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| 03:14.05 | fling | I have this thing each time on one of the patas after a disconnect > 2013-10-21T10:12:18+07:00 ata_22 84C9B2CF146F-sipg <local0.emerg> Reg Fail [301] |
| 03:14.17 | fling | But nothing in asterisk log hmm hmmmm |
| 03:18.06 | fling | The only difference I see in it's config is "Proxy server realm: Asterisk" |
| 03:18.15 | fling | just removed it and rebooted. |
| 03:18.23 | fling | Do I need a new device? :D |
| 03:22.04 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.84) |
| 03:22.20 | fling | One of two sip registered after this reboot. Hmmhmhmmhm |
| 03:25.05 | fling | after second reboot > -- Unregistered SIP '301' -- Registered SIP '302' at 10.0.2.22:5060 |
| 03:25.17 | fling | so they are not working at the same time :D |
| 03:27.22 | fling | 5 reboots and they both become reachable! yay! :> |
| 03:27.38 | fling | d-link sucks |
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| 03:51.45 | ChannelZ | dlink makes ATAs? |
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| 03:55.13 | Penguin | Probably. They make a wifi SIP phone that looks like a clamshell cell phone (2005 model). |
| 03:58.26 | ChannelZ | Huh. Didn't know. They have a small range of voip devices. |
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| 03:58.55 | ChannelZ | and a bunch of new wifi routers that look like the stupid new mac pro |
| 03:59.50 | Penguin | I'd guess they all suck. |
| 04:00.15 | ChannelZ | They already look like trashcans |
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| 09:55.03 | admin0 | hi all |
| 09:55.20 | admin0 | when i am making a call, |60|HRrL is going to my outbound proxy system .. how do I remove this ? |
| 09:57.04 | kaldemar | fix your Dial app to use the correct syntax. you're using pipes instead of a comma. |
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| 10:06.00 | admin0 | kaldemar: thank you .. so the line should be xten => _X.,n,DeadAgi(a2billing.php,1) instead of xten => _X.,n,DeadAgi(a2billing.php|1) > |
| 10:06.58 | admin0 | still goes though |
| 10:07.04 | admin0 | sorry .. i am very new into this one |
| 10:08.46 | kaldemar | admin0: i said Dial, not DeadAgi. if it is the a2billing.php that uses pipes for app Dial, you need to modify that. |
| 10:09.47 | admin0 | ok |
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| 10:45.21 | phpboy | hi, I'm trying to originate a call to the IP of a SIP phone instead of it's extension number... can this be done? |
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| 11:45.39 | phix | phpboy: so you want to call the device directly instead of going via asterisk? |
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| 12:13.00 | salz212 | Hello all, I am looking to use same queues on different boxes with distributive agent states. Need to hear from community the best practices about such customization in a rather large scale. |
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| 12:19.55 | phpboy | phix: I need it to go via asterisk |
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| 12:21.26 | phpboy | what I want to do is keep the call permanently connect an agent and transfer calls to him as and when needed |
| 12:21.32 | phpboy | trying to build a dialer |
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| 12:27.51 | [TK]D-Fender | phpboy: can you pastebin the conversation to date? |
| 12:32.06 | Greenlight | phpboy: You can just specify the SIP endpoint in the Originate to originate a call from/to it |
| 12:32.25 | Greenlight | Keeping the call permentantly connected is a little trickier |
| 12:33.12 | phix | phpboy: ummmm I think I am missing something, from what you have said you want a SIP device to register to your asterisk server then transfer calls to it when needed. As long as the SIP client is maintaining / updating the registration I don't see what the issue would be |
| 12:34.12 | phix | You want to only transfer some calls? or certain calls to it? based on some type of balancing between other agents / SIP clients or by callerid? |
| 12:35.29 | [TK]D-Fender | phix, could you PB the log I asked him for? |
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| 12:36.10 | Greenlight | [TK]D-Fender: Prior to your join the only thing he asked was: |
| 12:36.11 | Greenlight | [11:45am] <phpboy> hi, I'm trying to originate a call to the IP of a SIP phone instead of it's extension number... can this be done? |
| 12:36.55 | [TK]D-Fender | How does he have an IP and not a devicename? |
| 12:37.15 | [TK]D-Fender | And he should already know he can dial an IP as it is... |
| 12:37.29 | Greenlight | Yea, I think this is a question of running before he can walk.. |
| 12:37.44 | Greenlight | And using a devicename makes more sense to me |
| 12:39.56 | [TK]D-Fender | I'm just curious what he's doing where he'd somehow have better access to the former, or why he wouldn't want or be able to do the latter |
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| 12:44.12 | Chainsaw | [TK]D-Fender: Perhaps he's using the power of the internet to provide free telephony. |
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| 12:45.29 | jmetro | sup |
| 12:49.33 | phpboy | The agent shouldn't have an option to answer the call or not... the 'system' will make one call to the agent and send calls through to that channel when outbound calls are connected to clients |
| 12:50.19 | *** join/#asterisk asteriskmonkey (~Tardis@74-51-38-204.telnetcommunications.com) |
| 12:50.47 | phpboy | there will be say 10 agents connected and calls will be randomly sent to them as the outbound calls are established |
| 12:51.06 | Greenlight | Okay, that's your broad requirement. What part are you struggling with ? |
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| 12:52.14 | Bkhan | Hi |
| 12:52.36 | phpboy | the first one is establlishing the initial call to an agents IP which the agent will answer and then hold until a call gets passed through |
| 12:53.01 | Greenlight | So, during that "phase", what will the agent's call be connected to. |
| 12:53.07 | phpboy | once I've managed to get this right dialing and connecting the outbound calls should be quite straight forward |
| 12:53.08 | Bkhan | I want to listen Sip on multiport e.g 5060 and 5432 |
| 12:53.33 | Greenlight | phpboy: From someone who's coded what your trying to do; this is the easy bit... |
| 12:53.41 | [TK]D-Fender | Bkhan: You can't. Why are you looking to do this? |
| 12:53.46 | phpboy | Greenlight: the agent as a sip phone on his pc that listens for a sip call... how asterisk knows which IP to 'dial' is through a webinterface which tells asterisk which IP to dial |
| 12:53.56 | phpboy | the reason why I don't want to use extensions is because it |
| 12:54.15 | phpboy | it's going to be VERY difficult to manage... couple of thousand users is the plan |
| 12:54.41 | phpboy | Greenlight: then help with with the easy bit please :P |
| 12:54.43 | Bkhan | D-Fender> want to login agents not on default port because of limitation and provider sends calls on 5060 |
| 12:54.59 | Greenlight | phpboy: If your phones are going to register with asterisk, you'll need entries in sip.conf for them anyway. |
| 12:55.10 | phpboy | no |
| 12:55.22 | phpboy | no entries in sip.conf, that's the point here |
| 12:55.38 | phpboy | none of these agents must exist within asterisk |
| 12:55.54 | phpboy | I literally want asterisk to dial an IP and that's it |
| 12:55.54 | Greenlight | So they all have public IP's or have 5060 port forwarded ? |
| 12:56.06 | Greenlight | Or all are on a private network? |
| 12:56.12 | phpboy | private network |
| 12:56.24 | [TK]D-Fender | phpboy: So go dial an IP |
| 12:56.25 | Bkhan | not on live IP |
| 12:56.26 | Greenlight | Okay, so just dial. |
| 12:56.39 | phpboy | [TK]D-Fender: that part I've got |
| 12:56.46 | Greenlight | Lets see your originate command ? |
| 12:56.57 | Greenlight | Although it may be easier testing with a Dial |
| 12:57.04 | phpboy | but it's what I connect the IP to within asterisk to keep it in a 'holding' state while waiting inbetween calls |
| 12:57.04 | [TK]D-Fender | Bkhan: You'll have to run something in between to act as a proxy then |
| 12:57.04 | Greenlight | To be honest, as the syntax is thr same |
| 12:57.27 | [TK]D-Fender | phpboy: Ther isn't a "call"... what is there to hold? |
| 12:57.36 | phpboy | ok sorry, some stuff has happened inbetween now and asking the question it first |
| 12:57.45 | Greenlight | phpboy: Try a Wait() loop. |
| 12:58.04 | phpboy | my plan is for the Asterisk to establish this one call for say an 8 hour shift |
| 12:58.25 | phpboy | then as and when the outbound dialer reaches a human it will connect it to the first available agent |
| 12:58.26 | Greenlight | exten => 8000,1(start),NoOp(Agent waiting ...) |
| 12:59.04 | Greenlight | same => n,Wait(600) |
| 12:59.11 | Greenlight | same => n,GoTo(start) |
| 12:59.17 | Greenlight | Something like that |
| 12:59.18 | phpboy | when that call is hung up from the client side or by the agent (via the web interface) then it must go into the waiting state again |
| 12:59.36 | [TK]D-Fender | phpboy: I'm not understanding what you're trying do having this "non connection" as a thing that is "up" without being connected to anything. |
| 12:59.36 | phpboy | Greenlight: good thinking... I'll definitely give that one a shot |
| 13:00.06 | phpboy | [TK]D-Fender: the point is the agent must NOT have an option to answer a call connected to him |
| 13:00.06 | *** join/#asterisk vlad_starkov (~vlad_star@217.26.6.70) |
| 13:00.15 | phpboy | that's why he'll have his headset on and wait for a call |
| 13:00.39 | phpboy | so when asterisk calls him it will either be silence or on hold music of some sort |
| 13:00.44 | [TK]D-Fender | phpboy: But you're calling that IP with an originate. that is a "call". |
| 13:00.52 | phpboy | yes |
| 13:00.55 | [TK]D-Fender | phpboy: which implies that their state before is NOT being on a call. |
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| 13:01.08 | [TK]D-Fender | What side are you trying to lie to and why? |
| 13:01.12 | Greenlight | Yea, but that call will remaing active for say the 8 hour shift |
| 13:01.22 | [TK]D-Fender | Your description is very bad at clarifying this... |
| 13:01.35 | phpboy | Greenlight: you're getting it |
| 13:01.48 | phpboy | [TK]D-Fender: I need an active 'call' for the agent to be connected to the entire shift |
| 13:01.51 | Greenlight | It's common practice for diallers to work like this |
| 13:01.58 | phpboy | exactly |
| 13:02.07 | phpboy | all I'm trying to do is write my own dialer |
| 13:02.23 | [TK]D-Fender | Stop saying dialer and start clarifying when things actually get sent to it |
| 13:02.24 | phpboy | what I'm struggling with now is the initial call to the agent |
| 13:02.26 | [TK]D-Fender | You said ORIGINATE. |
| 13:02.36 | [TK]D-Fender | is this ONCE for the span of that entire shift? |
| 13:02.37 | phpboy | I know |
| 13:02.39 | [TK]D-Fender | You are not being specific |
| 13:02.43 | *** join/#asterisk admin0 (~admin0@5356416B.cm-6-7b.dynamic.ziggo.nl) |
| 13:02.49 | phpboy | [TK]D-Fender: one for the span of the shift |
| 13:02.59 | Greenlight | He neesd to connect the "agent" to somewhere for the duration of their shift. |
| 13:03.09 | Penguin | That's what chan_agent is for. |
| 13:03.09 | [TK]D-Fender | Fine, so that IS a single call then |
| 13:03.09 | Greenlight | He does not yet know where or how. That's what he's asking. |
| 13:03.36 | *** join/#asterisk generalhan (~generalha@about/windows/staff/generalhan) |
| 13:03.39 | phpboy | [TK]D-Fender: yes one single call |
| 13:03.46 | Greenlight | In my experience chan_agent is buggy at best |
| 13:03.47 | phpboy | connected to probably an infinite hold loop |
| 13:04.28 | phpboy | then as and when the dialer establishes calls to clients it will connect that call to an available agent |
| 13:04.54 | [TK]D-Fender | phpboy: so set your hangup handlers on the call so that the caller keeps getting thrown into a loop similar to what he showed (or MoH, etc) and hijack it to connect (AMI reditect, Bridge, etc) |
| 13:05.10 | Penguin | If you don't like chan_agent, emulate what it does with other tools. |
| 13:05.29 | phpboy | Penguin: I'm trying to write this from scratch |
| 13:05.41 | phpboy | busy with theory and planning at the moment |
| 13:05.49 | [TK]D-Fender | Penguin: that was just a login mechanism to dynamical set a chan_local dialout... he'd still have to bridge the channel at some point |
| 13:06.12 | phpboy | I've used chan_agent for inbound stuff |
| 13:06.20 | phpboy | but this is for outbound stuff |
| 13:06.30 | Penguin | Your agent calls INBOUND. |
| 13:07.04 | [TK]D-Fender | Penguin: His dialer picks the # and the agent and he just wants to bridge it imemdiately. |
| 13:07.05 | *** join/#asterisk dronacid (~dronacid@p549745D8.dip0.t-ipconnect.de) |
| 13:07.06 | phpboy | Penguin: i want the system to dial the agent NOT the other way around |
| 13:07.46 | phpboy | Penguin: one of the reason why it needs to bridge immediately is because if the dialer finds a human it needs to connect it to another human asif the agent dialed him |
| 13:07.55 | phpboy | there cannot be a delay because of this |
| 13:08.20 | Greenlight | YOu'll want to use the AMI Bridge |
| 13:08.56 | Greenlight | As a handy "feature" you'll noticed throgh testing that if you have the two Channels the right way around, the "agent" gets automatically returned to that wait loop. |
| 13:08.59 | [TK]D-Fender | You'll want to be coding your own dialplan App for this actually. |
| 13:09.18 | Greenlight | Bridge works fine. |
| 13:09.26 | Greenlight | No need to code a new App |
| 13:10.04 | Greenlight | Well, unless you wanted a lot of the logic to happen inside Asterisk rather than your AMI based app |
| 13:15.34 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 13:15.34 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 13:15.43 | phpboy | I'm trying to make it as 'light' as possible |
| 13:15.57 | phpboy | so as much logic as I can in asterisk |
| 13:16.19 | [TK]D-Fender | Actually the agent could set an AstDB value for the channelname it's on, agent ID, etc and the other end pull that instead of having to poll somewhere more painful. |
| 13:16.27 | [TK]D-Fender | And they could use the Bridge dialplan appas well. |
| 13:16.31 | [TK]D-Fender | so less AMI |
| 13:17.04 | phpboy | I'll consider that in the plans |
| 13:18.15 | phpboy | the dialer part is quite tricky.. for example if I dial a client and the line is busy I want to schedule that call for later again, say after an hour |
| 13:19.02 | Greenlight | You'll need to give careful considering to how you are "locking" as for example you may have 5 customers "answer" at exactly the same moment; you don't want to have them all try and connect to the same customer |
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| 13:21.47 | Penguin | Why would 5 customers who answer at the exact same time all connect to the same customer? |
| 13:22.10 | Greenlight | If you're routing logic didn't "lock" |
| 13:22.17 | Greenlight | Clearly you don't want that |
| 13:22.28 | Penguin | Usually, customers don't connect to customers. |
| 13:22.37 | Greenlight | Sorry, agent |
| 13:22.47 | Greenlight | ... connect to the same agent. |
| 13:22.49 | Greenlight | Typo. |
| 13:24.35 | phpboy | yeah |
| 13:24.45 | phpboy | it's quite a mission to build a dialer |
| 13:24.50 | Greenlight | Indeed |
| 13:24.56 | phpboy | lots of shit to take into consideration |
| 13:25.12 | phpboy | I'm busy downloading an open source dialer called OSdial |
| 13:25.31 | phpboy | wanna check how they do stuff maybe even roll out a live install |
| 13:28.02 | Bkhan | can we use multiple USB GSM dongle device of Huawei with asterisk |
| 13:29.30 | WIMPy | yes |
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| 13:39.47 | anonymouz666 | I wonder how stable it is chan_dongle to consider as something that could be sold or production use |
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| 13:46.01 | coppice | anonymouz666: as stable as your cellular operation :-) |
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| 14:10.15 | bananapie | my sip provider hangs up my calls when asterisk updates caller id with the provider after a call transfer on my PBX. How can I specify for my provider that no callerid updates should be done ? |
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| 14:20.09 | BorjaGVO | Hello everyone. Question: where are extension's state (hint) stored? I looked for it executing 'asterisk -rx "database show all"' but I don't see the keyword "Idle" (as my extension is). |
| 14:20.29 | [TK]D-Fender | BorjaGVO: in RAM |
| 14:21.04 | [TK]D-Fender | BorjaGVO: AstDB is persistent.... states are not |
| 14:24.28 | BorjaGVO | [TK]D-Fender: Thanks. However, sometimes the "folloewd" states do not get distributed by jabber. When does jabber receieve teh state info? Sometimes I have to make some reloads (I'm running FreePBX) until states get distributed. |
| 14:26.04 | [TK]D-Fender | I'd say when the other side sends the info... |
| 14:27.48 | BorjaGVO | [TK]D-Fender: I mean in the first point (when does jabber get the info from the local server to distribute it) |
| 14:29.11 | [TK]D-Fender | BorjaGVO: perhaps when the entire module loads it might poll.... |
| 14:30.34 | mirela666 | Hello, Has anyone done some BLF for static Queue members? |
| 14:31.19 | mirela666 | I'm trying to set up a feature code as PauseQueuMember(,SIP/XXX) but that coma for argument is making problem for feature code |
| 14:31.56 | mirela666 | is there any other wqay to notify Queueu or user status (DND for example) |
| 14:32.35 | mirela666 | I set up puqm => *99,peer,PauseQueueMember,,SIP/998 |
| 14:32.41 | mirela666 | feature code |
| 14:33.04 | mirela666 | and it is seen as == Mapping Feature 'puqm' to app 'PauseQueueMember()' with code '*99' |
| 14:33.08 | mirela666 | no argument |
| 14:33.28 | mirela666 | tried with ",', but not working |
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| 14:38.15 | [TK]D-Fender | Why would you be doing this with a feature code and not a notmal extension? |
| 14:38.23 | [TK]D-Fender | normal* |
| 14:39.48 | mirela666 | I need optin to login/logout on button |
| 14:40.01 | mirela666 | like DND |
| 14:40.05 | Qwell | [TK]D-Fender: I could see pause being extremely useful, during a call |
| 14:40.33 | Qwell | mirela666: iirc, you can escape the comma in the feature code. \, |
| 14:40.44 | mirela666 | maybe if I set dynamic members |
| 14:40.47 | [TK]D-Fender | mirela666: Doesn't matter on having it "on a button", the question is WHEN do you have to be able to do this... |
| 14:41.14 | mirela666 | Qwell: tried, not working, same separator :( |
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| 14:41.31 | Qwell | what are you putting, exactly? |
| 14:42.13 | mirela666 | [TK]D-Fender: for example 8AM agent login and 16PM agent log out |
| 14:42.28 | mirela666 | so IT is inposible with static agent |
| 14:42.29 | [TK]D-Fender | mirela666: you only need features.conf to trigger it DURING a call |
| 14:42.38 | Qwell | ; AppArguments -> These are the arguments to be passed into the application. If you need commas in your arguments, you should use either the second or third syntax, above. |
| 14:42.42 | [TK]D-Fender | mirela666: If you just want a basica login then use the dialplan like everyone else |
| 14:42.53 | mirela666 | I gues sI need a feature code to AgentLogin |
| 14:43.11 | [TK]D-Fender | no need for feature codes at all |
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| 14:43.33 | [TK]D-Fender | The method you are using is not a good approach for this. |
| 14:44.11 | mirela666 | but for the ON/OFF button on the phone functionality I need a feature code I guess |
| 14:44.17 | mirela666 | as sip uri |
| 14:44.20 | BorjaGVO | One more question: what happens with an extension state if I enable keeping track of states (make changes in the .conf files) and reload. Is the extension state checked or it will be updated when the extension state changes (that is, the device associated to it changes)? |
| 14:44.32 | mirela666 | aaah sorry no |
| 14:44.34 | [TK]D-Fender | [10:44]mirela666but for the ON/OFF button on the phone functionality I need a feature code I guess <- no, it's a dumb speed-dial like anything else |
| 14:44.56 | mirela666 | [TK]D-Fender understood , thanks :D |
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| 16:06.04 | redotis | Hey how do you check voicemail remotely with comedia mail? |
| 16:06.22 | jmetro | you program it. |
| 16:08.55 | ChannelZ-Wk | VoiceMailMain |
| 16:09.12 | ChannelZ-Wk | same as you check it "locally" |
| 16:11.41 | redotis | Sorry, I'm confused. Comedia mail takes care of everything the way I do it now. How do I just check the mail when I dial in? |
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| 16:18.19 | [TK]D-Fender | VoiceMailMain is it. |
| 16:18.25 | [TK]D-Fender | Get to it however you want |
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| 16:25.24 | redotis | So I have no damn clue how to call in and check voicemail. |
| 16:25.27 | redotis | Anyone? |
| 16:25.56 | _Corey_ | redotis: Did you set this Asterisk box up? |
| 16:26.01 | jblack | Make an extention called voicemail. |
| 16:26.12 | redotis | yes |
| 16:26.23 | jblack | pardon, that calls voicemail |
| 16:26.45 | _Corey_ | redotis: How is it receiving calls from the outside world? |
| 16:26.47 | redotis | Guys...that just enters you into comedia mail. |
| 16:26.49 | redotis | I've done that. |
| 16:26.53 | redotis | How do I CHECK it? |
| 16:27.49 | _Corey_ | You've been given the answer. You need to create something that executes VoiceMailMain. |
| 16:27.56 | *** part/#asterisk Blashyrkh (~michaeld@043-054-094-081.as39912.net) |
| 16:29.58 | redotis | So there's no way to do it from something like this after entering the voicemail menu? same => n,Voicemail(287@default,su) |
| 16:30.47 | _Corey_ | redotis: Is your question how to create a way to retrieve messages while listening to the voicemail greeting...? |
| 16:30.57 | redotis | yeah |
| 16:31.47 | [TK]D-Fender | "core show application voicemail" shows how to exit for this |
| 16:31.56 | _Corey_ | You can dial 0 or * while listening to the greeting and execute an 'o' or 'a' extension (respectively). Put your VoiceMailMain there. I normally use * for this. |
| 16:32.51 | redotis | so after this same => n,Voicemail(287@default,su) |
| 16:33.48 | redotis | 0 and * tries to execute o or a |
| 16:33.59 | _Corey_ | redotis: Read this: http://ofps.oreilly.com/titles/9781449332426/asterisk-Voicemail.html |
| 16:34.05 | redotis | I have |
| 16:34.38 | redotis | Having trouble grasping what's going on here |
| 16:35.05 | [TK]D-Fender | make extension. press the key during the greeting |
| 16:35.15 | [TK]D-Fender | does what you put in the extension. |
| 16:35.19 | redotis | so voicemail is listening for 0 or *? |
| 16:35.33 | _Corey_ | redotis: See 'exitcontext' in the link. |
| 16:35.36 | redotis | and will kick you over to the extension o or a respectively if it gets it? |
| 16:35.51 | _Corey_ | Yes. |
| 16:35.54 | redotis | ok thanks |
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| 16:48.06 | redotis | welp..that didn't work. |
| 16:48.18 | redotis | wait... |
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| 16:52.51 | redotis | ok...thanks a lot |
| 16:52.53 | redotis | derp |
| 16:53.26 | redotis | Works like a champ. Thanks for the help. |
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| 17:52.01 | herculey | i need help pls, i user asterisk 11.2 but i alwayes get this erros : [2013-10-21 13:50:53] ERROR[56002]: pbx.c:3972 ast_func_write: Function LANGUAGE not registered .. any suggestion pls? thanks in advance ... |
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| 17:52.58 | file | probably should use CHANNEL(language) instead |
| 17:53.26 | herculey | file, thanks a lot, but where i can fix this? |
| 17:53.53 | file | that would be in extensions.conf probably - where exactly... I have no idea as that is a configuration file controlled by you |
| 17:54.34 | herculey | so i must set a line in extentions.conf ? |
| 17:54.50 | herculey | i`am using en insteed |
| 17:55.54 | file | have to fix a line. |
| 17:55.59 | [TK]D-Fender | herculey: LANGUAGE is not a function, it is a value of the CHANNEL function you set the way file has indicated |
| 17:57.32 | herculey | [TK]D-Fender , i guessed that, i tried all the metods after migrationg to a new version of asterisk but no chance :( |
| 17:58.00 | [TK]D-Fender | herculey: ujust go use the CHANNEL function... it has been the proper way for a long time... |
| 17:59.42 | herculey | [TK]D-Fender , thanks for your answer, but from where i can set the channel function pls? |
| 18:02.48 | [TK]D-Fender | herculey: in extensions.conf. .... it is a DIALPLAN function... |
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| 18:04.51 | herculey | [TK]D-Fender , thanks again, i gonna see now |
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| 18:14.47 | *** topic/#asterisk by mjordan -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 11.6.0 (2013/10/21), 10.12.3 (2013/08/27), 1.8.24.0 (2013/10/21), DAHDI-linux 2.6.2 (2013/03/08), DAHDI-tools 2.6.2 (2013/03/08), libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
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| 18:36.40 | herculey | [TK]D-Fender , thanks a lot , solved. the error gone.. have a great time :) |
| 18:36.56 | [TK]D-Fender | herculey: You're welcome. |
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| 21:23.20 | Blashyrkh | i just installed asterisk from source and now iam running it as user asterisk but it isnt loading any modules |
| 21:23.46 | Blashyrkh | where do i set the variables for logging, so that i can maybe see what is going wrong |
| 21:25.00 | Blashyrkh | when i do a module load chan_sip.so in the cli it just tells me that it failes |
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| 21:28.44 | ChannelZ-Wk | did you change the perms on all the files? and it has access to the whole path? |
| 21:29.44 | Blashyrkh | /usr/lib/asterisk is set to 755 asterisk:asterisk |
| 21:29.57 | Blashyrkh | wich is my user |
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| 21:35.06 | mitchrodrigues | Afternoon all, so im trying to hunt down a memory leak, and after sifting through 20000k lines of *alloc/free lines, I am only comming up with |
| 21:35.07 | mitchrodrigues | http://pastebin.com/nneDDYQ9 |
| 21:36.25 | realbass1 | Hello All, I'm trying to figure out what is wrong with my asterisk setup today- I'm on 11.5.1, and while I can log into the CLI, sip show peers shows everything as unreachable/unknown, nothing is showing up in my CLI though I have verbosity set to 10... however, my calls are getting processed. Anyone run into this before? |
| 21:36.30 | mitchrodrigues | its a nasty memory leak too, after processing 8000 or so puts it up at about 700Mib and continues climbing (just my example ive seen it go alot higher then that |
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| 21:36.55 | mitchrodrigues | ALSO valgrand has forsaken me and asterisk sigills when trying to be run in valgrind |
| 21:37.24 | Qwell | mitchrodrigues: See MALLOC_DEBUG in menuselect. |
| 21:37.31 | mitchrodrigues | I am using that |
| 21:37.41 | Qwell | So what does memory show allocations show? |
| 21:37.50 | mitchrodrigues | one sec |
| 21:37.55 | mitchrodrigues | i didnt know about that command |
| 21:38.05 | mitchrodrigues | hell thats far better then the ast_mmlog i added |
| 21:38.49 | mitchrodrigues | i will paste |
| 21:38.53 | mitchrodrigues | in pb |
| 21:38.56 | mitchrodrigues | its brutal |
| 21:41.27 | mitchrodrigues | lol the show allocation |
| 21:41.30 | mitchrodrigues | is bigger then 500kb |
| 21:41.36 | mitchrodrigues | and i cant upload to pb hahaha |
| 21:41.36 | mitchrodrigues | one sec |
| 21:42.15 | Blashyrkh | i have no chan_sip.so in my asterisk/modules |
| 21:42.25 | Blashyrkh | isnt that compiled by default |
| 21:43.24 | mitchrodrigues | you need libss/openssl :) |
| 21:43.39 | Blashyrkh | for chan_sip? |
| 21:44.08 | Blashyrkh | i see its not even choosabel in menuconfig |
| 21:44.53 | mitchrodrigues | http://taazoo.net/memory.txt |
| 21:45.00 | mitchrodrigues | @ Qwell |
| 21:46.09 | Qwell | only 8M? |
| 21:47.17 | realbass1 | nvm on my question- found that a developer made a mistake and started another asterisk process on my machine which took over all the registrations and call processing |
| 21:47.41 | mitchrodrigues | that was on one call |
| 21:47.42 | mitchrodrigues | locally |
| 21:47.47 | mitchrodrigues | I can push it back into production |
| 21:47.53 | mitchrodrigues | and watch it goto 21% of the total ram |
| 21:47.55 | mitchrodrigues | in less then a day |
| 21:48.07 | Qwell | I mean, if you're trying to find a memory leak... |
| 21:48.18 | Qwell | There's got to actually be something leaking. |
| 21:48.19 | mitchrodrigues | Ok let me push more calls throug hti |
| 21:48.35 | *** join/#asterisk newtonr (~newtonr@nat/digium/x-kyyvtcoicyyrcjbg) |
| 21:48.35 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 21:50.02 | mitchrodrigues | <PROTECTED> |
| 21:50.03 | mitchrodrigues | 25543 root 15 0 657m 48m 9376 S 0.7 0.9 0:31.14 asterisk |
| 21:50.05 | mitchrodrigues | no calls |
| 21:50.10 | mitchrodrigues | currently |
| 21:50.20 | mitchrodrigues | and thats after only 100 |
| 21:51.24 | Qwell | 48M seems pretty reasonable to me. |
| 21:51.29 | mitchrodrigues | with no calls? |
| 21:51.43 | Qwell | du -hs /usr/lib/asterisk/modules /usr/sbin/asterisk |
| 21:52.14 | mitchrodrigues | I forget about that, Ok i will get back to you in an hour or so |
| 21:52.20 | mitchrodrigues | once i have more calls |
| 21:52.35 | mitchrodrigues | should only be about 20minutes before i get it above 300m |
| 22:01.57 | *** join/#asterisk bipul (~bipul@unaffiliated/bipul) |
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| 22:41.00 | Ice_Strike | How well does asterisk handle with 100 channels on ESXi host? |
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| 22:49.25 | *** join/#asterisk powerunits (272adbfe@gateway/web/freenode/ip.39.42.219.254) |
| 22:49.35 | powerunits | hi team, |
| 22:53.33 | *** join/#asterisk dfighter (~someone@arcemu/staff/dfighter) |
| 23:00.26 | powerunits | team i need some help please.. actually i have sip user in asterisk realtime ... that sip user is register in my system with his mobile number. and he calls on our inbound number his call get rejected by asteirsk.. if he dial our inbound number from his other mobile his call get connect with our system. |
| 23:00.29 | powerunits | http://pastebin.com/JNQSKNwf |
| 23:00.39 | powerunits | here are logs .. please can some one help me on this? |
| 23:02.33 | paulc | When he calls your inbound number, Asterisk matches his caller ID as presented by your telco against a peer.. and fails.. because the telco doesn't send the right authorisation.. I think.. |
| 23:02.41 | paulc | this is why it's bad to use numbers as peer names |
| 23:02.57 | paulc | get his mobile to register as paulc and have a [paulc] peer - the problem will go away. |
| 23:03.41 | powerunits | thanks let me try |
| 23:07.20 | powerunits | great help .... thank you very much.. |
| 23:08.12 | paulc | doffs hat - glad to be able to help |
| 23:11.39 | mitchrodrigues | Qwell: thanks for your help, ill have more information tomorrow, to late in the day to put a server in production. |
| 23:11.53 | mitchrodrigues | Secondly, i like your guys' memory stuff its pretty clever. |
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