| 00:08.52 | luka12345 | hi guys |
| 00:08.59 | luka12345 | anybody awake? |
| 00:09.15 | luka12345 | i'm developing a new channel driver for asterisk |
| 00:10.00 | luka12345 | i have made only minimal stuff - to log a message when i load/unload the driver |
| 00:10.20 | luka12345 | anybody knows how to manually load/unload driver from the CLI? |
| 00:22.59 | *** join/#asterisk freeedrich| (friedrich@perplexa.be) |
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| 00:37.00 | darkdrgn2k3 | hey |
| 00:37.09 | darkdrgn2k3 | any one know of a sip trunk provider that can receive faxes well? |
| 00:37.14 | darkdrgn2k3 | ie T.38 |
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| 00:59.49 | p3nguin | ~trunk |
| 00:59.50 | infobot | i heard trunk is a word with varying definitions. In Asterisk, a trunk is a "stream of UDP packets containing IAX2 frames from more than 1 call"; in telecom, a trunk is a "single voice channel between two pieces of switching equipment."; in Ethernet a trunk carries more than one 802.1q VLAN. There is no such thing as a "SIP Trunk" -- Don't use the term. the nose of an elephant |
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| 02:07.05 | darkdrgn2k3 | LOL ok... then what IS it called |
| 02:14.47 | beardy | The Ethernet definition above is Cisco-specific. In other worlds it means LACP/Etherchannel "trunking". |
| 02:16.12 | beardy | It is a good question. As a service "SIP Trunk" is what most people that sells the service calls it. I have seen FAR worse service names. "SIP Trunk" is quite descriptive and explanatory, even if it might be Wrong on many levels. But so many things are, and more, so.. |
| 02:17.14 | beardy | What is an "Internet connection"? That is as bad. I can go on. |
| 02:18.28 | darkdrgn2k3 | kinda like a digital MoDem :-P |
| 02:18.42 | beardy | darkdrgn2k3: But to find specific commercial companies and their services, I suggest searching the web and see what people has to say about them, and talk with the companies. If they seem to know what T.38 is it's a decent start. |
| 02:19.04 | darkdrgn2k3 | LoL does t.38 actualy work well? |
| 02:19.21 | beardy | Not that I know. :) |
| 02:20.00 | beardy | I've only seen trouble tickets about it, not heard anyone saying how well it works. But then again I only hear of what doesn't work, where I work. |
| 02:20.35 | beardy | Maybe it is a fine protocol, and works excellent if configured properly at both ends by competent people, like so many things. |
| 02:20.41 | beardy | I haven't played with it myself. |
| 02:21.27 | darkdrgn2k3 | i just need it for incomming faxes |
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| 02:41.14 | ryduh | hey everybody. i've got asterisk + iaxmodem + hylafax and i'm getting a bunch of no carrier errors. i've figured out that hylaxfax over voip won't work as well as i'd like. |
| 02:41.47 | ryduh | has anyone had a good experience sending faxes without any hardware? |
| 02:42.23 | ryduh | i'm thinking of upgrading my asterisk 1.4 to 1.6/10 to use t.38 with flowroute |
| 02:42.51 | ryduh | has anyone heard of successful stories faxing with flowroute over t.38? |
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| 02:43.52 | darkdrgn2k3 | ryduh: t.38 should work but im looking for a sip provider that supports it |
| 02:44.05 | ryduh | i've heard flowroute does |
| 02:44.11 | ryduh | at least on inbound |
| 02:44.27 | ryduh | i haven't tested, and can't find anything on their site about it |
| 02:45.46 | darkdrgn2k3 | ill have to look into it |
| 02:46.56 | ryduh | have you found any others darkdrgn2k3 |
| 02:46.59 | darkdrgn2k3 | ouch a little exp on the 800 |
| 02:48.33 | darkdrgn2k3 | http://www.voip-info.org/wiki/view/T.38 |
| 02:49.01 | darkdrgn2k3 | sorry |
| 02:49.01 | darkdrgn2k3 | http://www.voip-info.org/wiki/view/VOIP+Service+Providers+T.38 |
| 02:49.12 | darkdrgn2k3 | buti just started looking |
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| 04:42.28 | gundy | Can anyone tell me what the default user-agent string in Asterisk is? I need to re-write it my server to match what the AA50 has to get support :) |
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| 04:44.18 | gundy | (and does the AA50 use the same user-agent string as the full package?) |
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| 06:40.14 | gavimobile | morning folks |
| 06:49.51 | sawgood | Hi |
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| 07:10.17 | tokozedg | hi, what is the best way to compare incoming call's callerid num to a list, and add custom actions if in the list or not? |
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| 07:52.30 | schmidts | good morning |
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| 08:33.34 | qakhan | hi all |
| 08:33.59 | qakhan | is there any voice recogniation application that work with asterisk |
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| 08:52.34 | *** join/#asterisk prasanthprabalan (~root@218.248.24.19) |
| 08:52.38 | prasanthprabalan | hi all |
| 08:52.57 | thebomb | hello ? |
| 08:53.18 | prasanthprabalan | how to block certain callerid in asterisk |
| 08:54.06 | dym | Is this a google search? |
| 08:54.16 | prasanthprabalan | I have done it using comparison with GotoIf statement. |
| 08:54.26 | dym | brilliant. |
| 08:54.36 | prasanthprabalan | is there any better method |
| 08:56.04 | wdoekes2 | you can use the dialplan matching |
| 08:56.21 | wdoekes2 | Gosub(blacklist,${CALLERID(num),1) |
| 08:56.35 | wdoekes2 | +} |
| 08:56.36 | qakhan | is there any voice recogniation application that work with asterisk |
| 08:57.29 | prasanthprabalan | @wdoekes2: ok i will try that |
| 09:06.22 | kaldemar | qakhan: didn't you get some answers for that yesterday? |
| 09:10.01 | *** join/#asterisk MarKsaitis (~MarKsaiti@195.59.185.18) |
| 09:12.11 | qakhan | no |
| 09:12.18 | qakhan | do u know? |
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| 09:23.20 | kaldemar | qakhan: go try them and see that they probably don't work for names and addresses. |
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| 09:33.56 | *** join/#asterisk nickfennell (~nick@unaffiliated/nickfennell) |
| 09:34.15 | nickfennell | Hi. Is there a known bug where 'inuse' doesn't reset back to 0 making asterisk think the line is busy? |
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| 09:48.25 | *** join/#asterisk ruben23 (~RLACUMBA@121.97.111.142) |
| 09:48.32 | ruben23 | hi guys |
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| 09:50.25 | ruben23 | any have idea how to setup CDR with mysql on asterisk and be able to fill it up with data.. |
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| 10:06.12 | nickfennell | ruben23, there's like a million FAQs on that |
| 10:06.54 | nickfennell | It's also pretty easy. Configure the MySQL Server with a User and DB. Configure Asterisk to use the User and DB. Restart Asterisk. |
| 10:09.05 | kaldemar | cdr_mysql must be installed from addons if not already done. |
| 10:11.39 | ruben23 | <PROTECTED> |
| 10:13.02 | kaldemar | ruben23: what did you do to "add sterisk addon"? |
| 10:14.17 | ruben23 | kaldemar:i downloaded the tarball then compile it, ./configure, make , make install.. then i created a table tahts all..the table is not being fileld up while asterisk is running |
| 10:15.35 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 10:15.40 | kaldemar | ruben23: what does "module load cdr_mysql.so" in CLI tell you? |
| 10:15.57 | nickfennell | ruben23, when you're ready -> http://www.voip-info.org/wiki/view/Asterisk+cdr+mysql |
| 10:17.48 | ruben23 | this is the error output---> http://pastebin.com/f3mEPzU2 |
| 10:17.59 | ruben23 | yes that link i follow |
| 10:18.30 | *** join/#asterisk arnuld (~chatzilla@unaffiliated/arnuld) |
| 10:19.26 | arnuld | After successfulyl playing on extension ASTERISK says something like this: |
| 10:19.36 | arnuld | Spawn extension (.....) exited non-zero on 'SIP/..... |
| 10:19.46 | arnuld | how can I capture that non-zero value ? |
| 10:21.34 | ruben23 | nickfennell: do i need to recompile asterisk also..? |
| 10:21.54 | ruben23 | kaldemar:do i need to recompile asterisk also..? |
| 10:22.11 | kaldemar | ruben23: no. |
| 10:23.04 | kaldemar | arnuld: that just means the execution of the extension is finished. don't worry about it. |
| 10:23.41 | *** join/#asterisk nutxase (~nutxase@unix-02.lair.co.za) |
| 10:23.44 | arnuld | kaldemar: I know |
| 10:23.56 | arnuld | kaldemar: I just want to know how can I get that value if I want |
| 10:24.11 | nutxase | any found a problem with Dahdi when you do a reload in asterisk u need to do a dahdi restart else dahdi stops working? |
| 10:24.32 | arnuld | kaldemar: Its like you give input to some app e.g. MP3Player(file-location) |
| 10:24.44 | arnuld | kaldemar: same way I want the output back |
| 10:24.47 | kaldemar | ruben23: you don't have the module installed. |
| 10:25.29 | ruben23 | kaldemar:ok, how should i install that module..? im using ubuntu server |
| 10:25.38 | kaldemar | arnuld: what do you really want to do? |
| 10:25.59 | arnuld | kaldemar: thats my job ;( |
| 10:26.14 | kaldemar | arnuld: what is? |
| 10:26.44 | arnuld | kaldemar: to get whatever output that app is returning with |
| 10:27.11 | arnuld | kaldemar: so far I see it only CLI only |
| 10:27.39 | kaldemar | ruben23: you have the addons tarball, right? do a make menuselect and select the cdr_mysql module. then make and make install. |
| 10:28.40 | ruben23 | kaldemar:ok thank you somuch ill do it now |
| 10:28.48 | kaldemar | arnuld: MP3Player does not output anything. |
| 10:29.17 | arnuld | so who output that "Spawn extension (.....) exited non-zero on 'SIP/....." ?? |
| 10:29.36 | *** join/#asterisk MarKsaitis (~MarKsaiti@195.59.185.18) |
| 10:29.47 | arnuld | kaldemar: I mean who gave back non-zedro value ? |
| 10:29.54 | arnuld | zero* |
| 10:30.34 | kaldemar | arnuld: the extension, not any application. why are you worrying about it? |
| 10:31.41 | arnuld | kaldemar: I am not worried about it, I have been given the job of fidning out how to capture that value. thats it. I don't know why they need it |
| 10:32.03 | arnuld | kaldemar: so how to get that output of extension ? (am I correct now ?) |
| 10:32.10 | kaldemar | arnuld: you don't. |
| 10:32.53 | kaldemar | arnuld: and trying to get that value is pointless. you should be concetrating on something that actually means something. |
| 10:33.29 | arnuld | kaldemar: how to know that application is successfully executed or not ? |
| 10:33.42 | arnuld | kaldemar: when no one is sitting and watching CLI |
| 10:35.58 | kaldemar | arnuld: if you use an extension as a subroutine, you must produce a meaningful value yourself. for example execute the extension with GoSub() and return from it with Return(), which takes a return value as an argument. the value is stored in GOSUB_RETVAL. |
| 10:37.13 | arnuld | kaldemar: I think that will do the trick |
| 10:37.47 | kaldemar | arnuld: do you mean the MP3Player application? it will always get executed, maybe a check on the target file is something you require? |
| 10:38.36 | arnuld | kaldemar: lets say target file does not exist. so the right way wil be |
| 10:38.55 | kaldemar | to check if the file exists before executing MP3Player. |
| 10:39.10 | arnuld | lets say mpg123 is not there |
| 10:39.18 | arnuld | kaldemar: right way --> GoSub(MP3layer(location)) ? |
| 10:39.25 | kaldemar | wrong. |
| 10:39.33 | arnuld | newbie signs |
| 10:39.34 | kaldemar | core show application GoSub |
| 10:40.06 | arnuld | jumps to a lable |
| 10:40.09 | arnuld | wats a lable ? |
| 10:40.27 | arnuld | oh.. wait |
| 10:40.38 | kaldemar | read past the first line. |
| 10:40.42 | arnuld | documentaion says clearly: GoSub(my_context(my_extension)) |
| 10:41.08 | kaldemar | or Gosub([[context,]exten,]priority[(arg1[,...][,argN])]) |
| 10:41.44 | kaldemar | but gosub won't help you with anything related to checking whether the file exists. |
| 10:42.08 | arnuld | not intersted in that. only if the extension falied to execute or not |
| 10:42.14 | arnuld | whatever the reason is |
| 10:42.28 | kaldemar | then be specific on what you want to do. |
| 10:42.35 | arnuld | okay |
| 10:42.56 | kaldemar | the extension will always execute if you execute it. no exceptions. what do you really want to check? |
| 10:43.10 | arnuld | so if mpg123 is not installed on that machine |
| 10:43.16 | arnuld | then it won't execut |
| 10:43.21 | arnuld | execute* |
| 10:43.46 | kaldemar | use app System of func SHELL |
| 10:44.04 | arnuld | okay, let me check documentation |
| 10:44.06 | kaldemar | to check before trying to execute MP3Player app. |
| 10:44.28 | kaldemar | app System OR func SHELL was what i meant. |
| 10:45.36 | kaldemar | checking for things like mpg123 is lunacy in my opinion. what kind of environment are you running if you can't be sure that all needed SW is installed during operation? |
| 10:45.56 | arnuld | I can't tell that to my boss |
| 10:46.00 | arnuld | :-\ |
| 10:46.36 | arnuld | okay, extension will always be executed, I agree |
| 10:46.39 | kaldemar | you should if you're given pointless tasks. dialplan is really not the place for that kind of checks. |
| 10:47.32 | arnuld | I dont want to waste your precious time which can be invested in other tasks which have some sense |
| 10:48.19 | arnuld | still... lets say everything executed and MP3Player() or Dial() or ANyAppHere() successfully worked |
| 10:49.41 | arnuld | and CLI prints: "Spawn extension (.....) exited non-zero on 'SIP/....." |
| 10:49.43 | arnuld | thats it |
| 10:50.17 | kaldemar | being able to determine whether something was successful depends on the used application and what it does. "Spawn extension (.....) exited non-zero on 'SIP/....." will not help you with that in any way and is not useful for anything. |
| 10:51.25 | arnuld | Here is the literal sentence I have been told, now see what I did not get: |
| 10:52.11 | arnuld | "This ASTERISK app has taken an input e.g. mp3 file location to play it. Now that app also gives some output. We want that output" |
| 10:52.22 | arnuld | so I dont know whats asked here |
| 10:52.33 | arnuld | after al this discussion |
| 10:55.25 | arnuld | I can only see that he is referring to the output of extension |
| 10:55.29 | arnuld | that non-zero value |
| 10:55.35 | arnuld | thats all I can make out |
| 10:56.04 | kaldemar | MP3Player has nothing to do with the non-zero value. stop concentrating on the non-zero value. it is useless. |
| 10:56.15 | arnuld | okay |
| 10:56.24 | arnuld | so what do you think that sentence mean ? |
| 10:56.29 | arnuld | means* |
| 10:56.51 | kaldemar | it means someone wants something that does not exist. |
| 10:57.00 | arnuld | ha ha ha.... |
| 10:57.04 | arnuld | best answer :) |
| 10:57.12 | arnuld | likes kaldemar |
| 10:57.56 | kaldemar | MP3Player does not output anything that you could catch in dialplan. you should ask them what they really want. |
| 10:58.20 | arnuld | they want to know whether extension successfuly executed or not |
| 10:58.25 | arnuld | and that shows in CLI actually |
| 10:58.48 | kaldemar | the answer is always yes. |
| 10:59.03 | arnuld | hmm.. BRB |
| 10:59.06 | arnuld | ... |
| 10:59.07 | kaldemar | if the extension exists in dialplan, it will always be executed successfully. |
| 10:59.30 | arnuld | okay |
| 10:59.32 | arnuld | I get it |
| 10:59.34 | arnuld | completely |
| 11:05.52 | arnuld | back |
| 11:06.09 | arnuld | kaldemar: I talked to them again here is what they told me |
| 11:06.48 | arnuld | kaldemar: take Transfer() app. in documentation it says this: |
| 11:06.50 | arnuld | The result of the application will be reported in the The result of the application will be reported in the None - TRANSFERSTATUS channel variable |
| 11:07.05 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 11:07.33 | arnuld | <PROTECTED> |
| 11:07.52 | arnuld | so where I can find TRASFERSTATUS ? |
| 11:08.16 | arnuld | thats the quesiton |
| 11:09.24 | kaldemar | that has nothing to to with MP3Player. TRANSFERSTATUS is a variable that is set by app Transfer. you can see its value with ${TRANSFERSTATUS} after app Transfer has been executed. |
| 11:09.26 | j0b | oh man, this is broken by design, how ever. we have a astra pbx that talks SIP. This need a converter, a IP-module that converts voice streams that resides on another IP. så rtp signaling goes thorugh another ip than sip |
| 11:09.31 | j0b | how can we solve this |
| 11:09.35 | j0b | im confused |
| 11:10.03 | *** join/#asterisk MarKsaitis (~MarKsaiti@195.59.185.18) |
| 11:13.22 | arnuld | kaldemar: u mean in extensions.ael I can get the value e.g. |
| 11:13.33 | arnuld | 1006 => { Answer(); |
| 11:13.43 | arnuld | Transfer(SIP/Bob); |
| 11:13.59 | arnuld | ${TRANSFERSTATUS}; |
| 11:14.00 | arnuld | }; |
| 11:14.02 | arnuld | ?? |
| 11:15.02 | *** join/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 11:15.04 | kaldemar | ${TRANSFERSTATUS} does not do anything by itself. you need an application on that line. if you only want to see it in CLI, use NoOp or Verbose. e.g. NoOp(${TRANSFERSTATUS}) |
| 11:16.05 | arnuld | is there an app to print that value ? |
| 11:17.54 | kaldemar | i just have you two. :) |
| 11:18.45 | arnuld | oh.. sorry... |
| 11:19.05 | arnuld | <PROTECTED> |
| 11:19.07 | arnuld | :( |
| 11:20.07 | j0b | sip will handle the negotiation |
| 11:20.25 | j0b | so sending rtp to another server is a no go |
| 11:20.37 | j0b | because it should work |
| 11:20.56 | j0b | anyway, we get the classical, we can only hear one side talking |
| 11:22.40 | arnuld | sends thanks to kaldemar |
| 11:31.08 | *** join/#asterisk N1x1 (~n1x@mail.orient-logic.com) |
| 11:31.11 | N1x1 | hi guys |
| 11:31.30 | N1x1 | im trying to add pickup in dialplan like this exten => _7.,1,Pickup(${EXTEN:1}) |
| 11:31.42 | N1x1 | and i need to work it for all extension |
| 11:31.48 | N1x1 | but not working... |
| 11:32.02 | N1x1 | any ideas? :) |
| 11:32.54 | kaldemar | show it. |
| 11:33.43 | N1x1 | asterisk said just , number 71234 not found in context "blablabla |
| 11:34.21 | N1x1 | when i calling to exactly extension it works |
| 11:34.33 | N1x1 | i dont need to write this in every extension |
| 11:34.38 | N1x1 | in every context* |
| 11:36.27 | kaldemar | N1x1: if context is omitted in Pickup argument, the current context is used. |
| 11:36.56 | kaldemar | if the extension you want to pick up is in another context, that won't work. |
| 11:37.35 | N1x1 | kaldemar: i need to write in every context yes? |
| 11:37.47 | kaldemar | depends on your dialplan. |
| 11:37.59 | N1x1 | sec |
| 11:38.23 | kaldemar | writing that to every context won't help anything if you include the contexts in another. the first occurrence will always match but it is not necessarily the right context. |
| 11:39.17 | kaldemar | time to re-design your dialplan or use PICKUPMARK. |
| 11:40.06 | N1x1 | kaldemar: include statement wonn't help? |
| 11:40.10 | N1x1 | in context |
| 11:40.20 | kaldemar | depends on your dialplan. |
| 11:40.31 | N1x1 | kaldemar: i'll show you |
| 11:40.36 | N1x1 | sec please |
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| 11:45.30 | catphish | could anyone speculate as to what might cause the mysql cdr to start writing blank rows to the database instead of rows with data |
| 11:45.41 | catphish | starting after a restart and recovering after a reload |
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| 11:54.37 | N1x1 | kaldemar: http://pastie.org/3634298 |
| 11:54.48 | N1x1 | see it |
| 11:55.15 | N1x1 | kaldemar: i need to work this pickup to "toko" and a2billing |
| 11:55.22 | N1x1 | in both context |
| 11:56.01 | N1x1 | how i can do it? |
| 11:56.23 | N1x1 | just write include => toko , in a2billling context? may help it? |
| 11:56.49 | *** join/#asterisk Bullmoose (~Bullmoose@75-174-79-132.bois.qwest.net) |
| 12:00.01 | kaldemar | N1x1: core show application Pickup |
| 12:00.09 | *** join/#asterisk serafie (~erin@75.76.38.159) |
| 12:01.06 | kaldemar | N1x1: and think about what ${EXTEN} is. do you even have such an extension in those contexts? |
| 12:01.33 | N1x1 | kaldemar: yes |
| 12:01.46 | N1x1 | i need to match every extension |
| 12:02.06 | kaldemar | N1x1: what do you think exten => _*1X.,1,Pickup(${EXTEN}) does? |
| 12:03.00 | N1x1 | i have edited because exten => _*1X.,1,Pickup(${EXTEN:1}) |
| 12:03.05 | N1x1 | this not working |
| 12:03.12 | N1x1 | just for test purpose |
| 12:03.32 | kaldemar | why are you showing me irrelevant outdated stuff? |
| 12:04.09 | N1x1 | this is my current config :) |
| 12:04.41 | kaldemar | exten => _*1X.,1,Pickup(${EXTEN}) would try to pick itself up. exten => _*1X.,1,Pickup(${EXTEN:1}) would try to pick up extension that starts with 1. you don't have such an extension. |
| 12:05.23 | kaldemar | reword this: < N1x1> kaldemar: i need to work this pickup to "toko" and a2billing |
| 12:05.30 | kaldemar | it is not understandable. |
| 12:06.13 | flyingbull | Hi there:) I have a question, why would it that if I call out on an external number, the person called can hear me, but I can't hear them, But if I transfer a call to an external number there isn't a problem? |
| 12:06.18 | N1x1 | kaldemar: yes right , i'll asking you exactly this. how i can do it , i need to match every extension |
| 12:06.27 | N1x1 | i dont need to write specific extension |
| 12:07.24 | kaldemar | "i need to match every extension" does not make sense. |
| 12:07.42 | N1x1 | kaldemar: i'll explain you :D so listen :) |
| 12:08.02 | N1x1 | exten => _*1X., this mean |
| 12:08.36 | N1x1 | when one people calling to 75 from 65 , when i dialing *165 from my phone |
| 12:08.41 | N1x1 | i need to pickup this call |
| 12:08.42 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 12:09.00 | kaldemar | N1x1: that's not how it works. |
| 12:09.12 | N1x1 | kaldemar: sorry i mean *175 |
| 12:09.21 | N1x1 | pickup ring from 65 to 75 |
| 12:09.37 | kaldemar | N1x1: that's better. in what contexts are the extensions that match 75? |
| 12:11.41 | N1x1 | kaldemar: in other context |
| 12:11.52 | N1x1 | maybe "test" |
| 12:12.19 | N1x1 | can i do it from only one context? write this extension |
| 12:12.21 | kaldemar | exten => *175.,1,Pickup(${EXTEN:2}@test) |
| 12:12.25 | N1x1 | and make pickup from other context |
| 12:12.55 | N1x1 | ah :-) |
| 12:13.02 | kaldemar | actually, that's bullshit. exten => _*1XX,1,Pickup(${EXTEN:2}@test) |
| 12:13.16 | N1x1 | why? it works right? |
| 12:13.49 | kaldemar | exten => *175. has errors. no _ to denote a pattern and it expects something after 75. |
| 12:14.05 | N1x1 | yes |
| 12:14.10 | N1x1 | good |
| 12:14.12 | N1x1 | well thanks :) |
| 12:14.18 | *** join/#asterisk vipkilla (~chatzilla@unaffiliated/t-dot-zilla/x-2830497) |
| 12:14.44 | kaldemar | if you have two contexts, exten => *175.,1,Pickup(${EXTEN:2}@test&${EXTEN:2}@anothercontext) |
| 12:15.34 | N1x1 | kaldemar: and what this EXTEN:>2< this mean? |
| 12:15.54 | kaldemar | it cuts two first characters of variable EXTEN content. |
| 12:16.03 | kaldemar | so it removes the *1 part. |
| 12:16.06 | N1x1 | aa |
| 12:16.15 | N1x1 | ok |
| 12:16.33 | N1x1 | thanks thanks |
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| 12:32.15 | flyingbull | good morning everyone, I'm trying to figure out a problem I have, I am able to receive calls without a problem, and I am able to take these calls and transfer them to an external number without a problem. But if I make a call to an external number, I cannot hear the person on the other side, but they can hear me. I see no errors in my logs, so I think it has to be somewhere in my configuration, can anyone give me a clue where to loo |
| 12:32.22 | *** part/#asterisk vipkilla (~chatzilla@unaffiliated/t-dot-zilla/x-2830497) |
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| 12:33.21 | CommaCrazy | helloooo all :) |
| 12:37.26 | CommaCrazy | I got a bit of a problem, it's not a Asterisk problem (unfortunately, because that I would know how to solve). My topex softswitch pbx is connected to a voip provider who also has topex pbx and everything was working fine, nothing in the config is changed but I cannot make a call, it tells me that the q850 cause code is 41 (temporary failure) and from what I can see from the tcp dump the correct #s are sent out and then I see that the call is canceled wit |
| 12:37.26 | CommaCrazy | h the q850 41 cause |
| 12:38.36 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 12:38.49 | CommaCrazy | I know that this is a long shot that some 1 can help but hey when a man is desperate... |
| 12:45.31 | *** part/#asterisk arnuld (~chatzilla@unaffiliated/arnuld) |
| 12:46.36 | thebomb | can someone please look at this dialplain http://pastebin.com/KJfHanRA my problem is its not going over the SmartNote or international includes ? |
| 12:46.58 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:46.59 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 12:49.56 | kaldemar | thebomb: the extension in your context or one of the included ones before them match the call. |
| 12:51.48 | thebomb | kaldemar, so it matches the call on the _[089]XXXX and does not continue ? |
| 12:51.58 | thebomb | to the includes ? |
| 12:54.25 | kaldemar | thebomb: either that or an extension in the contexts that are included before SmartNote and international. |
| 12:55.37 | thebomb | kaldemar, so how to I "force" it to keep continue with the context ? |
| 12:56.01 | thebomb | kaldemar, Goto ? |
| 12:56.06 | kaldemar | you don't. |
| 12:56.23 | thebomb | damn ... |
| 12:57.15 | kaldemar | if there is a better match in SmartNote or international, include them before others. |
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| 13:15.00 | jaytee | good morning all |
| 13:15.24 | [TK]D-Fender | thebomb, If you want to prioritize, then take the loose dialplan in that context and put it in one of its own, and then change the include order |
| 13:15.50 | *** join/#asterisk faithlove (proxyuser@188.27.133.9) |
| 13:18.14 | *** join/#asterisk jacc0 (~jacc0@D522448D.static.ziggozakelijk.nl) |
| 13:18.45 | jacc0 | hi all!! |
| 13:18.49 | jacc0 | good day |
| 13:19.25 | jacc0 | the spring started today overhere :) |
| 13:19.47 | dym | We've had sunny weather for a couple of days now |
| 13:19.58 | jacc0 | we to over here |
| 13:20.13 | dym | did the cheese melt? :O |
| 13:20.30 | jacc0 | why does asterisk always show a warning after softhangup(${CHANNELNAME})? |
| 13:20.46 | jaytee | I'm having a issue with no audio to an asterisk server from flowroute. I had a test extension setup to play a sound file, then record a new file and play that back. It was working at one point and now fails with "maximum retries exceeded" and "no reply to our critical packet" |
| 13:21.13 | dym | RTP not coming through? |
| 13:21.13 | dym | NAT |
| 13:21.13 | dym | ? |
| 13:21.13 | jacc0 | nat |
| 13:21.13 | [TK]D-Fender | jaytee, networking issues |
| 13:21.17 | dym | ... |
| 13:21.20 | dym | [TK]D-Fender: parrot |
| 13:21.21 | dym | :P |
| 13:21.53 | jacc0 | ~nat |
| 13:21.53 | infobot | methinks nat is Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
| 13:21.59 | jaytee | yeah, that's my first suspect. The server is behind a firewall and I've been told that RTP ports are nat'd to the inside address |
| 13:22.37 | jaytee | it did work about 2 weeks ago but we had other issues to deal with and when I got back to it yesterday it wouldn't work. |
| 13:22.54 | jaytee | I have nat=yes in the general section of my sip.conf |
| 13:23.09 | jacc0 | doe you have 'externip' configured? |
| 13:23.17 | jaytee | yes |
| 13:23.48 | jaytee | here's a sip debug of a test call: http://pastebin.com/DrhwPHcz |
| 13:23.49 | jacc0 | couldn't you put the asterisk in the DMZ? |
| 13:24.09 | jacc0 | or forward port 10000 up to 20000 |
| 13:26.39 | [TK]D-Fender | jaytee, Reliably Transmitting (NAT) to 216.115.69.144:5060: <- flowroute is NOT behind nat. Set nat=no in their peer |
| 13:27.29 | [TK]D-Fender | <--- SIP read from UDP:216.115.69.144:5060 ---> <-- call arrived from |
| 13:27.44 | [TK]D-Fender | Contact: <sip:+13173855975@4.55.17.35:5060> <--- return contact requested |
| 13:28.09 | [TK]D-Fender | Your NAT setting prevents getting media from their very separate media servers |
| 13:28.20 | *** part/#asterisk serafie1 (~erin@75.76.38.159) |
| 13:28.35 | [TK]D-Fender | c=IN IP4 4.55.17.2 <--- actual media |
| 13:29.59 | thebomb | [TK]D-Fender, made a macro, then includes that worked |
| 13:30.05 | thebomb | :) |
| 13:30.40 | p3nguin | Not putting nat=no in the flowroute peer is not going to break it. We went over this the other day. New best practice is to define your nat setting only in the general section and in none of the peer sections. |
| 13:31.37 | [TK]D-Fender | p3nguin, NAT settings is a peer setting and yes it will break it. |
| 13:31.49 | p3nguin | No it won't. |
| 13:32.03 | [TK]D-Fender | It forces * to ignore the requested IP's for signalling and media and only send to the IP the call initially arrived on. |
| 13:32.09 | [TK]D-Fender | This will completely break it. |
| 13:32.42 | [TK]D-Fender | This is to deal with dumb devices that do things like passing private IP's for contact, etc. |
| 13:32.58 | [TK]D-Fender | eg: just about every phone behind a remote NAT |
| 13:33.49 | p3nguin | I follow the new recommendation. I have nat=yes in my general section because I support devices which are behind NAT and asterisk itself is also behind NAT. I have not overridden that general setting in any of my peers, ITSP or phone. |
| 13:34.28 | p3nguin | That means all my ITSP peers are now nat=yes due to the general setting. |
| 13:34.34 | p3nguin | And guess what... |
| 13:34.40 | p3nguin | It still works! |
| 13:34.40 | [TK]D-Fender | p3nguin, it is typically fine.. until you deal with ITSP's that separate signalling and media servers |
| 13:35.12 | [TK]D-Fender | p3nguin, If yours work... then lucky you, for the rest of people however... DOA |
| 13:36.43 | jaytee | having nat=no in my peer definition for flowroute made no difference. So you're saying that the media from 4.55.17.2 is being blocked by the nat settings in the firewall? |
| 13:36.49 | p3nguin | There's no way mine is a special case. I configure with standard setting and values. |
| 13:37.28 | *** join/#asterisk nighty- (~nighty@static-68-179-124-161.ptr.terago.net) |
| 13:37.30 | kaldemar | p3nguin: in your scenario the whole nat option would be pointless. it might aswell not exist and always be yes. |
| 13:37.42 | p3nguin | It is always yes. |
| 13:38.05 | [TK]D-Fender | jaytee, No, * settings. PB your configs mask only passwords, ensure 5060, 10000-20000 UDP are forwarded to your server and we'll go from there. And please include a new call debug |
| 13:39.34 | p3nguin | Oh, you're saying non-configurable. Well, if you don't support any NATted devices anywhere, set nat=no; otherwise, leave it default (nat=force_rport or nat=yes, depending on version) or explicitly set it to yes. |
| 13:39.42 | jaytee | [TK]D-Fender, does RTP have to be 10000-20000? can it be 10000-11000 as long as it matches rptstart and rtpend in rtp.conf? |
| 13:39.49 | [TK]D-Fender | jaytee, Sure |
| 13:40.26 | kaldemar | p3nguin: what you set in [general] gets inherited to peers unless otherwise configured. |
| 13:40.34 | p3nguin | I understand that. |
| 13:40.58 | jaytee | ok, they're in Ft. Wayne and the IT tech I've dealt with is usually very busy. They are also tight on security and won't let me view their nat rules on their firewall to verify. |
| 13:41.02 | p3nguin | And the default for the nat setting it now force_rport or yes. |
| 13:41.15 | [TK]D-Fender | jaytee, trust[-1] |
| 13:41.44 | jaytee | so I'll have to get back to this after I manage to get in contact with him. Hard to get him on the phone. He seems to prefer email :-( |
| 13:42.12 | *** join/#asterisk ggtest (~ggtest@c207.134.154-241.clta.globetrotter.net) |
| 13:42.56 | jaytee | [TK]D-Fender, p3nguin and dym, thanks very much for your input and advice. |
| 13:43.01 | p3nguin | And it is recommended to set the nat value ONLY in the general section. If you support ANY nat devices, leave it default or set explicitly to yes. If you do not support any nat devices, explicitly set it to no. If you have a special case where you NEED to override the default yes value, such as certain Cisco phones, you _can_ override the setting per peer. |
| 13:44.01 | [TK]D-Fender | "If you have a special case where you NEED to override the default yes value, such as certain Cisco phones, you _can_ override the setting per peer." <-- ITSPs |
| 13:44.26 | [TK]D-Fender | The rest you may as well leave as NAT=YES |
| 13:44.32 | kaldemar | p3nguin: yes. so a per peer definition is needed in some cases. |
| 13:44.46 | [TK]D-Fender | But you are trying to stretch the application of that notice to "at all times" |
| 13:45.04 | p3nguin | No, you just interpreted it that way. |
| 13:47.06 | *** join/#asterisk _Corey_ (~chatzilla@mail1.infradapt.com) |
| 13:47.50 | [TK]D-Fender | p3nguin, Between today and previous chats on the topic that is an impression you're leaving on it. But we've covered it here again in detail so hopefully that'll do for a while... |
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| 13:50.01 | ggtest | Hi! I wonder if asterisk binaries will be available for centos 6 soon |
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| 14:00.02 | *** join/#asterisk Sktenau (~textual@118.11.233.220.static.exetel.com.au) |
| 14:01.47 | Sktenau | Morning all, a quick Q I hope someone can confirm - if HANGUP is executed within AGI, is it expected that the channel would hangup and the dial plan stop running? |
| 14:02.06 | Andee | yes |
| 14:02.54 | *** join/#asterisk eicto (~morgoth@144-71.dsl.aichyna.com) |
| 14:03.36 | Sktenau | Hrmm, I would have thought so. I'm finding that regardless of how I execute it within the AGI (Ie. agi->hangup or agi->exec(hangup)) the channel is not dropped, the dial plan continues to run. What should the return code be? AGI debug suggests that its -1, I would have expected 0. |
| 14:04.42 | p3nguin | Just so you know I'm not lying or trying to make up something which isn't necessarily accurate... http://pastebin.com/Trg7ShSZ |
| 14:04.56 | Sktenau | I can however request hangup of the channel from the CLI without issue, and I've tested from both 1.6 and 1.8 branches. |
| 14:07.34 | Andee | https://issues.asterisk.org/jira/browse/ASTERISK-16525 <-- is this any help? the channel stays open until the AGI exists, even though it is hungup but not destroyed? |
| 14:08.23 | p3nguin | AGI hangup is buggy? |
| 14:08.35 | Sktenau | Yeah I came across that too, doesn't seem to be the same issue as if I either cleanly exit the script or forcefully kill it there is no difference. |
| 14:08.44 | Andee | no idea, that jira ticket says it's the expected behaviour |
| 14:09.30 | Sktenau | Possibly. I'm using PHP AGI so also have tested with exec(hangup) which AFAIK won't rely on their own implementation (Which is working with Noop and returning the response I'd expect) |
| 14:09.54 | p3nguin | It almost seems silly to allow hangup to be executed within the agi but not hang up the channel. What could be the reason for that to be normal behavior? |
| 14:10.51 | Andee | i don't know, the documentation is poorly worded |
| 14:11.20 | leifmadsen | the channel is likely only hung up once the AGI returns since the channel is actually executing the AGI |
| 14:11.49 | leifmadsen | probably a question for a devleoper who understands the internal workings of channels though |
| 14:12.14 | *** join/#asterisk radic (~radic@dslb-094-216-247-010.pools.arcor-ip.net) |
| 14:12.53 | p3nguin | I wondered if that might be the reason. |
| 14:15.01 | *** join/#asterisk fprior (be0bbc86@gateway/web/freenode/ip.190.11.188.134) |
| 14:15.43 | Sktenau | Interesting. I'd prefer to use exec(hangup) as so I can specify a causecode - If I use PHPAGI's hangup implementation and specify the channel name it actually seems to work. |
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| 14:26.25 | *** join/#asterisk Katty (~Katty@mail.copi-rite.com) |
| 14:26.42 | Katty | HELLO MY ASTerisk does not work at all how to fix plz?? |
| 14:27.02 | Katty | Qwell: ping. |
| 14:28.50 | Sktenau | Are you sure you are holding down shift when pressing the number 8 key? Its a rookie mistake. |
| 14:30.26 | *** join/#asterisk Alagar (~helpdesk@vsusg1.vernalissystems.com) |
| 14:32.25 | Alagar | any one can help me, is any good sip trunk provider for outgoing sip calls |
| 14:32.42 | Katty | Sktenau: you're a rookie mistake |
| 14:32.44 | Katty | Sktenau: OH SNAP |
| 14:32.51 | Katty | Sktenau: <3 |
| 14:32.51 | *** join/#asterisk MarKsaitis (~MarKsaiti@195.59.185.18) |
| 14:33.04 | Sktenau | Heh. I'll have to play with it some more, as noted if I use PHPAGI hangup and specify the channel the call is dropped correctly, but there is no facility to specify a causecode. If I exec(hangup, [causecode]) prior to the working hangup though, the call is dropped with the desired cause code. Go figure. |
| 14:33.11 | Katty | infobot: itsp |
| 14:33.11 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 14:33.25 | Katty | infobot: itsplist-us |
| 14:33.25 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
| 14:33.33 | Katty | ^- Alagar |
| 14:33.53 | Katty | we've used bandwidth.com. their tech support has been more or less, tolerable. |
| 14:34.06 | Katty | but rumor has it that's because i'm a girl. and tech support is generally nicer to girls who call in. |
| 14:34.51 | Katty | where is Qwell |
| 14:34.59 | Katty | i need Qwell |
| 14:35.21 | Katty | russellb: where is mister qwellery. |
| 14:35.25 | *** join/#asterisk ledesmam (~ledesmam@192.6.114.30) |
| 14:35.35 | russellb | Katty: out i suppose? |
| 14:35.50 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 14:35.52 | Katty | that boy doesn't go out >.< |
| 14:38.04 | Katty | yay crisis averted. |
| 14:40.39 | file | he's maybe still asleep |
| 14:40.48 | Katty | file: -> |
| 14:40.49 | file | he is on vakay |
| 14:41.34 | Chex | wow, Broadvoice is still around, are they still in the Wang building in Lowell, I wonder? |
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| 14:49.27 | leifmadsen | Katty: ya I've had pretty decent luck with bandwidth.com over the last 3 years |
| 14:54.19 | j0b | hmmm, ok we have it like this. everything runs on the same layer2 network, so it does not need to be routed. we have a pbx1, which is the main outgoing exhange. we have pbx2 that only servers one telephone number. i we call this number, the flow is as..... pbx1 receives a call for a phone, i forwardes it to pbx2, pbx2 checks its dialplan and calls VIA pbx1 with a number that we choose. pbx1 is an Aastra exchange 2065 and have a ipmodule that resides on another |
| 14:54.32 | j0b | no firewalling |
| 14:54.38 | j0b | what so ever |
| 15:01.02 | catphish | could anyone speculate as to what might cause the mysql cdr to start writing blank rows to the database instead of rows with data |
| 15:01.34 | j0b | wrong coulmnames? empty array? |
| 15:02.13 | j0b | which sql-scheme regarding columnnames are you using? |
| 15:02.58 | catphish | j0b: this started after a reboot, and recovered after a reload |
| 15:03.39 | catphish | j0b: http://paste.codebasehq.com/pastes/kbp1tvrma04f2shidc |
| 15:04.31 | catphish | so it's unlikely to be related to the schema |
| 15:05.19 | j0b | unlikely, indeed |
| 15:05.53 | catphish | nothing in the logs, which unfortunately had no verbosity set at the time |
| 15:07.17 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 15:09.05 | [TK]D-Fender | j0So far you have not described a problem, and it seems you are referring to 2 non-* systems..... |
| 15:09.14 | [TK]D-Fender | j0b, So far you have not described a problem, and it seems you are referring to 2 non-* systems..... |
| 15:11.22 | fprior | Hi, is there anyone who worked with chan_dongle ? my dongle always connect a "storage disk", instead of "modem gsm". I've used usb_modeswitch but with no success. |
| 15:23.38 | *** join/#asterisk DaveCL (~Dave@201-246-93-133.baf.movistar.cl) |
| 15:24.01 | DaveCL | hello |
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| 15:32.54 | DaveCL | hey guys! |
| 15:32.55 | DaveCL | I have a question, I have a TDM400 card and 1 FXO module ... I can make calls but I can not receive calls ... when I call the PSTN appears as busy ... and do not show me anything in the asterisk console, I can be doing wrong? |
| 15:33.51 | *** join/#asterisk WindBack (~quassel@190.123.122.17) |
| 15:34.31 | j0b | [TK]D-Fender: sorry man, could i give you some more information? |
| 15:34.33 | j0b | :) |
| 15:34.42 | WindBack | Is there any way to return more than 1 column in a function using func_odbc?? |
| 15:34.48 | j0b | the problem is that we get one way voice only |
| 15:36.52 | DaveCL | I ask again ... with an FXO module I can receive calls from the PSTN? |
| 15:36.56 | DaveCL | (sorry for my english) |
| 15:38.29 | *** part/#asterisk TheCops (~mdb@72.55.132.180) |
| 15:38.54 | [TK]D-Fender | j0b, So far I don't see where this is an * issue.... |
| 15:39.28 | [TK]D-Fender | DaveCL, PASTEBIN you dahdi configs, and your complete failed call attempt from CLI |
| 15:39.29 | [TK]D-Fender | ~pb |
| 15:39.29 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 15:39.31 | [TK]D-Fender | ^^^ |
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| 16:02.06 | WindBack | [TK]D-Fender: Is there any way to return more than 1 column in a function using func_odbc?? |
| 16:02.21 | WindBack | [TK]D-Fender: I cant find any documentation about this |
| 16:05.32 | [TK]D-Fender | WindBack, ODBC_FETCH() |
| 16:06.32 | WindBack | [TK]D-Fender: odbc_fetch let you retrieve more than one row, but i want to retrive more than one column |
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| 16:23.52 | *** join/#asterisk DaveCL (~Dave@201-246-111-80.baf.movistar.cl) |
| 16:26.39 | thurin | Hello all. I'm looking for ideas re: a problem I'm seeing with an Asterisk setup I've inherited. Seeing a hangup detection issue with IAX. SIP is working fine. If the IAX channel is in AgentLogin and the agent hangs up, the hangup is not detected. |
| 16:27.00 | thurin | If the agent joins a meetme conference and then hangs up, it is detected. |
| 16:27.25 | *** join/#asterisk Gaiax (~Gaiax@unaffiliated/gaiax) |
| 16:27.47 | thurin | This is in asterisk 1.4 |
| 16:37.28 | *** join/#asterisk paolosupino (~paolo-sup@net-2-38-113-188.cust.dsl.vodafone.it) |
| 16:37.30 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
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| 16:54.00 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 17:00.17 | [TK]D-Fender | WindBack, http://astbook.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/getting_funky.html |
| 17:01.49 | pigpen | thurin, and if anyone disagrees, please chime in, in 1.4 there were a mess of queue/agent issues. |
| 17:02.09 | pigpen | in fact, we abandoned all queues due to these in 1.4 years ago. |
| 17:02.09 | leifmadsen | WindBack: just request more than one column in your select statement then.... |
| 17:02.43 | leifmadsen | Set(ARRAY(Column_One,Column_Two)=${ODBC_MULTI_COLS()}) |
| 17:02.44 | pigpen | thurin, in general, IAX is very good regarding disconnection notification. |
| 17:03.29 | pigpen | thurin, so my bet would be your issues are more Queue/Agent issues than IAX. |
| 17:03.45 | pigpen | thurin, and you will quickly hear this?.upgrade. |
| 17:04.25 | thurin | pigpen, thanks, i'll take a look. I was focusing on IAX since the same agent connecting with SIP vs IAX works fine. |
| 17:05.15 | pigpen | thurin, yeah, it may be an IAX issue, but with so many Queue/Agent issues, it is an open field. |
| 17:05.34 | pigpen | There were also a ton of IAX fixes since then too. |
| 17:05.39 | pigpen | what ver of 1.4 ? |
| 17:05.45 | *** join/#asterisk navaismo (~navaismo@fixed-203-96-202.iusacell.net) |
| 17:05.57 | pigpen | man, I haven't even typed "1.4" for a very long time. |
| 17:06.19 | thurin | 1.4.20.1 |
| 17:06.48 | pigpen | at least it is one of the later ver's. |
| 17:08.24 | [TK]D-Fender | No, 1.4.42 <- |
| 17:08.34 | [TK]D-Fender | not even half-way there... |
| 17:09.13 | pigpen | heh. I think I went to 1.6 mid game and never looked back. |
| 17:12.13 | pigpen | [TK]D-Fender, fyi, I have been running very nicely on 10.1.2 |
| 17:12.21 | pigpen | thus far, very stable. |
| 17:17.06 | paolosupino | I'm looking for implementation ideas for the following problem: have groups of extensions that in the absence of the extension it rings at the head of the group… Any ideas how to implement such an idea? |
| 17:32.35 | [TK]D-Fender | paolosupino, Please rephrase. |
| 17:36.26 | paolosupino | I was asked to create extensions 20[00-30], 21[00-30], 22[00-30], 23[00-30] etc, but the company at the moment has only enough phone for extension 2000, 2100, 2200, 2300, 2400 etc.. so I was asked to setup the extension now in a manner that all the extensions already exists, but in the case they aren't registered in the PBX to forward the call to 2x00 number… |
| 17:40.24 | aberrios | anyone use res_cepstral and the lexicon.txt file? Running swift with -l takes notice but res_cepstral doesn't seem to. |
| 17:41.14 | aberrios | ignore that. module reload res_cepstral sorted it |
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| 17:43.06 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 17:45.44 | [TK]D-Fender | paolosupino, "core show application chanisavail", "core show dunction DEVICE_STATE" <- check before dialing. |
| 17:46.31 | paolosupino | [TK]D-Fender: Thanx :-) |
| 17:46.43 | paolosupino | [TK]D-Fender: will check… |
| 17:49.36 | *** join/#asterisk remuda (~Tim@173.243.243.172.rhinocommunications.net) |
| 17:55.40 | *** join/#asterisk FLeiXiuS (~FLeiXiuS@unaffiliated/fleixius) |
| 17:55.53 | FLeiXiuS | Is there a fullproof way of checking on the status of a call made through AMI? |
| 17:56.16 | p3nguin | fullproof? |
| 17:56.24 | p3nguin | Do you possibly mean fool proof? |
| 17:57.34 | [TK]D-Fender | FLeiXiuS, Set a unique channel variable when you create it |
| 18:00.57 | FLeiXiuS | [TK]D-Fender, From there I can query to see if the call was made? My problem now is Im sending an originate through AMI - sometimes it works and sometimes it doesnt |
| 18:01.49 | *** join/#asterisk twodogs (~twodogs@telok.vitiate.me) |
| 18:02.08 | FLeiXiuS | Either through Status or? |
| 18:04.34 | FLeiXiuS | Let me try status with a unique ActionID - I'll get back to you |
| 18:09.29 | *** join/#asterisk wonderworld (~ww@dsdf-4db55bb7.pool.mediaWays.net) |
| 18:10.27 | paolosupino | [TK]D-Fender: From reading www.voip-info.org I don't see that DEVICE_STATE unmonitored can be checked… which is what I see in sip show peers for extensions that were never connected before. How do I check those? |
| 18:11.10 | *** join/#asterisk hehol (~Adium@2a01:198:71d:0:21f:d0ff:fea1:568e) |
| 18:13.34 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 18:16.43 | paolosupino | I have another question: How do I put the '+' sign in dial plan? |
| 18:17.28 | paolosupino | for international dialing… |
| 18:17.37 | *** join/#asterisk jofry (~procube@port-212-202-156-85.static.qsc.de) |
| 18:18.31 | *** join/#asterisk its_jeremy_ (~omghax@gateway/tor-sasl/itsjeremy/x-75806909) |
| 18:19.40 | *** join/#asterisk nighty- (~nighty@static-68-179-124-161.ptr.terago.net) |
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| 18:24.42 | [TK]D-Fender | paolosupino, You just put it in. |
| 18:25.48 | *** join/#asterisk bkruse (~bkruse@75.76.105.124) |
| 18:26.14 | leifmadsen | paolosupino: you just put it in the dialplan :) |
| 18:26.27 | leifmadsen | exten => _+011.,1,NoOp(Ohai) |
| 18:33.57 | ruben23 | <PROTECTED> |
| 18:38.04 | *** join/#asterisk bkruse (~bkruse@75.76.105.124) |
| 18:38.37 | ruben23 | hi guys i installed asterisk add_on to implement cdr +mysql on asterisk, problme after i install asterisk addons i get this errro----> http://pastebin.com/f3mEPzU2 , they say the module is not install and during the asterisk addon install i make menuselect and found this options only---> http://pastebin.com/AiDp2ywD ---> are this not enough, im i lacking something..? |
| 18:39.23 | ruben23 | any idea guys who implemented cdr+mysql |
| 18:40.12 | [TK]D-Fender | ruben23, Module 'cdr_mysql.so' could not be loaded. |
| 18:40.21 | [TK]D-Fender | ruben23, [LD] cdr_addon_mysql.o -> cdr_addon_mysql.so |
| 18:40.30 | [TK]D-Fender | ruben23, Those files do not look at all the same to me. |
| 18:40.39 | [TK]D-Fender | ruben23, Maybe you should look a lot closer at your actual files. |
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| 19:07.45 | flyingbull | Did you do a configure before doing the menuselect? |
| 19:08.01 | flyingbull | Sorry, that was meant for Ruben. |
| 19:11.32 | *** join/#asterisk LemensTS (~matthew@adsl-70-238-135-101.dsl.stlsmo.sbcglobal.net) |
| 19:11.51 | LemensTS | Can you transfer voicemail messages from one mailbox to another in Asterisk 1.10? |
| 19:13.48 | LemensTS | I mean asterisk 10 |
| 19:17.44 | [TK]D-Fender | Same as every version |
| 19:20.14 | paolosupino | leifmadsen: I did _+., but it didn't accept it… |
| 19:20.30 | leifmadsen | LemensTS: yes |
| 19:20.38 | leifmadsen | paolosupino: then you'll have to show why it didn't work, but it will |
| 19:22.01 | paolosupino | I think I know the answer: There were 2 lines before it in the context: the first one: _00,1,callerid |
| 19:22.25 | paolosupino | the second line: _00,n,dial(sip/...) |
| 19:22.55 | paolosupino | and then I added _+.,n,dial(sip/...) |
| 19:23.21 | *** join/#asterisk nosaj (~jbarinas@gwb.anditel.com.co) |
| 19:23.24 | paolosupino | which would simply not work… |
| 19:24.13 | [TK]D-Fender | paolosupino, You can't just use "n" You have no "1" priority to start the use of that pattern |
| 19:24.18 | WindBack | leifmadsen: thanks for the explanation: Set(ARRAY(Column_One,Column_Two)=${ODBC_MULTI_COLS()}) |
| 19:24.20 | j0b | anyone in here that is running asterisk in qemu/kvm machine? |
| 19:24.29 | j0b | and does it work well for you? |
| 19:24.47 | leifmadsen | paolosupino: right, that won't work |
| 19:24.54 | paolosupino | [TK]D-Fender: looking at it now it seemed obvious that I need 1 for it… |
| 19:24.59 | leifmadsen | you need to have a priority 1 for every extension |
| 19:25.00 | WindBack | leifmadsen: I didn't see it in the book |
| 19:25.51 | paolosupino | leifmadsen: for some reason at 2AM it didn't seem like it's a new extension ;-) |
| 19:25.55 | leifmadsen | WindBack: it'st here |
| 19:27.05 | paolosupino | works :-) |
| 19:27.21 | paolosupino | thanx [TK]D-Fender and leifmadsen :-) |
| 19:27.48 | leifmadsen | WindBack: http://asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/getting_funky.html |
| 19:29.25 | WindBack | leifmadsen: thanks |
| 19:29.37 | [TK]D-Fender | [13:00] <[TK]D-Fender> WindBack, http://astbook.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/getting_funky.html |
| 19:29.50 | [TK]D-Fender | Where did those 2.5 hours just go... |
| 19:34.45 | *** join/#asterisk alexoc (~alexoc@187.106.48.119) |
| 19:35.03 | alexoc | what is the better method to do a dinamic ivr based on web? agi or generate confs ? |
| 19:35.10 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 19:35.40 | [TK]D-Fender | alexoc, What does "based on web" really mean? |
| 19:37.58 | alexoc | [TK]D-Fender: thanks for the help, I was not clear, sorry, need to develop an ivr that can be configured by web, using php. Do not know what is more advantageous: to use agi or generate conf files. |
| 19:38.49 | [TK]D-Fender | alexoc, Suppose it depends what you're going to actually allow it to do. |
| 19:38.58 | paolosupino | can I put template in files that are included in sip.conf? |
| 19:40.29 | leifmadsen | alexoc: there are several methods... you can use func_odbc to lookup info from the database and build a flat file where the data is dynamic. You could do a similar thing with func_curl but retrieve data via an http backend. The other method is to program it outside of asterisk using some other language, and call it from FastAGI |
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| 19:43.17 | alexoc | [TK]D-Fender: the User should be able to create as many levels you want in IVR. I believe that generate files Factory settings is very costly. On the other hand, do not know what the performance of IVR by AGI, estimated peak of about 50 simultaneous calls incoming. |
| 19:44.00 | leifmadsen | do you mean IVR or auto-attendant |
| 19:45.52 | Who-m3 | Anyone here got experience with a Asterisk + Freeswitch combo? Trying to get Asterisk and FreeSwitch to work together to allow me to dial from ext 109 through a second google voice number (fs is my gvoice/gtalk interface). |
| 19:46.16 | ajg | okay I get those two mixed up.. IVR is go through a menu to direct to the right destination, and auto-attendant is like a program to pull up info and such (like say account balance or whatnot), correct? |
| 19:46.17 | alexoc | leifmadsen: thanks for the help, ivr, i need external db queries, you said `The Other method is to program it outside of using asterisk Some other language, and call it from FastAGI` I thought about using php phpagi, there are some project that I can use as a starting point? The performance of IVR using AGI is well below the conf files? |
| 19:47.49 | WindBack | leifmadsen: I dont see the function ODBC_MULTI_COLS() in this chapter. Also I dont see this function in my asterisk version (1.8.7). Is it on asterisk 10? |
| 19:48.16 | leifmadsen | WindBack: you build the functions in func_odbc.conf -- it was just a made up function name |
| 19:48.38 | leifmadsen | the only requirement is that you use a SELECT statement that returns more than 1 column to get multiple columns |
| 19:48.53 | leifmadsen | alexoc: sorry, unknown |
| 19:49.05 | *** join/#asterisk Gaiax (~Gaiax@unaffiliated/gaiax) |
| 19:49.36 | leifmadsen | ajg: http://asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-AA-SECT-1.html |
| 19:50.23 | ajg | aha |
| 19:50.27 | alexoc | ajg: a definicao é contrario ao que voce disse. |
| 19:50.31 | ajg | I did have them backwards :) |
| 19:50.36 | alexoc | ajg: the definition is contrary to what you said. |
| 19:50.40 | WindBack | leifmadsen: Ok, Now I understand. the trick is in ARRAY function |
| 19:51.04 | ajg | thanks! |
| 19:51.24 | WindBack | leifmadsen: for any function in func_odbc which returns more than I column I should get it using ARRAY... Thanks for the help |
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| 19:54.34 | alexoc | leifmadsen: thanks! |
| 19:54.43 | LemensTS | What is the most efficient way to pattern match 800-814 ? |
| 19:56.58 | [TK]D-Fender | LemensTS, _8{01][01234] |
| 19:57.06 | [TK]D-Fender | LemensTS, _8[01][01234] |
| 19:57.42 | LemensTS | TKD: That wouldnt match 805-809 would it? |
| 19:58.05 | [TK]D-Fender | oops |
| 19:58.16 | [TK]D-Fender | 2 patterns required |
| 19:58.36 | LemensTS | Ok. I couldn't figure a way to do it in 1 either. |
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| 20:52.02 | *** join/#asterisk [sr] (~kvirc@pal-213-228-140-150.netvisao.pt) |
| 20:55.52 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 21:09.46 | paolosupino | I keep getting the message: doing dnsmgr_lookup for 'sip.voipcheap.com' printed on my asterisk console. How do I disable it? |
| 21:12.14 | leifmadsen | paolosupino: you don't unless you alter the code -- it's just a notification message |
| 21:12.32 | paolosupino | leifmadsen: annoying once at that… |
| 21:12.37 | leifmadsen | sure |
| 21:12.58 | *** join/#asterisk timahvo1 (~rogue@197.176.36.177) |
| 21:36.34 | *** join/#asterisk Maliuta (~nobusines@kiev.lusan.id.au) |
| 21:37.32 | [sr] | going to sleep |
| 21:46.45 | Kobaz | Asterisk 1.4.21.2. System uptime: 28 weeks, 5 days, 7 hours, 41 minutes, 19 seconds |
| 21:46.49 | Kobaz | hehe. still chugging along |
| 21:47.16 | *** join/#asterisk TimeRider (~steve@188-220-34-144.dsl.cnl.uk.net) |
| 21:52.05 | *** join/#asterisk thurin (~nwykes@174-29-209-91.hlrn.qwest.net) |
| 21:52.40 | *** part/#asterisk flyingbull (~Adium@cpe-065-190-158-078.nc.res.rr.com) |
| 21:54.31 | *** join/#asterisk timahvo1 (~rogue@197.176.36.177) |
| 21:55.02 | *** join/#asterisk dijib (~root@bas10-kitchener06-1279682199.dsl.bell.ca) |
| 21:55.11 | dijib | howdy ya'all |
| 21:55.59 | dijib | so how do i load an asterisk server that was build from source with all the .so wanted/available? |
| 21:58.52 | paolosupino | how do I put DEVICE_STATE = UNKNOWN in a GotIf statement? |
| 21:59.27 | devdvd | dijib, im not sure i understand your question. You asking how to load all the modules in? |
| 21:59.32 | devdvd | when you build it |
| 21:59.41 | devdvd | if so, try running make menuselect |
| 21:59.49 | devdvd | then you can select all the modules you want to add |
| 22:00.29 | kaldemar | paolosupino: do you mean how to check if DEVICE_STATE is UNKNOWN? |
| 22:00.41 | paolosupino | kaldemar: yes. |
| 22:01.21 | kaldemar | paolosupino: GotoIf($["${DEVICE_STATE(device)}" = "UNKNOWN"]?true:false) |
| 22:02.06 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
| 22:04.48 | *** join/#asterisk jibcage (~jack@99-118-153-239.lightspeed.stlsmo.sbcglobal.net) |
| 22:05.07 | jibcage | hello! is there any way that I can detect in my dialplan if a call is being forwarded to my pbx? |
| 22:09.07 | [TK]D-Fender | depends |
| 22:09.26 | jibcage | on what? |
| 22:10.15 | *** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek) |
| 22:11.18 | [TK]D-Fender | On the nature of the call coming in of course |
| 22:12.26 | *** join/#asterisk classix (salven@silenceisdefeat.com) |
| 22:12.30 | jibcage | I want asterisk to point a caller to a voicemail box when somebody calls my phone and my phone forwards the call over |
| 22:12.46 | [TK]D-Fender | Still doesn't show anything about the call itself |
| 22:13.04 | jibcage | I don't understand |
| 22:13.20 | [TK]D-Fender | what KIND of call is it? |
| 22:13.39 | jibcage | oh, the number is forwarded to a did, which then rings asterisk over IAX2 |
| 22:13.49 | jibcage | *call, not numbre |
| 22:14.24 | [TK]D-Fender | Where was the call being sent to originally? What service is it being forwarded to? |
| 22:16.35 | jibcage | The call was originally being sent to my cell phone, at which point, after 4 rings, AT&T would ring my DID, which would then ring my server over IAX2. |
| 22:16.58 | jibcage | I hope you pardon my ignorance if I'm missing any crucial information, I'm fairly new to this |
| 22:17.20 | *** join/#asterisk [netman] (~netman@40.224.76.188.dynamic.jazztel.es) |
| 22:18.11 | [TK]D-Fender | jibcage: getting better.... You'll need to look at the IAX2 debug to see if the carrier is passing it on. |
| 22:19.05 | jibcage | Passing what on? The forwarding information? How do I view the IAX2 debug log? |
| 22:19.23 | jibcage | Not much is in /var/log/asterisk except for messages, which isn't that helpful |
| 22:19.45 | [TK]D-Fender | iax set debug |
| 22:20.01 | jibcage | do I run that from the asterisk cli? |
| 22:20.05 | [TK]D-Fender | yes |
| 22:20.23 | jibcage | "no such command" |
| 22:20.32 | [TK]D-Fender | iax set debug on |
| 22:20.34 | [TK]D-Fender | should be it |
| 22:20.39 | kaldemar | iax2 set debug on |
| 22:20.48 | jibcage | there we go |
| 22:20.51 | jibcage | thanks! :) |
| 22:20.58 | jibcage | now let's see |
| 22:22.22 | *** join/#asterisk jofry (~procube@port-212-202-156-85.static.qsc.de) |
| 22:22.41 | *** join/#asterisk netman (netman@40.224.76.188.dynamic.jazztel.es) |
| 22:25.13 | *** join/#asterisk gunnarar (~gunnarar@96.49.96.81) |
| 22:25.58 | paolosupino | I have the following exempt: exten => _22XX,1,verbose(SIP/${EXTEN} state is ${DEVICE_STATE(SIP/${EXTEN})}) same => _22XX,n,GotoIf($["${DEVICE_STATE(SIP/${EXTEN})}" = "UNKNOWN"]?dial_head) |
| 22:25.59 | paolosupino | <PROTECTED> |
| 22:25.59 | paolosupino | <PROTECTED> |
| 22:25.59 | paolosupino | <PROTECTED> |
| 22:26.24 | gunnarar | I have a digium analog card with 4 channels, 2 of which work fine but 2 show up as "inAlarm: 1" when I do "dahdi show channels" in cli |
| 22:26.38 | paolosupino | when I try to dial an extension that is connected I get fall through because my extension is in state UNKNOWN. Why? |
| 22:27.56 | gunnarar | not seeing why that is, lines seem fine (swithced them around), this has worked before, restarted amportal and dahdi |
| 22:30.07 | gunnarar | how do I find out what this alarm is telling me? |
| 22:31.09 | jibcage | [TK]D-Fender: How can I access the IAX2 debug? |
| 22:31.28 | kaldemar | paolosupino: do you have such a device defined in sip.conf? |
| 22:31.35 | [TK]D-Fender | jibcage: we're you already looking at it? |
| 22:32.51 | paolosupino | kaldemar: yes, I have both 22xx extensions and 2305 extension defined in sip.conf and I have multiple soft phones configured on my laptop to mimic actual use… |
| 22:34.52 | kaldemar | paolosupino: show a CLI output of a call and sip.conf |
| 22:35.11 | kaldemar | ~pb |
| 22:35.12 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 22:35.21 | jibcage | [TK]D-Fender: http://pastebin.com/Z07E99Hh |
| 22:36.41 | paolosupino | kaldemar: sip.conf for extension 2105: |
| 22:37.05 | [TK]D-Fender | jibcage: doesn't look like a complete call in there |
| 22:37.08 | ruben23 | hi guys i installed asterisk add_on to implement cdr +mysql on asterisk, problme after i install asterisk addons i get this errro----> http://pastebin.com/f3mEPzU2 , they say the module is not install and during the asterisk addon install i make menuselect and found this options only---> http://pastebin.com/AiDp2ywD ---> are this not enough, im i lacking something..? |
| 22:37.12 | paolosupino | kaldemar: [2105](defaults,nat-opts) |
| 22:37.12 | paolosupino | secret=ePHUuRqdYZ |
| 22:37.39 | paolosupino | kaldemar: templates are: [defaults](!) |
| 22:37.40 | paolosupino | type=friend |
| 22:37.40 | paolosupino | host=dynamic |
| 22:37.40 | paolosupino | context=phones |
| 22:37.41 | paolosupino | DirectMedia=no |
| 22:37.41 | paolosupino | [nat-opts](!) |
| 22:37.42 | paolosupino | nat-opts=yes |
| 22:37.42 | paolosupino | qualify=yes |
| 22:37.46 | kaldemar | paolosupino: stop |
| 22:37.58 | ruben23 | sorry guys i missed the messge about this |
| 22:38.12 | ruben23 | kaldemar: hi there |
| 22:38.21 | kaldemar | paolosupino: 1. do NOT paste here, read about pastebin above. 2. [2105] has nothing to do with your extension. |
| 22:38.26 | jibcage | [TK]D-Fender: my bad. http://pastebin.com/wmAnyucC |
| 22:38.52 | paolosupino | kaldemar: give me 5 minutes and I will paste into paste bin. |
| 22:39.18 | paolosupino | kaldemar: sorry about pasting here :/ |
| 22:39.43 | [TK]D-Fender | jibcage: still doesn't look quite right.... also make sure to be at verbose 10 |
| 22:39.51 | ruben23 | on my make menu select i dont see any cdr_mysql module being need to eb selecetd...any idea, why its not included i got version ---> asterisk-addons-1.4.13 |
| 22:40.23 | kaldemar | ruben23: you should be loading cdr_addon_mysql.so, not cdr_mysql.so. |
| 22:40.27 | jibcage | [TK]D-Fender: how do I set that? |
| 22:40.27 | ruben23 | kaldemar: any idae and hel somehow |
| 22:41.26 | kaldemar | ruben23: cdr_addon_mysql.so has been renamed to cdr_mysql.so in a newer version than yours. |
| 22:42.26 | ruben23 | kaldemar: yes its already seelcetd when i check the menuselect, but after install and i did ---> module load cdr_mysql.so i get this errror-----> http://pastebin.com/f3mEPzU2 |
| 22:42.59 | kaldemar | ruben23: read my two previous comments again. especially the first one. |
| 22:44.22 | paolosupino | kaldemar: http://pastebin.com/SgCUn0hx |
| 22:45.07 | [TK]D-Fender | kaldemar: I told him this earlier |
| 22:45.20 | paolosupino | kaldemar: I know that initially I wrote about 22xx but 22xx and 21xx are configured the same in the dial plan (and I get the same result when I try to call from 2305). |
| 22:45.44 | ruben23 | kaldemar; i got this when i type on teh CLI----> WARNING[23037]: loader.c:711 load_resource: Module 'cdr_addon_mysql.so' already exists |
| 22:45.57 | kaldemar | [TK]D-Fender: maybe a little repetition is required. :) |
| 22:46.06 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-krepfygqbsqyrifr) |
| 22:46.39 | paolosupino | kaldemar: BTW: I also have extension 2500 which is a simple echo application and that one hangs up too and an extension 2999 which is an IAX soft phone and that one doesn't hangup. |
| 22:46.46 | paolosupino | kaldemar: thank you. |
| 22:46.48 | kaldemar | paolosupino: and you wrote that your device state is UNKNOWN, but it is UNAVAILABLE instead. set callcounter=yes and busylevel=1 for the peer. |
| 22:46.59 | jibcage | [TK]D-Fender: debug set to 10. http://pastebin.com/wuX4buNv |
| 22:47.12 | ruben23 | <PROTECTED> |
| 22:48.44 | kaldemar | ruben23: configure it in cdr_mysql.conf |
| 22:49.32 | kaldemar | ruben23: you'll find a sample under configs/ in the addon tarball. |
| 22:50.52 | *** join/#asterisk Sktenau (~textual@monitor.ic.ipera.net.au) |
| 22:51.02 | *** join/#asterisk Maliuta (~nobusines@kiev.lusan.id.au) |
| 22:54.00 | ruben23 | <PROTECTED> |
| 22:55.03 | ruben23 | remove the sample* word by renaming it.. |
| 22:55.08 | *** join/#asterisk timahvo1 (~rogue@197.176.36.177) |
| 22:56.01 | kaldemar | ruben23: and edit the contents. |
| 22:57.14 | ruben23 | kaldemar: thanks ill try now |
| 22:58.01 | gunnarar | I have a digium analog card with 4 channels, 2 of which work fine but 2 show up as "inAlarm: 1" when I do "dahdi show channels" in cli, not seeing why that is, lines seem fine (swithced them around), this has worked before, restarted amportal and dahdi. |
| 22:58.19 | gunnarar | can anyone help me with that? |
| 23:01.06 | ruben23 | kaldemar: can i use root user and passord for teh datasbed user on cdr table access..?is it possible |
| 23:02.44 | kaldemar | ruben23: don't do it, it's bad practice. |
| 23:04.44 | paolosupino | kaldemar: what do you mean? |
| 23:05.12 | kaldemar | paolosupino: what is unclear? |
| 23:08.36 | paolosupino | kaldemar: OK I changed the unknown to unavailable in the dial plan but no change. |
| 23:10.01 | kaldemar | paolosupino: it is unavailable because your peer does not have callcounter and busylevel configured. do that and try again, you'll see a different result. |
| 23:16.24 | ruben23 | kaldemar:ok i configure cdr_mysql.conf and load the cdr_addon_mysql.so on teh modules.conf with this load => cdr_addon_mysql.so , i restart asterisk then it should be ok right, no other step, table is already created.. |
| 23:17.45 | paolosupino | kaldemar: I put callcounter=yes and and busylevel=N (where N was 0,1,3,5) in sip.conf (in th defaults template), but it didn't shane anything. |
| 23:31.45 | *** join/#asterisk nix8n82 (~backbox@65.161.180.230) |
| 23:36.50 | paolosupino | kaldemar: any thoughts? |
| 23:43.25 | paolosupino | Everyone (and I apologize for not doing it personally): thank you for your help. |
| 23:43.30 | paolosupino | good night… |
| 23:44.02 | *** part/#asterisk paolosupino (~paolo-sup@net-2-38-113-188.cust.dsl.vodafone.it) |
| 23:46.28 | *** join/#asterisk rx4change (~Joshua@216.241.45.82) |
| 23:47.10 | rx4change | I am having an odd Asterisk 10.2.1 issue. Straight build from source, NO sip traffic getting to Asterisk itself |
| 23:47.20 | rx4change | netstat shows that I'm listening on 5060 |
| 23:47.22 | tzanger | just out of curiosity... where would I find a ground start PSTN connection? |
| 23:47.28 | rx4change | ngrep shows I'm actually getting connections through |
| 23:47.31 | rx4change | anyone seen this?? |
| 23:47.53 | rx4change | selinux off, firewall disabled... nothing in full debug logs - totally stumped |
| 23:49.05 | [TK]D-Fender | rx4change: show us |
| 23:49.07 | [TK]D-Fender | ~pb |
| 23:49.07 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 23:50.49 | rx4change | what do you want to see? |
| 23:50.54 | rx4change | i ngrep the socket, I see traffic |
| 23:51.01 | rx4change | no response from Asterisk whatsoever |