about sip retransmission timeout. what am i doing wrong or is that just how pstn Dibbler |
| 15:28.57 | SeRi | <PROTECTED> |
| 15:30.10 | slav3_kitten | searches through the book for sip an nat |
| 15:30.22 | SeRi | and I never said that. in any case good luck getting help you aint getting shit from me. you and your mental issues. |
| 15:32.25 | slav3_kitten | wait what? i forgot about sip debug i feel like a dumbass. it was more a lol garnering response not any feeling that you at all insulted me |
| 15:33.05 | slav3_kitten | i was trying to get a laugh man. i'm just trying to work this through myself as that's how i best learn. |
| 15:33.22 | slav3_kitten | if i can't figure it out in a few minutes i was going to ask for some more assistance |
| 15:39.03 | ghost75 | kaldemar: its working now |
| 15:39.21 | ghost75 | print "SET VARIABLE RESULTOERT \"" . $linkoert->as_trimmed_text . "\""; |
| 15:43.05 | *** join/#asterisk enzo (~jerome@89-93-205-195.hfc.dyn.abo.bbox.fr) |
| 15:43.08 | enzo | hello |
| 15:44.20 | enzo | I have a strange thing, I put a script in: /usr/share/asterisk/agi-bin/src-rh.agi, I restart asterisk, but the logs say: WARNING[20611]: pbx.c:4218 pbx_extension_helper: No application 'agi,src-rh.agi' for extension (Incoming, s, 5) |
| 15:44.39 | enzo | Where is the official dir to put some agi scripts ? |
| 15:52.39 | slav3_kitten | SeRi, i added externip and nat=yes to my sip.conf in the flowroute section an thought it'd fix it. still is a no go, can i show you my verbose output an get some help? |
| 15:53.43 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
| 15:53.49 | SeRi | slav3_kitten: I am a bit bussy right now. |
| 15:54.22 | slav3_kitten | SeRi, whenever you get a chance. i honestly meant no offense earlier and was just trying to get a laugh with the dumbass comment. seriously thought i had it figured out though |
| 15:55.03 | enzo | well syntax is now agi(...) instead of s,5,agi,xxx.agi |
| 15:59.26 | *** join/#asterisk Mango45 (~Mango45@d209-89-214-26.abhsia.telus.net) |
| 16:00.10 | Mango45 | How do I view messages in the CLI like this: |
| 16:00.17 | Mango45 | Call from '200' to extension '9011972599111142' rejected because extension not found. |
| 16:00.30 | Mango45 | All I see is "Using SIP RTP CoS mark 5 " |
| 16:05.38 | *** join/#asterisk SeRi (~wtf@pdpc/supporter/professional/seri) |
| 16:05.41 | SeRi | ok back |
| 16:05.55 | SeRi | still not working :/ |
| 16:05.57 | SeRi | damn |
| 16:05.59 | SeRi | brb |
| 16:07.13 | *** join/#asterisk SeRi (~wtf@pdpc/supporter/professional/seri) |
| 16:07.25 | slav3_kitten | wb SeRi, what ya working on that's not going as planned? |
| 16:07.26 | SeRi | ok now working |
| 16:08.06 | SeRi | slav3_kitten: dbus notification |
| 16:08.17 | SeRi | It turn out a stale terminal stuck on dbus |
| 16:08.19 | mathis_ | soo |
| 16:08.25 | slav3_kitten | ah |
| 16:08.26 | SeRi | one sec going to do one more test |
| 16:08.30 | SeRi | brb |
| 16:08.59 | mathis_ | say I modified an asterisk app to be much more awesome (app_jack can now be used with a wideband channel) |
| 16:09.17 | mathis_ | what would be the best way to get the patch into the asterisk tree? ;) |
| 16:09.19 | *** join/#asterisk bmg505 (~leon@196.209.99.11) |
| 16:09.45 | enzo | in order to code agi script in php, which php class would you advise ? |
| 16:13.31 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 16:15.19 | slav3_kitten | feels like he's hitting his head against a wall |
| 16:16.45 | Mango45 | Hi, [TK]D-Fender |
| 16:17.14 | *** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 16:19.54 | slav3_kitten | yep, still not working |
| 16:20.09 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 16:23.32 | *** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com) |
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| 16:48.31 | slav3_kitten | anyone use flowroute with static IP? |
| 16:50.15 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 16:52.57 | slav3_kitten | nevermind got it working |
| 16:56.07 | *** join/#asterisk timahvo1 (~rogue@41.212.120.182) |
| 17:13.10 | sanman | is there a way to clear the CLI history |
| 17:15.39 | *** join/#asterisk TimeRider (~steve@188-220-34-144.dsl.cnl.uk.net) |
| 17:20.03 | *** join/#asterisk wdoekes2 (~walter@wjd.osso.nl) |
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| 17:28.45 | *** join/#asterisk SeRi (~wtf@pdpc/supporter/professional/seri) |
| 17:32.54 | file | mathis_, any patches need to be submitted to http://issues.asterisk.org/jira with a valid license agreement signed |
| 17:33.35 | mathis_ | file: okay, thanks! I was just about to create an account there |
| 17:33.48 | mathis_ | file: did send it to asterisk-dev in the meantime though |
| 17:33.57 | file | I saw, I can't look at it though |
| 17:34.19 | file | prefers to keep away from any potential legalness, as do others |
| 17:34.44 | file | https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines there's a handy dandy "Patch and Code submission" part there |
| 17:34.55 | mathis_ | okay |
| 17:35.13 | mathis_ | well, this patch will not cause any legal/license troubles :) |
| 17:35.40 | mathis_ | it just replaces some hard coded values with changeable ones |
| 17:35.56 | slav3_kitten | wb SeRi |
| 17:38.16 | SeRi | thanks |
| 17:38.18 | SeRi | lol |
| 17:38.34 | slav3_kitten | pm? |
| 17:39.41 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 17:39.41 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 17:39.43 | SeRi | sure |
| 17:43.23 | *** join/#asterisk Alex_Bkash (b4ea1b97@gateway/web/freenode/ip.180.234.27.151) |
| 17:43.41 | Mango45 | How do I view messages in the CLI when someone with a bad username/password attempts to route a call via my PBX? |
| 17:43.54 | Mango45 | At the moment I see: == Using SIP RTP CoS mark 5 |
| 17:44.09 | Mango45 | but I would like to see something with more useful details. |
| 17:44.52 | [TK]D-Fender | "sip set debug on" |
| 17:45.01 | [TK]D-Fender | If you aren't loking with that on... you aren't really looking... |
| 17:45.38 | ghost75 | are there filters while debugging? |
| 17:46.47 | [TK]D-Fender | you can select by peer/ip |
| 17:46.51 | [TK]D-Fender | read the CLI help |
| 17:47.07 | Mango45 | [TK]D-Fender: I've turned that on, but it displays every single sip packet, 95% of which is good traffic. |
| 17:47.13 | Mango45 | and it makes the log files huge. |
| 17:47.27 | Mango45 | I was thinking of something like "Call from '200' to extension '9011972599111142' rejected because extension not found." |
| 17:47.49 | [TK]D-Fender | no |
| 17:47.50 | ghost75 | why you worry someone using your pbx |
| 17:49.20 | Mango45 | [TK]D-Fender: No? |
| 17:52.04 | ghost75 | normaly it should be impossible to do that if sip.conf is safe no? |
| 17:52.21 | [TK]D-Fender | I told you can can select by peer/ip. That is all. |
| 17:52.25 | [TK]D-Fender | no other filtering |
| 17:54.37 | Mango45 | How do I get the message I pasted to appear? |
| 17:55.06 | ghost75 | not at all |
| 17:55.14 | WIMPy | By having someone call an extension he can;t call. |
| 17:55.42 | Mango45 | I've tried that. The message I see is: == Using SIP RTP CoS mark 5 |
| 17:56.36 | WIMPy | If you see that, you will se it on every call. |
| 17:57.58 | cusco | Mango45: messages like those come up in /var/log/asterisk/messages using the default commented on logger.conf |
| 18:02.39 | Mango45 | cusco: Unfortunately they don't. |
| 18:03.30 | ghost75 | syslog.local0 => notice,warning,error |
| 18:04.28 | ghost75 | whats that for |
| 18:06.13 | ghost75 | my itsp gives me a lot of those: <--- SIP read from UDP:88.79.152.249:5060 ---> |
| 18:06.13 | ghost75 | SIP/2.0 403 Forbidden |
| 18:07.04 | ghost75 | is that because of qualify=yes ? |
| 18:09.27 | cusco | Mango45: they do here |
| 18:09.55 | cusco | I also have alwaysauthreject=yes and match_auth_username=yes in sip.conf |
| 18:10.35 | Mango45 | Maybe I should just upgrade to 11.0.1 and see if that "solves the problem. |
| 18:12.19 | *** join/#asterisk Neptu (~Neptu@c213-89-2-159.bredband.comhem.se) |
| 18:14.44 | *** join/#asterisk Alex_Bkash (b4ea25d5@gateway/web/freenode/ip.180.234.37.213) |
| 18:15.56 | Alex_Bkash | i've installed asterisk on my usb linuxmint. but asterisk wont run while i boot usb on another pc. How can i overcome this? |
| 18:18.57 | *** join/#asterisk tompaw (~tompaw@tompaw.xxx) |
| 18:19.04 | *** join/#asterisk Alex_Bkash (b4ea25d5@gateway/web/freenode/ip.180.234.37.213) |
| 18:19.19 | *** join/#asterisk [TK]D-Fender (~chatzilla@70.24.182.120) |
| 18:19.34 | tompaw | Hello. |
| 18:21.21 | Alex_Bkash | I've installed asterisk on my usb linuxmint. but asterisk wont start while i boot usb from another PC. is thr any way i can overcome this? |
| 18:21.57 | tompaw | OK guys, shit stopped being funny. I reinstalled everything on asterisk box and the problem still persists. Whe the hell would AMI buffer overflow via the network? http://pastebin.ca/2252749 |
| 18:22.20 | *** join/#asterisk imox (~imox@91-66-32-57-dynip.superkabel.de) |
| 18:24.17 | tompaw | ectospasm: your idea of using ncat proved to be correct, but I won't run production via ncat simply because Mr. Asterisk has problems handling Internet. |
| 18:26.39 | mathis_ | mhm |
| 18:27.15 | mathis_ | file: patches should be submitted as Improvement issues, right? |
| 18:28.09 | WIMPy | tompaw: Have you already opened an issue on jira? |
| 18:29.19 | tompaw | WIMPy: I don't even know what to report here, there's too many variables. |
| 18:29.54 | tompaw | Besides ectospasm tried this on his asterisk and it worked fine :/ |
| 18:30.48 | *** join/#asterisk vlad_starkov (~vlad_star@83.149.9.228) |
| 18:32.46 | SeRi | Mango45: have you try verbos on your dialplan context? |
| 18:33.03 | SeRi | s/verbos/verbose/ |
| 18:34.50 | SeRi | Mango45: same => n,Verbose |
| 18:34.55 | dijib | SeRi: how are you awake already? |
| 18:35.07 | SeRi | dijib: lol. |
| 18:35.11 | SeRi | sense 7AM |
| 18:35.18 | dijib | since |
| 18:35.23 | dijib | ouch.. i woke up at 1pm |
| 18:35.44 | dijib | eating my bowl of cerial an fruit right now |
| 18:36.34 | jpsharp | cereal killer! |
| 18:36.47 | SeRi | Mango45: maybe this can help... http://leifmadsen.wordpress.com/2011/03/04/debugging-the-asterisk-dialplan-with-verbose/ |
| 18:37.23 | dijib | im in conference |
| 18:37.36 | SeRi | kill him! |
| 18:37.39 | SeRi | lol |
| 18:37.42 | SeRi | ill jump in a bit |
| 18:37.44 | SeRi | lunch time |
| 18:38.01 | dijib | lol |
| 18:38.11 | dijib | cereal killer = hackers |
| 18:39.37 | Mango45 | SeRi: Thanks. |
| 18:42.21 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 18:44.37 | SeRi | Mango45: did you posted @dslr? |
| 18:44.51 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 18:45.04 | SeRi | dijib: ^^ |
| 18:45.06 | Mango45 | Yes, that was me. |
| 18:45.26 | SeRi | ok cool. glad to see you around here. |
| 18:45.38 | Mango45 | What's your name on DSLR? :) |
| 18:46.05 | SeRi | I am very hated there... :P |
| 18:46.10 | SeRi | XCOM |
| 18:46.12 | Mango45 | lol |
| 18:48.03 | SeRi | Mango45: I thought you were a freeswitch user/admin |
| 18:48.09 | SeRi | I thought wrong :) |
| 18:50.01 | Mango45 | I've always wanted to learn FS, but never got around to it. |
| 18:50.12 | Mango45 | Asterisk does everything I need, so the incentive isn't really there. |
| 18:50.24 | *** join/#asterisk felipealmeida (~user@mvx-187-16-79-187.mundivox.com) |
| 18:50.24 | SeRi | ah. |
| 18:52.09 | ghost75 | A new option, match_auth_username in sip.conf changes the matching of incoming requests. If set, and the incoming request carries authentication info, the username to match in the users list is taken from the Digest header rather than from the From: field. |
| 18:52.14 | ghost75 | this option makes any sense? |
| 18:52.34 | cusco | you should ask yourself that |
| 18:52.50 | dijib | http://pastebin.com/0AGPYWVn |
| 18:53.03 | ghost75 | i dont know how authentication works in detail |
| 18:53.09 | [TK]D-Fender | Does it work? |
| 18:57.17 | slav3_kitten | hey dijib ... i'm having that issue where if i hang up an in progress call it continues to attempt the call until the dialed party answers in one form or another |
| 18:57.21 | *** join/#asterisk dfgas-cr48 (~user@71-90-33-37.dhcp.ftbg.wi.charter.com) |
| 18:58.03 | dfgas-cr48 | dijib and p3nguin, are you guys here |
| 18:58.13 | dfgas-cr48 | good news and bad news |
| 18:58.30 | *** join/#asterisk OneNarrowWay (~OneNarrow@ip4da1344b.direct-adsl.nl) |
| 18:59.06 | slav3_kitten | which is dfgas-cr48 ? |
| 18:59.52 | mathis_ | eh wtf |
| 19:00.03 | dfgas-cr48 | my carrier is working again, i can do echo text on voip.ms but now my asterisk server is not getting call yet |
| 19:00.05 | mathis_ | do issues just get deleted?! |
| 19:01.31 | mathis_ | ah noes. it just got moved to an area I am not allowed to see -.- |
| 19:03.52 | dijib | dfgas-cr48: i just got in |
| 19:05.13 | kaldemar | ~ask |
| 19:05.13 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 19:05.18 | kaldemar | dfgas-cr48: ^ |
| 19:10.51 | dijib | http://sphotos-b.ak.fbcdn.net/hphotos-ak-ash3/561518_302571386524871_1063140192_n.jpg |
| 19:12.42 | slav3_kitten | pokes dijib |
| 19:13.15 | *** join/#asterisk dfgas-cr48 (~user@71-90-33-37.dhcp.ftbg.wi.charter.com) |
| 19:13.33 | slav3_kitten | SeRi, said you were helping someone with an issue when using their itsp to dial out and they hang up their local set while it's ringing. it will continue to ring the callee until they pick up |
| 19:14.36 | dfgas-cr48 | dijib, |
| 19:14.40 | dfgas-cr48 | from my cell it doesn't work |
| 19:14.44 | dfgas-cr48 | wife gets busy too |
| 19:14.48 | dfgas-cr48 | but when i turn voip.ms to echotest that works |
| 19:16.12 | datarecall | is it possible to set you pbx up to send sms messages |
| 19:17.32 | sanman | how do i fix this issuer |
| 19:17.34 | sanman | configure: *** configure: *** The IMAP_TK installation appears to be missing or broken. configure: *** Either correct the installation, or run configure configure: *** including --without-imap. |
| 19:19.30 | *** join/#asterisk fritz09 (~Adium@pop1-224.catv.wtnet.de) |
| 19:20.06 | mathis_ | install the appropriate dev package |
| 19:20.17 | mathis_ | or run configure with --without-imap |
| 19:20.18 | mathis_ | :) |
| 19:22.10 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 19:24.53 | *** join/#asterisk vlad_starkov (~vlad_star@83.149.9.228) |
| 19:25.04 | jpsharp | datarecall: If you have either a GSM card in your Asterisk box or if you have a late enough version of Asterisk to support SIP messaging *and* an ITSP that supports sending SMS over SIP. |
| 19:27.09 | SeRi | http://www.voipsupply.com/manufacturer/audiocodes |
| 19:33.00 | datarecall | ok my DID is SMS enabled jpsharp how would I check if my asterisk is able to do it |
| 19:39.37 | *** join/#asterisk Neptu (~Neptu@c213-89-2-159.bredband.comhem.se) |
| 19:43.47 | *** join/#asterisk jblack_ (~jblack@pool-108-57-61-167.sctnpa.east.verizon.net) |
| 19:50.42 | jpsharp | datarecall: What version of Asterisk are you running? |
| 19:52.00 | datarecall | 2.10.0.1 |
| 19:52.21 | datarecall | Asterisk 1.6.2.4, |
| 19:52.47 | jpsharp | I don't think it is available that far back. |
| 19:53.11 | slav3_kitten | uh datarecall you do know asterisk 1.6 is older then dirt right? 11 is the current |
| 19:53.51 | jpsharp | slav3_kitten: I think he's running FreeBSD |
| 19:53.54 | jpsharp | er, FreePBX. |
| 19:53.59 | slav3_kitten | ah |
| 19:56.04 | jpsharp | Yep. SendText support for SIP is only available in Asterisk 10+. |
| 19:56.33 | jpsharp | Or rather, it would be MessageSend for SIP. |
| 19:59.09 | jpsharp | Now I wonder if I can send an SMS via GoogleTalk using Asterisk and MessageSend. |
| 20:00.36 | slav3_kitten | jpsharp, if you succeed at that, i'd love see how you do it |
| 20:01.03 | *** join/#asterisk BrettB (~Brett@75.92.223.35) |
| 20:01.13 | datarecall | yeah im running freepbx |
| 20:01.17 | slav3_kitten | i know a guy who was asking me about sending texts |
| 20:02.04 | slav3_kitten | datarecall, i highly recommend you switch to something using a modern version |
| 20:02.44 | datarecall | slav3_kitten is there a good gui using latest asterisk ? |
| 20:02.48 | datarecall | ive only ever used freepbx |
| 20:04.09 | slav3_kitten | uh that's a good question i can't answer. honestly configuring it all in vi is pretty easy |
| 20:04.39 | slav3_kitten | but yea the version of asterisk you're using is from feb of 2010 |
| 20:05.12 | p3nguin | 1.8 is also current |
| 20:05.18 | p3nguin | 10 is also current |
| 20:05.32 | [TK]D-Fender | datarecall: No GUI out here cares about SMS at all. You're going to have to deal with that aspect by hand yourself |
| 20:06.06 | slav3_kitten | what version is asterisk now running? |
| 20:10.43 | dijib | p3nguin: good day sir. |
| 20:11.00 | dijib | [TK]D-Fender: did i hear you a fellow canuck |
| 20:11.31 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 20:11.31 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 20:12.14 | *** join/#asterisk tonikasch (~tonikasch@unaffiliated/tonikasch) |
| 20:14.53 | datarecall | is there a repo for centos 6 for asterisk |
| 20:15.04 | [TK]D-Fender | dijib: Correct |
| 20:15.10 | dijib | QC? |
| 20:27.04 | jpsharp | datarecall: Building Asterisk from source isn't that hard. |
| 20:28.17 | ghost75 | i will update soon to 1.8 |
| 20:29.02 | ghost75 | its in apt |
| 20:30.08 | Nivex | it's not that hard, but I'm also lazy |
| 20:30.30 | ghost75 | why they have still astcanary |
| 20:34.23 | datarecall | just upgraded asterisk now all my files can not be found : WARNING[2190]: file.c:962 ast_streamfile: Unable to open vm-goodbye (format 0x4 (ulaw)): No such file or directory |
| 20:37.20 | slav3_kitten | datarecall, you went from version 1.6 to 11? |
| 20:37.40 | datarecall | 1.6 ->1.8 |
| 20:37.55 | slav3_kitten | that's likely going to break everything |
| 20:38.25 | datarecall | just seems that it cant find the sound files |
| 20:38.57 | jpsharp | they should be in /var/lib/asterisk/sounds/ |
| 20:39.03 | jpsharp | If not, you'll need to reinstall them. |
| 20:39.06 | slav3_kitten | /var/lib/asterisk/sounds/en/ |
| 20:39.18 | jpsharp | That too. |
| 20:39.21 | datarecall | hmm yeah all the sound files are there |
| 20:39.46 | jpsharp | And, alas, I cannot seem to use MessageSend along with GV to send SMS. |
| 20:40.01 | slav3_kitten | aww |
| 20:40.32 | datarecall | ok so if the sound files are all in /var/lib/asterisk/sounds wonder why 1.8 can't find them |
| 20:40.43 | slav3_kitten | correct me if i'm wrong, but isn't freepbx basically made up of lots of scripts an such that are pretty much broken by doing major version changes of asterisk? |
| 20:41.16 | ChannelZ | more or less |
| 20:41.35 | slav3_kitten | datarecall, incorrect permissions on the asterisk crap would be my guess |
| 20:43.51 | datarecall | permissions look good : http://dl.dropbox.com/u/2397195/2012-11-18_1343.png |
| 20:45.42 | slav3_kitten | datarecall A) i see a lot of links to files an not actual files. B) i see everything in sln an not ulaw like was requested |
| 20:46.54 | datarecall | slav3_kitten can you just install the sound files ? |
| 20:47.10 | *** join/#asterisk dfgas-cr48 (~user@71-90-33-37.dhcp.ftbg.wi.charter.com) |
| 20:47.20 | slav3_kitten | download the sound files from asterisk.org and put them into that directory |
| 20:48.39 | dfgas-cr48 | SeRi, yo |
| 20:52.58 | ghost75 | /usr/share/asterisk/sounds i have |
| 20:59.12 | *** join/#asterisk Brokedcomputer (~ce@184.71.199.202) |
| 21:00.35 | Brokedcomputer | Hey Guys |
| 21:04.07 | jpsharp | Greetings, program |
| 21:04.33 | Brokedcomputer | Greetings |
| 21:06.37 | ghost75 | greetings infidel :) |
| 21:08.29 | Brokedcomputer | :P |
| 21:10.21 | datarecall | which gui would you guy's reccomend using elastix, asterisknox, freepbx ? gonna redo this install i guess |
| 21:11.22 | jpsharp | None of the above. GUIs just limit you, force you to use very specific installation versions, and are just a general all around pain in the processor. |
| 21:11.54 | datarecall | jpsharp dealing with asterisk conf's is out of my league. If you had to use one which would you use :) |
| 21:12.49 | *** join/#asterisk k610 (~Instantbi@host-78-129-3-116.brutele.be) |
| 21:12.55 | jpsharp | I cannot recommend one over another. I don't use them, so couldn't judge them on their merits. |
| 21:15.08 | Brokedcomputer | jpsharp is a master of conf configuration, you should just ask what you need :P |
| 21:15.24 | ghost75 | gui is for users, not for admins |
| 21:15.45 | datarecall | file.c:667 ast_openstream_full: File vm-goodbye does not exist in any format i downloaded the sound files but still getting these errors |
| 21:16.24 | ghost75 | try with including pathname in dialplan |
| 21:17.14 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 21:18.34 | datarecall | http://paste2.org/p/2492619 is the log of whats going on when i dial a number |
| 21:19.20 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 21:20.19 | jpsharp | There's no /var/lib/asterisk/sounds/custom/cred_intro.something |
| 21:20.53 | jpsharp | or /var/lib/asterisk/sounds/en/custom/cred_intro.something. |
| 21:21.16 | datarecall | but it is there: http://dl.dropbox.com/u/2397195/2012-11-18_1420.png |
| 21:21.45 | ghost75 | lol |
| 21:21.51 | ghost75 | its looking for gsm not wav |
| 21:22.15 | datarecall | 1.6 played those fine though |
| 21:23.07 | kaldemar | the sound directory structure has changed. |
| 21:23.31 | slav3_kitten | datarecall, you've upgraded one major version. there were drastic changes between 1.6 an 1.8 from what i read in the book |
| 21:23.37 | kaldemar | sounds/<language>/custom would be the right place. |
| 21:24.15 | *** join/#asterisk fritz09 (~Adium@pop1-224.catv.wtnet.de) |
| 21:25.01 | *** join/#asterisk dfgas-cr48 (~user@71-90-33-37.dhcp.ftbg.wi.charter.com) |
| 21:25.07 | datarecall | slav3_kitten would you just recommend doing a fresh system them starting from scratch ? |
| 21:25.34 | datarecall | ghost75 i converted those files to gsm and it still cant find them |
| 21:25.36 | kaldemar | datarecall: not needed. just put your sound files in the right directory. |
| 21:25.52 | ghost75 | try full path |
| 21:26.35 | slav3_kitten | ghost75, he's upgraded his free PBX from version 1.6 of astersik to version 1.8 |
| 21:27.42 | ghost75 | and? |
| 21:28.01 | p3nguin | Doesn't matter if it is the full path or not. Asterisk is already looking for the file in the path where it is supposed to look. Relative paths are what it uses. |
| 21:28.26 | datarecall | ghost what do you mean by full path ? |
| 21:28.34 | Brokedcomputer | Does anyone know why fuser is loading my processor with requests? (Sorry I know its not asterisk related) |
| 21:28.59 | slav3_kitten | ghost75, and i can't imagine freebpx is happy with the upgrade |
| 21:30.24 | datarecall | <PROTECTED> |
| 21:30.31 | jaytee | I'm having an issue with inbound calls from my sip provider, Flowroute, where I get two way audio for about 3 to 5 seconds and then the call drops due to retransmission timeouts. Here is a sip debug of the call, http://www.pastebin.com/Gmk6MEbh |
| 21:33.20 | ghost75 | do: find -name *.gsm |
| 21:38.00 | ghost75 | or move files to /usr/share/asterisk/... |
| 21:38.24 | *** join/#asterisk trumee (~parul@93-96-159-40.zone4.bethere.co.uk) |
| 21:39.10 | datarecall | ghost75 that fixed it |
| 21:40.27 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 21:41.23 | datarecall | now I see this error WARNING[3071]: pbx.c:4235 pbx_extension_helper: No application 'VoiceMail' for extension (macro-vm, s-NOANSWER, 2) |
| 21:44.46 | ghost75 | i think is called voicemailmain |
| 21:48.09 | datarecall | and this one here WARNING[3171]: res_agi.c:1621 launch_script: Failed to execute '/usr/share/asterisk/agi-bin/dialparties.agi': File does not exist. |
| 21:49.01 | *** join/#asterisk Neptu (~Neptu@c213-89-2-159.bredband.comhem.se) |
| 21:49.32 | ghost75 | shouldnt be hard to find |
| 21:50.45 | datarecall | yeah i found it and moved it into the share directory |
| 21:51.38 | jpsharp | You're ending up with a Frankenbyte system. |
| 21:51.52 | datarecall | yeah it seems so |
| 21:51.57 | datarecall | think i might just start fresh |
| 21:56.13 | *** join/#asterisk vfabi (~fabi@host-static-89-41-121-42.moldtelecom.md) |
| 21:59.27 | p3nguin | dijib: Is that you fucking up Tony's system? |
| 22:00.19 | p3nguin | modules.conf has been deleted, chan_sip isn't loaded, etc. |
| 22:00.34 | dijib | no im not in there |
| 22:00.37 | dijib | are you in there? |
| 22:00.45 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 22:00.54 | dijib | i think he is rebuilding it line by line at current |
| 22:01.00 | dijib | good learning process |
| 22:01.18 | dijib | i dont have ssh creds |
| 22:01.21 | dijib | make me an account |
| 22:01.32 | dijib | share your un/pw |
| 22:02.36 | dijib | voipms came back to him and said its not their end |
| 22:02.52 | dijib | he said he was able to call the echo test setting on the did from his cellphone |
| 22:03.01 | p3nguin | I know. |
| 22:03.04 | dijib | he has switched it back to the main account |
| 22:03.07 | dijib | how do you know?!?!?! |
| 22:03.22 | p3nguin | I read the ticket and replied to them. |
| 22:03.24 | dijib | your an omnipresent god |
| 22:03.30 | dijib | lol |
| 22:03.33 | dijib | how you do that? |
| 22:03.41 | dijib | through your account? |
| 22:04.02 | dijib | ive changed remotesecret=password back to secret=password |
| 22:04.04 | p3nguin | I also tested echo test and routed calls. echo works, but the calls do not get to him when routed. I say it is a router issue. |
| 22:04.08 | p3nguin | That's wrong. |
| 22:04.17 | p3nguin | remotesecret is the correct setting. Like I told you. |
| 22:04.32 | dijib | yeah thats what were thinking but why then yesterday did it work |
| 22:04.38 | dijib | flatten the router??? |
| 22:04.38 | p3nguin | I really shouldn't have to tell you more than one time. |
| 22:04.47 | dijib | :/ |
| 22:05.13 | *** join/#asterisk [TK]D-Fender (~chatzilla@70.24.182.120) |
| 22:05.14 | p3nguin | Someone has totally fucked it up, now. The sip.conf is all jacked up. |
| 22:05.27 | p3nguin | So I'm out. I don't need to keep cleaning up after someone else. |
| 22:05.35 | dijib | its him |
| 22:05.37 | slav3_kitten | datarecall, throwing in the towel an building from scratch? |
| 22:05.46 | dijib | he is rewriting everything |
| 22:06.01 | dijib | starting out with a fresh sample configs |
| 22:06.02 | p3nguin | It didn't need to be rewritten. |
| 22:06.08 | dijib | <PROTECTED> |
| 22:06.09 | p3nguin | I personally checked it myself last night. It was fine. |
| 22:06.14 | dijib | i know. |
| 22:06.20 | dijib | thats what i told him... |
| 22:06.27 | p3nguin | Now he only has extensions.conf and sip.conf. Everything else is gone. |
| 22:06.32 | dijib | should i black list my conf? |
| 22:06.36 | dijib | did you fix your conf? |
| 22:06.45 | p3nguin | My conf wasn't broken. |
| 22:06.48 | dijib | he is copying back everything? |
| 22:06.50 | dijib | one by one. |
| 22:07.02 | dijib | my conference has been reconfigured also |
| 22:07.05 | p3nguin | He has a jacked up sip.conf on there now. |
| 22:07.10 | dijib | i cant see |
| 22:07.15 | dijib | share your creds |
| 22:07.20 | dijib | make me an acocunt |
| 22:07.46 | p3nguin | There was no reason to delete the sip.conf nor the extensions.conf. Both were 95% or better in my opinion. |
| 22:08.54 | p3nguin | Not up to me, though. The first time I looked, it was a freebie. Next time I have to fix it, it's going to cost someone. |
| 22:08.54 | dijib | he didnt delete he backed them up |
| 22:09.03 | dijib | :D |
| 22:09.11 | dijib | my fucking nightmare |
| 22:09.17 | dijib | need some dough |
| 22:09.55 | dijib | i need to campaign businesses... sell some asterisk, get some cashflow in this channel |
| 22:09.56 | p3nguin | I'll bet an hour's pay on it being the router at fault. |
| 22:10.09 | dijib | you dont want to know what i make an hour |
| 22:10.10 | dijib | lo |
| 22:10.12 | dijib | lol |
| 22:10.17 | p3nguin | Either a configuration in the thing or the router itself. |
| 22:10.43 | dijib | 3.36 hourly |
| 22:10.52 | dijib | it was working yesterday |
| 22:10.55 | dijib | breifly |
| 22:10.58 | dijib | and the night before |
| 22:11.21 | p3nguin | I'm putting part of my chips on his fixation with port triggering instead of port forwarding. |
| 22:11.24 | dijib | although thats not confirmed becuase i was justphoning through my asterisk... i thought he said he was able to call in though or a friend |
| 22:11.44 | dijib | i checked that... they seemed to be configured as port forwarding |
| 22:12.04 | dijib | there were two radio buttons and the triggering took you to a page with no config.. the forward = populated |
| 22:12.44 | dijib | he has upgraded to a new version of the firmware. |
| 22:13.08 | dijib | it has changed the whole gui and seri couldnt find the alg stuff now after |
| 22:13.14 | dijib | or maybe eventualyl did |
| 22:13.20 | dijib | remote management is on 8089 |
| 22:13.24 | dijib | for his router |
| 22:13.30 | dijib | you know the creds |
| 22:15.59 | *** join/#asterisk dfgas-cr48 (~user@71-90-33-37.dhcp.ftbg.wi.charter.com) |
| 22:16.47 | *** join/#asterisk dfgas-cr48 (~user@71-90-33-37.dhcp.ftbg.wi.charter.com) |
| 22:18.00 | dijib | dfgas-cr48: back in eh |
| 22:18.11 | dfgas-cr48 | hey |
| 22:18.15 | dfgas-cr48 | read your pm |
| 22:18.22 | dfgas-cr48 | found the issue kinda |
| 22:19.13 | *** join/#asterisk ghost75 (~trechber@dslb-088-064-220-074.pools.arcor-ip.net) |
| 22:21.25 | p3nguin | SIP ALG is under the WAN setup page. |
| 22:21.44 | p3nguin | I think I found the issue. |
| 22:22.13 | p3nguin | I bet the router restarts on this change. |
| 22:24.34 | p3nguin | I'm interested to know what he thinks the problem is now. |
| 22:25.59 | dijib | i think it did |
| 22:26.00 | dijib | lol |
| 22:26.10 | p3nguin | What did he think the problem is? |
| 22:26.19 | dijib | 17:25 -!- dfgas-cr48 [~user@71-90-33-37.dhcp.ftbg.wi.charter.com] has quit [Ping timeout: 244 seconds] |
| 22:28.48 | jaytee | figured out my dropped call issue. It was an issue with the router. |
| 22:29.15 | p3nguin | dijib: I got the calls making it to asterisk now. |
| 22:34.45 | dijib | yes he is reporting this to me now. |
| 22:35.54 | p3nguin | It works, now. |
| 22:36.36 | p3nguin | I just verified that he put back the sip.conf, which I had to edit back to the way it was last time I edited it, and fixed some of the extension stuff that was still out of whack (that I ignored before). |
| 22:38.26 | dijib | ok he says he saw his cell phones callerid when he tested it |
| 22:38.29 | p3nguin | It isn't "done," but calls get to where they are supposed to go. |
| 22:38.38 | dijib | but i just tried and its still congestion |
| 22:38.50 | dijib | to voipms-inbound |
| 22:38.55 | dijib | thank you p3nguin |
| 22:39.42 | p3nguin | I'm kind of wanting to revert that change that I think made the difference. |
| 22:40.40 | *** join/#asterisk dfgas-cr48 (~user@71-90-33-37.dhcp.ftbg.wi.charter.com) |
| 22:40.51 | dfgas-cr48 | p3nguin, yo |
| 22:40.56 | p3nguin | Yep. |
| 22:41.14 | dfgas-cr48 | so idk if you fixed it or if its what i tried |
| 22:41.19 | dfgas-cr48 | but calling in works |
| 22:41.29 | [sr] | going to sleep |
| 22:41.36 | p3nguin | What did you change that you think fixed it? |
| 22:41.49 | dfgas-cr48 | i moved all the config files out and created samples and put basic voip.ms in |
| 22:41.57 | dfgas-cr48 | and then it showed me calling in |
| 22:42.00 | dijib | its still not working for me |
| 22:42.20 | dfgas-cr48 | so i removed all those files and put in the config files you guys have been working on |
| 22:42.28 | dijib | you need to fix your confbridge.conf with something like this, and then fix the dialplan for confbridge |
| 22:42.32 | dijib | http://pastebin.com/0AGPYWVn |
| 22:42.32 | p3nguin | And then I had to re-edit the files after you got rid of them and then put them back. |
| 22:43.09 | dfgas-cr48 | in the extensions.conf i found my other DID 8187 line was still in the inbound stuff so i removed all those lines and then it started to show me connecting to asterisk when i would call in |
| 22:43.30 | p3nguin | I changed a setting on the router. |
| 22:43.49 | dfgas-cr48 | wait what did you edit? |
| 22:43.57 | p3nguin | Having more extensions configured than you use does not prevent calls from reaching asterisk. Having bad router settings does. |
| 22:44.20 | dfgas-cr48 | ahh |
| 22:44.23 | dfgas-cr48 | what setting was bad? |
| 22:45.05 | p3nguin | Someone enabled DMZ and supplied asterisk's address. |
| 22:45.19 | p3nguin | ~dmz |
| 22:45.19 | infobot | [~dmz] De-Militarized Zone, or usually a separate physical or logical network that has limited access to your internal systems and is accessible in limited ways from untrusted networks such as the Internet. Putting Asterisk in the DMZ is not an acceptable alternative to properly forwarding the appropriate ports, so don't do it. Plastic router appliances generally do not implement DMZ well. |
| 22:45.21 | dfgas-cr48 | yah seri wanted me to try that today |
| 22:45.29 | p3nguin | Never ever use that. |
| 22:45.41 | p3nguin | You don't use DMZ and you don't ever use sudo su. |
| 22:45.48 | dfgas-cr48 | k, it wasn't working before so we thought we would try it |
| 22:45.49 | *** join/#asterisk imox (~imox@91-66-32-57-dynip.superkabel.de) |
| 22:46.07 | dfgas-cr48 | p3nguin, why no sudo su? |
| 22:46.14 | p3nguin | It wasn't working before because there was a problem with the delivery of the DID. |
| 22:46.21 | dfgas-cr48 | i have used that for years |
| 22:46.22 | p3nguin | Because you must learn to use sudo correctly. |
| 22:46.30 | p3nguin | sudo or su, not both. |
| 22:46.32 | dfgas-cr48 | weird |
| 22:46.47 | dfgas-cr48 | yah they emailed me and said everything worked fine |
| 22:47.02 | dfgas-cr48 | sudo - right |
| 22:47.09 | p3nguin | sudo allows you to run commands as another user. Without specifying a user, it assumes root. |
| 22:47.24 | p3nguin | su allows you to become another user. Without specifying a user, it assumes root. |
| 22:47.41 | p3nguin | So sudo su means precisely this: As root, become root. |
| 22:47.42 | dfgas-cr48 | nvm, su - |
| 22:48.08 | dfgas-cr48 | ahh, true |
| 22:48.12 | p3nguin | If you are already running things as root, you don't need to also run things as root. YOU ARE ALREADY ROOT. |
| 22:48.18 | dijib | yeh seri was testing the dmz |
| 22:48.24 | dfgas-cr48 | sorry habbit i picked up a long time ago |
| 22:48.28 | p3nguin | If you want a root shell, sudo -i or sudo -s will take care of that. |
| 22:48.43 | dfgas-cr48 | ahh |
| 22:48.52 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 22:48.52 | dfgas-cr48 | so everything is all good now? |
| 22:49.08 | p3nguin | I didn't see too much else that was a problem. |
| 22:49.10 | dfgas-cr48 | expect something with my confbridge? |
| 22:49.25 | slav3_kitten | p3nguin, sudo su is typically how you become root on ubuntu boxes |
| 22:49.54 | dfgas-cr48 | yah when they changed not having root password that is what they told me to do |
| 22:49.57 | dfgas-cr48 | brb |
| 22:50.04 | p3nguin | slav3_kitten: Nope. |
| 22:50.07 | p3nguin | lrn2sudo |
| 22:50.16 | p3nguin | And read what I just typed about it. |
| 22:50.47 | slav3_kitten | p3nguin, i did. but the most common thing people will tell you us sudo su. not saying it's right just what's commonly done |
| 22:51.05 | p3nguin | They are retarded and don't need to be calling themselves admins. |
| 22:51.15 | slav3_kitten | yea i'm not going to disagree with you there |
| 22:51.46 | p3nguin | It's like running sudo sh bash sudo bash. NONSENSE. |
| 22:52.20 | slav3_kitten | that's an amazing idea... i love recursion |
| 22:52.24 | dfgas-cr48 | dijib, whats up with confbridge |
| 22:52.37 | dijib | you need a confbridge.conf |
| 22:53.09 | dijib | and you need to fix your dialplan for the confbridge specifying the bridge,user,menu (new *11 format) |
| 22:53.18 | dijib | i had not had that configured correctly |
| 22:53.19 | dfgas-cr48 | oh put in mine what yours says |
| 22:53.28 | dijib | yeh if you want |
| 22:55.08 | dfgas-cr48 | on confbridge is changed, then i need to change extensions.conf? |
| 22:56.03 | dijib | yes with the new format for the ConfBridge application instance in there |
| 22:56.32 | dijib | same => n,ConfBridge(2663,norec_bridge,admin_user,user_menu); |
| 22:56.38 | dfgas-cr48 | not sure what you mean |
| 22:56.41 | slav3_kitten | i need to get some of that cord protector stuff for the network cable to the phone |
| 22:56.44 | p3nguin | I'm still waiting for you to fix your stuff so you can connect to my conf. |
| 22:56.55 | dijib | me?! |
| 22:56.58 | p3nguin | yep |
| 22:57.05 | p3nguin | You're the only one who can't reach me. |
| 22:57.13 | p3nguin | And I checked the firewall; you are not blocked. |
| 22:57.19 | dijib | hmm |
| 22:57.49 | dijib | i dont see the problem |
| 22:57.51 | dfgas-cr48 | dijib, do i add that line to the conference? |
| 22:58.02 | dijib | yes in extensions.conf |
| 22:59.38 | dijib | i dont knwo where it is... i cant even originate to it |
| 22:59.40 | dfgas-cr48 | dijib, in your pm, like that? |
| 23:00.02 | dijib | you can change the 2663 to ${EXTEN} |
| 23:00.12 | p3nguin | You don't HAVE TO. |
| 23:01.06 | dijib | lol |
| 23:01.08 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 23:01.10 | *** join/#asterisk j4jes (~root@bas2-montreal42-3096486061.dsl.bell.ca) |
| 23:01.19 | dijib | change admin_user to default_user |
| 23:02.24 | p3nguin | Both his DIDs are working fine, now. |
| 23:02.45 | j4jes | hello, anyone here know where to plug in the 11 digit international dial rule for skyconnect in trixbox (entensions_custom.conf??) |
| 23:02.57 | p3nguin | ~trixbox |
| 23:02.57 | infobot | Trixbox is unable to be supported here. It is a closed-source distribtuion of Asterisk and FreePBX containing proprietary code that its users do not have access to, making it difficult to support. Try joining #trixbox and asking your questions there. |
| 23:03.38 | p3nguin | If you know what number you want to dial and what needs to be sent to the peer, we can help you with an appropriate extension. |
| 23:03.42 | j4jes | anyone know how to get into #trixbox if it's invite only? |
| 23:05.06 | p3nguin | (1704.42) -!- p3nguin has joined #trixbox |
| 23:05.10 | p3nguin | It isn't invite only. |
| 23:05.26 | j4jes | I used this skype-out context in extensions.conf in a basic asterisk install, and it works exten => _X.,n,Dial(SIP/skype/+1${EXTEN},90) |
| 23:05.49 | j4jes | probably because skype-connect needs the +1 prepend |
| 23:06.02 | j4jes | and the 11 digit international dialing |
| 23:06.27 | j4jes | wish I could just plug the same thing into the trixbox |
| 23:07.44 | j4jes | oh? |
| 23:07.45 | *** join/#asterisk dfgas-cr48 (~user@71-90-33-37.dhcp.ftbg.wi.charter.com) |
| 23:07.51 | dijib | p3nguin: i still cannot phone from my direct to voipms account |
| 23:08.09 | dijib | although i can reach a at&t 1800# |
| 23:08.12 | p3nguin | I don't know what you mean. |
| 23:08.30 | p3nguin | If you dial his number(s) from your cell phone, you get what? |
| 23:08.33 | dfgas-cr48 | dijib, can you check my files and is there a way when i call from outside line in that i can reach your conference from mine maybe using a password maybe |
| 23:08.45 | p3nguin | I called both of the DIDs, and the calls hit asterisk 100%. |
| 23:08.51 | dijib | when i dai from ata-> voipms i get congestion |
| 23:09.03 | *** part/#asterisk j4jes (~root@bas2-montreal42-3096486061.dsl.bell.ca) |
| 23:09.05 | p3nguin | Misconfiguration <---------- |
| 23:09.17 | dijib | hold on ive got a call |
| 23:13.40 | *** join/#asterisk j4jes (~root@bas2-montreal42-3096486061.dsl.bell.ca) |
| 23:24.24 | j4jes | can someone show me what dialpattern would append +1 infront of a 10 digit number? |
| 23:31.30 | p3nguin | 10-digit North American number? |
| 23:32.18 | p3nguin | _NXXNXXXXXX is the pattern. +1{EXTEN} makes it add +1 to the number dialed. |
| 23:34.18 | j4jes | hmmm ok, so in freepbx I can add _NXXNXXXXXX to the dial pattern, but the +1{EXTEN} i don't know, maybe in the outbound route |
| 23:35.38 | slav3_kitten | j4jes, what service are you using? |
| 23:36.02 | slav3_kitten | for your itsp |
| 23:36.13 | j4jes | trixbox <sigh> |
| 23:36.33 | slav3_kitten | trixbox is your pbx, voip.ms or flowroute would be your itsp |
| 23:36.53 | slav3_kitten | i know flowroute wants countrycode + number |
| 23:37.47 | j4jes | oh right.. I'm using skypeconnect (skype for sip) which needs 11 digit dialing, basically +1, XXXXXXXXXX |
| 23:38.02 | slav3_kitten | appending +1 would actually be 01+10digits or 12 digits |
| 23:38.46 | j4jes | so in my basic asterisk install adding exten => _X.,n,Dial(SIP/skype/+1${EXTEN},90) to the skype-out context in extensions.conf does well |
| 23:39.02 | j4jes | because i insist on trixbox, i am stuck |
| 23:39.12 | ectospasm | tompaw: I did? I do not recall trying to confirm your AMI issue on my system (not that I didn't, just that I don't remember doing so) |
| 23:39.34 | j4jes | but i think i just have to put the same string in extensions_custom.conf |
| 23:39.43 | slav3_kitten | j4jes, that's a terrible pattern match.... |
| 23:39.51 | slav3_kitten | hold up let me show you what i got |
| 23:39.55 | p3nguin | (1732.18) <p3nguin> _NXXNXXXXXX is the pattern. +1{EXTEN} makes it add +1 to the number dialed. |
| 23:40.09 | p3nguin | I meant +1${EXTEN} |
| 23:40.26 | p3nguin | I was eating a sandwich and missed the $. |
| 23:40.45 | slav3_kitten | p3nguin, yea but he said skype expects 11 digits and +1 then the 10 dialed would send 12 digits |
| 23:41.01 | j4jes | right. so I can add the pattern in my freepbx, but the +1 i have to add to my skype-out context ... |
| 23:41.05 | j4jes | wherever that is |
| 23:42.39 | slav3_kitten | j4jes, you do realize A) trixbox is dead. B) we are an asterisk channel. |
| 23:43.42 | j4jes | yes, i have sinned sorry |
| 23:44.21 | Nivex | I much prefer $EDITOR extensions.conf to some GUI monstrosity. |
| 23:44.30 | slav3_kitten | http://pastie.org/5398743 |
| 23:44.46 | slav3_kitten | j4jes, for north american dialing |
| 23:44.51 | Nivex | oh, I know I'm late to the game, but whoever came up with templates for SIP peers, THANK YOU! |
| 23:45.09 | slav3_kitten | iirc mine is based of either SeRi, p3nguin, or dijib's conf they posted earlier |
| 23:45.10 | j4jes | was hoping there might be one or two trixbox users left, since the #trixbox channel is kinda dead |
| 23:45.32 | p3nguin | dijib probably copied his from my example in the pastebin. |
| 23:45.33 | slav3_kitten | #trixbox is dead because the project has died |
| 23:45.48 | j4jes | I'm transitioning to bare asterisk, just using trixbox to get me past the learning curve |
| 23:45.54 | slav3_kitten | p3nguin, i don't remember who it was.... but it was one of you three |
| 23:46.14 | p3nguin | That looks like what I have in my example. |
| 23:47.29 | j4jes | anyways, skype-out context is usually in extensions.conf just have to find where trixbox keeps that context .. then I can edit it to prepend the +1$(EXTEN) |
| 23:47.54 | p3nguin | Maybe they have some type of "custom" extensions where you can do whatever you want. |
| 23:48.24 | j4jes | checking .. |
| 23:49.44 | slav3_kitten | probably you then p3nguin |
| 23:50.44 | p3nguin | There's a problem with it, though. |
| 23:51.01 | p3nguin | exten => _1NXXNXXXXXX,1,Dial(${TOLL}/1${EXTEN}); ;sends 11 digits |
| 23:51.31 | p3nguin | That actually sends 12 digits -- 11 followed by the number. |
| 23:52.41 | j4jes | that looks like the dial international context |
| 23:53.22 | p3nguin | I don't have a context just for international. |
| 23:53.40 | j4jes | sure it's 11 digit dial pattern but the +1$ isn't really a part of that |
| 23:55.08 | j4jes | ahh! i pressed the cloaking button |
| 23:55.15 | j4jes | what the hell is that |
| 23:56.50 | carrar | never press the cloading button |
| 23:56.55 | carrar | cloaking |
| 23:57.19 | slav3_kitten | err p3nguin i actually removed my dialing prefix for flowroute. guess i missed a digit |
| 23:58.36 | slav3_kitten | earlier SeRi was giving me a hand an i pastebinned my sip.conf while forgetting to redact my phone secret |
| 23:59.33 | *** join/#asterisk UnixDev_ (~UnixDev@unaffiliated/unixdev) |
| 23:59.40 | j4jes | in bitchx the delete button activates cloaking |
| 23:59.47 | j4jes | strange |
| 23:59.56 | UnixDev_ | when using asterisk realtime, how can you specify more than 1 codec if you can only have 1 column "allow" ? |
