| 00:01.36 | navaismo | hmm there is no ps_transport table, can I create and it will work? |
| 00:01.46 | file | you can't create transports in realtime |
| 00:02.02 | navaismo | too bad |
| 00:03.36 | file | using static realtime the information could be stored in a database |
| 00:04.00 | navaismo | yes, i use more static realtime |
| 00:04.23 | navaismo | so if i crete it it will work for static realtime? |
| 00:04.54 | file | nothing has to be written to use static realtime, it's part of the normal configuration functionality |
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| 00:25.39 | navaismo | but to the database |
| 00:27.03 | WIMPy | Hmm. Doesn't WaitExten go to t any more if it doesn't receive any input? |
| 00:30.57 | navaismo | thank you all see you later |
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| 03:15.34 | DarthExpeditor | OK. I have two sip accounts that I created and I have soft clients registered to both |
| 03:15.46 | DarthExpeditor | Next step is the extensions.conf right? |
| 03:18.30 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.76) |
| 03:22.00 | Penguin | You'll want extensions if you plan to do anything useful. |
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| 03:25.21 | DarthExpeditor | yeah. working on my extensions file now |
| 03:25.38 | DarthExpeditor | this has been very educational so far though. |
| 03:26.10 | Penguin | ~book |
| 03:26.10 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 03:26.27 | Penguin | ~primer |
| 03:26.27 | infobot | rumour has it, primer is http://burner.com/asterisk-primer |
| 03:26.34 | DarthExpeditor | I have the book. |
| 03:26.52 | DarthExpeditor | It jumps around and is pretty frustrating |
| 03:27.08 | DarthExpeditor | I have found exploring on my own and using google easier. |
| 03:27.14 | Penguin | Read it by section from the TOC and not like a novel. |
| 03:28.21 | DarthExpeditor | Hmmm... Can you give me an example? |
| 03:29.01 | Penguin | Example 1: Want to learn about extensions? Look at the table of contents for that topic, go to the section, read it. |
| 03:29.25 | Penguin | Example 2: Want to learn how to make an Auto attendant? Look in the TOC for that topic, go to the section, read it. |
| 03:29.30 | DarthExpeditor | Ahhh, yeah, I will be at that level soon. |
| 03:29.44 | Penguin | Don't read it from cover to cover. You'll just get annoyed. |
| 03:29.45 | DarthExpeditor | Not quite to the desk reference level yet. |
| 03:30.11 | DarthExpeditor | Yeah. The reason I'm doing this is to understand how to do this from scratch |
| 03:32.55 | DarthExpeditor | Any thoughts on fixing this, |
| 03:32.56 | DarthExpeditor | <PROTECTED> |
| 03:33.47 | Penguin | That makes me think you don't have the ast DB created and present. |
| 03:33.58 | DarthExpeditor | in MySQL? |
| 03:34.08 | Penguin | No, it should be sqlite. |
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| 03:34.44 | DarthExpeditor | Crap |
| 03:34.47 | DarthExpeditor | How do I check that? |
| 03:35.01 | Penguin | Do you have sqlite installed? |
| 03:35.04 | DarthExpeditor | If it were MySQL I wouldn't have to ask. LOL |
| 03:35.08 | DarthExpeditor | let me check |
| 03:36.04 | DarthExpeditor | Package sqlite-3.6.20-1.el6.x86_64 already installed and latest version |
| 03:36.13 | DarthExpeditor | Misconfiguration? |
| 03:37.52 | DarthExpeditor | <PROTECTED> |
| 03:37.55 | DarthExpeditor | Path in the res_config_sqlite.conf points to |
| 03:38.43 | DarthExpeditor | but that file doesn't exist |
| 03:39.24 | DarthExpeditor | I think I figured it out |
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| 04:11.10 | DarthExpeditor | Ok |
| 04:11.31 | DarthExpeditor | I'm stuck, I can't solve this SQLite issue. |
| 04:11.50 | DarthExpeditor | The file that it needs doesn't exist and I'm not sure how to create it |
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| 04:19.12 | Penguin | I couldn't tell you. Something I haven't used is asterisk >1.8. |
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| 04:38.06 | ChannelZ | Did you build asterisk yourself or is it a package? |
| 04:45.00 | MaliutaLap | using butterflies to alter the magnetic properties of the earth to write to the disks |
| 04:49.50 | ChannelZ | 'going commando' |
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| 04:51.17 | MaliutaLap | ok, that shows how tired I am. I read ChannelZs last question as "Did you build asterisk yourself, in your garage?" |
| 05:05.45 | ChannelZ | You never know |
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| 05:16.57 | MaliutaLap | ChannelZ: you're saying I was reading your mind? |
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| 10:12.23 | Rico | hi there |
| 10:13.08 | Rico | I want to execute an action after a Dial in a perl AGI. But right after the caller / callee hangups, the AGI script is terminated and action is not done |
| 10:13.37 | Rico | how can I do taht ? I saw that deadAGI is deprecated in latest 1.8 |
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| 10:27.25 | b7 | Hi everyone. I have a strange problem: some peers can't register against the asterisk server. I have the two absolutely identical spa921 phones with the same settings and one of them can register but another can not. |
| 10:27.53 | b7 | Can anyone give a tip? |
| 10:28.16 | Chainsaw | Plug the network cable into the other phone. |
| 10:28.51 | Chainsaw | See if the problem moves. If it does, connectivity. If it doesn't, a setting problem more subtle then you anticipated. |
| 10:28.56 | bulkorok | Rico: check h-extension |
| 10:29.07 | Rico | bulkorok: I want to do it in the AGI |
| 10:29.18 | Rico | I don't want to return in the extensions.conf dialplan |
| 10:29.30 | bulkorok | then do it in agi |
| 10:29.40 | bulkorok | afair this should be possible |
| 10:30.14 | Rico | mmmh, I don't know how |
| 10:33.29 | bulkorok | is deadagi deprecated?! |
| 10:33.51 | Rico | [Nov 20 11:15:52] WARNING[3167]: res_agi.c:3930 deadagi_exec: DeadAGI has been deprecated, please use AGI in all cases! |
| 10:34.00 | Rico | it seems yes |
| 10:34.02 | bulkorok | :-D |
| 10:35.42 | kaldemar | Rico: core show application AGI <-- AGISIGHUP |
| 10:40.02 | Rico | kaldemar: thakns, it seems to work |
| 10:40.17 | Rico | but there's no do cabout that in core show app AGI |
| 10:40.57 | b7 | The network cable is plugged in. I did "sip set debug on" and I see the following: http://pastebin.com/8msqtWzj |
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| 10:42.20 | b7 | The phone do not send the password to server, as far as I can understand. Another same phone send it. Anyone have an idea why? |
| 10:46.56 | Rico | b7: same firmware ? have you reset the phone ? does the two asterisk peers have exactly the same config ? no error or warning messages in asterisk CLI (with core set debug 10 and core set verbose 10) ? |
| 10:48.20 | b7 | Yeah, absolutely same firmware, settings, I did reset the both phones and the config for peers are exactly same and debug and verbose enabled and no error messages. |
| 10:50.36 | b7 | That is really mysterious for me. |
| 10:50.54 | Rico | your phone does not send correct response to 401 reply |
| 10:51.14 | Rico | it does not send www-authenticate response to asterisk |
| 10:51.42 | Rico | check the settings of your phone once again |
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| 11:04.52 | b7 | Yes, I think same. But I did the checks countless times and can't find any differences. |
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| 11:26.53 | b7 | Please, somebody help. I'm stuck. :( |
| 11:30.35 | petn-randall | Hi, is the info that there can only be one SIP endpoint per account still valid for Asterisk 11? (http://lists.digium.com/pipermail/asterisk-users/2011-March/260080.html) |
| 11:33.01 | LooserOuting | petn-randall: pjsip in Asterisk 12 can have many aors for one username. i am pretty sure in asterisk 11 it's not possible. |
| 11:33.02 | kaldemar | petn-randall: yes. |
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| 12:11.35 | petn-randall | LooserOuting, kaldemar, thanks for the info. |
| 12:22.36 | b7 | Ugh, it seems like I solved my problem. |
| 12:24.12 | b7 | That was a network misconfiguration between the server and the second phone, the server was sending the "unathorized" package, but in some reasons the package never got it to the phone, so the phone was asking for registry again and again... |
| 12:28.18 | cusco | gi folks |
| 12:29.07 | cusco | having a peer from a 3rd party sip provider, what is the best way to dial? Dial(SIP/${EXTEN}@peer) or Dial(SIP/peer/${EXTEN}) ? |
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| 12:33.11 | MaliutaLap | cusco: are you trunked to the peer? (i.e. registered as a sip peer) |
| 12:36.19 | MaliutaLap | cusco: for example I have DID from a sip provider, I set variables for my various channels (SIP, IAX and DAHDI - e.g. HANDSET="SIP/phone". I also use ael for my dialplan), then refer to the variables in dial: e.g. DIAL(${HANDSET}/${EXTEN}) |
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| 12:40.43 | MaliutaLap | cusco: it's a bit different if you're just "dialing" into a sip host you're not registered with. You might have to do a DIAL(SIP/${EXTEN}@host) |
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| 13:28.34 | ghost75 | is this correct for AMI? Variable = '__SIPADDHEADER51=Call-Info: <sip:asterisk@192.168.2.2>\;answer-after=0' |
| 13:30.06 | Greenlight | No. |
| 13:30.15 | Greenlight | AMI needs actions. The action you need is SetVar |
| 13:30.40 | ghost75 | the action is originate |
| 13:31.09 | Greenlight | Yup, you can set them there as well |
| 13:31.28 | ghost75 | that __ and 51 is needed? |
| 13:31.42 | Greenlight | The "__" allows unlimited inheritence |
| 13:32.57 | Greenlight | The function is just SIPAddHeader, so I'm not sure about the 51 |
| 13:33.03 | ghost75 | http://www.voip-info.org/wiki/view/Asterisk+cmd+SipAddHeader there it says: 51 is because the asterisk limits do 50 calls to the func "SipAddHeader" and this way it wont interfere |
| 13:33.34 | Greenlight | Yea, voip-info can be outdated at times |
| 13:34.01 | ghost75 | yeah :< |
| 13:34.14 | Greenlight | I would try without it |
| 13:34.28 | Greenlight | And firstly, are you sure that header is exactly what you want ? |
| 13:34.32 | ghost75 | is the header only active during the originate or for all calls? |
| 13:34.44 | Greenlight | What do you mean ? |
| 13:35.15 | ghost75 | i want to make an originate which calls my phone and somebody outside but i want that my phone automatically picks up and doesnt ring |
| 13:35.35 | Greenlight | And you're sure (eg you've tested) that your phone supports it ? |
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| 13:35.54 | ghost75 | support the sip header? |
| 13:36.06 | Greenlight | Yes. |
| 13:36.13 | ghost75 | will just try :) |
| 13:36.27 | Greenlight | Try with normal dialplan, *then* try with AMI |
| 13:36.30 | Greenlight | And, tbh |
| 13:36.42 | Greenlight | I would be tempted to just use a Local channel, and add the header in the Dialplan |
| 13:36.46 | Greenlight | Makes it clearer imo |
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| 13:37.32 | tomodachi | hi im using asterisk 1.8 , conference keeps complaning about no dahdi channel available (im using a dummy one) and conference works just not the introduction, any idea what I could try? |
| 13:37.34 | Greenlight | So, say your are device "1000". Originate to Local/1000@autoanswer |
| 13:37.47 | ghost75 | so for what the header is active, for all calls on this extension? |
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| 13:38.25 | Greenlight | ghost75: DOn't understand what you mean. |
| 13:38.28 | sjs205 | Hello all, I've got a problem with an inbound sip trunk, I have allowguest=no in sip.conf, but when i call on the inbound trunk I get "Sending fake auth rejection for device"... how can i rectify this? |
| 13:38.55 | arnotixe | hi all I'm wondering if this is correct behaviour: I set up in extensions.conf "exten => 9907,1,Dial(SIP/arno)" but forgot to specify the user "arno" in sip.conf. Now, I don't get "invalid extension" or something, the call just hangs. Never HangsUp(). |
| 13:39.01 | Greenlight | sjs205: Rectify what? |
| 13:39.16 | ghost75 | the extension should not autoanswer every call, only the ones i want from originate |
| 13:39.26 | Greenlight | arnotixe: It's likely trying to resolve arno as a host name |
| 13:39.32 | arnotixe | aha |
| 13:39.43 | sjs205 | Greenlight, make it so I can calls from known numbers are "allowed in" |
| 13:39.44 | Greenlight | ghost75: Yes, so only calls that reach it via the "autoanswer" context. |
| 13:39.52 | sjs205 | Sorry, should have been more specific :) |
| 13:40.14 | Greenlight | sjs205: Defined "known numbers" ? Is this from a known peer ? |
| 13:40.26 | ghost75 | ok |
| 13:40.53 | sjs205 | Greenlight, basically, at the moment it is my mobile number, I have tried defining this as a peer but I get the message above |
| 13:41.20 | Greenlight | Um... you don't define a number as a peer. How is that call being delivered to you ? |
| 13:41.46 | sjs205 | Greenlight, through our inbound sip trunk... |
| 13:42.03 | arnotixe | Greenlight: Problem is, it also hangs on things like "9907,1,Dial(SIP/930&SIP/932&SIP/arno)" - I expected it to just ignore the extension SIP/arno, as it would if the peer was not registered. If 930 isn't registered, it simply isn't tried. How can I make asterisk try registered peers only? |
| 13:42.05 | Greenlight | Doesn't sound like it's being matched against that trunk. |
| 13:42.17 | sjs205 | Greenlight, I know it works since when I allowguest=yes, the call is routed correctly |
| 13:42.30 | Greenlight | As I said, the peer isn't being matched.... |
| 13:42.39 | Greenlight | Hence when you allow guests it works. |
| 13:42.42 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 13:43.52 | sjs205 | Greenlight, hmmm, that is sort of what i thought... I have the peer definied in the correct context... could it be that the peer is host=dynamic? |
| 13:44.16 | Greenlight | sjs205: Your ITSP should give you the details about this |
| 13:45.02 | sjs205 | Greenlight, We are using gradwell... I will have a look see what they say... |
| 13:45.26 | Greenlight | https://support.gradwell.com/entries/23669416-Configuring-an-inbound-SIP-trunk-on-an-Asterisk-PBX |
| 13:45.38 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
| 13:46.14 | Greenlight | There's load of stuff on Google... |
| 13:46.28 | Greenlight | What do you have in sip.conf at present ? |
| 13:46.59 | arnotixe | Greenlight: this hanging complicate matters more when the call comes in through an iax2 trunk - then the dialplan hangs with this: http://pastebin.com/HK3hu7Mn and I actually have to restart the asterisk service to "unlock" the iax2 trunk. Which is bad... |
| 13:47.21 | Greenlight | Apparently they want you to allow ananymous inbound calls... that's nuts. |
| 13:47.26 | Greenlight | Get a decent ITSP ;/ |
| 13:51.01 | *** join/#asterisk darkbasic (~quassel@host37-245-static.119-2-b.business.telecomitalia.it) |
| 13:53.03 | *** join/#asterisk anonymouz666 (~anonymouz@189-25-59-196.user.veloxzone.com.br) |
| 13:53.27 | anonymouz666 | I am back |
| 13:58.58 | arnotixe | Greenlight: I'm not so sure asterisk actually tries to resolve SIP/arno to the hostname "arno". I added "arno" with the IP of my machine to its /etc/hosts and sniffed the my machine's net with wireshark, and see no connection attempts from the pbx to my machine. How can I tell wireshark to not try non-existing SIP/<whatever> accounts? |
| 13:59.46 | Greenlight | The better question, is why are you trying to dial a device that doesn't exists. |
| 14:00.36 | arnotixe | heh well, I deleted it from sip.conf but forgot about extensions.conf |
| 14:00.56 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
| 14:02.27 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 14:02.45 | arnotixe | I understand I should keep the pbx organized, but wouldn't it be good if asterisk handled that kind of errors? Could be bad side effects to this, of course, and those could be good reasons for /not/ implementing such error handling. |
| 14:02.59 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 14:04.32 | anonymouz666 | holiday today... |
| 14:05.49 | *** join/#asterisk endemic (~endemic@lynx.onvox.net) |
| 14:06.22 | *** join/#asterisk serafie (~erin@nat/digium/x-dwsxuedgukuepkct) |
| 14:06.28 | *** join/#asterisk tparcina (~tomo@cisco15.fesb.hr) |
| 14:07.11 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 14:13.00 | anonymouz666 | anyone in here using connected line stuff? |
| 14:14.18 | WIMPy | yes |
| 14:16.09 | *** join/#asterisk serafie (~erin@nat/digium/x-gwtaqsmfjeaeomrv) |
| 14:18.18 | tomodachi | hello , meetme complains about dahdi , i have the module loaded in my kernel but metme still complains. Im also unable to load the module in asterisk |
| 14:18.42 | WIMPy | Hmm. Is there no way to find out about the status of a queue from the dialplan? |
| 14:19.34 | Greenlight | https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_QUEUE_MEMBER_COUNT <-- Things like that ? |
| 14:19.40 | WIMPy | tomodachi: Did you (re)build Asterisk after installing dahdi? But maybe you should take a look at ConfBridge instead anyway. |
| 14:20.26 | WIMPy | Argh. I didn't expect 'core show functions like queue' to be case sensitive :-( |
| 14:20.39 | Greenlight | Heh :) |
| 14:21.41 | tomodachi | tomodachi: i installed asterisk from my distributions source, it comues with support for dahdi since it has meetme and i can see the module |
| 14:21.51 | tomodachi | WIMPy: |
| 14:22.06 | WIMPy | QUEUE_WAITING_GOUNT would be it then. |
| 14:22.49 | *** join/#asterisk zigg (~matt@unaffiliated/zigg) |
| 14:24.47 | WIMPy | Argh. That doesn't work as I have to play the announcement befor putting callers in to the queue :-( |
| 14:25.16 | WIMPy | Or is there something I have missed to play it immediately from Queue()? |
| 14:30.26 | WIMPy | Using Queue to ensure calls are delivered in the correct order was nice, but I think I get more issues than it solves :-( |
| 14:31.52 | WIMPy | So does anyone have a good idea how to ensure calls are delivered in the correct order without using Queue()? |
| 14:32.13 | ghost75 | Greenlight: now i have in internal context for testing: SIPAddHeader(Call-Info: answer-after=0) but doesnt seem to work on cisco phone or on zoiper soft phone |
| 14:32.16 | anonymouz666 | WIMPy: so it works like... I call SIP/666 and then I can update the connected caller id name to "anonymouz" after answer? |
| 14:32.45 | WIMPy | anonymouz666: Any time. |
| 14:33.01 | anonymouz666 | sounds... COOL! |
| 14:34.04 | WIMPy | I used it to display muted/unmuted status in ConfBridge, for example. |
| 14:35.27 | colinwielga | Is it true that when pinging the GW from my PBX, the pings aren't necessarily a reliable indicator of connection quality since the gw can prioritize packets? I'm getting high ping spikes to my GW and I am being told by a tech not to look into the high ping spikes too much because they are inconclusive due to the fact that the GW can wait until more important packets go through before responding to the ICMP packets…Is |
| 14:35.27 | colinwielga | this correct? Just need a second opinion |
| 14:36.45 | WIMPy | That is correct. |
| 14:37.06 | WIMPy | Or can be, if your router is of the better kind. |
| 14:38.33 | sjs205 | In sip.conf, does allow guest only apply to calls, that is, if allowguest=yes, incoming calls will be allowed from anybody but unauthenticated users will not be able to make calls? |
| 14:39.21 | WIMPy | That question doesn't make sense. |
| 14:39.45 | WIMPy | allowguest will enable unauthenticated calls. |
| 14:39.52 | WIMPy | The rest is about using contexts. |
| 14:40.09 | colinwielga | WIMpy, so I guess my next question is what would be a reliable method in determining the connection between the PBX and the GW? |
| 14:40.35 | WIMPy | There isn't any. |
| 14:41.02 | WIMPy | You can try to chect the connection to what you actually want to connect to. |
| 14:41.05 | sjs205 | WIMPy, my question doesn't make sense? |
| 14:41.33 | WIMPy | sjs205: Correct |
| 14:41.42 | *** join/#asterisk bananapie (~david@static-dslcom4-242.express.oricom.ca) |
| 14:41.45 | WIMPy | All calls are incomming calls. |
| 14:42.00 | [TK]D-Fender | [09:38]sjs205In sip.conf, does allow guest only apply to calls, that is, if allowguest=yes, incoming calls will be allowed from anybody but unauthenticated users will not be able to make calls? <- yes |
| 14:43.09 | sjs205 | Incoming calls being coming in throguh my inbound sip trunk |
| 14:43.59 | WIMPy | If you dial on your phone, that's an incomming call to Asterisk as well. |
| 14:44.34 | bananapie | Hello, I called Set(CHANNEL(musicclass)=DefaultTest) in priority 1 in my extension, I called MusicOnHold(OtherTest) in priority 2 in my extension. Trouble is that it is playing DefaultTest and not OtherTest. If I put DefaultTest as the parameter to MusicOnHold and OtherTest as set music class on hold, when I call MusicOnHold I hear OtherTest. The doc says that the parameter to MusicOnHold takes priority over the default, is it pos |
| 14:44.35 | bananapie | sible that I have misunderstood the doc ? I want to have different music when the user is placed on hold and when i send a user into MusicOnHold app |
| 14:44.47 | WIMPy | incomming/outgoing only makes sense on a per peer(device) level. |
| 14:46.05 | sjs205 | WIMPy, that is what i mean. Sorry. Basically, with allowguest=yes, I can call my inbound sip trunk and my call is routed correctly, if allowguests=no, the call fails due to no authentication... |
| 14:46.55 | WIMPy | ITSPs usually don't authenticate to you. You might use their IP for authentication instead. |
| 14:46.58 | [TK]D-Fender | sjs205: because you didn't set your system up properly to accept their call |
| 14:47.35 | sjs205 | So it is working with allowguest=yes, so all i wanted to know was whether this also allowed unauthenticaed sip users to connect to my server and make calls |
| 14:47.43 | sjs205 | [TK]D-Fender, story of my life! :/ |
| 14:47.55 | WIMPy | It allows anyone to make calls. |
| 14:48.10 | [TK]D-Fender | sjs205: fix your peer. allowguest=no |
| 14:49.28 | arnotixe | sjs205: WIMPy has a very important point. I did this once (and only once) when I was new to asterisk. Got a $1000 bill for one month's worth of calls to Africa... |
| 14:49.33 | sjs205 | [TK]D-Fender, I have created a peer for the mobile number that I'm using to call the trunk. When i set this to host=dynamic it always fails, when i set it to a specific IP it works but only when the call is from that IP, it seems to change all the time. |
| 14:49.44 | *** join/#asterisk mattcen (~mattcen@c110-22-201-130.sunsh4.vic.optusnet.com.au) |
| 14:49.45 | [TK]D-Fender | sjs205: that makes no sense. |
| 14:49.51 | [TK]D-Fender | You don't make a peer for a "number" |
| 14:50.01 | sjs205 | arnotixe, I know this, only last week ÂŁ60 worth of calls were made! |
| 14:50.16 | WIMPy | arnotixe: That's becaue of your dialplan and impropper use of contexts., not because of allowguests. |
| 14:50.25 | arnotixe | sjs205: sorry for you. |
| 14:50.28 | *** join/#asterisk slicknick5181 (~slick@207-255-114-073-dhcp.unt.pa.atlanticbb.net) |
| 14:50.55 | sjs205 | [TK]D-Fender, so how would i go about trying to allow calls on an inbound trunk? |
| 14:51.04 | sjs205 | from a particular number? |
| 14:51.06 | [TK]D-Fender | sjs205: "it seems to change all the time." <- maybe they have a few IP's, but probably not that many |
| 14:51.09 | arnotixe | WIMPy: yes I fixed that too. I have allowguests on one trunk, but all calls go to the switchboard... |
| 14:51.21 | [TK]D-Fender | sjs205: there is no such thing |
| 14:51.46 | ghost75 | ghost |
| 14:51.48 | [TK]D-Fender | sjs205: forget the words "from a particular number". Asterisk matches by USERNAME, or by IP |
| 14:51.52 | ghost75 | ups |
| 14:51.52 | sjs205 | [TK]D-Fender, so i guess it is back to the drawing board... I neexd to fix my contexts i guess? |
| 14:52.00 | sjs205 | oh i see... |
| 14:52.01 | [TK]D-Fender | sjs205: PPERS, not CONTEXTS |
| 14:52.04 | WIMPy | You can't allowguests "on one trunk". Either it's on or off. Global. |
| 14:53.09 | sjs205 | [TK]D-Fender, so i would define my peer with username whatever, and make this host="the list of possible hosts" |
| 14:53.20 | bananapie | nevermind, I'll just do variable gymnastics and reset the class before and after the command |
| 14:53.30 | [TK]D-Fender | sjs205: Host could be dynamic |
| 14:53.40 | [TK]D-Fender | sjs205: and "could" depends on what your PROVIDER sends you |
| 14:53.50 | WIMPy | sjs205: You can only have one host per peer. If you need multiple hosts, you need multiple peers. |
| 14:54.04 | sjs205 | WIMPy, that helps somewhat... |
| 14:54.07 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 14:54.07 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 14:54.11 | arnotixe | WIMPy: interesting. Anyways, I deleted the entire dialplan, read about contexts, rewrote everything and it helped a lot. |
| 14:54.24 | sjs205 | [TK]D-Fender, I don't think they send me much! ;) |
| 14:54.30 | WIMPy | arnotixe: That's the way to go. |
| 14:56.01 | [TK]D-Fender | sjs205: I thnk you need to look at your calls more.... |
| 14:56.31 | sjs205 | [TK]D-Fender, what do you mean look at my calls more? as in the SIP dialog? |
| 14:56.44 | [TK]D-Fender | sjs205: clearly |
| 14:56.50 | sjs205 | sure... |
| 14:57.23 | slicknick5181 | sjs205, If you must allow guests into your machine you could make is more secure by having Asterisk match the DID before passing the cal |
| 14:57.28 | arnotixe | WIMPy: always good to know what you're doing, especially when someone bills you for your calls :P Also had a problem with a SIP/GSM device being too open - as long as it had registered to my asterisk pbx, ANYONE could access its port 5060 and make calls (not only my pbx). Incredibly stupid. Had to jail it on a separate network with pbx connectivity only. |
| 14:58.45 | sjs205 | slicknick5181, hmmm... interesting |
| 14:58.59 | WIMPy | arnotixe: Ok, so security is one more (big) point on my list of why I don;t like GWs. |
| 14:59.13 | slicknick5181 | sjs205, Hold on 1 sec and I'll send you a pastebin |
| 14:59.49 | [TK]D-Fender | If you allow unauthed calls OUT then that's a bad mark on you... |
| 15:02.14 | arnotixe | WIMPy: Tried calling my cell provider and ask if they offer SIP connections, and no one ever understood what that was. Tried calling numerous times to get to different agents. Now I see they DO offer SIP, but they call it "mobile ip". heh and their web page says "order this if you want to use a SIP application". Good luck trying to train those customer care agents. |
| 15:03.25 | ghost75 | do not all sip phones support modifying sip header? |
| 15:03.33 | slicknick5181 | sjs205, http://pastebin.com/LRnixpuK |
| 15:03.44 | slicknick5181 | sjs205, Who is your provider? |
| 15:04.02 | WIMPy | It can happen the other way round as well. One telco tried to sell us their IP PBX trunk, but when I started to ask technical questions, they said they had no idea, as they never implemented it so |
| 15:04.04 | WIMPy | far. |
| 15:04.11 | sjs205 | slicknick5181, gradwell |
| 15:04.26 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 15:04.56 | Penguin | More like grad barely. |
| 15:06.17 | slicknick5181 | sjs205, I used a shit provider callcentric for a while and this was the setup I used becasue they have 512 different IPs to send calls from |
| 15:06.23 | [TK]D-Fender | ghost75: Your question is too vague |
| 15:06.37 | arnotixe | WIMPy: cool/scary. At our place I also have a salesman selling misc items - he usually don't check if it works, but is very happy when having made the sale. And who gets the trouble of setting it up/securing it? |
| 15:07.35 | sjs205 | slicknick5181, this is the only provider i've used... they seem okay... most of the problems relate to my lack of experience! |
| 15:07.36 | sjs205 | ;) |
| 15:09.08 | slicknick5181 | sjs205, It wasn't all the long ago I was in the same boat |
| 15:09.57 | Penguin | exten => s,1,Goto(${SIP_HEADER(To):5:10},1); |
| 15:10.07 | Penguin | That's the only line I use for dumb providers. |
| 15:11.28 | WIMPy | That looks dangerous to me. |
| 15:11.34 | arnotixe | Is it secure enough to put SIP peers allowed to make external calls, in a context with the outgoing route? Strange question. I mean like http://pastebin.com/1yecUK5m - Or should I add extra authentication on all outgoing calls? |
| 15:12.18 | WIMPy | It's all about contexts, yes. |
| 15:12.33 | Penguin | arnotixe: Build your contexts in a hierarchical manner. |
| 15:12.50 | WIMPy | Make sure all calls arrive in the correct context and make sure to only include context that you really want there. |
| 15:13.14 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.54) |
| 15:14.03 | arnotixe | ok so this way the sip peer authentication is sufficient security then? (I mean, assuming the people at the office can trust eachother not using other people's phones) |
| 15:14.35 | arnotixe | WIMPy: yes I can see why including contexts here and there is dangerous |
| 15:14.46 | arnotixe | dialplan show <context> is very handy |
| 15:15.28 | Penguin | Build a context hierarchy. |
| 15:15.57 | arnotixe | Penguin: yep. I do this by hand. What gui/webapp alternatives are there? |
| 15:16.16 | WIMPy | Yes, the context for your devices should probably only have includes of other contexts. |
| 15:16.38 | Penguin | For example: Phones get a context of 'phones', and the phones context includes 'outgoing_calls' and 'internals' contexts. |
| 15:17.34 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 15:17.34 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:17.37 | Penguin | Then if your incoming calls context needs to include the internals context, it doesn't allow dialing back out. |
| 15:18.00 | Penguin | The internals context only has extensions for handsets. |
| 15:19.14 | arnotixe | Since I have routes out in different countries, I have also made a context for each country, and phones include the "correct" outgoing-context. My phone includes all, of course :D |
| 15:21.50 | Penguin | As long as only authenticated peers get access to any outbound contexts, and no incoming contexts include outgoing contexts, it should be fine. |
| 15:21.57 | arnotixe | nice |
| 15:22.24 | *** join/#asterisk LeandroDardini (~leandro@ip-224-67.sn2.eutelia.it) |
| 15:22.35 | LeandroDardini | hello all |
| 15:22.37 | Penguin | But that's the purpose for context hierarchy. It allows you to make one-way overlaps in contexts. |
| 15:23.55 | *** join/#asterisk whizzi (~whizzi@D4B2620B.static.ziggozakelijk.nl) |
| 15:24.07 | whizzi | hi :) |
| 15:29.00 | LeandroDardini | How can I use the Dial command to dial using SIP and providing username, password, server and extension? |
| 15:29.13 | arnotixe | Penguin: My biggest source of dialplan confusion was that I initially thought a SIP peer in one context couldn't be called from whatever other context. Like a context was some kind of jail to those peers. I used to work with analogue PBX systems and all usually are built up in peer "classes" with their set of privileges. The asterisk way of inheriting privileges was new to me. |
| 15:29.44 | arnotixe | I guess it *could* be set up like that in asterisk, too, but very redundant and not flexible. |
| 15:29.45 | LeandroDardini | I tried using Dial(SIP/username:password@server/ext) but I get an error about "Conflicting extension values given" |
| 15:29.47 | Penguin | leandrodardini: Do you always dial the same host and user? |
| 15:30.08 | LeandroDardini | the same host, but not using the same user and password |
| 15:30.35 | cusco | MaliutaLap: I'm registered, it works both ways, but wich one is best? is there any impact on using one or another? |
| 15:31.30 | [TK]D-Fender | [10:29]LeandroDardiniI tried using Dial(SIP/username:password@server/ext) but I get an error about "Conflicting extension values given" <- show us the call |
| 15:33.20 | LeandroDardini | <PROTECTED> |
| 15:33.20 | LeandroDardini | <PROTECTED> |
| 15:33.20 | LeandroDardini | [2013-11-20 16:09:05] NOTICE[8764][C-0000140d]: chan_sip.c:29745 sip_request_call: Conflicting extension values given. Using '4902****' and not '011390574****14' |
| 15:34.01 | Penguin | ~pb |
| 15:34.01 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 15:35.03 | whizzi | Is there any major difference in the Dial command between Asterisk 1.8 and Asterisk 11? The dial command I'm using which worked on 1.8, doesn't seem to work on Asterisk 11. I have an entry [st-x2-zipzap01] in my sip.conf (type=friend, this is registered and ok to receive and make calls). My Dial command is as simple as Dial (SIP/st-x2-zipzap01/phonnr,60,) but the only thing I see is that it's trying to resolve st-x2-zipzap01 (and failing) |
| 15:35.09 | Penguin | leandrodardini: How many different user names do you need to dial? |
| 15:35.41 | LeandroDardini | maybe an hundred... maybe more |
| 15:36.15 | LeandroDardini | whizzi, do you have an entry in your sip.conf for [st-x2-zipzap01]? |
| 15:36.17 | Penguin | All on flowroute? |
| 15:36.30 | LeandroDardini | yes, same provider |
| 15:36.52 | LeandroDardini | I'd like to recover the username/password from the DB instead of defining 100 different trunks in sip.conf |
| 15:36.54 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 15:36.59 | whizzi | LeandroDardini: Yes, I have.. registered with username/password, type=friend |
| 15:36.59 | Penguin | I can't imagine having that many user names of my own on the same provider. |
| 15:37.26 | LeandroDardini | whizzi, do you see it if you run "sip show peers"? |
| 15:37.36 | whizzi | LeandroDardini: Yes :) |
| 15:38.07 | Penguin | Maybe that trailing comma is causing trouble. |
| 15:38.31 | whizzi | I'll remove that to see if that fixes the issue |
| 15:39.00 | LeandroDardini | I usually use this way: SIP/phonenumber@provider |
| 15:39.06 | Penguin | That's wrong. |
| 15:39.23 | Penguin | If you have a defined peer, use SIP/peername/extension. |
| 15:39.56 | Penguin | If you do not have a defined peer, use SIP/username@sip.domain.com |
| 15:40.10 | *** join/#asterisk SushiB (~Thunderbi@200.77.217.106) |
| 15:40.36 | LeandroDardini | Penguin, interesting... how do you provide the password in your last example? |
| 15:40.50 | Penguin | I don't. |
| 15:41.00 | [TK]D-Fender | LeandroDardini: "@flowroute/" <- this is not a proper FQDN |
| 15:41.14 | [TK]D-Fender | LeandroDardini: And is probably the problem |
| 15:41.18 | Penguin | If I have to authenticate, I configure a peer entry and dial SIP/peername/extension. |
| 15:43.42 | LeandroDardini | TK, unfortunately it seems not: -- Executing [8888@authenticated:2] Dial("SIP/103-DEVEL-00001418", "SIP/4902****:cyz*******@sip.flowroute.com/011390574****14") in new stack |
| 15:43.42 | LeandroDardini | [2013-11-20 16:41:58] NOTICE[10865][C-00001416]: chan_sip.c:29745 sip_request_call: Conflicting extension values given. Using '4902****' and not '011390574****14' |
| 15:44.29 | Penguin | I'm wondering about using SIP/extension@username:password@sip.flowroute.com |
| 15:44.45 | *** part/#asterisk bulkorok (~Benjamin@85.183.61.47) |
| 15:44.45 | [TK]D-Fender | LeandroDardini: And do you have a peer named [sip.flowroute.com] in sip.conf? |
| 15:45.03 | Penguin | I can't remember the correct format on this, but I know we have talked about it before. |
| 15:45.19 | Penguin | maybe SIP/extension:username:password@sip.flowroute.com |
| 15:45.24 | whizzi | no, the comma doesn't seem to be the problem |
| 15:45.25 | [TK]D-Fender | it's "Dial(SIP/user:pass@fqnd/extension) |
| 15:45.57 | Penguin | Maybe I'm thinking of register statements using both username and authuser. |
| 15:47.00 | *** join/#asterisk whizzi (~whizzi@D4B2620B.static.ziggozakelijk.nl) |
| 15:47.23 | LeandroDardini | TK, I read the same in the docs... but it is not working for me in asterisk 11.6 |
| 15:47.23 | whizzi | ok, colloquy is also unstable :P |
| 15:47.59 | LeandroDardini | TK, I have a peer named [flowroute] |
| 15:48.38 | whizzi | <PROTECTED> |
| 15:48.38 | whizzi | [Nov 20 16:44:15] ERROR[24626][C-000000d0]: netsock2.c:269 ast_sockaddr_resolve: getaddrinfo("st-x2-zipzap01", "(null)", ...): Name or service not known |
| 15:48.46 | [TK]D-Fender | LeandroDardini: Pastebin the actual call with SIP DEBUG enabled... |
| 15:49.17 | Penguin | whizzi: The st-x2-zipzap01 peer doesn't seem to have a valid host name associated with ti. |
| 15:49.22 | Penguin | s/ti/it/ |
| 15:49.26 | *** join/#asterisk rogersja (~rogersja@206.191.104.90) |
| 15:50.06 | Penguin | whizzi: Pastebin the st-x2-zipzap01 peer entry from sip.conf. |
| 15:51.02 | LeandroDardini | TK, good idea! ... let me check it |
| 15:51.26 | whizzi | Penguin: http://pastebin.com/CsXLKv2H |
| 15:52.25 | [TK]D-Fender | whizzi: that name is NOT the same at all |
| 15:52.53 | [TK]D-Fender | whizzi: "st-x2-zipzap01" <> [st-sn-zipzap01] |
| 15:52.56 | Penguin | hmm |
| 15:53.20 | Penguin | I don't even understand how that mistake is made. |
| 15:53.23 | whizzi | sorry, typo in my description, see previous bit of logging |
| 15:53.41 | whizzi | it matches: Executing [s@macro-portal-out:17] Dial("SIP/bart-100-00000031", "SIP/st-x2-zipzap01/08000543,60") in new stack |
| 15:53.42 | Penguin | On another note, the host for THIS entry is valid and resolves. |
| 15:53.50 | Penguin | But you're using the wrong peer name. |
| 15:53.57 | [TK]D-Fender | whizzi: your PEER name does not match |
| 15:54.04 | Penguin | Either you showed me the wrong peer, or you are dialing the wrong peer. |
| 15:54.15 | whizzi | oh right |
| 15:54.16 | Penguin | Either way, it's wrong. |
| 15:54.39 | whizzi | wow, I'm amazed I didn't see that |
| 15:54.39 | whizzi | :P |
| 15:55.27 | whizzi | hits himself |
| 15:55.33 | Penguin | Need a hand with that? |
| 15:55.36 | whizzi | *and gets more coffee* |
| 15:57.52 | *** join/#asterisk jasonwert (~w3rt@75-134-81-98.static.aldl.mi.charter.com) |
| 15:59.47 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 16:00.20 | whizzi | thanks for all the help :). I'll be probably hanging around a lot more here, mostly for helping people. |
| 16:00.24 | LeandroDardini | TK, check http://pastebin.com/v5vbkRuu |
| 16:02.54 | [TK]D-Fender | LeandroDardini: "sip show peers" |
| 16:05.24 | LeandroDardini | 103-DEVEL/103-DEVEL 83.211.224.67 D N 56517 OK (319 ms) Cached RT |
| 16:05.24 | LeandroDardini | flowroute/4902**** 216.115.69.144 a 5060 Unmonitored |
| 16:07.10 | *** join/#asterisk toresbe (~toresbe@158.36.190.254) |
| 16:08.16 | mirela666 | Hi, what is the best practice (files/dialplans organisation) for the shared clients PBX (more customers that shouldn't dial eachother,...) |
| 16:08.24 | [TK]D-Fender | LeandroDardini: I need a proper sanity check for this. Please PM the actual call debug, no filtering |
| 16:08.41 | [TK]D-Fender | mirela666: Put them in separate contexts. The end. |
| 16:08.49 | Penguin | mirela666: The files don't have anything to do with that, but do use different contexts. |
| 16:09.16 | Penguin | mirela666: The files will only be for your own organization of the contexts. |
| 16:09.35 | mirela666 | exactly, to not lose my head around |
| 16:10.02 | mirela666 | and if possible keep as many contexts generic as possible |
| 16:10.26 | Penguin | I have different files for different entities. /etc/asterisk/extensions.d/some-company/extensions.conf |
| 16:10.53 | Penguin | Then in the main extensions.conf, I #Include /etc/asterisk/extensions.d/*/extensions.conf |
| 16:11.14 | Penguin | And I messed up the exact path... |
| 16:11.14 | petn-randall | Hi, is there a a half-way safe/sane way to rate-limit SIP registrations to Asterisk? Being UDP, it might be a little tricky to do that with iptables. |
| 16:11.37 | Penguin | /etc/asterisk/extensions.conf.d/ ... |
| 16:12.00 | Penguin | #Include extensions.conf.d/*/extensions.conf; |
| 16:12.03 | mirela666 | Penguin: yes that is what I want to do, but still , for contexts, make specific ones that will only set the needed variables to be recognised and them go to generic ones for dial_out for example |
| 16:12.53 | Penguin | If you have one set of extensions for outbound dialing that all phones will use, put those extensions in the main extensions.conf. |
| 16:13.59 | Penguin | I do the same with sip.conf.d/some-company/sip.conf for each tenant's phones. |
| 16:14.49 | Penguin | #Include sip.conf.d/*/sip.conf |
| 16:15.20 | Penguin | It works just fine for me. |
| 16:16.37 | *** part/#asterisk whizzi (~whizzi@D4B2620B.static.ziggozakelijk.nl) |
| 16:18.19 | pabelanger | Penguin, nice to see other people do that too |
| 16:19.27 | Penguin | It seems like a sensible way to handle multiple clients on a single asterisk instance. |
| 16:21.50 | pabelanger | https://github.com/kickstandproject/asterisk/tree/master/debian/ast_config |
| 16:21.59 | pabelanger | that's the stock configs we run |
| 16:22.04 | pabelanger | then have puppet provision everything else |
| 16:22.14 | pabelanger | been working good for the last 2 years |
| 16:28.10 | *** join/#asterisk endemic (~endemic@lynx.onvox.net) |
| 16:32.26 | *** part/#asterisk toresbe (~toresbe@158.36.190.254) |
| 16:32.53 | Penguin | pabelanger: What's the difference between the #Include I use and the #tryinclude that you use? |
| 16:38.30 | *** join/#asterisk MLNoah (~chatzilla@noc.metalink.net) |
| 16:38.32 | file | Shameless plug: If you're on Twitter and want a high level pulse on Asterisk development you can follow the AsteriskDev twitter account. |
| 16:39.53 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-pgaerpocxdtzdqva) |
| 16:39.53 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 16:41.00 | MLNoah | is there anything different about Voicemail exitcontext when you're using ODBC voicemail instead of File storage? I've got an Asterisk 11.5.1 server set up, didn't set the exitcontext= parameter in voicemail.conf (because I want to exit into the context that Voicemail() was called from), and there is a 0 and a extension in the context Voicemail() was called from, but when I press 0 or *,... |
| 16:41.02 | MLNoah | ...nothing happens. DTMF is otherwise working. |
| 16:44.15 | [TK]D-Fender | MLNoah: You didn't tell us you DID set a context anywhere... so that sounds like "none". Why would we expect it to go anywhere? |
| 16:44.37 | [TK]D-Fender | MLNoah: Show us configs and debug for this... |
| 16:44.40 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 16:44.43 | MLNoah | the sample config for voicemail.conf says that default is to exit to the context Voicemail() was called from |
| 16:44.55 | MLNoah | and I guess I'm incorrect, * is working to the a extension, but 0 isn't. |
| 16:45.18 | MLNoah | because it's supposed to be o, not 0, and I can't read. |
| 16:45.20 | MLNoah | gah. |
| 16:47.06 | WIMPy | Try a different font :-) |
| 16:47.18 | MLNoah | and yet o isn't working either. I hate my life right now. |
| 16:47.43 | Penguin | Maybe you have it in the wrong context. Look at the debug. |
| 16:51.39 | pabelanger | Penguin, #include will fail if the file does not exists (modules doesn't load). #tryinclude will still fail, but allow the module to load |
| 16:55.56 | MLNoah | http://pastebin.com/RmXCwScZ |
| 16:56.02 | *** join/#asterisk SuperNull (~supnul@24-148-101-238.ip.mhcable.com) |
| 16:56.50 | MLNoah | I didn't really think to press * on the one that I had the debugs on for... if nothing jumps out at anyone, I'll set things up so I don't bug the customer yet again and mess with *. |
| 16:57.25 | SuperNull | hey guys, having an issue where we have an Adtran TA900 doing sip -> pri and the customer is sending out #s that are not registered/explicitly authed to it .. anyone ever deal with this ? |
| 16:57.28 | *** join/#asterisk threesome (~threesome@ip-94-113-13-143.net.upcbroadband.cz) |
| 16:57.31 | Penguin | pabelanger: Ah, okay. I've always had the files present, so that's why I never saw a problem with it. |
| 16:59.02 | pabelanger | Penguin, Right, the base config files we include, users configs we tryinclude. |
| 16:59.23 | Penguin | [Nov 20 10:58:18] WARNING[13762]: config.c:1238 process_text_line: Unknown directive '#TryInclude' at line 1924 of /etc/asterisk/sip.conf |
| 16:59.27 | Penguin | Not going to work for me. |
| 16:59.55 | MLNoah | shouldn't #tryinclude be all lower? |
| 17:00.15 | Penguin | I wouldn't think so. |
| 17:00.28 | pabelanger | Penguin, I added the patch to trunk a few years ago |
| 17:00.35 | MLNoah | i always use all-lower in my config files. and given that your warning has case in it, I'd suggest trying all lower and see if it helps. |
| 17:00.39 | pabelanger | so, I think its in 11 now |
| 17:00.44 | Penguin | I use 1.8. |
| 17:00.48 | pabelanger | you need to backport into 1.8 if you want it |
| 17:00.55 | pabelanger | ya, that's what I do |
| 17:01.33 | pabelanger | https://github.com/kickstandproject/asterisk/blob/master/debian/patches/backport-r345735.diff |
| 17:05.37 | ghost75 | PCMA/8000 in sip packet is the g.711 alaw ? |
| 17:05.47 | [TK]D-Fender | yes |
| 17:05.52 | Penguin | pabelanger: Do you also have a backport for sip description field? With some help, I created one that I used on 1.8.8.x, but I recently tried to use it on 1.8.23.x and it failed horribly. I tried to fix the patch, but it was never successful. |
| 17:06.25 | pabelanger | Penguin, haha, that was like the first one we backported |
| 17:06.26 | pabelanger | https://github.com/kickstandproject/asterisk/blob/master/debian/patches/backport-r313528.diff |
| 17:06.48 | pabelanger | we use 1.8.7.1 |
| 17:06.52 | ghost75 | somebody using zoiper? there is no audio transmitted |
| 17:06.52 | Penguin | I don't know for sure why I can't fix it, but I think it has something to do with vim and tabs. |
| 17:07.20 | pabelanger | ghost75, we use zoiper for all softphones, no issues here |
| 17:07.37 | Penguin | Even using noexpandtabs and changing the tabstops, I still didn't have luck and eventually gave up. |
| 17:07.50 | [TK]D-Fender | ghost75: Just assuming the client is wrong and not your configuration is a bad start to debugging. |
| 17:08.09 | [TK]D-Fender | ghost75: And you're asking about random problem with no backup. |
| 17:08.10 | MLNoah | even modifying my voicemail_users table to include an exitcontext column and putting in the context I want to dump into doesn't fix exit on 0. |
| 17:08.11 | ghost75 | pabelanger: udp or tcp? |
| 17:08.23 | pabelanger | udp |
| 17:08.27 | [TK]D-Fender | ghost75: Voice doesn't go over TCP |
| 17:08.29 | Penguin | I have no problems with zoiper and udp. |
| 17:09.29 | Penguin | If there ever is a problem with zoiper and asterisk, it is a problem with a stupid NAT implementation. |
| 17:09.49 | ghost75 | its in lan |
| 17:09.59 | *** join/#asterisk Tokeiito (~quassel@78-62-103-201.static.zebra.lt) |
| 17:10.27 | pabelanger | Ya, disable stun support and try again |
| 17:10.36 | ghost75 | ok |
| 17:11.08 | ghost75 | i see also it has wrong rtp port |
| 17:11.53 | Penguin | No it doesn't. |
| 17:12.10 | Penguin | That is the client RTP port. Leave it set to default. |
| 17:12.24 | ghost75 | hmm it was set to 8000 |
| 17:12.41 | Penguin | There is no need to change it from default. |
| 17:12.52 | ghost75 | i didnt set it :) |
| 17:13.08 | Penguin | Then you're seeing the default, which you do not need to change. |
| 17:13.23 | ghost75 | default was 8000 |
| 17:13.37 | ghost75 | i removed and working :) |
| 17:14.23 | karl-s | ghost75, I would compare that with what is in /etc/asterisk/rtp.conf |
| 17:14.29 | karl-s | to be sure |
| 17:14.32 | Penguin | They are not related. |
| 17:14.40 | Penguin | No need to compare. |
| 17:14.46 | karl-s | oh |
| 17:14.48 | ghost75 | is "stun" doing this? |
| 17:15.21 | Penguin | rtp.conf sets asterisk's rtp port range. The setting in the client sets the client's rtp port or port range. |
| 17:15.22 | *** join/#asterisk jpoz (~jpoz@c-24-22-121-42.hsd1.or.comcast.net) |
| 17:15.35 | karl-s | huh, thats good to know |
| 17:17.53 | ghost75 | i guess this stun thing write a wrong rtp port, dunno |
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| 17:33.30 | *** join/#asterisk volga629 (~bendersky@CPE085b0e07d3f2-CM7cb21b15b251.cpe.net.cable.rogers.com) |
| 17:33.55 | volga629 | Hello Everyone, trying setup corosync |
| 17:33.57 | volga629 | [2013-11-20 18:32:47] ERROR[9648]: res_corosync.c:639 load_module: Failed to join (6) |
| 17:33.59 | volga629 | [2013-11-20 18:32:47] ERROR[9648]: res_corosync.c:609 cleanup_module: Failed to finalize cpg (6) |
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| 18:13.38 | As001 | Hello is it possible to tell Asterisk 1.8.23 to stop writing QueueMemberStatus to some AMI user while at the same time tell Asterisk to write to AMI Agentlogoff event ? Both depend on agents class in manager.conf and I need second and not need first event (it flood me with QueueMemberStatus). |
| 18:18.02 | As001 | Can I control which events Asterisk writes to AMI per event not per class because I need just few of them ? |
| 18:19.08 | WIMPy | Not on 1.8. |
| 18:19.51 | As001 | and in 1.11 ? |
| 18:20.11 | As001 | I meant 11 ? |
| 18:22.17 | *** join/#asterisk navaismo (~navaismo@189.241.101.4) |
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| 18:43.37 | dar123 | <PROTECTED> |
| 18:43.37 | *** join/#asterisk bmg505 (~leon@196-209-99-92.dynamic.isadsl.co.za) |
| 18:52.51 | pabelanger | dar123, cat and grep |
| 18:53.13 | Penguin | just grep, no cat. |
| 18:53.19 | outtolunc | or sed '/^\;/d' |
| 18:55.42 | outtolunc | i suppose that doesn't deal with the ones indented or after real settings tho |
| 18:55.54 | *** join/#asterisk dxd828 (~dxd828@host-92-24-127-29.ppp.as43234.net) |
| 19:06.24 | cusco | quick question |
| 19:06.28 | cusco | the more weight on a queue |
| 19:06.35 | cusco | the faster or slower it is picked up? |
| 19:06.57 | mjordan | WIMPy: Asterisk 1.8 has event filtering |
| 19:07.47 | cusco | higher weights get first shot at available |
| 19:08.02 | Penguin | cusco: Queues with higher weight get chosen before queues with lower weight. |
| 19:08.03 | WIMPy | Oh, I thought it was newer. |
| 19:08.26 | navaismo | is there a command like sip debug with chan_pjsip? |
| 19:08.54 | navaismo | pjsip set logger on isn't showing anything |
| 19:09.19 | cusco | thank you Penguin :) |
| 19:09.19 | Penguin | Or maybe I worded that wrong, but the answer is higher weight is first. |
| 19:09.43 | tm1000 | navaismo: tab is your friend |
| 19:09.48 | tm1000 | pjsip <tab> |
| 19:10.01 | navaismo | pjsip set logger on isn't showing anything |
| 19:16.54 | navaismo | hmmm also there is no pjsip reload. Not sure if I screwed the installation or missing something |
| 19:17.11 | file | it has no explicit reload command, it's module reload res_pjsip.so |
| 19:17.42 | Penguin | I would alias it. |
| 19:20.32 | navaismo | hmm ok there i no output because asterisk isn't listening on 0.0.0.0:5060 :S |
| 19:21.08 | file | there is no built-in default transport |
| 19:21.28 | navaismo | yea i created one |
| 19:22.04 | file | then I'd look at the console output at startup and see what it says |
| 19:22.20 | *** join/#asterisk kasanop (~kasanop@162.243.108.56) |
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| 19:25.10 | *** mode/#asterisk [+o angler] by ChanServ |
| 19:27.10 | slicknick5181 | What is the wget command for asterisk. I cant seem to find it |
| 19:27.28 | Penguin | Huh? What does that mean? |
| 19:27.29 | *** join/#asterisk jetlag (~jetlag@pool-71-168-196-225.cmdnnj.east.verizon.net) |
| 19:27.42 | slicknick5181 | to download the latest source |
| 19:27.47 | Penguin | Which branch? |
| 19:27.52 | slicknick5181 | 1.8 |
| 19:29.07 | Penguin | http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-1.8.24.0.tar.gz |
| 19:29.10 | Penguin | Try that. |
| 19:30.07 | Penguin | I reference it in my PKGBUILD by http://downloads.asterisk.org/pub/telephony/asterisk/${pkgname}-${pkgver}.tar.gz so just have to plug in the name and version. |
| 19:31.16 | file | you can also use http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-1.8-current.tar.gz if you always want the current one |
| 19:32.03 | Penguin | That's what I should have done instead of having to look for the current version before I gave him the URL. |
| 19:32.08 | Penguin | Next time, maybe. |
| 19:34.51 | navaismo | ok found the error, the protocol under transport only allow one, i.e: protocol=udp and not protocol=udp,tcp |
| 19:35.31 | file | correct. |
| 19:39.55 | slicknick5181 | A dialplan question, How do I set up to dial out on truck ABC and on fail try CBA. I remeber finding this a while ago but I dont remember where |
| 19:40.25 | slicknick5181 | Asterisk 1.8 |
| 19:41.03 | [TK]D-Fender | slicknick5181: Dial one right after the other |
| 19:41.16 | navaismo | awesome: Segmentation fault (core dumped) |
| 19:41.25 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
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| 19:44.49 | navaismo | file an endpoint cant have more than one transport? |
| 19:45.24 | file | it can't, but you aren't required to specify a transport |
| 19:45.37 | file | it's used to lock an endpoint to a specific transport |
| 19:46.42 | *** join/#asterisk dxd828 (~dxd828@host-92-24-127-29.ppp.as43234.net) |
| 19:49.24 | navaismo | hmmm |
| 19:49.26 | paulc | slicknick5181: Or look at the DIALSTATUS variable after dialing to see why it failed (legitimate busy end user versus congestion or channel unavailable if trunk ABC is down) (depending on whether "trunk" really means "trunk" or ITSP) |
| 19:51.44 | Penguin | I doubt it means trunk; I figure it just means peer. |
| 19:52.32 | volga629 | Hell everyone is anybody know about this message |
| 19:52.38 | volga629 | res_corosync.c:316 ast_event_cb: CPG mcast failed (6) |
| 19:53.27 | volga629 | ping multicast failed ? |
| 19:54.24 | slicknick5181 | Paulc: Intriguing, I'd like to attempt on DAHDI 4 then if in use dial on SIP trunk |
| 19:56.08 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 19:56.46 | paulc | slicknick5181: you probably DO want to check DIALSTATUS then, rather than falling through from one Dial to the next.. because "Busy" or "No Answer" are things that are valid and don't need a roll-over to try again? |
| 19:57.41 | Penguin | Sounds legit to me. |
| 19:57.48 | slicknick5181 | Would you pastebin an example. I've never used Dialstatus |
| 19:59.07 | Penguin | dialstatus example: http://pastebin.com/FEXndLUc |
| 20:00.56 | Penguin | This shows the concept. You can adapt it for your case. |
| 20:03.28 | navaismo | is normal to have 3 locks after start asterisk 12 no calls at all just after start the service? |
| 20:05.24 | paulc | slicknick5181: Here's another example - http://pastebin.com/zBJ03Nbr |
| 20:06.07 | paulc | Penguin: props for using DIALSTATUS as the priority labels - I hadn't thought of that :) |
| 20:06.50 | navaismo | and the Segmentation fault(core dumped) was my foolishness of store the transport inside the DB |
| 20:07.17 | Penguin | It works perfectly. |
| 20:08.51 | paulc | Yeah, nice.. don't you then have to define all possible values of DIALSTATUS though? |
| 20:08.57 | *** join/#asterisk jpoz (~jpoz@c-24-22-121-42.hsd1.or.comcast.net) |
| 20:08.59 | paulc | looking at it again, yeah, you've done that.. all good :) |
| 20:09.05 | Penguin | I did. |
| 20:09.28 | Penguin | If you don't want to define all the possible status, I suppose it could be worked around. |
| 20:10.24 | paulc | Yeah.. I did it with a result-xxxx extension.. match the ones I want, pattern match everything else |
| 20:10.37 | paulc | as Larry Wall says: There's more than one way to skin a cat |
| 20:10.57 | Penguin | Like this: http://pastebin.com/EERAjv0t |
| 20:11.32 | *** join/#asterisk joesmoe (~joesmoe@admins.phreefilez.com) |
| 20:12.06 | Penguin | I prefer the labels, but either way does work. |
| 20:13.33 | slicknick5181 | Wow thank you all very much |
| 20:14.12 | *** join/#asterisk julgr (~julgr@c-66-235-46-252.sea.wa.customer.broadstripe.net) |
| 20:15.28 | paulc | +1 for commenting the dialplan too.. I do that. Yeah, I built the office PBX but I don't OWN it.. someone else should be able to take over if (when?) I leave.. a few lines of inline commenting go a long way to getting people into the headspace you were in at the time. |
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| 20:47.17 | lordvadr | Where can I find a list of atoms asterisk's regular expressions supports? |
| 20:47.52 | *** join/#asterisk corretico (~luis@190.241.180.58) |
| 20:53.15 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 21:03.45 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 21:10.51 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 21:15.19 | navaismo | lordvadr, according to the old voip-info http://www.voip-info.org/wiki/view/Asterisk+func+regex |
| 21:15.52 | Chainsaw | Only three years old, that page. |
| 21:16.00 | Chainsaw | Young, by voip-info terms. |
| 21:16.25 | navaismo | last modification |
| 21:21.00 | lordvadr | navaismo: I know how to call the function, I need to know a list of atoms it understands. If I want whitespace, do I say [:space:] or \s? |
| 21:31.03 | mjordan | lordvadr: it uses this: http://pubs.opengroup.org/onlinepubs/009695399/functions/regcomp.html |
| 21:31.12 | mjordan | with extended regular expressions |
| 21:31.16 | mjordan | whatever that uses, it uses |
| 21:37.00 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.41) |
| 21:38.35 | navaismo | file if i didnt specify a trasnport for an endpoint then the endpoint can connect with all protocols: udp or tcp or ws or tls etc? |
| 21:42.45 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 21:46.31 | *** join/#asterisk danfromuk (~IceChat77@unaffiliated/danfromuk) |
| 21:47.37 | danfromuk | Hi, I'm logging cdr via odbc to mysql. It seems that there is an intermittent problem where the CDR(userfield) isn't written to the database. The rows all seem to be there, but that specific field is missing. The same dialplan. Nothings changed. Probably 1 in 10 is missing the userfield. |
| 21:47.44 | danfromuk | Any ideas how I can debug this? |
| 21:50.32 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 21:52.02 | *** join/#asterisk Martin` (martin@shell.ipv6.octocore.net) |
| 21:52.20 | Martin` | Hello guys :D |
| 21:52.24 | Martin` | (and girls?) |
| 21:54.00 | Chainsaw | (On IRC? Honestly...) |
| 21:54.24 | Martin` | I know some :P |
| 21:56.46 | Martin` | But I've a quetion. I've just installed a new server and compiled asterisk 11.6. All without any errors. But now when I try to start I get the message "Illegal instruction (core dumped)" what can be the problem? |
| 21:58.03 | navaismo | s/girls/G.I.R.L./ |
| 21:58.31 | Chainsaw | Martin`: You compiled it for a different CPU then what is in the server. |
| 21:58.32 | navaismo | danfromuk, enable the debug on the cli and take a look on the query |
| 21:59.06 | Chainsaw | Martin`: Good way to induce this is to compile with -march=native on a shiny new CPU elsewhere and then taking it back to your retro P4 in the corner. |
| 21:59.49 | Martin` | I do it on a virtual server. and it is compiled for x86_64 |
| 21:59.50 | Chainsaw | Martin`: But if you compiled it on the same machine, then -march=native should have been fine. Unless it's not an AMD64, but that would have been a fact worthy of mention. |
| 22:00.16 | Chainsaw | Martin`: Compiled for an x86_64 CPU you don't have. What did you set? |
| 22:00.29 | Martin` | model name: QEMU Virtual CPU version 1.5.0 |
| 22:00.43 | Martin` | I just did ./configure and it choosed that one :) |
| 22:00.49 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 22:00.59 | Chainsaw | Martin`: And that QEMU virtual CPU... is it even 64-bit? |
| 22:01.03 | Chainsaw | Hey Fender. |
| 22:01.11 | Martin` | looks like http://comments.gmane.org/gmane.comp.telephony.pbx.asterisk.user/275467 is the same problem |
| 22:01.16 | Martin` | yes |
| 22:01.26 | Martin` | installed a 64bit os on it, so I think it is 64bit :P |
| 22:02.22 | danfromuk | navaismo: debug 3 doesnt seem to show odbc / mysql calls. |
| 22:03.34 | Chainsaw | Martin`: So set -march for the underlying CPU, not the virtual CPU. Or use -mtune=native instead of -march=native. |
| 22:03.57 | Martin` | do I give that by ./configure? |
| 22:04.02 | Chainsaw | Martin`: Other alternatives include running on "bare metal", as the cool kids call it these days. |
| 22:04.17 | navaismo | danfromuk, check the full log |
| 22:04.26 | Chainsaw | Martin`: ./configure CFLAGS=-mtune=native |
| 22:04.31 | jeev | man, just shut off an as5300, saved 4 amps, plugged in an adtran, i bet i'm using <1amp |
| 22:05.01 | Martin` | ok lets try that :) |
| 22:06.11 | Martin` | I don't know the underlying cpu, I've just ordered a vps to run my site and voip. The bare metal I use for other things :P |
| 22:06.26 | danfromuk | navaismo: nothing in the full log about odbc or mysql. but it is logging to mysql properly. just that field gets left out every so often |
| 22:06.33 | Chainsaw | Martin`: The -mtune=native may well work. |
| 22:06.45 | Chainsaw | Martin`: If it doesn't, then you'll have to ask your VPS provider. |
| 22:06.57 | Martin` | ok, lets hope |
| 22:07.12 | Martin` | it is baking now |
| 22:07.30 | Chainsaw | (The VPS people must *love* QEMU. Now those pesky customers finally can't see it's all on a dusty P4 in the corner of the data centre) |
| 22:08.11 | Martin` | lol |
| 22:08.17 | Martin` | I hope it is not :P |
| 22:08.40 | Martin` | and I think you need 64bit hardware to run a 64bit vps? |
| 22:08.59 | Chainsaw | Martin`: I know of 64-bit Pentium IV CPUs. |
| 22:09.14 | Chainsaw | Martin`: Doesn't mean they're very good at it... But it's on there. |
| 22:09.54 | *** join/#asterisk jpoz (~jpoz@ec2-184-169-152-1.us-west-1.compute.amazonaws.com) |
| 22:10.05 | Martin` | <PROTECTED> |
| 22:10.06 | Martin` | <PROTECTED> |
| 22:10.09 | Martin` | hmm failed :( |
| 22:10.41 | Chainsaw | Martin`: Then start with a clean tarball. |
| 22:10.52 | Martin` | ok |
| 22:11.00 | Martin` | i did make clean, but not enough? |
| 22:11.06 | Chainsaw | No. |
| 22:11.11 | Chainsaw | mrproper |
| 22:11.19 | danfromuk | Martin`: if this is your first time, why not build from yum, and see if it works. |
| 22:11.55 | Martin` | apt-get do you mean? (ubuntu) It is only asterisk 1.8 |
| 22:12.16 | Chainsaw | Martin`: I support your choice of Asterisk 11 over 1.8 |
| 22:12.29 | Chainsaw | Martin`: And this is a minor inconvenience that can be overcome. Don't listen to the evil RedHat user. |
| 22:13.13 | danfromuk | navaismo: the userfield appears properly in the csv file. Just not in the db |
| 22:13.41 | slicknick5181 | Upgrading from 1.8 to 11 how much of my extensions.conf will need changed or is it all basically still the same |
| 22:14.01 | danfromuk | Not much but there are some changes. Look in changes.txt |
| 22:14.17 | slicknick5181 | danfromuk, Thank you |
| 22:14.22 | danfromuk | Sorry, upgrade.txt |
| 22:14.27 | danfromuk | Sorry, upgrade-1.8.txt |
| 22:14.37 | Martin` | :P |
| 22:18.14 | Martin` | and asterisk is running :D |
| 22:18.27 | Martin` | tnx Chainsaw :D |
| 22:18.33 | Martin` | now I can set it up :) |
| 22:18.54 | Chainsaw | Martin`: *folds hands* Excellent. |
| 22:23.44 | slicknick5181 | bes way to move from 1.8 to 11 apt-get remove? |
| 22:23.48 | danfromuk | what order is the CDR CSV in? |
| 22:24.28 | Martin` | 4 sip peers [Monitored: 4 online, 0 offline Unmonitored: 0 online, 0 offline] |
| 22:24.41 | Martin` | happy it did not changed :P |
| 22:24.52 | Martin` | now I only have to make the extensions :) |
| 22:25.56 | slicknick5181 | since I installed it that way initially |
| 22:26.18 | Martin` | slicknick5181: I did apt-get remove asterisk |
| 22:26.53 | Martin` | moved my config to a backup folder and compiled asterisk 11 |
| 22:28.39 | Martin` | hmm new files extensions.lua and extensions.ael |
| 22:28.42 | slicknick5181 | Matrin, Sweet thats what I was looking for |
| 22:28.53 | slicknick5181 | Martin* |
| 22:29.19 | Martin` | slicknick5181: oh and I also did apt-get autoremove to be sure all unused asterisk things are gone :) |
| 22:29.32 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 22:29.42 | slicknick5181 | Martin, apt-get autoremove asterisk?? |
| 22:29.56 | Martin` | no just autoremove after you removed asterisk |
| 22:30.05 | Martin` | it will remove packages that are no longer needed |
| 22:31.00 | Martin` | after that you can check google what packages you need for compile :) |
| 22:31.20 | Martin` | google said build-essential ncurses-dev libxml2-dev libsqlite3-dev libssl-dev to me :P |
| 22:31.48 | slicknick5181 | I have all of that as I just moved my 1.8 install to a new machine |
| 22:32.02 | slicknick5181 | I just want to upgrade before I'm in too deep |
| 22:36.11 | danfromuk | Weird. Some lines of the cdr aren't written intermittently. Even the cdr-csv is missing lines. |
| 22:36.32 | danfromuk | Cant' see anything obvious that makes the difference. Its the same dial plan. Sometimes works, sometimes doesnt. |
| 22:37.16 | *** join/#asterisk classix (salven@silenceisdefeat.com) |
| 22:39.40 | Martin` | and the tth sip connection is connected |
| 22:39.45 | Martin` | 5th |
| 22:39.48 | Martin` | or how to type it |
| 22:39.52 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.41) |
| 22:39.53 | Martin` | feels like a phone provider |
| 22:40.43 | *** join/#asterisk amizraa (~amizraa@gateway/tor-sasl/amizraa) |
| 22:40.46 | Martin` | 3 incomming, 1 outgoing, and my softphone |
| 22:45.44 | Martin` | I'm happy my outgoing provider accepts any telephone nr als callerid :P |
| 22:45.52 | apb1963 | getting one way audio... This was working... changed nothing at all. Log here: http://paste.ubuntu.com/6450630/ |
| 22:48.49 | Martin` | lets check if my deskphone still works :) |
| 22:50.27 | *** join/#asterisk g_r_eek (~g_r_eek@176.92.124.225) |
| 22:50.38 | *** join/#asterisk razu (~razu@2001:ad0:1:1:202:b3ff:fe36:921c) |
| 22:51.49 | danfromuk | Just upgraded to 11.6.0. CDR lines are still missing intermittently. |
| 22:52.08 | slicknick5181 | sqlite3 dev package? |
| 22:52.10 | slicknick5181 | Cant find it |
| 22:55.18 | *** join/#asterisk jpoz (~jpoz@ec2-184-169-152-1.us-west-1.compute.amazonaws.com) |
| 22:56.16 | Chainsaw | slicknick5181: It won't work without, the internal Asterisk database uses sqlite. |
| 22:56.47 | navaismo | apb1963, quick look and didnt found an explicit issue, maybe the GV right now have problems or became unstable |
| 22:58.52 | *** join/#asterisk glj12_ (4a6ecf89@gateway/web/freenode/ip.74.110.207.137) |
| 22:59.17 | slicknick5181 | Chainsaw, Where can I find the package sudo apt-get what |
| 22:59.39 | Martin` | I can finaly make a call again with that phone :D |
| 22:59.44 | Martin` | (sip phone) |
| 22:59.48 | Chainsaw | slicknick5181: Martin` gave you exact package names. |
| 23:00.03 | Chainsaw | Martin`: Phones are more fun when they work, yes. |
| 23:00.16 | glj12_ | Hi all -- I wanted to know if it is possible/how would I go about forwarding phone calls over Asterisk to land line numbers? Is this possible? I am currently running an Ettus B100, OpenBTS, and Asterisk. |
| 23:00.19 | Martin` | but the phoneline is eaten by weasels |
| 23:01.22 | Chainsaw | glj12_: Sure, you could add analog gateway and just talk SIP to it. |
| 23:01.28 | WIMPy | glj12_: You should be able to interface to OpenBTS via LCR. Otherwise it's about hardware or an ITSP. |
| 23:01.28 | Martin` | now I just have to config incomming calls. 3 numbers. but all with almost the same config :) |
| 23:01.41 | *** join/#asterisk g_r_eek (~g_r_eek@176.92.124.225) |
| 23:01.51 | Martin` | need to forward call to sip-phone and forward to my mobile phone |
| 23:02.32 | Martin` | and it needs to find the caller information (in my custommerdatabase and on the web) |
| 23:02.34 | glj12_ | Oh man, okay. Maybe I could use a resource? I am a bit unfamiliar with those terms. Is this something that is free, or is it a service I need to subscribe to in order to forward the calls appropriately? |
| 23:03.04 | WIMPy | glj12_: Both |
| 23:03.13 | glj12_ | Ah |
| 23:03.28 | glj12_ | Is this something I can do via Google Voice? Or am I misunderstanding terms |
| 23:06.03 | slicknick5181 | Keeping in mind that I has Asterisk 1.8 installed on this system. I removed it and menuselect for asterisk 11 has cn_sip X out it's not checked nor can it be checked |
| 23:06.24 | slicknick5181 | chan_sip* |
| 23:07.26 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 23:07.35 | Chainsaw | slicknick5181: Martin` gave you a complete list of packages you would need. Did you install them all? |
| 23:08.03 | Chainsaw | slicknick5181: <Martin`> google said build-essential ncurses-dev libxml2-dev libsqlite3-dev libssl-dev to me :P |
| 23:09.25 | Qwell | or just `contrib/scripts/install_prereq` |
| 23:10.47 | mjordan | danfromuk: if I had to guess, it isn't a CDR/ODBC/MySQL problem, specifically. It's probably the result of a transfer operation and masquerades going awry deep in the bowels of the Asterisk bridging core. There's some funky logic there with CDRs in general, and userfields in particular |
| 23:10.50 | *** join/#asterisk bkruse (~Adium@64.89.97.127) |
| 23:11.34 | danfromuk | mjordan: i've removed odbc from the equation. I'm now testing with cdr-csv, and there are lines missing from the cdr intermittently. |
| 23:11.36 | glj12_ | Chainsaw: So, when you say add an analog gateway, can you give an example of what you mean? |
| 23:12.00 | *** join/#asterisk classix (salven@silenceisdefeat.com) |
| 23:12.07 | danfromuk | mjordan: sometimes they are recorded, sometimes not. I've logged it as an issue and included my dialplan. |
| 23:12.27 | danfromuk | mjordan: https://issues.asterisk.org/jira/browse/ASTERISK-22874 for reference if it helps at all |
| 23:12.28 | LieutPants | [ASTERISK-22874] [Status: Triage] CDR Lines Missing - https://issues.asterisk.org/jira/browse/ASTERISK-22874 |
| 23:12.31 | mjordan | As I said, this is already pretty much known about |
| 23:12.36 | mjordan | ASTERISK-20010 |
| 23:12.37 | LieutPants | [ASTERISK-20010] [Status: Open] Bridge CDR fails to be created when, in a Local channel bridging two outbound channels that is optimized away, the CDR userfield setting is set on both outbound channels - https://issues.asterisk.org/jira/browse/ASTERISK-20010 |
| 23:12.43 | mjordan | ASTERISK-19165 |
| 23:12.44 | LieutPants | [ASTERISK-19165] [Status: Open] Empty CDR userfield and wrong UniqueID values stored in Master.csv when using Originate from AMI - https://issues.asterisk.org/jira/browse/ASTERISK-19165 |
| 23:12.47 | Chainsaw | glj12_: A device with an ethernet port (that talks SIP to you) and an FXO port, which connects to an analog telephone line. |
| 23:12.52 | mjordan | ASTERISK-16194 |
| 23:12.53 | LieutPants | [ASTERISK-16194] [Status: Open] [patch] In agent calls recording, CDR userfield is not updated when the call is transferred - https://issues.asterisk.org/jira/browse/ASTERISK-16194 |
| 23:12.56 | Chainsaw | glj12_: Perhaps even multiple FXO ports, if you are so inclined. |
| 23:13.07 | danfromuk | Ignoring userfield, cdr lines are totally missing. |
| 23:13.28 | *** join/#asterisk g_r_eek (~g_r_eek@46-137-51.adsl.cyta.gr) |
| 23:13.37 | mjordan | Looking at your issue, you're going to want to create a simpler test case that reproduces it. |
| 23:13.56 | glj12_ | Chainsaw: Ah, I see. Is it feasible for me to configure Asterisk with Google Voice in order to serve the same purpose? |
| 23:14.03 | danfromuk | Ok. I'll try to reduce the dialplan and see if it still causes problems. |
| 23:14.07 | Chainsaw | glj12_: I have no idea, that is a US-specific service. |
| 23:14.28 | mjordan | You've got some Local channels in there as well |
| 23:14.47 | mjordan | what specific CDR records are getting produced, and what ones are you missing? That is, what specific channels are not being represented? |
| 23:15.16 | glj12_ | Chainsaw: Gotchya, okay. I don't have a line I can hook up to (nor the tools, I would think) -- do you know of any free service similar to that of Google Voice with Asterisk? |
| 23:15.29 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 23:15.37 | Chainsaw | glj12_: Telephony isn't free, sorry. |
| 23:15.58 | danfromuk | mjordan: the first channel is always logged. Its the Local channel thats intermittently missing. |
| 23:17.06 | mjordan | k, that's useful info :-) Are you always Answering the Local channel? |
| 23:17.51 | mjordan | Because there's a GotoIf in there that auto-hangs up the local channel. By default, non-answered calls aren't logged to CDRs |
| 23:17.57 | mjordan | unless you have unanswered=yes |
| 23:18.15 | mjordan | in fact, it might be worthwhile testing with unanswered=yes just to see if they show up. |
| 23:18.35 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.41) |
| 23:19.29 | glj12_ | Chainsaw: No that's fine -- I'll keep working with Google Voice over here. My issue may be blocked ports... The calls go through, but I don't get any voice upon picking up on the receiver. =/ |
| 23:20.04 | Penguin | No audio? This is almost always a problem with NAT. |
| 23:20.12 | danfromuk | mjordan: ive removed some of the dialplan in the issue. I'll test unanswered=yes, however if it hits a queue, i would expect it to be marked as answered. In fact, the first channel is marked as ANSWERED in the cdr |
| 23:21.49 | danfromuk | can unanswered=yes be placed in [csv]? or can it only be placed in [general] |
| 23:23.08 | glj12_ | Penguin: How should I go about troubleshooting to see if that is indeed the case? |
| 23:23.25 | Penguin | glj12_: What channel tech are you using that isn't getting audio? |
| 23:24.22 | *** join/#asterisk Pullphinger (~Pullphing@c-24-13-69-42.hsd1.il.comcast.net) |
| 23:24.24 | glj12_ | Penguin: Sorry, don't hit me for this, but what do you mean by channel tech? |
| 23:25.05 | navaismo | great now a websocket connection generate a new segmentation fault(core dumped) :S |
| 23:25.05 | Penguin | The channel technology, such as SIP, IAX2, something analog using DAHDI, etc. |
| 23:25.48 | glj12_ | SIP -- I am attempting to route my GSM call over my Ettus radio |
| 23:26.04 | glj12_ | Or did I just name two different techs =/ |
| 23:26.33 | Penguin | I don't think Asterisk speak GSM tech, so we'll talk about the SIP part of your calling equation. |
| 23:26.47 | Penguin | Where does the SIP call go from and to? |
| 23:26.47 | glj12_ | Sure |
| 23:26.58 | glj12_ | I'll map it out |
| 23:27.18 | Penguin | Just tell me. I don't need a map. |
| 23:27.36 | glj12_ | Okay |
| 23:27.39 | Penguin | Does it go between your ITSP and your asterisk? |
| 23:27.53 | glj12_ | Yes |
| 23:28.05 | glj12_ | ITSP being Google Voice in this case |
| 23:28.24 | Penguin | That's not SIP, then. |
| 23:28.32 | glj12_ | Ah, ok |
| 23:28.48 | Penguin | But it still uses RTP as far as I know. And that's the important part when it comes to NAT. |
| 23:28.56 | danfromuk | mjordan: unanswered seems to have resolved it. Weird that it worked intermittently with unanswered=no |
| 23:30.29 | glj12_ | So, am I going about this the right way? I am connecting my GSm phone to my Ettus B100, then fowarding it to Asterisk and then the ITSP |
| 23:30.50 | glj12_ | Or am I going about it all wrong |
| 23:31.21 | Penguin | I don't know about the forwarding part. I figure you just want to Dial() somewhere else. |
| 23:31.30 | glj12_ | Right |
| 23:31.41 | glj12_ | I was following this before: https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google |
| 23:31.57 | Penguin | Which asterisk branch are you using? |
| 23:32.33 | glj12_ | 1.8 |
| 23:32.49 | Penguin | In your gtalk.conf, did you define your externhost or externaddr? |
| 23:33.49 | glj12_ | No |
| 23:35.25 | Penguin | Do you have a static IP address on the outside of your NAT router, or do you get a dynamic IP address? |
| 23:35.36 | glj12_ | I have a static IP |
| 23:35.53 | Penguin | Then set externaddr=your.static.addr |
| 23:35.56 | Penguin | in gtalk.conf |
| 23:36.20 | glj12_ | Alright, I'll give that a go. |
| 23:36.41 | glj12_ | Is there anything I should be looking for in verbose mode to see if things are not being sent because of the NAT? |
| 23:36.46 | glj12_ | within asterisk |
| 23:38.03 | *** join/#asterisk jpoz (~jpoz@ec2-184-169-152-1.us-west-1.compute.amazonaws.com) |
| 23:38.20 | Penguin | If it was SIP, we can easily debug that. But the tech that google voice uses isn't SIP. I don't know how detailed jabber/gtalk debugging is. I just configure things right and expect them to work. |
| 23:38.54 | glj12_ | Gotchya, okay |
| 23:38.57 | danfromuk | mjordan: I cant use unanswered. It creates duplicate lines in the cdr. Of all channels, not just the Local ones. |
| 23:39.08 | glj12_ | One last thought would be: it turns out I was using the older version of the link that I gave you |
| 23:39.25 | glj12_ | this is with asterisk 1.8. is it worth it a go to try the new article that I sent you with asterisk 11? |
| 23:39.44 | mjordan | I wasn't saying that you should use it, but if your mysteriously not present channels show up when unanswered=yes, then you have your answer as to why they aren't showing up |
| 23:40.14 | mjordan | keeping in mind that the reason why they aren't showing up then isn't a bug |
| 23:40.19 | Penguin | I'm pretty sure in 11 they are using a different channel driver. chan_motif or something instead of chan_jingle? |
| 23:40.26 | mjordan | they aren't showing up because they aren't Answered |
| 23:40.47 | mjordan | Penguin: yes, chan_motif. chan_jingle still exists, but should be avoided like malaria. |
| 23:41.19 | slicknick5181 | i know this is documented a couple places online bu some of it confuses me I have a TDM 400 with 1 FXO and 1 FXS. How do I set up the dahdi.conf so they work. Asterisk 11 |
| 23:41.35 | Penguin | glj12_: So make sure you use the guide for 1.8, not 11. |
| 23:41.52 | danfromuk | mjordan: I wouldnt class them as unanswered, because there is an Answer() cmd, and the caller hears prompts and menus before hitting a queue. |
| 23:43.04 | mjordan | What is the disposition of the channel in the CDR with unanswered=yes? |
| 23:43.06 | glj12_ | Penguin: Oh, good. I was doing that already (but it says that it's old -- I guess that doesn't make a difference then) |
| 23:43.42 | danfromuk | mjordan: ANSWERED |
| 23:44.50 | mjordan | For both the ;1 and the ;2? |
| 23:44.59 | Chainsaw | slicknick5181: You give it fxs_ks signalling on the FXO and fxo_ks on the FXS. |
| 23:45.38 | mjordan | and when you go into the Queue, what is the disposition of the channel that talks to the Queue member? |
| 23:46.06 | mjordan | I'd imagine you have something like: SIP/foo, Local/company_648;1 as one record |
| 23:46.16 | mjordan | and Local/company_648;2 ... ? as another record |
| 23:46.18 | danfromuk | mjordan: one moment. unanswered is now 'no' but i'm still seeing duplicate cdr lines. |
| 23:46.45 | danfromuk | I'm going back to 11.5.1. One moment. |
| 23:46.54 | mjordan | I'm not sure what you mean by duplicate CDR lines. And this behavior hasn't changed in any version recently. |
| 23:47.00 | mjordan | Or quite frankly, since 1.8.0-ish |
| 23:47.35 | mjordan | the last time we changed behavior of CDRs was Asterisk 12, and we rewrote most of it (and most people will probably hate it) |
| 23:47.59 | danfromuk | The same cdr record, twice. Identical fields. |
| 23:48.39 | mjordan | I'm not aware of that happening in any issue I've seen. At this point, I think if you want someone to look at this more in depth, you'll need to attach a full DEBUG log to the issue, along with sample records demonstrating what you're seeing. |
| 23:48.44 | mjordan | Include as well the channel names involved. |
| 23:49.10 | glj12_ | Penguin: I just had a thought... Can my bindaddr be 0.0.0.0 ? I assumed that this was okay since it means to let anyone connect to it? |
| 23:49.36 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 23:49.54 | Penguin | glj12_: 0.0.0.0 means all addresses on all interfaces. It doesn't have anything to do with who is allowed to connect. Yes, a bind address of 0.0.0.0 is valid, and even recommended. |
| 23:50.13 | glj12_ | Penguin: Okay, cool. Thanks for the clarification |
| 23:50.57 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.41) |
| 23:53.31 | danfromuk | mjordan: ok, if i cant figure it out, i'll build a test server and try to replicate the issue so i can submit the debug and channel names etc. |
| 23:53.55 | danfromuk | mjordan: thanks for your help |
| 23:55.14 | mjordan | np - sorry it wasn't a more obvious answer |
| 23:57.23 | slicknick5181 | It appears as though dahdi_channels set it sleft up but what do I have to do in Dahdi.conf |
| 23:59.44 | boom^time | I'm having a weird issue where if I set a CDR variable after a fork it sets the variable for the original CDR, not the forked one. And if I use the s(CDR(blah)=blah) option in forkcdr I get the same result. |
| 23:59.53 | boom^time | Does anyone have any experience with this issue? |