| 00:31.01 | pdtpatrick | p3nguin, yeah i've setup several asterisk boxes .. but i'll say im intermediate if anything. I've done things like VoiceXML and other fancy things like festival .. but i'd like to try out calendaring on 1.85 |
| 00:31.04 | pdtpatrick | that'd be fun |
| 00:33.40 | *** join/#asterisk santialel109 (~santialel@190.62.210.132) |
| 00:34.35 | santialel109 | Can some one help with some cdr bug for asterisk 1.4.42? |
| 00:34.54 | p3nguin | If you don't describe a problem, no. |
| 00:37.26 | santialel109 | OK, here is the problem, I hvae not identify in which version this bugs appears, but when transfer a call with in queues the userfield on the cdr store the file name for each recording on the queue, but on version 1.4.42 it only stores the last call record |
| 00:38.08 | p3nguin | Can you show me an example of one that you feel is correct, then show me one that you feel is incorrect? |
| 00:39.29 | santialel109 | for example the call enters queue A and its recorded, then the call its transfer to queue b, usually the userfield stores the files name separated by ; for each queue, but now the userfield stores the file name for the recording o queue b |
| 00:40.01 | santialel109 | only the file for queue b |
| 00:40.40 | santialel109 | there no filename on userfield for queue a, which was the first queue the call was answer |
| 00:45.07 | santialel109 | here is an example for asterisk 1.4.21-2 |
| 00:45.19 | santialel109 | userfield on cdr has agent-220-1313847986-493893.gsmagent-220-1313847986-493893.gsm;agent-240-1313849181-493926.gsm |
| 00:46.01 | santialel109 | here is an example for asterisk 1.4.42 |
| 00:46.30 | santialel109 | userfield on cdr only stores the last queue recording agent-220-1313849989-493966.gsm |
| 00:47.55 | santialel109 | ? |
| 00:50.44 | santialel109 | someone read what I post? |
| 00:52.52 | santialel109 | no answer? |
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| 01:02.59 | chuckf | santialel109: I didn't see an answer |
| 01:03.41 | santialel109 | I'm waiting for one, or I need to post something else? |
| 01:04.18 | chuckf | I'd say you have to keep waiting for one |
| 01:05.15 | santialel109 | thanks, hope to get one |
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| 02:57.03 | penguin | ok this is suck a newbbie question but i cant figure it out. My sip registration keepts registering with my provider every 30 seconds. i already set registertimeout=700 but it seems to ignore that |
| 02:57.07 | penguin | and ideas? |
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| 03:20.40 | brad_mssw | got an odd issue with asterisk 1.8 I just installed. When a voicemail is recorded, the duration reported is off, like if the duration is really 25s, it shows 8s. Normally this wouldn't be an issue, but it doesn't record a voicemail less than '4' seconds, but with the odd calculation a voicemail of 10-15s falls under that criteria |
| 03:22.10 | brad_mssw | I see other reports googling for asterisk 1.8 voicemail duration wrong ... but no resolution or confirmation |
| 03:27.09 | brad_mssw | 1.8.6.0-rc1 btw |
| 03:35.11 | p3nguin | Does it happen in 1.8.5.0? |
| 03:55.20 | penguin | does anyone have any idea how i can control the time my asterisk registers with my sip provider? |
| 03:55.32 | penguin | im using asterisk 1.8.5.0 |
| 03:57.03 | ChannelZ | What do you mean control the time it registers |
| 04:02.35 | *** join/#asterisk Tech_Travis (~Travis@cpe-76-168-191-127.socal.res.rr.com) |
| 04:03.46 | Tech_Travis | is non-md5 authentication still allowed in 1.8.5.0 for SIP and IAX? |
| 04:04.34 | penguin | yes control the time it registers with my provider flowroute |
| 04:05.01 | penguin | it keeps ast_get_serv: SRV lookup for sip |
| 04:05.06 | penguin | every 30 seconds |
| 04:05.27 | penguin | i added registertimeout=700 in sip.conf but apparently it still registering every 30 seconds |
| 04:13.09 | p3nguin | Just because it does a lookup every 30 seconds doesn't mean it is registering every 30 seconds. Does it? |
| 04:13.46 | p3nguin | With sip debug enabled, do you see a REGISTER every 30 seconds? |
| 04:15.29 | *** join/#asterisk tyman_ (~tyler@173-12-219-189-Fresno.hfc.comcastbusiness.net) |
| 04:24.51 | penguin | p3nguin: it just shows > ast_get_srv: SRV lookup for '_sip._udp.sip.flowroute.com' mapped to host sip.flowroute.com, port 5060 |
| 04:25.05 | p3nguin | Then that isn't a registration. |
| 04:25.25 | penguin | p3guin: oh i thought that it was registering becuz it keeps showing that every 30 seconds on cli |
| 04:25.25 | p3nguin | If sip debug doesn't show a REGISTER, it's not registration. |
| 04:25.38 | penguin | p3nguin: so what is that? what is it trying to do? |
| 04:25.47 | p3nguin | Just like if there is no INVITE, you didn't get a call. |
| 04:26.17 | penguin | p3nguin: makes sense. i just didnt want my sip provider to get mad becuz i thought it was sending sip registration requests all those times |
| 04:26.21 | p3nguin | I guess the DNS record has told you that the record is only valid for 30 seconds, which makes you look it up again after 30 seconds. |
| 04:26.55 | penguin | so the dns is telling my server is only valid for that length of time then asterisk tries find find dns host again |
| 04:26.57 | penguin | correct? |
| 04:27.10 | p3nguin | That's my guess, without investigating it for myself. |
| 04:27.23 | penguin | yea it seems to make more sense now |
| 04:28.02 | p3nguin | Resource records have a TTL. When the time expires, another lookup should be performed for potentially new and different data. |
| 04:28.29 | penguin | exactly |
| 04:28.31 | penguin | kool |
| 04:28.31 | penguin | thanks |
| 04:30.23 | p3nguin | It looks like the TTL for sip.flowroute.com is 14400 seconds. |
| 04:30.35 | p3nguin | I don't find any SRV for them. |
| 04:31.36 | penguin | yea well its doing it more than 14400 seconds |
| 04:31.38 | penguin | weird |
| 04:35.25 | p3nguin | That was my only idea. Maybe someone else know with certainty. |
| 04:41.51 | *** join/#asterisk godmachine-x6 (~godmachin@unaffiliated/godmachine-x6) |
| 04:48.58 | ChannelZ | penguin: change dnsmgr.conf |
| 04:49.05 | ChannelZ | refreshinterval |
| 04:49.23 | ChannelZ | or turn it off if the IP never changes |
| 04:49.54 | penguin | channelz: ok so then it wasnt registering it was just doing a dns lookup every 30 seconds? |
| 04:50.21 | p3nguin | We've already established that it was not registering. |
| 04:50.25 | ChannelZ | maybe. |
| 04:50.29 | p3nguin | (2325.25) <p3nguin> If sip debug doesn't show a REGISTER, it's not registration. |
| 04:51.10 | ChannelZ | I'm not sure if the SRV lookups, given no other activity, fall under DNS manager. |
| 04:51.11 | penguin | p3nguin: ok kool no i believed you i just like to be re-assured |
| 04:51.12 | penguin | sorry |
| 04:51.44 | p3nguin | The part I was not sure about was the reason DNS lookups were being done every 30 seconds. |
| 04:52.14 | ChannelZ | it seems like something else |
| 04:53.17 | ChannelZ | A qualify expiry or something, though I don't know why it keeps lookup up SRV records.. something I haven't seen before. It might help to see the peer entry in sip.conf. Could always turn off SRV lookups if you don't need them too |
| 04:53.51 | penguin | channelz: do i need srv lookups enabled? i currently have it set to =yes |
| 04:54.04 | p3nguin | It doesn't happen on 1.4 versions (which I use), but I see it on 1.8 installs continuously. |
| 04:54.20 | penguin | p3nguin: yes im using 1.8 maybe thats y |
| 04:54.33 | ChannelZ | penguin: unless you're dialing random SIP URIs a lot, chances are no you don't need it |
| 04:54.53 | p3nguin | There was an obvious change, but since I don't use 1.8, I haven't learned why it does things like that. |
| 04:55.32 | ChannelZ | I have srv lookups on but don't get such things |
| 04:57.08 | penguin | channelz: ok i turned it to =no so lets see if i get that ast_serv_lookup |
| 04:57.21 | penguin | ok i reload and got this: > doing dnsmgr_lookup for 'sip.flowroute.com' |
| 04:57.26 | penguin | thats is normal correct? |
| 04:57.33 | ChannelZ | do these messages come out at random or only when calls are placed? |
| 04:57.44 | p3nguin | They show up no matter what. |
| 04:57.54 | p3nguin | periodically |
| 04:58.06 | penguin | yea they just show up randomly |
| 04:58.15 | penguin | nothing to do with placing or receiving calls |
| 04:58.30 | penguin | i do not see it ast_serv_lookup anymore after i turned srvlookup=no |
| 04:58.36 | ChannelZ | Huh. I just turned on qualify too for a peer and still don't get it |
| 04:58.49 | ChannelZ | wonder what is triggering it to do the lookup |
| 04:59.14 | p3nguin | <PROTECTED> |
| 04:59.29 | p3nguin | It shows up at core verbose 4 (and above). |
| 04:59.29 | penguin | channelz: seems to not be doing it anymore.. i dont know if its because i set it to no or if its is what i did in the dnsmaner.conf |
| 04:59.47 | ChannelZ | well the dnsmgr lookup is different from a SRV lookup |
| 04:59.47 | penguin | p3nguin: yes so thats normal then |
| 04:59.57 | p3nguin | That's what I've been saying. |
| 05:00.11 | penguin | p3nguin: ok...but the srv lookup thing was weird |
| 05:00.18 | p3nguin | It's not weird. |
| 05:00.28 | penguin | p3nguin: seems to not be doing it anymore after i set srvlookup=no |
| 05:00.30 | p3nguin | VoIP uses SRV recors when configured to do so. |
| 05:02.17 | x86 | SRV records are the best thing since sliced bread |
| 05:02.32 | p3nguin | Are you sure? |
| 05:02.34 | penguin | what is everyone's opinion on billing software for asterisk? agilevoice? |
| 05:02.40 | penguin | a2billing sux |
| 05:03.08 | x86 | penguin: best thing to do is just give away the service for free ;) |
| 05:03.38 | penguin | x86: sorry cant im a reseller for carrier grade termination...i do offer free sip to sip tho ') |
| 05:03.43 | penguin | ;) |
| 05:04.55 | p3nguin | Now that I'm watching, I'm seeing a lookup for the ITSP at 1 min 45 seconds. |
| 05:05.20 | p3nguin | But I'm seeing another host getting a lookup much quicker -- like 20 seconds. |
| 05:05.32 | penguin | p3nguin: thats what i mean |
| 05:05.50 | penguin | p3nguin: even tho i set it to lookup up every 700 seconds its doing it every 30 |
| 05:10.33 | djuhl30 | woohoo asterisk runs in Xen |
| 05:12.13 | p3nguin | Like, in a guest? |
| 05:17.32 | penguin | has anyone used agilevoice for billing? |
| 05:17.39 | penguin | anything other than a2billing |
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| 05:39.20 | penguin | ??? |
| 05:39.25 | penguin | not everyone at once plz |
| 05:55.48 | x86 | penguin: how do you compete with Google Voice? they offer free VoIP termination to the US and Canada |
| 05:56.02 | x86 | penguin: and it intergrates quite nicely with Asterisk |
| 05:57.15 | penguin | x86: just because theres google voice doesnt mean ppl dont purchase minutes from other companies |
| 05:57.28 | penguin | look at skype, vonage, magic jack |
| 05:57.57 | carrar | Hows that google voice support |
| 05:58.05 | penguin | google voice isnt always gunna be around |
| 05:58.10 | x86 | carrar: works really, really well |
| 05:58.26 | penguin | mind you they already announced they will not be offering free outbound service in 2013 |
| 05:58.45 | x86 | true, but there will be more :) |
| 05:58.53 | x86 | free VSPs pop up all the time :P |
| 05:59.10 | x86 | remember dialpad.com? heh |
| 05:59.24 | penguin | hopefully |
| 05:59.36 | penguin | x86: anywayz i run a calling card service also |
| 05:59.40 | penguin | many ppl use that |
| 05:59.56 | x86 | that's also a dying market |
| 06:00.13 | x86 | no money to be made in domestic termination nor calling cards these days |
| 06:00.25 | x86 | only way to make money in termination is internation markets |
| 06:00.32 | x86 | international* |
| 06:00.33 | penguin | exactly |
| 06:00.39 | penguin | all international calling |
| 06:00.55 | coppice | only way to make money in termination is to be an assassin |
| 06:01.14 | carrar | voip assassin |
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| 06:23.01 | joako_ | Does anyone know if there's a way to use application Queue() without answering the calls? I can do Playback(file|noanswer) and it will play the sound without answering. But when the call enters the queue it is answered. |
| 06:27.46 | carrar | core show application Playback |
| 06:28.48 | carrar | doesn't look like it |
| 06:34.58 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 06:41.02 | joako_ | The solution (drum roll please): remove Answer() line in extensions.conf |
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| 08:22.48 | qakhan | hi all |
| 08:23.43 | penguin | wats uo qakhan |
| 08:24.15 | qakhan | can anyone help me to learn PHPAGI |
| 08:24.37 | qakhan | i want to learn asterisk AGI with PHP |
| 08:26.22 | penguin | get the book called asterisk the future of telepony |
| 08:26.27 | penguin | it will teach you all that |
| 08:27.33 | qakhan | yes i read this but can understand any single line |
| 08:32.14 | qakhan | can u give some other references? |
| 08:33.00 | penguin | viop.info |
| 08:33.36 | qakhan | i also tried this but ...:( |
| 08:34.30 | *** join/#asterisk singler (~singler@78-60-139-121.static.zebra.lt) |
| 08:38.51 | ChannelZ | AGi isn't that hard |
| 08:39.17 | ChannelZ | PHPAGI is kind of a prebuilt framework though that I've never used.. I just do it myself |
| 08:40.33 | qakhan | yes channelz i think soo but i m understanding its syntax |
| 08:40.55 | qakhan | i need a bit beginner level start |
| 08:41.30 | qakhan | or i need a lecture on php agi |
| 08:42.18 | ChannelZ | well I can't help you with PHPAGI specifically |
| 08:42.45 | ChannelZ | The syntax of AGI is pretty much all here: http://www.voip-info.org/wiki/view/Asterisk+AGI |
| 08:43.04 | ChannelZ | I can give you one of my little scripts as an example if you want one (again, not written under PHPAGI, just using PHP) |
| 08:43.41 | qakhan | yes sure it can help me to understand |
| 08:46.18 | ChannelZ | Here is a very simple example: http://pastebin.com/vGJ2txsG |
| 08:47.16 | ChannelZ | It just gets the CallerID of the channel it's run from, converts it to upper/lower case and re-sets the CID to it |
| 08:48.18 | ChannelZ | Just note the one preg_match() call in there is looking for "Home-PSTN" which (used to be) the name of the peer my calls came in on; if there was no CID it would just be set as the peer name from my ATA. |
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| 08:51.16 | qakhan | ya thankx channelz let me read this if i have queastion i will ask to u |
| 08:52.28 | ChannelZ | The basic concept with AGI is 1. Asterisk sends you a bunch of variables when it launches your script (over stdin); 2. You send commands over stdout, and 3. Asterisk replies back over stdin. |
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| 09:10.39 | qakhan | set_time_limit(5); |
| 09:10.39 | qakhan | error_reporting(0); |
| 09:10.39 | qakhan | $agi_in = fopen( "php://stdin", "r" ); |
| 09:10.40 | qakhan | $agi_out= fopen( "php://stdout", "w" ); |
| 09:10.54 | qakhan | what is the meaning of these line |
| 09:10.57 | qakhan | lines* |
| 09:16.23 | *** join/#asterisk jkroon (~jkroon@41.49.74.130) |
| 09:17.02 | singler | I guess you are opening stdin to read and stdout to write :) |
| 09:17.17 | jkroon | hi guys, i'm busy looking at the timerfd lockup issue. |
| 09:17.38 | WIMPy | Good man! |
| 09:18.00 | jkroon | from what i've managed to determine the only way this can happen is if either of the following happens: (a) the kernel returns a short read; or (b) no timer rate ever gets set up. |
| 09:18.27 | jkroon | in a standalone app, after timerfd_create() directly followed by a read it will block indefinitely. |
| 09:19.37 | jkroon | i'd like to know firstly if there is documentation of the "client" use of the timing code I can look at, and secondly, if there is existing debugging infrastructure for the timing code that I can use to verify that users of the timing code uses it correctly? |
| 09:19.59 | jkroon | (it looks like there is an issue with chan_sip not ever setting up the timer but then trying to _ack it which then blocks) |
| 09:20.43 | jkroon | I think (a) is unlikely as the read of the expected 8 bytes is in all probability atomic. |
| 09:24.08 | jkroon | so my suspicion is there is a mis-use bug in chan_sip, which simply gets masked by res_timing_dahdi and pthread. |
| 09:24.32 | djuhl30 | Is there anymore free training on asterisk? Well besides that book? |
| 09:25.06 | ChannelZ | qakhan: set_time_limit should keep the script from running away if something bizarre happens so it doesn't hang the call for a long time. error_reporting turns off PHP errors so it doesn't spit out anything (which probably actually doesn't matter.) |
| 09:25.28 | ChannelZ | then the two fopen lines open stdin and stdout for reading/writing to/from Asterisk. |
| 09:26.05 | WIMPy | jkroon: Do you have an idea, when that happens? |
| 09:26.19 | jkroon | WIMPy, none. |
| 09:26.56 | ChannelZ | djuhl30: a whole free book - what more do you want?! |
| 09:27.02 | jkroon | but i might be able to add code to timerfd to detect _when_ it would block and if there is infrastructure to dump core at that point. |
| 09:27.07 | *** join/#asterisk [sr] (~kvirc@pal-213-228-140-150.netvisao.pt) |
| 09:27.09 | [sr] | howdy |
| 09:27.23 | jkroon | and then just do what dahdi/pthread would do and return ... |
| 09:27.23 | WIMPy | jkroon: Sounds useful. |
| 09:27.25 | djuhl30 | Something more than the book |
| 09:27.27 | ChannelZ | oh hi |
| 09:27.43 | djuhl30 | It gets me working |
| 09:27.53 | WIMPy | [sr]: How the battle with FreePBX? |
| 09:27.53 | ChannelZ | Well there is every single .sample config file.. the wikis.. |
| 09:28.04 | [sr] | hi WIMPy |
| 09:28.16 | [sr] | yesterday went here to tell you but you weren't here |
| 09:28.22 | [sr] | i solved it, and already tested it |
| 09:28.44 | [sr] | just created an inbound route with no DID specification, with that freepbx does a catch-all |
| 09:28.55 | [sr] | and i set it to blackhole destination, and voilá |
| 09:28.59 | jkroon | not sure how to do the core dump and continue though ... |
| 09:29.04 | [sr] | problem solved |
| 09:29.16 | [sr] | after that i just set the inbound route for the DID i want to |
| 09:29.20 | djuhl30 | What wikis? The address in the topic has a bunch of cartoons |
| 09:30.01 | WIMPy | jkroon: NFI, but you don;t have to contimue, if you have a hit. |
| 09:30.22 | ChannelZ | That's because xkcd is... a web cartoon |
| 09:31.08 | ChannelZ | wiki.asterisk.org and www.voip-info.org/wiki/view/Asterisk |
| 09:31.11 | ChannelZ | GIYF |
| 09:31.23 | jkroon | WIMPy, that would need to be conditional IMHO. |
| 09:31.45 | jkroon | you don't want a production system to just crash. I'd rather just have a warning being issued. |
| 09:32.00 | jkroon | whereas if I explicitly set up a debug environment then I don't particularly care. |
| 09:32.37 | WIMPy | That was what I was thinking about. |
| 09:33.47 | jkroon | first things first - need to be able to detect the bork case... |
| 09:36.00 | jkroon | nano == 10^-9 right? |
| 09:37.17 | WIMPy | yes |
| 09:39.05 | jkroon | ok, would an extra syscall in timerfd_timer_ack be a problem? |
| 09:39.42 | jkroon | or should I rather add an extra state var to the timerfd_timer struct? |
| 09:42.58 | jkroon | WIMPy, do you know off hand if ao2_alloc clears the memory being allocated? |
| 09:43.26 | WIMPy | Not sure what you need, but if I am in theright picture, the state var should be alot faster. |
| 09:43.29 | qakhan | channelz do i need to add these four lines in every phpagi script? |
| 09:43.55 | WIMPy | No. I'm not really sure when what gets cleared. |
| 09:44.17 | WIMPy | My impression was that it's more of a reuse thing going on. |
| 09:46.09 | jkroon | rtfs then ... |
| 09:47.29 | jkroon | it seems if the rate gets set after continuous mode gets enabled, then there is a possibility that timerfd_settime(2) never gets called, in which case _ack would call read(2) blocking indefinitely waiting for a timer that will never fire. |
| 09:48.57 | WIMPy | I wonder how it happens so rarely then. |
| 09:50.39 | jkroon | agreed. |
| 09:52.45 | jkroon | semantics of _ack - is it supposed to wait for the next timer event before returning or to simply check what the last event was? |
| 09:53.13 | jkroon | a double _ack call (duplicate packet received?) might also trigger it. not sure on this one though. |
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| 09:58.22 | qakhan | channelz u there? |
| 10:00.16 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 10:02.15 | qakhan | .... |
| 10:04.55 | jkroon | WIMPy, where can I find the semantics on how the timing api is supposed to behave? |
| 10:05.31 | WIMPy | I think you should ask the guys in #asterisk-dev |
| 10:11.56 | *** join/#asterisk x1user (~x1user@host-212-75-8-69.bbccable.net) |
| 10:12.41 | x1user | Hi, how can i make AGI more verbose? |
| 10:27.59 | kaldemar | agi set debug on. use command completion with tab in CLI to get commands. |
| 10:43.47 | x1user | I got this error 511 from the AGI "Command Not Permitted on a dead channel" is this means unreachable peer? |
| 10:47.15 | *** join/#asterisk Subdolus (dexterity@creep.bur.st) |
| 10:53.04 | angryuser_laptop | x1user, deadchannels work with dead agi |
| 11:00.39 | *** join/#asterisk xnfinite (~xnfinite@176.188.223.87.dynamic.jazztel.es) |
| 11:01.24 | *** join/#asterisk qakhan (~qakhan@180.178.138.226) |
| 11:03.16 | qakhan | Channelz is there your belog on asterisk and agi php? |
| 11:03.28 | jkroon | WIMPy, #asterisk-dev seems unresponsive :p |
| 11:16.10 | *** join/#asterisk qakhan (~qakhan@180.178.146.38) |
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| 12:00.09 | x1user | When I hang up i have no sound. |
| 12:04.00 | *** join/#asterisk Subdolus (dexterity@creep.bur.st) |
| 12:05.37 | jkroon | WIMPy, I don't think timerfd complies with the spec as put out in timing.h ... |
| 12:05.45 | x1user | "511 Command Not Permitted on a dead channel" I still have this issue even with DeadAGI? |
| 12:06.18 | jkroon | so the deadlock will trigger if for whatever reason the need arrises to _ack() twice between two triggers. |
| 12:07.02 | jkroon | or one trigger gets skipped, so _ack() ... fires ... fires .. _ack() .. _ack() <-- the last _ack() will deadlock. |
| 12:09.16 | jkroon | i think this will cause a Expected to acknowledge %u ticks but got %llu instead message to be output on the first of the last two ack() calls ... |
| 12:13.27 | jkroon | not seeing any. |
| 12:14.28 | jkroon | but it will only output with core set debug >= 2 |
| 12:17.02 | *** join/#asterisk irroot (~irroot@197.170.233.204) |
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| 13:41.00 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 13:41.24 | puzzled | hi |
| 14:05.34 | *** join/#asterisk BuenGenio (~BuenGenio@cm61-10-82-188.hkcable.com.hk) |
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| 14:07.29 | *** join/#asterisk jwiggins (~James@gateway/tor-sasl/jwiggins) |
| 14:27.58 | [sr] | hei WIMPy |
| 14:29.18 | [sr] | people |
| 14:29.32 | [sr] | how can i calculate the time in CDR, that a call was ringing? |
| 14:30.04 | [sr] | CDR only has call date and duration total |
| 14:30.35 | [sr] | for unanswered calls, the duration is the time it was ringing, but for answered one's, i dont have the info, or do i? (where?) |
| 14:34.33 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 14:50.02 | brad_mssw | running asterisk 1.8.6.0-rc1 (first 1.8 series I've run, can't run 1.8.5 due to digium's build not supporting PRI for ubuntu 10.04), the voicemail duration is off. If I record a 10s voicemail, with a minimum voicemail time of 4s, I get _nothing_... Testing further, leaving a 25s voicemail, the voicemail does get logged, but as 8s (though the entire 25s message is there, just the calc is off) |
| 14:51.44 | brad_mssw | anyone know if this is something new in 1.8.6.0-rc1? 1.6.2 didn't have this issue, but I don't know if 1.8.5, for instance, had this issue or not |
| 17:04.28 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 17:04.28 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.0.0-beta1 (2011/07/22), 1.8.5.0 (2011/07/11), dahdi-linux 2.4.1.2 (2011/04/11), dahdi-tools 2.4.1 (2011/03/03), libpri 1.4.12 (2011/07/06) -=- Visit the new official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/ |
| 17:07.29 | *** join/#asterisk sorressean (~tyler@tds-solutions.net) |
| 17:07.42 | sorressean | I'm curious of something: Is there a way to make asterisk forward to skype? |
| 17:08.15 | *** join/#asterisk kaushal (~kaushal@14.99.157.252) |
| 17:08.18 | kaushal | Hi |
| 17:08.19 | mchou | chan_skype |
| 17:08.22 | kaushal | p3nguin: hi |
| 17:08.41 | p3nguin | If you have skype for asterisk, you can send calls to skype. It's not really "forwarding," though. |
| 17:08.41 | mchou | sorressean: but you'd have to pay for it |
| 17:09.11 | sorressean | well, here's the deal. if I don't answer my cell, I'd like it to forward (or just put the call through to skype). Something like that possible? |
| 17:09.15 | ChannelZ | not anymore, you can't buy it I don't think |
| 17:09.32 | mchou | oh right, msft bought them :) |
| 17:09.42 | mchou | too bad |
| 17:09.43 | p3nguin | If you already bought it, you can still use it, though. |
| 17:10.02 | ChannelZ | Yah |
| 17:10.06 | sorressean | lame. |
| 17:11.22 | mchou | part of msft nokia skype mobile strategy |
| 17:12.44 | ChannelZ | "strategy" |
| 17:15.12 | ChannelZ | Grrph. JIRA search is busted |
| 17:17.20 | drmessano | The Skype for Asterisk deal wasn't a part of the MSFT purchase |
| 17:17.41 | drmessano | That was decided beforehand... |
| 17:18.06 | mchou | yeah well, I still blame msft |
| 17:18.19 | tzafrir | JIRA is busted. no disk space available |
| 17:19.13 | drmessano | Giving MSFT credit for killing it completely removes the blame from Skype themselves |
| 17:19.42 | drmessano | Skype is evil. Period. They're worse about their API's than MSFT |
| 17:20.04 | mchou | pfft |
| 17:20.21 | mchou | those two organizatios dereve each other |
| 17:20.26 | mchou | deserve* |
| 17:21.15 | drmessano | Skype has only opened the door to their network SLIGHTLY, just enough to look like they have an opening, but not enough to do anything super useful with it |
| 17:31.06 | *** part/#asterisk mchou (~quassel@unaffiliated/mchou) |
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| 17:35.02 | *** join/#asterisk creativx (~creadurex@226.62-97-205.bkkb.no) |
| 17:36.14 | sorressean | I'm signing up with a teliax account, if you get discounts or something, I'll take the first referral code. |
| 17:37.31 | *** join/#asterisk Bipul (~bipul@unaffiliated/bipul/x-4918593) |
| 17:37.34 | ChannelZ | let me look |
| 17:37.48 | ChannelZ | oh nevermind |
| 17:38.00 | ChannelZ | is delirious, needs lunch |
| 17:38.53 | p3nguin | is having pizza |
| 17:39.26 | ChannelZ | Did that yesterday.. and Friday.. :) |
| 17:40.47 | p3nguin | Pizza is good. |
| 17:41.04 | p3nguin | (unless it is from Pizza Hut, then it isn't) |
| 17:55.11 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
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| 18:00.15 | drmessano | I guess Digium needs a bug tracker for the bug tracker |
| 18:02.25 | *** join/#asterisk tyman (~tyler@173-12-219-189-Fresno.hfc.comcastbusiness.net) |
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| 19:00.21 | ChannelZ | or a new hard drive |
| 19:02.33 | p3nguin | Sure, do things the easy way. |
| 19:07.06 | drmessano | 1GB should be enough for everyone |
| 19:07.54 | p3nguin | stripes a couple of 480MB disks in RAID 0 |
| 19:08.08 | *** join/#asterisk johnnyasterisk (~johnnyast@81.136.230.203) |
| 19:08.33 | *** part/#asterisk johnnyasterisk (~johnnyast@81.136.230.203) |
| 19:10.44 | [sr] | p3nguin: hope no one goes doen |
| 19:10.46 | [sr] | down |
| 19:10.46 | [sr] | :p |
| 19:10.52 | drmessano | I'd really like to see bugs that affect Google Talk/GV get more attention. Wonder if there's a "buzz" button |
| 19:18.38 | *** join/#asterisk salz212 (~chatzilla@182.178.171.61) |
| 19:24.17 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 19:28.34 | sorressean | I'm trying to figure something out, and not finding much info on it. I see plenty of examples for how to handle extensions, but how do I handle incoming calls? |
| 19:28.47 | ChannelZ | Depends how they're coming in |
| 19:28.49 | sorressean | So like, I want to play a sound, then dial a number. |
| 19:28.58 | sorressean | ChannelZ: what do you mean? |
| 19:29.14 | ChannelZ | It's still an extension.. if your calls come in SIP, 99% of the time it comes to the extension matching your DID |
| 19:29.17 | ChannelZ | Is that what you're asking? |
| 19:29.23 | sorressean | I think so. |
| 19:29.31 | sorressean | I'll try and see how that goes. :) |
| 19:29.45 | ChannelZ | If your phone number is 3334445555 most ITSPs will send your calls to extension 3334445555 |
| 19:30.55 | ChannelZ | The console should tell you how/where they are arriving |
| 19:31.14 | sorressean | sweet, thanks |
| 19:31.51 | p3nguin | A call comes from a peer. The peer is either going to be one that you have identified in sip.conf, or it will be anonymous. If you have identified it by creating a peer for it, that gives you the ability to control which context calls from that peer go to in your dial plan. |
| 19:32.20 | p3nguin | If it is anonymous, it will go to the context that you have configured in the general section. |
| 19:33.32 | p3nguin | Assuming you have configured a peer to match the call, send the call to a context of your choice, where an extension matches the destination extension of the call and does useful things. |
| 19:34.15 | p3nguin | All calls need to match an extension, or the call will die. |
| 19:34.48 | ChannelZ | Or go to 'i' |
| 19:35.16 | p3nguin | The call would have to have a destination of i to reach i initially. |
| 19:35.29 | p3nguin | No call will just end up on i all by itself. |
| 19:36.04 | sorressean | I set the context to default, so I have a default section in my config. |
| 19:36.16 | p3nguin | Never use default as the context. |
| 19:37.59 | sorressean | Last thing, I think. Right now I'm just trying to make this dial a number, but I'm kind of confused. I have to use iax2/device? do I need to associate a "device" with a number smoehow? |
| 19:38.19 | sorressean | p3nguin: anything else works though? |
| 19:38.28 | p3nguin | Ideally, contexts will be named with a reasonable reflection of what the context is used for. For calls from an itsp by the name of shitty, a reasonable context for call from it might be shitty-inbound. |
| 19:38.42 | ChannelZ | who cares? "default" isn't any more special than "foocontext" |
| 19:39.18 | p3nguin | Will "default" work? Sure. Is it smart to use it? Not at all. |
| 19:39.38 | p3nguin | Don't reinforce bad practices. |
| 19:39.45 | ChannelZ | Why because you don't like the word or something? |
| 19:40.09 | sorressean | sounds like it just keepps things cleaner. |
| 19:40.56 | sorressean | Anyway, it's already been changed. Do I need to add an entry to sip.conf for my outbound landline? I built asterisk from source, so my config /etc/asterisk folder is totally empty. |
| 19:41.10 | ChannelZ | It's arbitrary. He makes it sound like "default" is some special context that does horrible things. It's just a name. |
| 19:41.40 | sorressean | ChannelZ: Probably for the same reason that I scream at students that use variable names like a, b, c in their code. |
| 19:41.44 | p3nguin | If you have already created your peer in sip.conf, you can use it to dial out through it. |
| 19:42.14 | sorressean | nods. I'll keep digging through the wiki/tiliax to figure out how to set that up. |
| 19:42.23 | p3nguin | Create an extension in your dial plan to send a call through the peer: Dial(SIP/shitty/${EXTEN}) |
| 19:43.41 | sorressean | where shitty is the section in iax.conf? |
| 19:44.04 | p3nguin | For an 11-digit NANP number over IAX2, I'd probably use: exten => _1NXXNXXXXXX,1,Dial(IAX2/shitty/${EXTEN}) |
| 19:44.23 | p3nguin | where shitty is the name of the peer in iax.conf |
| 19:44.31 | p3nguin | (thought we were dealing with sip) |
| 19:45.02 | p3nguin | I'd guess you'd name your peer something more reasonable, like teliax. |
| 19:45.19 | p3nguin | Dial(IAX2/teliax/${EXTEN}) |
| 19:45.33 | sorressean | p3nguin: I receive the call through teliax, then I'm trying to forward it (I guess just open another channel and dial out) to my cell number. |
| 19:45.43 | sorressean | nods |
| 19:45.47 | p3nguin | That's standard dialing, not forwarding. |
| 19:46.00 | *** join/#asterisk jwiggins (~James@gateway/tor-sasl/jwiggins) |
| 19:46.17 | p3nguin | For an incoming call, you'd match your extension to the number that teliax is calling, then Dial() to your mobile number. |
| 19:46.56 | sorressean | ah. |
| 19:47.00 | sorressean | thanks. sorry, I'm really new to this mess. |
| 19:47.48 | p3nguin | If your number with teliax is 3145551212 and your mobile number is 3149691077, something like exten => 3145551212,1,Dial(IAX2/teliax/3149691077,30) might be what you need. |
| 19:47.54 | *** join/#asterisk Kobaz (~kobaz@its.kobaz.net) |
| 19:48.32 | p3nguin | That extension would need to be in the context you assigned to the peer. Hopefully it's something sensible. |
| 19:50.11 | sorressean | nods, thanks |
| 19:50.40 | p3nguin | That is in the most basic form. You can do other fancy things if necessary. |
| 19:50.57 | trumee | I am routing asterisk over openvpn but my calls are dropping with message, "chan_sip.c:3622 retrans_pkt: Retransmission timeout reached on transmission" |
| 19:51.12 | trumee | chan_sip.c:3651 retrans_pkt: Hanging up call 348eee3b5dbfe2b0045b4cac108b2267@172.16.1.132:5060 - no reply to our critical packet" |
| 19:51.28 | salz212 | I am willing to introduce Java Agi in production, heard a lot about dynamic Java IVRs. Can any one recommend me one? I am a C/C++ guy so dont know how JAVA would work with Agi .. and Is there any standard? |
| 19:51.32 | trumee | is there any way to make asterisk perform better with openvpn? |
| 19:51.39 | p3nguin | Does the vpn's other end point have a route back to your device? |
| 19:52.04 | *** join/#asterisk jmwpc (~jeremy@c-24-5-58-60.hsd1.ca.comcast.net) |
| 19:52.48 | trumee | p3nguin: my openvpn is running on an openwrt router |
| 19:53.04 | trumee | p3nguin: let me pastebin my config |
| 19:54.00 | trumee | p3nguin: this is my openvpn config, http://www.pastie.org/2407941 |
| 19:55.00 | jmwpc | Has anyone had an issue placing outbound calls through google voice starting the last couple of days? |
| 20:00.08 | ChannelZ | yes |
| 20:00.12 | ChannelZ | It's the flavor of the month |
| 20:00.16 | ChannelZ | or week as it were |
| 20:01.00 | jmwpc | Currently working on applying this change... will let you know how it goes http://forums.asterisk.org/viewtopic.php?f=1&t=79599 |
| 20:01.04 | ChannelZ | See if this rings any bells: https://issues.asterisk.org/jira/browse/ASTERISK-18301 |
| 20:01.24 | trumee | p3nguin: any idea how to solve my issue? |
| 20:05.09 | salz212 | >I am willing to introduce Java Agi in production, heard a lot about dynamic Java IVRs. Can any one recommend me one? I am a C/C++ guy so dont know how JAVA would work with Agi .. and Is there any standard? |
| 20:05.55 | ChannelZ | AGI is just a stdin/stdout interface |
| 20:06.13 | ChannelZ | Any language you can get access to those file handles you can use AGI with |
| 20:06.46 | jmwpc | goes to get coffee while asterisk compiles |
| 20:07.28 | ChannelZ | compiling on a Commodore 64? (and not the new one?) |
| 20:08.51 | jmwpc | amiga |
| 20:08.55 | ChannelZ | sweet |
| 20:09.09 | ChannelZ | I still have a 3000 in the closet |
| 20:09.19 | jmwpc | :) c2d e6320... kinda slow |
| 20:11.52 | jmwpc | That appeared to fix it |
| 20:12.26 | jmwpc | thanks again internet |
| 20:12.36 | ChannelZ | It knows everything |
| 20:15.44 | [sr] | people |
| 20:15.59 | [sr] | dialing from google talk to 10 digit phone numbers for US, are free, is this correct? |
| 20:18.02 | ChannelZ | Last I knew |
| 20:18.14 | *** join/#asterisk godmachine-x6 (~godmachin@unaffiliated/godmachine-x6) |
| 20:18.34 | jmwpc | That has been my experience |
| 20:19.00 | p3nguin | Yes, dialing in the US are free. |
| 20:19.18 | [sr] | both landline & mobile network? |
| 20:19.29 | p3nguin | As long as they are US numbers yes. |
| 20:19.46 | [sr] | cool! |
| 20:20.00 | [sr] | have a solution for my vacation in the us if someone wants to call me!!! |
| 20:20.10 | ChannelZ | Does DTMF work yet? |
| 20:21.54 | [sr] | damn have one week to configure this |
| 20:22.23 | [sr] | i have to create a google talk account, but sai that i live in the US for it to work, correct? does it need any kind of validation by phone or so? |
| 20:22.53 | p3nguin | Google knows geolocation, and it will ask for a validation phone number. |
| 20:23.32 | [sr] | damn, wont work them :( |
| 20:23.47 | [sr] | them/then |
| 20:23.51 | p3nguin | If you can get them to allow your registration, I can help you validate. |
| 20:24.31 | [sr] | that would be great |
| 20:24.52 | [sr] | tomorow i'll setup the account and ask you |
| 20:25.18 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 20:25.23 | [sr] | sorry for the offtopic, which is the best mobile network to buy a card for only 2 weeks? |
| 20:25.26 | [sr] | pre-paid |
| 20:25.46 | [sr] | t-online? at&t? other? |
| 20:26.02 | p3nguin | AT&T generally sucks. |
| 20:26.45 | p3nguin | Verizon and T-Mobile have a fairly good network, but I don't know how much it would cost you to get a prepaid phone. |
| 20:27.58 | *** part/#asterisk jmwpc (~jeremy@c-24-5-58-60.hsd1.ca.comcast.net) |
| 20:28.00 | [sr] | ok i have to see when i get there |
| 20:29.00 | p3nguin | I'd go to Walmart and see what they have. |
| 20:33.35 | [sr] | i'mm search for that |
| 20:33.40 | [sr] | i'mm=i'll |
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| 20:40.53 | [sr] | p3nguin: at least installing talk with the phone plugin, it doesn't appear... m,aybe 'cause im not in the us |
| 20:42.44 | *** join/#asterisk weinerk (~user@unaffiliated/weinerk) |
| 20:43.04 | p3nguin | *shrug* |
| 20:48.38 | ChannelZ | Is it just me or do PenTile displays look like they have a texture that actually moves through them? |
| 20:55.49 | *** join/#asterisk johnnyasterisk (~johnnyast@81.136.230.203) |
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| 21:06.09 | [sr] | p3nguin: i think i'll go for voipbuster... just have one week to studie gtalk... no time |
| 21:11.29 | [sr] | p3nguin: give me the number for example of the call center for at&t |
| 21:11.33 | [sr] | with country code |
| 21:21.12 | [sr] | ok never mind, this doesn't have the module for gtalk |
| 21:21.37 | ChannelZ | Go to www.att.com and even try to find a phone number |
| 21:29.24 | [sr] | tks for now ChannelZ, i'll go for buster, it's already setup |
| 21:34.11 | *** join/#asterisk godmachine-x6 (~godmachin@unaffiliated/godmachine-x6) |
| 21:49.46 | x1user | Hi, I make a call from sip client but i dont have sound. Any idea? |
| 21:51.47 | [sr] | going to sleep |
| 21:51.49 | [sr] | bye |
| 21:59.24 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 22:01.43 | *** join/#asterisk Atriks (~Hatrix@2a01:e34:ee51:4670:b058:12dc:c2e3:3a15) |
| 22:01.44 | Atriks | Hey |
| 22:06.42 | ChannelZ | x1user: firewall/routing issues |
| 22:07.13 | x1user | nope firewall is stopped. nat is enabled |
| 22:07.40 | Atriks | Does somehone have a sip.conf exemple ? Nothing works.. :/ |
| 22:07.43 | ChannelZ | It could be on either end, but you didn't say which way the audio works |
| 22:08.11 | ChannelZ | Atriks: That's like asking "does someone have a page out of the dictionary" |
| 22:08.29 | ChannelZ | The answer being "yes, but it probably doesn't have the word you're trying to spell on it" |
| 22:08.37 | Atriks | Humm ok |
| 22:08.53 | ChannelZ | Are you trying to setup a softphone to test with, or do you have an account with an ITSP already, or.... |
| 22:08.56 | Atriks | I found some tutorials, but I don't understand |
| 22:09.11 | ChannelZ | ~book |
| 22:09.12 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/ or ~buybook |
| 22:09.51 | Atriks | It's for the sip server, I don't know where to enter it with my sip login/username |
| 22:11.52 | ChannelZ | The SIP server of your ITSP you're trying to hook your asterisk to? |
| 22:12.43 | Atriks | yes |
| 22:15.08 | WIMPy | Good morning! |
| 22:15.15 | Atriks | not morning here :p |
| 22:15.24 | Atriks | whatever, hi |
| 22:15.26 | ChannelZ | And they have no instructions? Here's the basic of most basic not really knowing what you require - http://pastebin.com/eRMDs6fp |
| 22:15.51 | Atriks | Thanks ChannelZ |
| 22:16.14 | ChannelZ | You might also need to register with them... register => youruser:yourpassword@youritsp.com in [general] |
| 22:16.25 | Atriks | I have a sip account |
| 22:17.56 | ChannelZ | With whom |
| 22:18.27 | Atriks | sorry, what ? |
| 22:19.09 | ChannelZ | Who is your SIP account with |
| 22:19.14 | x1user | build_peer: '' is not a valid RTP hold time at line 0. Using default. ? |
| 22:19.26 | Atriks | Free, it's french |
| 22:22.47 | *** join/#asterisk godmachine-x6 (~godmachin@unaffiliated/godmachine-x6) |
| 22:23.29 | Atriks | So, in XLite in enter my server's IP, and it says "404 - Not Found", I don't understand why |
| 22:45.37 | *** join/#asterisk leejohn (~Administr@58.69.86.236) |
| 22:46.10 | leejohn | Any asterisk admin here................... |
| 22:46.24 | leejohn | I think we have a space crunch on bug tracker |
| 22:46.30 | leejohn | there was an error in JIRA |
| 22:46.34 | p3nguin | no kidding |
| 22:46.49 | leejohn | https://issues.asterisk.org/jira/rest/gadget/1.0/issueTable/jql?jql=assignee+%3D+currentUser()+AND+resolution+%3D+unresolved+ORDER+BY+priority+DESC%2C+created+ASC&num=10&addDefault=true&enableSorting=true&sortBy=null&paging=true&startIndex=0&showActions=true |
| 22:52.20 | ChannelZ | yes it's been broken all day |
| 22:53.02 | IsUp | No space left on device |
| 22:53.16 | leejohn | really |
| 22:53.28 | leejohn | it's broken all day then nothing? |
| 22:53.42 | IsUp | nothing |
| 22:53.45 | IsUp | just nothing |
| 22:54.24 | leejohn | ouch |
| 22:55.03 | IsUp | i'll call jerry springer |
| 23:01.08 | ChannelZ | Don't, the attention only encourages him |
| 23:03.43 | IsUp | i cant compile Asterisk on Amazon EC2, any ideas? |
| 23:04.19 | WIMPy | yes |
| 23:04.22 | SunTsu | IsUp: my first idea would be to paste a compile log onto pastey or something like that |
| 23:04.39 | p3nguin | I think the idea is to tell us what the symptoms are, then we can give suggestions to overcome them. |
| 23:05.14 | p3nguin | I don't think anyone ever got their stuff fixed by simply saying it doesn't work. |
| 23:06.06 | IsUp | well if it takes too much time, i'll remove instance. this is my first time with Amazon, 30 day trial thing. |
| 23:06.53 | IsUp | error: C++ preprocessor "/lib/cpp" fails sanity check |
| 23:07.25 | ChannelZ | try Ritalin |
| 23:07.40 | ChannelZ | haHA |
| 23:07.43 | p3nguin | I think that's for hyperactivity. |
| 23:08.07 | ChannelZ | Yeah I'm not up on anti-psychotics |
| 23:08.08 | IsUp | lol |
| 23:08.31 | p3nguin | It might need something more like Zoloft so it can have babies with birth defects. |
| 23:08.33 | ChannelZ | But ADD sure makes kids act crazy so it's close enough |
| 23:20.05 | *** join/#asterisk Gugge (~gugge@46.30.120.1) |
| 23:23.04 | *** join/#asterisk Cesare (~Adium@creati59.lnk.telstra.net) |
| 23:28.37 | *** part/#asterisk leejohn (~Administr@58.69.86.236) |
| 23:29.52 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 23:30.09 | *** join/#asterisk Atriks (~Hatrix@2a01:e34:ee51:4670:61c4:c8e5:69b7:a868) |
| 23:31.38 | Atriks | is IsUp here ? |
| 23:33.43 | WIMPy | 5 min too late |
| 23:33.49 | Atriks | :( |
| 23:33.54 | Atriks | Did he say something ? |
| 23:35.00 | *** join/#asterisk kerx (~kerx@76-240-161-60.lightspeed.irvnca.sbcglobal.net) |
| 23:36.29 | p3nguin | No, he left. |
| 23:36.35 | Atriks | erf |
| 23:36.41 | Atriks | He was helping me to configure my asterisk |
| 23:37.07 | Atriks | So, can you replace him ? :p My error in xlite is Call Failed : service unavailable |
| 23:37.25 | Atriks | I have set verbose to 10, to see logs, I think |
| 23:37.29 | Atriks | so if you want it.. |
| 23:38.17 | p3nguin | I do not like x-lite. |
| 23:38.30 | p3nguin | But it only has a couple things to configure in it. |
| 23:38.46 | Atriks | Yes but the problem seems to come from asterisk |
| 23:39.02 | p3nguin | Configure a peer for your x-lite. |
| 23:39.08 | Atriks | already done |
| 23:39.33 | Atriks | http://pastebin.com/6J7cLJQw |
| 23:39.40 | Atriks | Asterisks says that |
| 23:40.32 | p3nguin | Looks like x.lite works to me. |
| 23:40.46 | Atriks | Yes |
| 23:40.49 | p3nguin | Looks to me like x.lite works, rather. |
| 23:40.50 | Atriks | It's asterisk |
| 23:40.55 | p3nguin | Silly English. |
| 23:41.05 | Atriks | IsUp said that it's a problem with mi itsp |
| 23:41.24 | p3nguin | It could be. Asterisk doesn't look like it is giving any problem. |
| 23:41.46 | Atriks | But I don't understand where the problem is |
| 23:42.00 | p3nguin | Did you configure any extensions? |
| 23:42.04 | Atriks | Yes |
| 23:42.07 | p3nguin | Show me. |
| 23:42.12 | p3nguin | all of them. |
| 23:42.18 | p3nguin | The entire file, actually. |
| 23:42.27 | Atriks | I just changed one thing |
| 23:42.38 | Atriks | [out] |
| 23:42.38 | Atriks | exten => _.,1,Dial(SIP/freephonie.net/${EXTEN},60) |
| 23:42.39 | Atriks | that |
| 23:42.50 | Atriks | at the end of the file |
| 23:42.51 | p3nguin | That's a terrible choice of pattern. |
| 23:43.01 | Atriks | IsUp sent me that |
| 23:43.09 | p3nguin | Let's address your poor choice of pattern first. |
| 23:43.16 | p3nguin | _. matches EVERYTHING. |
| 23:43.29 | p3nguin | It even matches things that you should not be matching. |
| 23:43.45 | p3nguin | Extensions s, h, i, and t... just for a few examples. |
| 23:44.12 | Atriks | replacI don't understand |
| 23:44.16 | p3nguin | If you want to match all dialed numbers, _X. is usually okay. |
| 23:44.22 | Atriks | o_ô |
| 23:44.53 | p3nguin | So don't use _. as an extension... at least not until you know what you're doing and you know why you're doing it. |
| 23:45.04 | Atriks | Ok |
| 23:45.30 | Atriks | Hum |
| 23:45.34 | Atriks | I found that on internet |
| 23:45.35 | Atriks | exten => _9.,1,Dial(SIP/freephonie-out/${EXTEN:1}) |
| 23:45.43 | p3nguin | Now that you have changed that, let's move on to the peer name. In that Dial(), you have freephone.net as the peer name. Did you configure a peer in sip.conf called [freephone.net] ? |
| 23:45.56 | Atriks | no |
| 23:45.58 | p3nguin | Don't start using _9. either. |
| 23:46.04 | Atriks | Freephonie.net is my itsp |
| 23:46.15 | p3nguin | Did they give you a user name and password? |
| 23:46.23 | Atriks | Yes, of course |
| 23:46.29 | Atriks | Username is ok |
| 23:46.32 | p3nguin | Have you configured a peer for them in sip.conf? |
| 23:46.42 | Atriks | yes yes |
| 23:46.43 | p3nguin | What name did you put on that peer in sip.conf? |
| 23:47.12 | p3nguin | the one in square brackets |
| 23:47.26 | Atriks | host, fromuser, secret, type |
| 23:47.27 | p3nguin | I would have called it freephone |
| 23:47.31 | p3nguin | negative |
| 23:47.36 | p3nguin | the one in square brackets. |
| 23:47.52 | Atriks | [freephonie.net] ? |
| 23:47.55 | p3nguin | The peer name looks like [this] |
| 23:48.17 | p3nguin | I asked if you named it freephone.net and you said no. |
| 23:48.22 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 23:48.26 | Atriks | I didn't understand |
| 23:48.27 | p3nguin | freephonie, whatever |
| 23:48.31 | Atriks | And it's freephonIe :p |
| 23:48.47 | p3nguin | So the peer name really is freephonie.net? |
| 23:48.55 | Atriks | yes |
| 23:49.01 | p3nguin | as in [freephonie.net] |
| 23:49.11 | Atriks | euh wait |
| 23:49.14 | Atriks | let me understand |
| 23:49.17 | Atriks | I've to [] |
| 23:49.27 | Atriks | [Hatrix] and [freephonie.net] |
| 23:49.27 | p3nguin | [freephonie] is likely to be a better choice. |
| 23:49.33 | p3nguin | Hatrix is your phone. |
| 23:49.54 | Atriks | freephonie.net is for the itsp so |
| 23:50.03 | p3nguin | Do me a favor and change [freephonie.net] to [freephonie]. |
| 23:50.11 | Atriks | done |
| 23:50.18 | Atriks | is it really important ? |
| 23:50.24 | p3nguin | maybe |
| 23:50.42 | p3nguin | Then change your extension to exten => _X.,1,Dial(SIP/freephonie/${EXTEN}) |
| 23:51.24 | p3nguin | If asterisk was trying to call the host directly without bothering with the peer information you provided in sip.conf, it's certainly important. |
| 23:51.46 | p3nguin | Since I don't know if it will do that or not, I'm not going to suggest leaving it that way. |
| 23:51.55 | Atriks | service still unavailablr |
| 23:51.58 | Atriks | -r +e |
| 23:52.24 | p3nguin | Did you save extensions.conf and sip.conf? And did you run dialplan reload and sip reload? |
| 23:52.38 | Atriks | I'm dumb |
| 23:52.42 | Atriks | I forgot to reload |
| 23:52.43 | Atriks | .... |
| 23:52.50 | p3nguin | dialplan reload , sip reload |
| 23:53.23 | Atriks | service still unavailable :/ |
| 23:53.40 | p3nguin | I suppose now is the time to do sip debugging to see why. |
| 23:53.43 | p3nguin | sip set debug on |
| 23:53.47 | p3nguin | Then make a call. |
| 23:53.56 | p3nguin | Copy the ENTIRE output and paste in pastebin. |
| 23:54.12 | p3nguin | sip set debug off to turn off debugging. |
| 23:55.30 | Atriks | wait 2 sec |
| 23:55.40 | *** join/#asterisk vinhdizzo (~vinh@pool-173-60-112-55.lsanca.fios.verizon.net) |
| 23:57.35 | Atriks | I think it is only that |