| 00:00.39 | *** join/#asterisk jpoz (~jpoz@ec2-184-169-152-1.us-west-1.compute.amazonaws.com) |
| 00:01.49 | PMantis | No, but the context that originate connects it to plays, waits, goto play. |
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| 00:04.28 | *** mode/#asterisk [+o sruffell] by ChanServ |
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| 00:06.38 | nny | setting up Vitelity with a NAT connection to a client. I have all the normal things done (externip, etc.) I have tested with nat=yes and no for the sip provider. I usually try to avoid NAt but this scenario may require it. Thoughts? I have one audio (none to remote end) http://pastebin.com/eRs7dCTY |
| 00:07.12 | nny | I have NAt set to yes in vitelity's interface to inform them the device is behind NAT on our end. |
| 00:07.14 | Penguin | Vitelity is not behind nat, so the answer is nat=no for the provider's peer entry. |
| 00:07.25 | nny | Penguin I knew that but got |
| 00:07.40 | nny | [2013-11-13 19:04:14] NOTICE[20437]: chan_sip.c:20257 handle_response_invite: Failed to authenticate on INVITE to whenever nat=no |
| 00:08.01 | Penguin | Is that a call going inbound to asterisk or a call going out to the provider? |
| 00:08.04 | nny | let me kick debug on |
| 00:08.06 | nny | outbound |
| 00:08.29 | Penguin | Fix your defaultuser and secret. That is not related to NAT. |
| 00:08.49 | [TK]D-Fender | nat=no <- this should be YES in [general] |
| 00:08.58 | Penguin | and no for the provider's entry. |
| 00:09.04 | [TK]D-Fender | line 21 has it wrong |
| 00:09.24 | [TK]D-Fender | remove canreinvite and use directmedia=no |
| 00:09.42 | [TK]D-Fender | type=friend <- better off being peer |
| 00:09.50 | nny | arrgg outdated vitelity support configs |
| 00:10.03 | nny | yeah that is directly from their config page |
| 00:10.04 | [TK]D-Fender | You are also not specifying codec in your peer.. not great |
| 00:10.07 | nny | which is always outdated as hell |
| 00:10.12 | *** join/#asterisk btcquant (~noahnoah@99-41-172-179.lightspeed.irvnca.sbcglobal.net) |
| 00:10.16 | nny | copy pasted, let me correct |
| 00:10.18 | [TK]D-Fender | line 48 ... wrong |
| 00:10.20 | Penguin | And why do you need to trustrpid on the outbound peer definition? |
| 00:10.24 | nny | ask vitelity |
| 00:10.55 | Penguin | Providers have never provided accurate configs for clients. They just simply don't know asterisk. |
| 00:11.13 | [TK]D-Fender | nny: And the worst part ..... |
| 00:11.16 | [TK]D-Fender | nny: -- Executing [8436942813@sip:3] Dial("SIP/6760-0000616c", "SIP/18436942813@vitel-outbound,,rg") in new stack |
| 00:11.25 | Penguin | wrong format |
| 00:11.25 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.62) |
| 00:11.38 | [TK]D-Fender | nny: there is NO WAY I'm going to trust that SIP/6760 is anything resembling "right" at this point :) |
| 00:11.52 | nny | it's a peer on the local network |
| 00:11.59 | nny | but i'll post debug from that |
| 00:12.06 | [TK]D-Fender | nny: Don't see debug, trust level=0 |
| 00:12.13 | nny | roger |
| 00:12.26 | [TK]D-Fender | Remember ... that's only because it's an UNSIGNED INT |
| 00:12.40 | Penguin | You're also allowing all codecs in general, but I can see that this was not your intention. |
| 00:12.48 | Penguin | (there is no disallow before two allows) |
| 00:13.32 | nny | why would nat=yes be in general? |
| 00:16.17 | nny | http://pastebucket.com/22532 |
| 00:16.19 | nny | new configs |
| 00:16.21 | [TK]D-Fender | nny: Your server is behind NAT, isn't it? |
| 00:16.48 | nny | yes but not to the peers. nat=yes is defined only in places where the peer is behind nat |
| 00:16.58 | [TK]D-Fender | nny: that is why nat=yes for general |
| 00:17.00 | nny | so default is no, remote peers have nat=yes on their config |
| 00:17.03 | [TK]D-Fender | that is a GENERAL state |
| 00:17.08 | Penguin | This is more like how _I_ would have configured it: http://pastebin.com/Ungi19gd |
| 00:17.12 | nny | and all the peers who use general are nat=no |
| 00:17.26 | [TK]D-Fender | that's fine |
| 00:17.36 | [TK]D-Fender | but general should be "yes" |
| 00:17.52 | nny | isn't general "unless overridden based on peer"? |
| 00:18.06 | nny | and 90% of the peers in the network have no concept of NAT to the PBX |
| 00:18.10 | Penguin | except s/username/defaultuser/g, since I forgot to update that for you. |
| 00:18.19 | nny | the PBX does to the provider, as well as a few remote peers |
| 00:18.43 | PMantis | "yes" doesn't mean exactly what you think |
| 00:19.01 | [TK]D-Fender | nny: new call, complete debug |
| 00:19.04 | nny | ok |
| 00:19.05 | nny | one sec |
| 00:19.28 | Penguin | fixed it. |
| 00:19.41 | Penguin | See my paste to see how I would have configured it for starters. |
| 00:19.57 | *** join/#asterisk threesome (~threesome@ip-94-113-13-143.net.upcbroadband.cz) |
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| 00:25.20 | nny | new config, http://pastebucket.com/22533 new dump: http://pastebucket.com/22534 |
| 00:25.34 | nny | inbound isn't used so it's not included |
| 00:26.50 | nny | is directmedia=no required at the peer AND in general, isn't that redundant? |
| 00:27.54 | [TK]D-Fender | nny: we clearly don'e see vitelity outbound call debug there |
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| 00:28.13 | nny | i enabled it not sure why i'll post the debug log |
| 00:29.06 | nny | is there a way to show debugs that are enabled? |
| 00:29.23 | nny | sip set peer vitel-outbound on is what I sent to console |
| 00:30.36 | Penguin | I see the circuits busy message there. When I get that, it is because I am sending bad info for the call or because I am not registered. |
| 00:30.50 | Penguin | Maybe you aren't supposed to send rpid? |
| 00:31.00 | Penguin | I would have thought you would want to. |
| 00:31.21 | nny | [TK]D-Fender: any way to confirm the debug status? I can't see why it wouldn't work with the command posted to console sip set peer vitel-outbound |
| 00:32.42 | [TK]D-Fender | <--- Reliably Transmitting (no NAT) to 10.1.0.109:5060 ---> |
| 00:32.48 | [TK]D-Fender | we are clearly getting debug for this peer |
| 00:32.52 | nny | yes |
| 00:32.58 | [TK]D-Fender | You CANNOT debug multiple specific peers |
| 00:33.03 | [TK]D-Fender | ^ |
| 00:33.04 | nny | ahh well shit |
| 00:33.06 | [TK]D-Fender | One or all |
| 00:33.14 | nny | i'll have to do two calls |
| 00:33.22 | [TK]D-Fender | So if you though you would restricrt it to those TWO ... then no |
| 00:33.23 | Penguin | Like a third of the three musketeers. |
| 00:33.46 | nny | ok [TK]D-Fender running it again with only the vitel debug on |
| 00:35.42 | nny | http://pastebucket.com/22535 |
| 00:35.45 | nny | vitel debugged |
| 00:36.28 | nny | Proxy Authentication Required |
| 00:36.43 | nny | says online in the vitelity interface |
| 00:37.02 | Penguin | It says that in the debug, too. |
| 00:37.14 | nny | Penguin: no |
| 00:37.16 | Penguin | yes |
| 00:37.21 | nny | i am repeating what the edbug says |
| 00:37.23 | nny | stop please |
| 00:37.37 | Penguin | Line 66 says so. |
| 00:37.47 | nny | Vitelity interface says the pbx is ojnline not the error message stop confusing it |
| 00:37.55 | Penguin | I see. |
| 00:38.20 | nny | [TK]D-Fender: http://pastebucket.com/22535 |
| 00:38.28 | Penguin | Online would mean it registered. |
| 00:40.29 | nny | http://i.imgur.com/47lcnJ7.png |
| 00:40.39 | nny | I'll change the user name after this mess is over.. god i hate nat |
| 00:41.22 | nny | this all worked before they decided that having the box on a public IP wasn't as good as behind this pfsense firewall I am having them check for siproxd on the pfsense or some kind of automagic BS setting |
| 00:43.08 | [TK]D-Fender | nny: and invert your dial syntac as I'm sure Penguin probably told you to do |
| 00:43.20 | [TK]D-Fender | nny: SIP/18436942813@vitel-outbound ----> SIP/vitel-outbound/18436942813 |
| 00:43.24 | Penguin | I didn't specifically say it. |
| 00:43.37 | Penguin | I just said it was wrong. |
| 00:43.40 | [TK]D-Fender | I'm sure it didn't escape being implied ;) |
| 00:43.45 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.50) |
| 00:43.49 | nny | i don't follow |
| 00:43.57 | Penguin | SIP/peername/extension |
| 00:44.09 | Penguin | in your dialplan. |
| 00:44.14 | nny | Dial(SIP/XX@Peer is how every one of my pbxs has worked for 6 years |
| 00:44.22 | nny | is this really relevant? |
| 00:45.31 | [TK]D-Fender | I've seen things go wonky before |
| 00:45.36 | [TK]D-Fender | DO IT :) |
| 00:46.00 | nny | yeah no, I have ~ 30 lines of dial strings in that plan. I'll just revert it back, I am going no where here with the blind shots |
| 00:46.19 | nny | it's almost 8 pm I don't really have time to guess at issues |
| 00:46.50 | ChannelZ-Wk | Search @ replace on 30 lines! When will we have computers to do this mundane work!? |
| 00:47.26 | nny | sigh not the point |
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| 00:48.01 | Penguin | While it is wrong syntax, I don't believe it is causing this issue. |
| 00:48.12 | nny | wonrg, deprecated, whatever |
| 00:48.15 | nny | wrong |
| 00:49.02 | Penguin | What caught my attention, and I don't know how to correct it, is the failed INVITE from SMITH,CARSON. |
| 00:49.18 | Penguin | I feel like the name there is a problem. |
| 00:49.47 | Penguin | sorry, INVITE to, not from. |
| 00:52.27 | ChannelZ-Wk | I'm not Vitelity but use static IP so I've not had to deal with their auth |
| 00:52.42 | nny | ChannelZ-Wk: I did as well before going behind NAT |
| 00:52.48 | ChannelZ-Wk | s/not V/on V/ |
| 00:55.44 | ChannelZ-Wk | I've not been paying that close attention but looking at some of your previous pastes I'm curious why it's sending multiple 407s back, like asterisk isn't ACKing |
| 00:55.49 | ChannelZ-Wk | And doesn't even bother trying to auth |
| 00:56.16 | nny | I've gotten it to 403 now changing the outbound server to a non DNS address |
| 00:56.24 | nny | i suspect round robin may be part of it |
| 00:57.00 | ChannelZ-Wk | I assume you've registered? |
| 00:57.39 | nny | it was checking again. |
| 00:58.33 | nny | it is |
| 00:58.57 | nny | ok screw this bs. Back to no NAT |
| 01:07.33 | nny | ha oddly enough I can get vitelity to work with IP auth fine but no audio stream because they have no way to set static IP accounts as NAT yes/no. Is it some kind of industry standard that every voip provider excel at 50% of things and suck at the other 50? |
| 01:09.28 | m0sphere | what would be the most efficient way to maintain a customer database that is allowed to use DISA? |
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| 01:16.31 | nny | annnd 16 minutes later everything works again |
| 01:19.19 | nny | methinks this pfsense router is possibly part of the issue |
| 01:20.07 | nny | oddly my 20 dollar linksys running ddwrt sitting in my house would handle this fine |
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| 01:41.10 | Katty | HI LADS. |
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| 04:00.58 | PMantis | nny, I use pfsense and vitelity |
| 04:01.45 | PMantis | Hmmm, gone. waaaay late |
| 04:02.41 | Penguin | I'm interested in it. |
| 04:03.00 | Penguin | The problem was not related to NAT. |
| 04:03.08 | Penguin | The problem was related to authentication. |
| 04:04.01 | Penguin | Authentication was failing, but I wasn't able to determine why. |
| 04:04.06 | PMantis | I'm rarely in this channel, so perhaps I should share my config with you then... |
| 04:04.52 | Penguin | Do you have an inbound and an outbound context for vitelity? It seems they use a different proxy for taking calls from the customer. |
| 04:05.01 | PMantis | I'm sure TK would find someting wrong with it... but it works. lol |
| 04:05.02 | PMantis | Yes |
| 04:05.19 | Penguin | I'll probably find something wrong, too, but that isn't the point. |
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| 04:06.11 | Penguin | Vitelity is the only one I have seen regularly the users two different hosts for in and out. Most ITSPs use one host for both ways. |
| 04:06.26 | PMantis | I wrote my own XML config file that's interpreted by perl and spit out over STDOUT with an #exec. I'll generate the sip config and pastebin it. |
| 04:06.37 | Penguin | s/y the/y that/ |
| 04:06.49 | Penguin | s/users/uses/ |
| 04:06.58 | Penguin | fnck! |
| 04:07.00 | PMantis | lol |
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| 04:11.26 | PMantis | ?pastebin |
| 04:11.44 | Penguin | ~pb |
| 04:11.44 | 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. |
| 04:12.28 | PMantis | Eh, I just went to .ca.. :) |
| 04:12.31 | PMantis | http://pastebin.ca/2475990 Good for 1 day |
| 04:13.19 | PMantis | At some point, I had an issue with a host, so I hard-coded the IP address. |
| 04:13.38 | PMantis | Forgot about that. I even forgot WHAT the issue was. lol |
| 04:19.53 | Penguin | He mentioned that changing from name to IP resulted in a different problem. I figured it had something to do with the wrong Dial syntax, but he had already decided he wasn't about to change that. |
| 04:21.26 | PMantis | I have my Vitelity trunk set to send 11 digits |
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| 04:28.13 | Penguin | I would change type to peer, nat to no, and replace canreinvite with directmedia. |
| 04:28.30 | Penguin | Other than that, it's pretty much what the other guy should have been using. |
| 04:29.00 | PMantis | Yeah, I wasn't paying attention to his issue at that time... |
| 04:29.57 | PMantis | I'm behind a nat firewall, and this works. Why change to no? |
| 04:31.13 | PMantis | peer/friend/user I get, though. :) |
| 04:31.56 | Penguin | The nat setting in a peer definition decides if asterisk needs to treat the remote host as if it is behind a NAT. Vitelity is clearly not behind a nat, so nat=no would be the appropriate setting for that peer entry. |
| 04:32.09 | PMantis | Ahh |
| 04:32.19 | Penguin | Will it break it to set it to yes? Probably not. But why set it to the wrong value and risk it? |
| 04:32.40 | PMantis | There was always a little vagueness in the nat setting for me. |
| 04:33.24 | Penguin | I've ran my ITSP entries without an explicit nat setting for a long time, and that means it inherits from the general section, which is set to nat=yes on my box. |
| 04:33.50 | Penguin | So it obviously won't break it. |
| 04:34.11 | Penguin | But I went ahead and explicitly configured nat=no on those entries when I saw that I hadn't already done it. |
| 04:34.14 | PMantis | It figures it out most times... depending on how good the other end is. |
| 04:34.44 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 04:34.56 | Penguin | I haven't seen a case, to my knowledge, where setting it to yes does break it; setting it to no when it needs to be yes, though, will break it. |
| 04:41.00 | Penguin | For the inbound entry, I would change type to peer and delete the insecure setting. |
| 04:41.39 | Penguin | Other than that, it seems almost like what the other guy had. I don't have a copy of his values handy. |
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| 04:48.39 | PMantis | On my outbound entry, I get one-way audio unless nat=yes |
| 04:49.35 | PMantis | Inbound seems to be ok with nat=no |
| 04:50.08 | Penguin | You aren't using SIP registrations, are you? |
| 04:50.21 | PMantis | Yes. |
| 04:50.41 | Penguin | I think someone mentioned there's a setting on the provider to nat yourself. |
| 04:51.20 | PMantis | I don't remember seeing that... |
| 04:51.34 | Penguin | I don't quite understand how setting nat to yes for a host that is not behind nat would fix one-way audio. |
| 04:53.06 | PMantis | Yeah, agreed. |
| 04:53.26 | Penguin | I think of the nat settings from a point of view where your asterisk is a "client" device and the provider is your "main asterisk." Since your "main asterisk" is not behind a nat, but your client is, you'd set your account to nat enabled. In asterisk, we always disable nat settings on the "client," so that would mean setting nat=no in that peer definition. |
| 04:53.28 | PMantis | When calling out, don't get audio BACK from the remote (Vitelity) end. |
| 04:53.51 | Penguin | Maybe that's not the clearest way to describe it, but that's how I view it when thinking about the settings. |
| 04:53.59 | PMantis | s/, / I/ |
| 04:55.00 | Penguin | But you said it's working, so leave it that way for now. |
| 04:55.59 | Penguin | I wouldn't be able to leave it that way. I'd ponder it until I came up with the reason for the behavior and then I'd make adjustments until the settings made sense AND worked. |
| 04:56.48 | PMantis | Hehe, I've learned to not question things sometimes. It'll eventually make sense, just not always when I want them to. |
| 04:57.54 | PMantis | I can call in and out, business is functional. I'm ok with a nat setting for now. |
| 04:57.55 | Penguin | I have to have things make sense. |
| 04:58.20 | Penguin | I would be able to leave it that way on a temporary basis. It would have to be corrected at some point. |
| 04:59.02 | PMantis | Mild OCD? :) |
| 04:59.13 | Penguin | Pretty much, yeah. |
| 04:59.36 | *** join/#asterisk elcontrastador (~textual@173-13-184-97-sfba.hfc.comcastbusiness.net) |
| 04:59.40 | PMantis | I'm convinced we are ALL OCD about something.. |
| 05:11.44 | elcontrastador | I'm having significant delays in processing menu exten input during Background(). It waits around 5 secs after the user exten entry. I'm not using WaitExten() so it was my understanding matching extens should match immediately. http://pastie.org/8479256 |
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| 05:12.40 | PMantis | Neither should have a delay unless the entered digits match multiple options, and it waits for input first, to be sure the user is done. |
| 05:13.44 | elcontrastador | yeah...I thought maybe default behavior had changed since 1.8 or something (running 11.5.1 here) |
| 05:14.10 | Penguin | If you are entering, say, 5, and you have extension 5 and extension 50 available, when you put in 5 it will wait longer to see if you will be entering more. |
| 05:14.29 | elcontrastador | ahhh...the include at the top... |
| 05:14.31 | PMantis | Look in the ces context, too. |
| 05:15.42 | elcontrastador | hmmm...nothing i can do about it I guess as they want the full campus dial context to be avail during menu |
| 05:16.10 | elcontrastador | BackgroundDetect didn't work for me as expected... |
| 05:16.26 | Penguin | Consider adjusting TIMEOUT(digit) if it waits too long. |
| 05:16.53 | Penguin | BackgroundDetect() listens for sound while playing a file. |
| 05:16.59 | PMantis | OR, add an entry. "If you know the extension you want, press 1" |
| 05:17.19 | elcontrastador | ah yes... |
| 05:18.42 | elcontrastador | thx guys |
| 05:18.51 | PMantis | YW! |
| 05:19.58 | Penguin | When BackgroundDetect() is being used, sound on the channel causes the call to jump to extension 'talk'. |
| 05:21.32 | elcontrastador | Penguin: Clearly not what I needed. Just was dabbling in things...my asterisk chops are extremely rusty now |
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| 06:40.19 | D30 | hi all.. need some input and advice from you guys.. good day btw.. |
| 06:40.50 | D30 | how can we trace or output dialed number per extension in asterisk? |
| 06:41.01 | D30 | is there a way we can look that on the log files? |
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| 07:07.04 | ChannelZ | Well the CDR should show you who dialed what |
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| 07:12.25 | D30 | hi i know this is a newbie question but how to find files with this filename? OUT00-20131114-020517-1384412717.4343.wav |
| 07:12.47 | D30 | in /var/spool/asterisk/monitor i cant seem to find that recording :( |
| 07:13.09 | D30 | so i wanna know what directory used for storing recordings |
| 07:13.17 | D30 | ChannelZ: ? any idea? |
| 07:23.56 | ChannelZ | find / -name "OUT*.wav" |
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| 07:52.18 | bulkorok | hi |
| 07:52.31 | ChannelZ | jo |
| 07:52.36 | mirela666 | hi |
| 07:52.48 | ChannelZ | moves his chair slightly to the left |
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| 09:36.41 | wasanzy | hello |
| 09:38.02 | MaliutaLap | hellno ;) |
| 09:40.56 | wasanzy | I installed Sangoma w400 card and the driver with this chan_hahdi.conf |
| 09:42.22 | wasanzy | I have also inserted sim card but when I call the number, it doesn't go through, do I need to do anything more? |
| 09:42.24 | MaliutaLap | I can't help with that, I only use Digium cards :) |
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| 09:43.33 | wasanzy | MaliutaLap: but with the Digium card, do you have to edit the extension.conf before it worked? |
| 09:43.33 | MaliutaLap | although I have been toying with the idea of trying to put a GSM link into my * box using an old phone and on of the many programs that let you work with them |
| 09:43.46 | MaliutaLap | wasanzy: well yes |
| 09:44.05 | MaliutaLap | wasanzy: you need to have the correct channels set up to send things to |
| 09:44.50 | MaliutaLap | wasanzy: in extenstions.ael I set "HANDSET=DAHDI/2;" in globals |
| 09:45.43 | wasanzy | oh ok |
| 09:48.36 | MaliutaLap | wasanzy: you should be able to get some helpful information from the cli (core show channeltypes) :) |
| 09:52.43 | wasanzy | MaliutaLap: ok |
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| 10:27.13 | ocholetras | Hi guys! |
| 10:27.40 | ocholetras | Looking for some advice to best or most common way to handle fax with asterisk |
| 10:27.52 | ocholetras | i have seen that some people use iaxmodem + hylafax setup |
| 10:28.12 | Greenlight | IMO, don't. Get a normal PSTN line if you must have a fax machine. |
| 10:28.43 | ocholetras | we only want to get pdf emails from incoming faxes |
| 10:28.58 | bulkorok | http://youtu.be/tv63cFp8N5U |
| 10:29.28 | Greenlight | There are a number of online services offering fax-to-email if that's all you need |
| 10:30.08 | ocholetras | umm my data on my home please xD |
| 10:30.29 | ocholetras | ill have a look |
| 10:31.03 | ocholetras | main goal is to dont waste paper and to have digital archive of the faxes |
| 10:31.15 | ocholetras | otherwise you will have to scann each fax paper.. |
| 10:31.17 | ocholetras | -_-' |
| 10:31.54 | ocholetras | So, as far as i can see, asterisk community people recomend do not use asterisk for faxing.. |
| 10:32.04 | ocholetras | right? |
| 10:33.05 | mirela666 | ocholetras: iaxmodem + hylafax will do just fine |
| 10:33.14 | *** part/#asterisk bulkorok (~Benjamin@85.183.61.47) |
| 10:33.51 | mirela666 | + for bulkorok :D |
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| 10:37.58 | Greenlight | Many people have asterisk working great with fax. BUt I also see a lot of people having issues and pain with it. |
| 10:39.19 | mirela666 | I'm just trinig to find out why first 17 pages out of 35 of fax job didn't came to destination, and reported as sucessfull job |
| 10:41.47 | fling | How to log few calls to mysql? |
| 10:42.47 | mirela666 | fling: http://www.voip-info.org/wiki/view/Asterisk+cdr+mysql |
| 10:43.05 | fling | is already reading this link ^ |
| 10:43.15 | fling | mirela666: but is not it for logging every call? |
| 10:43.27 | fling | mirela666: I want only for few dids |
| 10:44.10 | Greenlight | https://wiki.asterisk.org/wiki/display/AST/CDR+Applications |
| 10:44.35 | fling | Greenlight: thanks. |
| 10:44.50 | Greenlight | Specifically, NoCDR - Make sure no CDR is saved for a specific call |
| 10:45.10 | fling | Greenlight: oh! ok. Now I want examples :D |
| 10:45.29 | Greenlight | I'll leave that to your Googling skills ;) |
| 10:45.40 | fling | is now googling… |
| 10:47.09 | mirela666 | http://bit.ly/17vLLcY :D |
| 10:47.44 | Greenlight | ^^ |
| 10:48.44 | fling | :> |
| 10:49.20 | Greenlight | My god the pain that is VNC over Remote Desktop. I feel like screaming. |
| 10:50.08 | Greenlight | I made the mistake of opening a webpage with animated graphics... |
| 10:54.56 | mirela666 | hehehe |
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| 12:13.22 | mirela666 | hey, is it possible (well it is, but I wonder how) to barge in on ten phone |
| 12:13.48 | Greenlight | ChanSpy or ExtenSpy |
| 12:13.52 | Greenlight | ^^ |
| 12:13.54 | mirela666 | to do a dial that doesn't need answer, phone automaticly stats the call |
| 12:14.22 | Greenlight | You mean make a phone automatically answer a call ? |
| 12:15.02 | mirela666 | yes a saw that was working like a group of phones started on speakerfone at the same time |
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| 12:15.28 | Greenlight | Yes, some phones support this functioanlity |
| 12:15.37 | mirela666 | for example 10 is dialing 11 extension and directly starting on it;s speakerphone |
| 12:15.39 | Greenlight | There are SIP "auto-answer" type headers that you can send |
| 12:15.58 | Greenlight | Or "answer-after". Consult your phones manual. |
| 12:16.01 | mirela666 | so you think it is autoanswer on phone setup nothing to do with asterisk |
| 12:16.11 | Greenlight | Yes, and no. |
| 12:16.24 | Greenlight | Asterisk can send the SIP header if your phone supports it. |
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| 12:16.35 | mirela666 | aha maybe was that |
| 12:16.43 | Greenlight | SOme phones will also provide auto-answer features that don't require headers to be sent. |
| 12:16.55 | kaldemar | app Page might be useful too. |
| 12:17.14 | kaldemar | in case you want to dial more than one phone at a time. |
| 12:17.16 | Greenlight | Most hardphones support the needed headers. |
| 12:17.42 | mirela666 | ok thx I'll read up on Page |
| 12:18.14 | mirela666 | thx Green I used SIP headers for changing ringtone and forgot it could make auto answer too :) |
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| 12:40.39 | bunglex | i am going mad trying to set up SPA3102 as a trunk using asteriskGUI - there seems to be no real info online - can anyone help |
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| 13:09.22 | mirela666 | If I add sip header + Dial it works but only on one phone, and with Page it just rings :| |
| 13:10.40 | Greenlight | Use Local channels |
| 13:11.01 | Greenlight | SO you page to the Local channels, and then loopback and add the header then Dial. |
| 13:11.23 | mirela666 | aha |
| 13:12.27 | mirela666 | header for each dialed extension |
| 13:12.38 | Greenlight | ? |
| 13:12.51 | wasanzy | my extension is defaulting to s, how do I change that? |
| 13:13.24 | Greenlight | Page(Local/<EXTEN>@mycontext) ... |
| 13:13.56 | Greenlight | Then inside mycontext, set the SIP header, and Dial ... |
| 13:14.54 | Greenlight | For example: exten => _X.,1,Dial(SIP/${EXTEN}) |
| 13:15.07 | Greenlight | Assuming your SIP device name mirrors it's "extension" number . |
| 13:16.17 | mirela666 | still one phone answeres and rest channels are killed |
| 13:16.46 | Greenlight | Lets see the call... |
| 13:18.04 | kaldemar | you need to add the header with Page aswell. |
| 13:19.40 | mirela666 | http://pastebin.com/rW7UNqVd |
| 13:20.26 | mirela666 | kaldemar: how? |
| 13:20.42 | kaldemar | forget about that, you're using dial anyway since you use local channels. |
| 13:20.56 | Greenlight | mhm |
| 13:21.04 | Greenlight | Lets see the force_conf context |
| 13:21.07 | kaldemar | adding the header before the Dial is enough, adding it before the page would be useless (and probably not work anyway). |
| 13:21.10 | mirela666 | I can use just dial or queue doesn't matter |
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| 13:21.32 | mirela666 | yep doesn't work |
| 13:21.32 | kaldemar | Dial("Local/force_conf@incoming_calls-9a88;2", "SIP/995&SIP/999&SIP/998&SIP/996") <-- this is at fault |
| 13:21.38 | wasanzy | any help please? |
| 13:21.45 | Greenlight | ~ask |
| 13:21.45 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 13:22.04 | kaldemar | mirela666: the Page should be paging multiple local channels, and each local channel should add the sip header and Dial _one_ channel only. |
| 13:22.43 | mirela666 | aha I did one local for multiple dials |
| 13:22.50 | mirela666 | ok got it |
| 13:23.13 | wasanzy | Greenlight: I asked |
| 13:23.26 | wasanzy | my extension is defaulting to s, how do I change that? |
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| 13:23.58 | wasanzy | No 's' extension in context 'mtn' |
| 13:24.03 | kaldemar | wasanzy: extensions don't default to anything, reword your question. or even better, show what happens. |
| 13:24.28 | MaliutaLap | wasanzy: as in the incoming SIP/IAX session is looking for 's'? or an outgoing? |
| 13:24.41 | wasanzy | Incoming |
| 13:25.08 | wasanzy | Span 1: Incoming Call, Type = Real, CallingNum = '"0266103372"', CallingName = '' |
| 13:25.08 | MaliutaLap | wasanzy: then you need to create an s extension. It's a pretty standard thing |
| 13:25.26 | mirela666 | wasanzy: s stands for start, so you are using application which is looking for s extension since you didn't point it to explicit one |
| 13:25.53 | mirela666 | you can add exten => s,1,NoOp() and see if it will get there |
| 13:26.31 | wasanzy | ok, am using chan_dadhi |
| 13:26.35 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 13:27.41 | wasanzy | when I add this exten => s,1,Set(SCODE=${EXTEN}) it worked, but I want that to be exten => +233546829659,1,Set(SCODE=${EXTEN}) |
| 13:35.04 | [TK]D-Fender | "s" is not "+233546829659" |
| 13:35.31 | *** join/#asterisk CassiusSmith (~textual@66.192.165.33) |
| 13:35.34 | [TK]D-Fender | exten => s,1,Set(SCODE=${EXTEN}) <-- if you're in "s" and run this.. the only thing you'r going to get is scode=s |
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| 13:40.33 | wasanzy | [TK]D-Fender: I don't think am clear |
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| 14:06.11 | Penguin | wasanzy: ${EXTEN} = whatever extension you are currently running. Does that help? |
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| 14:33.23 | bunglex | does the SPA3102 have to be registered with asterisk to use it as a trunk? |
| 14:34.19 | [TK]D-Fender | no |
| 14:38.22 | bunglex | thanks...im completely stumped |
| 14:38.30 | bunglex | <PROTECTED> |
| 14:39.34 | Greenlight | Doesn't it then re-attempt using credentials ? |
| 14:39.44 | bunglex | the spa says its registered on its status page... but asterisk doesn't have anything in show sip registry |
| 14:39.58 | MaliutaLap | I thank god I got rid of the NAT for VoIP on my network |
| 14:40.03 | Penguin | Where did you register it? |
| 14:40.16 | [TK]D-Fender | bunglex: "sip show registry" is showing things that ASTERISK registered to. |
| 14:40.20 | [TK]D-Fender | Not the other way around |
| 14:40.45 | bunglex | i see |
| 14:44.09 | bunglex | im not really sure how to follow the sip trace but i think you are right.... after the 401 it seems to carry on and then look for the number i dialled in the dial plan and then rejecting it saying user not found in dial plan |
| 14:45.22 | Penguin | Users are not found in dial plan. |
| 14:45.29 | Penguin | Extensions are found in dial plan. |
| 14:45.32 | Penguin | ~pb |
| 14:45.32 | 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. |
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| 14:50.35 | [TK]D-Fender | "sip set debug on" |
| 14:50.40 | [TK]D-Fender | "core set verbose 10" |
| 14:52.55 | bunglex | http://pastebin.com/huyPyA4n |
| 14:54.30 | Penguin | extension '1471' rejected because extension not found in context 'DLPN_DialPlan1'. |
| 14:54.54 | Penguin | This reminds me of something a GUI created. |
| 14:55.53 | Greenlight | 1471 isn't going to work natively |
| 14:56.05 | Greenlight | That's a PSTN feature |
| 14:56.10 | Penguin | What is 1471? |
| 14:56.34 | Greenlight | In the UK it gives your details of your most recent incomming call |
| 14:56.40 | Penguin | Neat. |
| 14:56.57 | Penguin | But he can make that work with asterisk. |
| 14:57.01 | *** join/#asterisk aberrios (~aberrios@195.130.201.200) |
| 14:57.07 | Penguin | He's got an adapter to connect to the phone line. |
| 14:57.08 | Greenlight | He can, sure, but it wont out the box :) |
| 14:57.18 | Penguin | Nothing works out of the box. |
| 14:57.44 | Penguin | So now he just has to create relevant dial plan to send that to the PSTN. |
| 14:59.57 | [TK]D-Fender | This has nothing to do with the SPA3102 |
| 15:00.06 | [TK]D-Fender | This is an Asterisk-GUI dialplan issue |
| 15:00.10 | [TK]D-Fender | And Ast4erisk GUI is DEAD. |
| 15:00.20 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-enmaogywdamqokjv) |
| 15:00.20 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 15:00.26 | [TK]D-Fender | I highly recommend you pick another strategy for configuring your system |
| 15:00.28 | bunglex | i just realised that id not included the outgoing call rule in the dial plan..... - ahhhh |
| 15:00.36 | bunglex | asterisk GUI is not good then |
| 15:00.59 | [TK]D-Fender | It's been dead for about 3 years now |
| 15:01.19 | bunglex | i have had some success doing it manually - but was hoping to find a gui |
| 15:01.22 | Penguin | It's better to learn asterisk first. Then, later, if you want to learn a GUI, pick one to learn. |
| 15:01.29 | Greenlight | By far your best option is to use raw asterisk intalled from source. If you really really need a GUI, use FreePBX |
| 15:01.36 | [TK]D-Fender | Hpoefully one that is maintained |
| 15:02.02 | Penguin | "installed from source" isn't really all that important. Qwell builds some fine packages. |
| 15:03.27 | Greenlight | Then they come in here needing a particular application or function that's not included in their packaged install... |
| 15:03.41 | Greenlight | And they've never even see the menuselect screen ever. |
| 15:03.48 | [TK]D-Fender | Greenlight: We'll burn that bridge when come across it ;) |
| 15:03.58 | Greenlight | :) |
| 15:04.19 | bunglex | thanks for the advice...manually it is then...its took so long to get the SPA configured without echo i can do without GUI |
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| 16:04.47 | gartral | hey all. Aside from a poorly broken line, which is the result of pasting being dumb, will the following work? http://paste.ubuntu.com/6416471/ |
| 16:05.40 | Qwell | no |
| 16:06.16 | *** join/#asterisk lvlolvlo (~lvlolvlo@unaffiliated/lvlolvlo) |
| 16:06.21 | gartral | Qwell: please excuse my ignorance, but how might I make it work? |
| 16:06.23 | Qwell | g,d? Dial,(? Dial, 6003? |
| 16:06.46 | Qwell | Also, your callerid won't have [] in it. |
| 16:06.54 | jmetro | Dial,( ... no comma there. |
| 16:07.07 | jmetro | 6002 wont work, its dial(Tech/Exten) |
| 16:07.46 | *** join/#asterisk vNistelroot (~secure@81.Red-88-6-157.staticIP.rima-tde.net) |
| 16:07.46 | gartral | so Dial(SIP/6002) |
| 16:07.48 | vNistelroot | hi people |
| 16:08.22 | jmetro | or Local/6002 if youre dialing something internally |
| 16:08.37 | vNistelroot | could I set "Waiting Call" in display while Im currently talking ? |
| 16:08.40 | gartral | jmetro: it's a sip extension |
| 16:08.46 | *** join/#asterisk kresp0_ (~kresp0@gateway/tor-sasl/kresp0) |
| 16:08.49 | vNistelroot | if someone calls me, see it in display |
| 16:08.49 | jmetro | there you go then |
| 16:08.51 | vNistelroot | sum like that |
| 16:09.17 | gartral | jmetro: I'm trying to make it so that if that number rings in.. it goes straight to two extensions simultaniosly.. |
| 16:09.33 | jmetro | so dial SIP/exten&exten |
| 16:09.56 | gartral | jmetro: and will that hang up on the extension that didn't answer? |
| 16:10.01 | jmetro | yes |
| 16:10.09 | gartral | jmetro: PERFECT |
| 16:10.33 | jmetro | test it first, it might be SIP/exten&SIP/exten |
| 16:11.05 | gartral | jmetro: I have no way of testing it.. at least not without locking myself out of my building. |
| 16:11.24 | gartral | the number is a callbox in the foyer for entry |
| 16:11.39 | jmetro | all in the line of duty. |
| 16:12.14 | [TK]D-Fender | vNistelroot: That's up to your phone |
| 16:12.21 | gartral | jmetro: I'm the only one here for the next 3 hours.. kinda don't want to be outside.. that long |
| 16:13.22 | gartral | so, in theroy.. same => n,GotoIf($[${CALLERID(num)} = [2165211574]?d:allow) |
| 16:13.37 | gartral | followed by same => n(d),Dial(SIP/6002&6003) should work? |
| 16:14.00 | jmetro | i think the []'s are wrong |
| 16:14.33 | jmetro | i suggest using a doorstop so you can test it. |
| 16:15.04 | gartral | jmetro: box won't call if the maglock isn't on |
| 16:15.23 | jmetro | oy |
| 16:15.41 | gartral | yea.. |
| 16:15.41 | [TK]D-Fender | gartral: your expression goes inside of $[] |
| 16:15.43 | *** join/#asterisk Dpunkt (~Dpunkt@p5DE5F85E.dip0.t-ipconnect.de) |
| 16:15.54 | [TK]D-Fender | gartral: you aren't counting your braces and just opening new opens all over the pace |
| 16:16.08 | gartral | [TK]D-Fender: I saw that, fixing it now |
| 16:16.15 | vNistelroot | ohh |
| 16:16.16 | [TK]D-Fender | http://www.voip-info.org/wiki/view/Asterisk+Expressions |
| 16:16.26 | vNistelroot | [TK]D-Fender: thank you so much |
| 16:16.31 | vNistelroot | im using yealink phones |
| 16:17.10 | *** join/#asterisk vlad_starkov (~vlad_star@217.26.6.70) |
| 16:17.56 | *** join/#asterisk NephFL (6c09e362@gateway/web/freenode/ip.108.9.227.98) |
| 16:18.00 | NephFL | is there a module or script that I can use to monitor call quality MOS style so that I can use it to change my default trunk? |
| 16:18.39 | gartral | so then same => n,GotoIf($[${CALLERID(num)}=2165211574]?d:allow) \n same => n(d),Dial(SIP/6002&6003) should do it? |
| 16:18.47 | Penguin | Dial(SIP/6002&6003) 6003 has no Tech defined. |
| 16:19.27 | gartral | Penguin: I was going by jmetro's sugestion |
| 16:19.34 | jmetro | yeah so it was SIP/exten&SIP/exten |
| 16:19.38 | jmetro | i couldnt remember if you had to do it twice |
| 16:19.47 | [TK]D-Fender | [11:10]jmetrotest it first, it might be SIP/exten&SIP/exten |
| 16:19.57 | [TK]D-Fender | And yes, you have to specify the FULL thing to dial for each |
| 16:20.03 | [TK]D-Fender | It doesn't assume a tech |
| 16:20.16 | Penguin | Well, no, that's still not exactly right. SIP/device. Device, not exten. Don't confuse extensions with phones. |
| 16:20.17 | gartral | ok, I had missed that line |
| 16:20.25 | *** join/#asterisk tomii (~tomii@128.54.168.125.sta.wbroadband.net.au) |
| 16:21.33 | *** join/#asterisk ghost75 (~quassel@dslb-088-064-049-241.pools.arcor-ip.net) |
| 16:22.50 | gartral | so same => nd,Dial(SIP/device&SIP/device) or would I have to define ring-times too? |
| 16:23.22 | gartral | s/nd/n/\(d\) |
| 16:23.29 | Penguin | Referring to phones or devices as extensions is only hurting your own understanding of asterisk. |
| 16:23.51 | Penguin | If you do not define a ring timeout, there will be no timeout. |
| 16:26.27 | *** join/#asterisk w9sh (~sph@50-79-224-193-static.hfc.comcastbusiness.net) |
| 16:27.39 | Penguin | s/nd/n(d)/ |
| 16:27.56 | gartral | wait.. I have a perfect way of testing this! if the rule works.. add a SendDTMF() to my extension |
| 16:28.16 | ghost75 | conferencing needs to be allowed in * ? |
| 16:29.15 | gartral | Penguin: eh.. my shell tends to interperet parentheses weirdly |
| 16:29.21 | Penguin | ghost75: Not allowed, no. It needs to be created. |
| 16:29.35 | Penguin | gartral: And slashes in the wrong place, as well? |
| 16:29.54 | gartral | Penguin: yes.. it's odd.. |
| 16:29.56 | ghost75 | any config files or on the phone? |
| 16:30.13 | Penguin | I know sed would blow up if you tried that. |
| 16:30.53 | [TK]D-Fender | [11:28]ghost75conferencing needs to be allowed in * ? <- what conferencing? There are multiple kinds... |
| 16:31.05 | gartral | Penguin: yea, but i'm used to using a different, "in house" regex parser.. |
| 16:31.24 | ghost75 | when conference button is pressed on phone then * sends 401 unauthorized |
| 16:31.37 | Penguin | Regardless of the kind of conference, it isn't something that is allowed/disallowed. Conferences need to be created with dial plan. |
| 16:31.39 | ghost75 | maybe because its cisco :> |
| 16:31.56 | Penguin | No, because you didn't configure any conferences. |
| 16:32.00 | gartral | regardless, off to test! |
| 16:32.01 | ghost75 | oh ok dialplan |
| 16:32.31 | ghost75 | then never mind |
| 16:33.14 | Penguin | Never mind... you're not going to do it because making a dial plan for conferencing is too hard? |
| 16:33.49 | ghost75 | no, never mind my question |
| 16:33.50 | [TK]D-Fender | I'm wonding what the actual method being used is |
| 16:34.04 | ghost75 | have to dig ... |
| 16:34.20 | gartral | wait.. before I do this, wouldn't it Dial(6002&6003) work because those are my extension numbers in extensions.conf? |
| 16:34.23 | Penguin | I use SCCP with my Cisco phones, and the confernce button is programmed in sccp.conf to dial an exten. |
| 16:34.47 | Penguin | gartral: No, Dial() has to have Tech/device. |
| 16:35.20 | Penguin | Extensions contain Dial() commands, not the other way around. |
| 16:36.20 | Penguin | Using extension numbers as your device names also hurts your understanding of asterisk's operation. |
| 16:36.41 | Penguin | ~devicenames |
| 16:36.41 | infobot | Devices, extensions, and people should be entirely abstracted. Extension numbers are applied to people, and people are applied to devices. This means you should name your devices something unique to each device, such as an ID tag or asset tag number, or a MAC address. |
| 16:37.03 | *** join/#asterisk endemic (~endemic@lynx.onvox.net) |
| 16:37.07 | jmetro | or a combination like asset-exten-mac |
| 16:37.35 | gartral | Penguin: that I'm not.. device names are the names of the person, we all use softphones so there's no physical devices to base an id off of |
| 16:37.53 | gartral | except of course, the computers temselves |
| 16:37.58 | Penguin | I've never met anyone named 6002. |
| 16:38.28 | gartral | Penguin: I've met a 1 before.. |
| 16:38.43 | Penguin | I'm number 1. |
| 16:39.04 | gartral | Penguin: that was a bad pun, his name is Juan.. |
| 16:39.09 | Penguin | haha |
| 16:40.06 | Penguin | That reminded me of a joke... it needs to be said not typed. Regarding Jose (hose A) and hose B. |
| 16:40.39 | Penguin | Anyway... |
| 16:40.47 | gartral | Penguin: oh lord.. i think I've heard that one before |
| 16:41.17 | br_2013_ | Hey all, I've got a question about including the SIP method "MESSAGE" in the Allow SIP header line. I found a post (http://forums.asterisk.org/viewtopic.php?f=1&t=83638) on this question, but there was no response. |
| 16:41.22 | Penguin | You can use the MAC address of the primary interface on the computer which runs the soft phone, if you intend to use MAC addresses for your device naming scheme. |
| 16:41.25 | br_2013_ | The problem is that the client, the softphone Jitsi, includes the MESSAGE method in its SIP Allow header, but Asterisk does not, so Jitsi thinks that Asterisk does not support the SIP method MESSAGE, so won't send SIP messages. |
| 16:41.35 | br_2013_ | I thought I had configured sip.conf correctly to support SIP messages (accept_outofcall_message = yes, outofcall_message_context = messages, auth_message_requests = yes, subscribecontext = users), but maybe I'm missing something. I can supply specific configuration sections on a pastebin site if needed. Anyone have any ideas ? |
| 16:41.51 | NephFL | Is there a call quality monitoring script/application someplace for asterisk? |
| 16:42.33 | ghost75 | voipmonitor for everything you need |
| 16:42.36 | Penguin | But you shouldn't use the extension used to reach a device in the device's name. It's bad practice and it breaks the abstractions that you should be making. |
| 16:42.51 | br_2013_ | (I am running Asterisk version 1.8) |
| 16:44.04 | gartral | Penguin: ok, then just simply put, what *IS* the method I should ude to ring the extensions, not the devices? |
| 16:44.10 | gartral | s/ude/use |
| 16:44.17 | Penguin | You don't ring extensions. |
| 16:44.24 | Penguin | Extensions ring devices. |
| 16:44.43 | Penguin | Phones ring. |
| 16:44.56 | Penguin | Asterisk extensions make those phones ring by using Dial(). |
| 16:45.03 | [TK]D-Fender | Extensions execute DIALPLAN APPLICATIONS |
| 16:46.39 | Penguin | If you want a phone to ring, create an extension to do it. exten => 1234,1,Dial(SIP/000011112222,34); In this example, extension 1234 will dial a device by the name of 000011112222, and let it ring for 34 seconds before exiting. |
| 16:47.04 | gartral | Penguin: ... how do I bounce the incoming call up the extensions.conf to where exten => 6002,1,Dial(SIP/gartral) exists, so that way I can have it auto-pick up and send DTMF(9)? |
| 16:47.36 | [TK]D-Fender | gartral: that's up to that device to be capable of picking up automatically or not |
| 16:47.36 | Penguin | You'd enter in 6002 on a phone and press send or dial, depending on the phone. |
| 16:48.19 | gartral | ugh.. |
| 16:48.25 | Penguin | Some phones can auto-answer. I haven't seen a soft phone that does, though. |
| 16:48.33 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 16:48.41 | [TK]D-Fender | I've seen the option in one of the coutnerpath ones... |
| 16:48.55 | Penguin | I guess that's the one to look for. |
| 16:49.02 | Kobaz | dialplan needs a * match for 0 or more matches (non-immediate completion) |
| 16:49.26 | Penguin | ! isn't what you want? |
| 16:49.30 | Kobaz | nope |
| 16:49.34 | Kobaz | ! is immediate completion |
| 16:49.39 | Penguin | Are you using patterns, otherwise? |
| 16:49.49 | Kobaz | yeap |
| 16:49.59 | Penguin | Drop the last pattern character and replace with a dot. |
| 16:50.29 | Kobaz | i suppose that accomplishes the same thing, but it doesn't look as pretty as 55XX* |
| 16:51.54 | gartral | Penguin [TK]D-Fender I have same => n(d),Dial(SIP/gartral,45&SIP/jitsi,45) in my [incoming] section of my dialplan in extensions.conf, what would I replace that Dial() application with so that it would jump to exten => 6002,1,Dial(SIP/gartral,20) and walk down THAT dialplan so I can test this without locking myself out of my building? |
| 16:52.01 | [TK]D-Fender | Kobaz: might as well do 55X. there then |
| 16:52.07 | Kobaz | yeah but it's not pretty |
| 16:52.16 | Penguin | Dial(SIP/gartral&SIP/jitsi,45) |
| 16:52.19 | [TK]D-Fender | Kobaz: considering if you're numeric... might as well use the "." instead of the last X |
| 16:52.25 | Penguin | two devices, one timeout value. |
| 16:52.28 | *** join/#asterisk justdave_ (~dave@unaffiliated/justdave) |
| 16:52.56 | [TK]D-Fender | gartral: "core show application goto" |
| 16:53.00 | gartral | Penguin: that's because I only want it to ring jitsi for 45 seconds, and forever on me |
| 16:53.33 | *** join/#asterisk Vendigroth (41254d61@gateway/web/freenode/ip.65.37.77.97) |
| 16:53.55 | [TK]D-Fender | gartral: then you'd have to dial 2 local channels with no timeout. Each of those would be responsible for their respective device dials with their own timeouts |
| 16:54.00 | Penguin | Dial() only takes ONE TIMEOUT VALUE for all devices in the dial string. All dials to devices will obey that timeout, unless otherwise configured to not obey it. |
| 16:54.40 | [TK]D-Fender | gartral: Or just a dial to you and a local channel that dials Jitsi with a timeout |
| 16:55.28 | jmetro | yep |
| 16:55.36 | gartral | ok |
| 16:55.37 | jmetro | dial local/jitsitimeout&sip/exten |
| 16:55.51 | Penguin | not SIP/exten, SIP/device. |
| 16:55.54 | jmetro | jitsi timeout being dial(sip/jitsi,45) |
| 16:55.57 | Penguin | You're confusing extensions and devices still. |
| 16:56.00 | Penguin | Stop it. |
| 16:56.06 | *** join/#asterisk jpoz (~jpoz@c-24-22-121-42.hsd1.or.comcast.net) |
| 16:56.17 | jmetro | Penguin: its his fault. |
| 16:56.23 | Penguin | He made you do it? |
| 16:56.39 | Vendigroth | What is the current dahdi_tool version? i have 2.7.0.1, but source seems to stop at 1.6. |
| 16:56.51 | Penguin | I think I would have blamed katty. |
| 16:57.14 | gartral | guys.. all of the numercal, dialable extensions are is extensions.conf with the rest of my dialplan |
| 16:57.33 | *** join/#asterisk lvlolvlo (~lvlolvlo@unaffiliated/lvlolvlo) |
| 16:57.47 | *** join/#asterisk jpoz_ (~jpoz@ec2-184-169-152-1.us-west-1.compute.amazonaws.com) |
| 16:57.50 | gartral | I'm using names for the devices because.. well.. it was the best solution I could come up with |
| 16:58.16 | Penguin | It'll be fine. |
| 16:58.25 | Penguin | It's not ideal, but it isn't the worst possible case. |
| 16:58.53 | Penguin | If the phones were permanently implanted in the persons, I would say it is perfect. |
| 16:59.25 | Penguin | At least with using a person's name for the device, you make the disconnection between extension and device. |
| 17:00.32 | Penguin | Whe people create device names that ARE the extension numbers, they usually cannot form the disconnection mentally. And they always have problems explaining things. |
| 17:01.24 | *** part/#asterisk Blashyrkh (~michaeld@043-054-094-081.as39912.net) |
| 17:03.04 | gartral | extensions are the number the device is configured to listen for.. the person is who has the device sitting in front of them. I get that.. but in my case.. there's only 4 people here.. so I don't really care much if the device is identified as the person to whome it belongs. |
| 17:03.09 | Penguin | no |
| 17:03.15 | Penguin | Extensions are unrelated to phones. |
| 17:03.49 | gartral | extension > phone < monkey |
| 17:03.51 | gartral | grr |
| 17:03.56 | gartral | extension > phone > monkey |
| 17:04.27 | Penguin | Extensions execute dial plan applications. An extension can play a sound file, ring a phone, or make [tk]d-fender's coffee. |
| 17:04.34 | ghost75 | monkey island |
| 17:05.47 | *** join/#asterisk ChannelZ-Wk (~bobm@spark.idolum.com) |
| 17:05.51 | gartral | in my mind, the extension is the phone number extension to which you are calling.. I get that Asterisk Extensions can be applications, but I'm not using many applications. |
| 17:06.18 | Penguin | Explicitly disconnect the term extension from the term phone in your mind. They are not related. |
| 17:06.37 | Penguin | The only thing extensions do is execute applications. |
| 17:10.52 | gartral | ok, so then if I put in same => n(d),GoTo(users,6002,3) under my GotoIF will that jump to extension 6002,3,Wait(5) and walk down to 6002,4,SendDTMF(9)? |
| 17:11.37 | Penguin | I would say yes. |
| 17:11.51 | Penguin | Assuming all prerequisites have been met. |
| 17:11.54 | gartral | thank you.. then I'm ready to test. |
| 17:13.22 | gartral | well I know for a fact that everything else under context [incoming] works, including running the Directory which allows people to calling in to find and instruct asterisk to ring their phone. |
| 17:15.44 | Penguin | My biggest concern is why you want to call in and hear the number 9 in your ear. |
| 17:15.57 | gartral | and I know that when I dial an extension for someone, thier phone ring rings |
| 17:16.32 | gartral | Penguin: because that's what the box at the door accepts as it's "authorized for entry" tone, it's outside of my control |
| 17:17.22 | gartral | so blah |
| 17:17.25 | Penguin | If you made it so the door phone called you on your phone, you answer the call from the person standing outside... |
| 17:17.38 | Penguin | And if you want them to come in, you just press 9 while talking to them? |
| 17:18.08 | file | yup |
| 17:18.26 | ChannelZ-Wk | I would make them sing or answer a riddle or something first |
| 17:18.28 | gartral | Penguin: not me, the company that installed and mantains it, and yes.. the stupid thing is wired directly into a POTS jack and gos through the PSTN |
| 17:18.31 | file | I had that at my apartment, although I hacked the unit a bit and gave myself my own entry code as well... |
| 17:19.16 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 17:20.02 | gartral | trust me, if I could, I would make the dumb thing listen for a voice-authorization and match against an voice imprint.. |
| 17:20.03 | Penguin | If you can make the call from the doorphone to a person's phone inside, and pressing 9 opens the door, I don't see why SendDTMF(9) won't open the door. I'd say send someone outside to try it. |
| 17:20.38 | gartral | Penguin: that's what I'm doinf, but I'm the only one here.. so it I lock myself out, I'm stuck for two hours |
| 17:20.46 | Greenlight | lol |
| 17:20.54 | Penguin | Long lunch break! |
| 17:20.57 | Greenlight | There's an incentive to get it right first time |
| 17:21.22 | gartral | Greenlight: that's what I've been here doing, making sure I understanf the dialplan and building it correctly |
| 17:22.29 | [TK]D-Fender | [12:15]gartraland I know that when I dial an extension for someone, thier phone ring rings <- that is the thought process to eliminate. When you dial a number.. it does SOMETHING. It could happen that the number is expected to make somethng ring, but that isn't always the case. |
| 17:22.42 | [TK]D-Fender | gartral: You dial an extension... and whatever it chooses to do is what it will do |
| 17:22.52 | *** join/#asterisk tomii (~tomii@128.54.168.125.sta.wbroadband.net.au) |
| 17:23.11 | Greenlight | Can't you jam the actual door open, and then "debug" safe in the knowledge you're not going to get locked out? |
| 17:24.24 | Penguin | He said the lock has to be engaged to make the phone work. |
| 17:24.35 | Greenlight | Oh. |
| 17:25.24 | Penguin | Which makes a test that a normal person would try to perform... void. |
| 17:25.55 | *** join/#asterisk war9407 (war@c-71-62-63-105.hsd1.va.comcast.net) |
| 17:26.30 | Penguin | So... |
| 17:27.23 | Penguin | After getting tired of asterisk 1.8 showing messages about fake auth and not being able to see the IP address of the offender, I came up with a solution that allows me to ban those addresses that were previously invisible. |
| 17:28.45 | Penguin | Setting allowguest=no is what makes those attempts get fake auth replies. So allowguest=yes stops the fake auth replies, thereby eliminating the hidden offender IP address within it. |
| 17:29.07 | jmetro | Penguin: what about fail2ban |
| 17:29.11 | *** join/#asterisk human39 (~human39@c-71-236-110-155.hsd1.pa.comcast.net) |
| 17:29.22 | Greenlight | fail2ban needs the IP ... |
| 17:29.27 | jmetro | Penguin: or update asterisk, my ast10 and 11 sees ips |
| 17:29.38 | Penguin | Make sure you have a sensible context set in the general section, capture all extensions dialed from devices not authenticated, and use the Log() application. |
| 17:29.53 | Penguin | Now fail2ban can match the Log info and ban accordingly. |
| 17:30.32 | Penguin | When it was just showing the fake auth messages, the IP address of the offender was not available, so fail2ban never knew there was a problem that needed dealt with. |
| 17:30.37 | *** join/#asterisk amizraa (~amizraa@gateway/tor-sasl/amizraa) |
| 17:30.37 | gartral | wait.. derp, I can replace the number in my GoTo application with my cell phone number and test that way |
| 17:30.44 | Penguin | 1.8 doesn't have the security logging mechanism. |
| 17:31.03 | Penguin | And I won't be upgrading any time soon. |
| 17:31.23 | Penguin | So the solution is to make it log the IP address in another manner. |
| 17:31.27 | human39 | Hi all, Has anybody played around with or have any insight with using the asterisk queues (or the call distribution system) with non-calls? In this example.. chats. |
| 17:32.08 | file | human39, erm it won't work |
| 17:32.23 | human39 | file, that is what I'm afraid of. :-) |
| 17:33.01 | Penguin | jmetro: 10 is where the security mechanism was added to logger. That functionality is what I desire, but I won't upgrade from 1.8 at this time. |
| 17:33.21 | human39 | I was kind pondering of a solution that using AMI to originate a call and send it to the queue.. and that call would be a nudge for the call center agent to check the chat queue. |
| 17:33.27 | gartral | eww |
| 17:33.36 | gartral | SO glad I wasn't outside.. |
| 17:33.37 | human39 | but there are too many holes.. and too much to go wrong. |
| 17:34.10 | Greenlight | human39: You could use the AMI to ChanSpy *whsiper* a beep or other tone to the agent, to give them an indication. |
| 17:34.23 | gartral | phone number rang in as +xxxxxxxxxx not xxxxxxxxxx |
| 17:34.25 | Greenlight | And then manage your "chat" in some other application. |
| 17:34.34 | gartral | how do I hadle that +? |
| 17:34.45 | Penguin | Remove it in dial plan. |
| 17:35.04 | jmetro | or make the chat program "nudge" |
| 17:35.09 | jmetro | ala shaking window. |
| 17:35.14 | jmetro | stealing focus, alert tones |
| 17:35.32 | human39 | Greenlight: thanks.. but we use a call back type system (with dynamic queues).. Also, I'd like them to not get calls when they're on a chat. |
| 17:35.38 | [TK]D-Fender | gartral: match it |
| 17:35.48 | gartral | Penguin: how? that was my cell phone number ringing it, that's how asterisk sees it |
| 17:35.48 | Penguin | Is the doorphone sending the phone number to call with a + on the front? |
| 17:36.03 | Penguin | I don't understand who is sending what. |
| 17:36.23 | gartral | Penguin: I changed the number that should be the doorphone with the number for my cell.. so I can test while inside :P |
| 17:36.34 | jmetro | gartral: whoever pushes your call to your server added a + |
| 17:36.37 | Greenlight | human39: Yea, you're getting into more complex territory there. You can of course use the AMI to Pause them in the queue |
| 17:36.42 | Penguin | So the door phone DIALED +yourcellnumber? |
| 17:37.00 | jmetro | gartral: match XXXXXXX and +XXXXXXX |
| 17:37.04 | gartral | Penguin: no, I dialed my asterisk number from my cell |
| 17:37.07 | jmetro | and have +XXXXXXX goto(XXXXXXX,1) |
| 17:37.34 | human39 | Greenlight: yeah.. that's a solution I was starting to poke at. I figure I would ask if there was an easier way.. or a already rolled solution to save me a bunch of time. |
| 17:37.35 | [TK]D-Fender | [12:34]gartralphone number rang in as +xxxxxxxxxx not xxxxxxxxxx <-- are yoi talking about the number that was DIALED IN? Or the CALLERID? |
| 17:37.43 | Penguin | You've being extremely vague in what has the plus sign on it. |
| 17:37.56 | Penguin | Be. More. Clear. |
| 17:37.58 | *** join/#asterisk felipealmeida (~user@177.41.16.10) |
| 17:39.37 | gartral | Penguin: http://paste.ubuntu.com/6416916/ here's what I'm seeing |
| 17:40.00 | *** join/#asterisk navaismo (~navaismo@189.241.101.4) |
| 17:40.24 | human39 | I thought of a hot second that maybe I could use SIP's chat protocol to help me out here, but I'm not seeing much open source software using it.. |
| 17:40.30 | Penguin | I need to see what was before that. |
| 17:40.38 | [TK]D-Fender | gartral: I'd go look at that expression real close |
| 17:41.15 | [TK]D-Fender | human39: * can send SIP Messages |
| 17:41.39 | Penguin | gartral: Show me the GotoIf() that you are using. |
| 17:42.00 | human39 | [TK]D-Fender: right, but can it easily be rolled into a full customer support chat system? :-) |
| 17:42.06 | gartral | same => n,GotoIf($[${CALLERID(num)}=+number]?d:allow) |
| 17:42.18 | human39 | well, of course it can.. but.. in my time frame |
| 17:42.39 | gartral | Penguin: ^^^ |
| 17:42.48 | [TK]D-Fender | human39: That has no useful detail to go on. |
| 17:43.07 | [TK]D-Fender | human39: So lets just say "yes, shall we? Would that make you feel good? |
| 17:44.03 | Penguin | gartral: So the comparison is the caller id number to a static number. The number being received in callerid contains a + and a 1 before the 10-digit number you are comparing it against. |
| 17:44.54 | human39 | [TK]D-Fender: I'm not asking anybody to do my job for me. I figured #asterisk would be a place to ask for possible tips or already rolled solutions. |
| 17:45.27 | [TK]D-Fender | human39: I don't see enough detail to say how easy it will be. |
| 17:45.39 | gartral | Penguin: yes, that appears to be the case, I had forgotten about that |
| 17:45.40 | Penguin | gartral: If you know for a fact that the callerid number from you phone will always have that plus on it, you could strip the plus from the number and then compare without it... or you could duplicate the actual value that is sent and use it in your comparison. |
| 17:45.58 | [TK]D-Fender | human39: Details make the difference between it being "easy", and "Oh God I'm completely screwed" |
| 17:46.39 | Penguin | gartral: GotoIf($[${CALLERID(num):1}=1number]?...) |
| 17:46.43 | *** join/#asterisk saint_ (~saint@c-50-166-85-78.hsd1.nj.comcast.net) |
| 17:46.58 | Penguin | gartral: GotoIf($[${CALLERID(num):2}=number]?...) |
| 17:47.36 | gartral | Penguin: I had tired that initially, and with the plus in the GotoIf I had given you, another problem is the random junk added *after* the phone number |
| 17:48.01 | Penguin | There's more stuff in the callerid number? |
| 17:49.19 | [TK]D-Fender | gartral: Show a complete call with a nOOp on the callerid prior |
| 17:49.25 | gartral | Penguin: non-issue.. adding the :2 where you said appears to work fine |
| 17:49.44 | human39 | [TK]D-Fender: fair enough. I'm currently in the discovery phase anyhow. Perfect world solution is that I can, somehow, accept a chat, much like I accept a call, obey agent weights, show as "on call" and have it all work. But again, that's pretty much what management wants. Right now, I'm focusing on if I can somehow distribute calls and chats in a similar fashion. |
| 17:50.05 | Penguin | If you have +1xxxxxxxxxx and compare it to xxxxxxxxxx, stripping the first two characters makes the comparison more "fair." |
| 17:51.04 | [TK]D-Fender | human39: You mean you want the queue to NOT distribute calls to you when a chat comes in? |
| 17:52.03 | human39 | [TK]D-Fender: If an agent is on a call, I don't want them to get a chat. Vice versa, if they are on a chat. no call. |
| 17:52.17 | gartral | Penguin: http://pastebin.com/MsPPvGtb |
| 17:52.23 | [TK]D-Fender | then have your chat system PauseQueueMember() them. |
| 17:52.50 | [TK]D-Fender | and have your queue dial local channels that execute a macro on agent answer that will signal your chat system accordingly |
| 17:53.42 | gartral | Penguin: now I have a new issue.. asterisk just hangs up, no waiting 5 secongs and playing a 9 in my ear |
| 17:54.09 | [TK]D-Fender | gartral: "core show application goto" <---- |
| 17:54.15 | [TK]D-Fender | gartral: Do read the instructions this time |
| 17:54.20 | Penguin | incoming-motif,users,6002 |
| 17:54.26 | [TK]D-Fender | nope |
| 17:54.36 | gartral | [TK]D-Fender: I read them, yes |
| 17:54.42 | Penguin | incoming-motif context; users extension; 6002 priority |
| 17:54.44 | human39 | [TK]D-Fender: thanks. Should I originate a Local change into the queue, then? |
| 17:54.47 | [TK]D-Fender | gartral: Where's your PRIORITY there? |
| 17:54.47 | Penguin | Probably not what you wanted. |
| 17:55.14 | Penguin | Your Goto() is broken. |
| 17:55.17 | [TK]D-Fender | human39: I'd use AMI, or as a CLI command call |
| 17:55.34 | *** join/#asterisk SushiB (~Thunderbi@200.77.217.106) |
| 17:55.55 | gartral | [TK]D-Fender Penguin the GoTo reads same => n(d),GoTo(users,6002,3) the priority, for what ever reason, is being ignored |
| 17:56.14 | Penguin | You forgot to dialplan reload? |
| 17:56.23 | gartral | I may have, hang on |
| 17:56.48 | Penguin | Asterisk says your Goto is Goto(users,6002). |
| 17:57.03 | [TK]D-Fender | "maybe" looks more like a certainty |
| 17:57.35 | gartral | Penguin: yea, I reloaded and now it reads Executing [s@incoming-motif:9] Goto("Motif/+1number-e11b", "users,6002,3") in new stack |
| 17:57.44 | Penguin | Good. |
| 17:57.48 | Penguin | Next problem? |
| 17:58.07 | gartral | Penguin: it's just hanging up, not waiting 5 seconds and sending DTMF 9 |
| 17:58.10 | human39 | [TK]D-Fender: yeah.. that would solve the agent not getting a call when busy portion. Any idea how you would handle a new, incoming chat that asterisk should distribute like a call? |
| 17:58.14 | Penguin | New debug? |
| 17:58.49 | [TK]D-Fender | human39: I fail to see how asterisk is part of the actual chat itself |
| 17:59.00 | [TK]D-Fender | human39: it is not a chat distribution system |
| 17:59.08 | Penguin | Oh, I have an idea what the problem might be. You answer and then right away send 1. I would answer, wait 2-8 seconds, and then send 1. |
| 17:59.19 | [TK]D-Fender | it can send/receive from individul SIP devices, but it is not a conversation redistributer |
| 17:59.21 | gartral | Penguin: pastebin.com/Ktkvtpc3 |
| 17:59.25 | [TK]D-Fender | THAT is not happening |
| 17:59.54 | [TK]D-Fender | gartral: Show us your dialplan |
| 18:00.06 | Penguin | users,6002,3 doesn't exist. |
| 18:00.35 | [TK]D-Fender | That's what it's pretty much saying... |
| 18:00.45 | [TK]D-Fender | So I'd like to see why he'd think it should |
| 18:00.45 | human39 | [TK]D-Fender: That's true. The issue is, though, if possible I'd like to somehow use asterisk to distribute the chats, along side with calls. |
| 18:00.56 | human39 | But again, it could be not possible. |
| 18:01.02 | Penguin | At the least, the users context and the entire 6002 extension. |
| 18:01.08 | [TK]D-Fender | human39: correct. that part is not. |
| 18:01.10 | gartral | http://paste.ubuntu.com/6417024/ |
| 18:01.29 | human39 | [TK]D-Fender: I appreciate your feedback, thanks! |
| 18:01.40 | [TK]D-Fender | gartral: I see 1 , 1, 2 |
| 18:01.41 | Penguin | gartral: No wonder. |
| 18:01.42 | Penguin | It is broken. |
| 18:01.47 | [TK]D-Fender | \garYou should watch your numbering |
| 18:01.57 | Penguin | Nah, that's a different extension. |
| 18:02.05 | Penguin | 6001,1; 6002,1 |
| 18:02.12 | gartral | read it again, first is a different extension |
| 18:02.16 | Penguin | It's the exten => same part that is busted. |
| 18:02.18 | [TK]D-Fender | gartral: and exten => same,3 is an NAMED extension called "same" |
| 18:02.40 | [TK]D-Fender | gartral: that is NOT "same => 3,App() |
| 18:02.43 | Penguin | Either stop using exten => after the first line, or don't use same in the extension. |
| 18:02.51 | [TK]D-Fender | gartral: You are mashing your syntax |
| 18:03.01 | gartral | oh.. derp |
| 18:03.06 | gartral | ok.. my bad |
| 18:03.14 | Penguin | And consider not using numbered priorities. |
| 18:03.20 | Penguin | That is a pain later on. |
| 18:03.24 | [TK]D-Fender | Or consider math :p |
| 18:04.12 | [TK]D-Fender | Numbered priorities are good. Because if you generate enough screwups that way maybe they'll take it as a sign they shouldn't be responsible for deploying solutions like this :) |
| 18:04.25 | Penguin | :D |
| 18:04.27 | gartral | I actually don't I changed them because I thought that the GotoIf was choking on that, not the same part |
| 18:04.55 | [TK]D-Fender | Penguin: Can't count your step #'s? GTFO!!! :p |
| 18:06.27 | gartral | there we go.. ok, I didn't explicitly *hear* the 9, but I see it was sent on the console |
| 18:07.12 | gartral | I heard a faint click sound from my phone then the call hung up |
| 18:07.16 | gartral | so i think this will work |
| 18:08.26 | *** join/#asterisk u0m3 (~u0m3@109.96.137.96) |
| 18:19.48 | *** join/#asterisk serafie (~erin@nat/digium/x-iovjoneuzmxdslyp) |
| 18:23.36 | gartral | ok, that works and someone else is now here.. so all I *should* need to do change the goto to read GoTo(users,6002&6003,1) yes? |
| 18:24.48 | [TK]D-Fender | gartral: no. |
| 18:24.56 | [TK]D-Fender | gartral: You are inventing syntax again |
| 18:25.08 | [TK]D-Fender | gartral: there is no "&" for a Goto |
| 18:26.04 | gartral | [TK]D-Fender: then what is the operator for defining multiple extensions? |
| 18:26.04 | gartral | just CSVs? |
| 18:26.15 | [TK]D-Fender | gartral: No. |
| 18:26.45 | [TK]D-Fender | [12:54][TK]D-Fendergartral: "core show application goto" <---- |
| 18:27.20 | gartral | [TK]D-Fender: I did! and it says in it's syntax "Goto([[context,]extensions,]priority)" |
| 18:27.51 | gartral | [TK]D-Fender: _extensions_ |
| 18:27.59 | gartral | there is an S there |
| 18:28.22 | [TK]D-Fender | it isn't plural |
| 18:28.34 | [TK]D-Fender | EXTENSION |
| 18:28.37 | [TK]D-Fender | ONE |
| 18:28.40 | [TK]D-Fender | there is no multiple |
| 18:28.42 | [TK]D-Fender | that is a typo |
| 18:29.12 | gartral | well then that isn't _MY_ fault, now is it? :P |
| 18:29.34 | Penguin | No, someone made a typo in there. |
| 18:29.34 | [TK]D-Fender | gartral: it is for you not reading the actual body of the instructions that show it as singulatr EVERY time |
| 18:30.16 | [TK]D-Fender | gartral: Read the title off the cover of the book, that means I know the whole story right? |
| 18:30.47 | *** join/#asterisk endemic (~endemic@lynx.onvox.net) |
| 18:31.11 | gartral | [TK]D-Fender: that doesn't matter, sytactually, MOST people would read that to mean you can use GoTo([[context,] extension, extension, etc..] priority) |
| 18:31.49 | [TK]D-Fender | gartral: that doesn't have the [..] indication of multiple like other syntax would show |
| 18:32.23 | gartral | [TK]D-Fender: you're not like much where you work, are you? |
| 18:32.24 | Penguin | It's an error. Let's accept it and move on. |
| 18:32.30 | gartral | liked* |
| 18:33.30 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 18:33.30 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 18:33.39 | gartral | Penguin [TK]D-Fender regardless, I have a working solution, use Dial, as I don't want the door-phone going a voicemail anyway |
| 18:34.35 | Penguin | So you're going to create a new extension just for the door phone? |
| 18:34.45 | Penguin | And not use Voicemail()? |
| 18:36.51 | ChannelZ-Wk | Ding dong, Avon lady! |
| 18:37.01 | gartral | Penguin: no, i'm using Dial(SIP/ext&SIP/ext, 45) |
| 18:37.54 | Penguin | Except that you're indicating dialing extensions, again. |
| 18:38.26 | Penguin | That says to me that you're still having trouble with that concept. |
| 18:39.31 | navaismo | ChannelZ-Wk, at least not a jehova guy |
| 18:44.50 | gartral | Penguin: ok, s/ext/dev regardless.. it works.. |
| 18:52.50 | gartral | Penguin [TK]D-Fender thank you for baring with my ignorance on this and helping me, you've both been invauable |
| 18:52.57 | *** join/#asterisk tomii (~tomii@128.54.168.125.sta.wbroadband.net.au) |
| 18:53.57 | *** join/#asterisk budmang (~budmeng@ip72-194-86-41.oc.oc.cox.net) |
| 18:54.09 | budmang | would there be any reason, just parked calls would have trouble hearing? |
| 18:54.14 | budmang | no right? |
| 18:54.53 | [TK]D-Fender | not specifically |
| 18:58.02 | ChannelZ-Wk | Old age? |
| 18:58.05 | jmetro | i have an error transferring extensions |
| 18:58.46 | jmetro | ext 101 dials 105, they talk, 101 tries to transfer the call, asterisk console shows "Got OK on Refer notify" and then sends a 400 error to 101 |
| 18:58.56 | jmetro | call drops |
| 18:59.35 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-cwyxvvfvuisgsqiz) |
| 18:59.38 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 19:00.01 | ChannelZ-Wk | Some people like to do attended transfers and just hang up before the person answers thinking they're doing a blind.. |
| 19:00.16 | jmetro | This is me doing the transfer on two test phones <.< |
| 19:01.09 | jmetro | ive also never seen this Got OK on Refer Notify in my console until this started happening] |
| 19:02.11 | ChannelZ-Wk | Can't say I ever have either. What kind of phones? |
| 19:03.21 | jmetro | 2 aastra 6757i's, ive also reproduced on a spa509 |
| 19:04.29 | ChannelZ-Wk | you say on "a" spa509", is that between one of the aastras? |
| 19:04.46 | jmetro | SPA to Polycom, Spa fails. |
| 19:04.56 | jmetro | aastra <> aastra and aastra <> cisco also fails |
| 19:05.28 | ChannelZ-Wk | Neat. |
| 19:05.31 | *** join/#asterisk yoavz (yoavz@yoavz.net) |
| 19:05.58 | jmetro | its always PartyA that fails, partyb has no problem |
| 19:13.19 | ChannelZ-Wk | * version? |
| 19:13.53 | *** join/#asterisk serafie (~erin@nat/digium/x-kizimtqaddgdttkh) |
| 19:19.13 | jmetro | happens on 10 and 11 |
| 19:19.16 | jmetro | we have both |
| 19:19.23 | jmetro | brb |
| 19:20.29 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-cbxfxoezyjprbuyi) |
| 19:20.30 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 19:20.58 | *** join/#asterisk kpettit (~keith@99-116-144-119.lightspeed.hstntx.sbcglobal.net) |
| 19:22.14 | kpettit | is there anything other than "registerattempts=0" I can do to force asterisk to do to keep trying to connect to a sip provider? |
| 19:28.32 | *** join/#asterisk jmetro (~nickserv@75-150-221-198-Illinois.hfc.comcastbusiness.net) |
| 19:33.17 | *** join/#asterisk jpoz (~jpoz@ec2-184-169-152-1.us-west-1.compute.amazonaws.com) |
| 19:42.59 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 19:50.52 | *** join/#asterisk d45h (~d45h@unaffiliated/dash-/x-7576607) |
| 19:51.33 | *** join/#asterisk dms (~dms@c-68-58-240-113.hsd1.sc.comcast.net) |
| 19:51.42 | dms | [Nov 14 19:50:22] WARNING[10834][C-00000005]: channel.c:5337 set_format: Unable to find a codec translation path from (nothing) to (ulaw) |
| 19:51.42 | dms | [Nov 14 19:50:22] WARNING[10834][C-00000005]: bridge.c:880 bridge_make_compatible: Failed to set channel PJSIP/anonymous-0000000a to read format ulaw |
| 19:51.48 | dms | file: putnopvut: ^^ |
| 19:51.51 | dms | help |
| 19:51.53 | dms | pls |
| 19:51.54 | dms | :) |
| 19:52.00 | dms | driving me nunts |
| 19:52.04 | putnopvut | uhhh |
| 19:52.22 | dms | 4 19:50:22] WARNING[10834][C-00000005]: channel.c:5106 ast_write: Codec mismatch on channel PJSIP/anonymous-0000000a setting write format to ulaw from unknown native formats (nothing) |
| 19:52.22 | dms | [Nov 14 19:50:22] WARNING[10834][C-00000005]: channel.c:5337 set_format: Unable to find a codec translation path from (nothing) to (ulaw) |
| 19:52.22 | dms | [Nov 14 19:50:22] WARNING[10834][C-00000005]: chan_pjsip.c:814 chan_pjsip_write: Asked to transmit frame type ulaw, while native formats is (nothing) (read/write = unknown/unknown) |
| 19:52.22 | dms | [Nov 14 19:50:22] WARNING[10834][C-00000005]: channel.c:5106 ast_write: Codec mismatch on channel PJSIP/anonymous-0000000a setting write format to ulaw from unknown native formats (nothing) |
| 19:52.22 | dms | [Nov 14 19:50:22] WARNING[10834][C-00000005]: channel.c:5337 set_format: Unable to find a codec translation path from (nothing) to (ulaw) |
| 19:52.22 | dms | [Nov 14 19:50:22] WARNING[10834][C-00000005]: chan_pjsip.c:814 chan_pjsip_write: Asked to transmit frame type ulaw, while native formats is (nothing) (read/write = unknown/unknown) |
| 19:52.22 | dms | [Nov 14 19:50:22] WARNING[10834][C-00000005]: channel.c:5106 ast_write: Codec mismatch on channel PJSIP/anonymous-0000000a setting write format to ulaw from unknown native formats (nothing) |
| 19:52.23 | dms | [Nov 14 19:50:22] WARNING[10834][C-00000005]: channel.c:5337 set_format: Unable to find a codec translation path from (nothing) to (ulaw) |
| 19:52.24 | dms | [Nov 14 19:50:22] WARNING[10834][C-00000005]: chan_pjsip.c:814 chan_pjsip_write: Asked to transmit frame type ulaw, while native formats is (nothing) (read/write = unknown/unknown) |
| 19:52.25 | dms | [Nov 14 19:50:22] WARNING[10824]: res_pjsip_sdp_rtp.c:234 set_caps: No joint capabilities between our configuration((nothing)) and incoming SDP((ulaw|alaw|g729)) |
| 19:52.26 | putnopvut | Do you have any allow/disallow settings on your pjsip endpoint? |
| 19:52.28 | [TK]D-Fender | PASTEBIN |
| 19:52.34 | dms | sorry, sorry |
| 19:52.35 | [TK]D-Fender | dms: do NOT flood like that |
| 19:52.36 | dms | been a while |
| 19:53.02 | dms | [TK]D-Fender: who are you even? |
| 19:53.04 | dms | lol |
| 19:53.21 | putnopvut | dms: by default, there are no formats set on a PJSIP endpoint. So you have to explicitly have allow=<format> to use them. |
| 19:53.30 | putnopvut | I'm guessing that's what's up here, but not sure. |
| 19:53.55 | dms | putnopvut: Gotcha. I didn't originally, but I put them in. Let me just confirm and pastebin the config |
| 19:54.04 | putnopvut | Okay |
| 19:56.49 | dms | putnopvut: http://pastebin.com/Z95P5a24 |
| 19:57.06 | putnopvut | looks |
| 19:57.48 | putnopvut | dms: try adding allow=ulaw to your anonymous endpoint. |
| 19:57.58 | dms | ok |
| 19:58.07 | WIMPy | a/ign 1-* |
| 19:58.14 | WIMPy | oops |
| 19:58.40 | WIMPy | (and a space missing as well) |
| 20:00.18 | *** join/#asterisk can0beans (~can0beans@unaffiliated/can0beans) |
| 20:00.22 | dms | putnopvut: no dice. Here is the logging output . . http://pastebin.com/2ksFnDX4 .. |
| 20:00.24 | *** join/#asterisk Shinobi (~wjtaylor@c-76-23-208-102.hsd1.ct.comcast.net) |
| 20:00.41 | *** part/#asterisk can0beans (~can0beans@unaffiliated/can0beans) |
| 20:01.08 | dms | what struck me was this . . [Nov 14 19:57:33] WARNING[10896]: pjsip:0 <?>: tsx0x857ee4 ...Failed to send Request msg INVITE/cseq=11356 (tdta0x8012d0)! err=120001 (Operation not permitted) |
| 20:01.45 | Shinobi | is it possible to be your own trunk, if you purchase your own DIDs? |
| 20:02.15 | ChannelZ-Wk | Trunk of what? |
| 20:03.20 | Shinobi | SIP |
| 20:03.47 | WIMPy | ~siptrunk |
| 20:03.50 | ChannelZ-Wk | I guess I don't understand what you're really asking. |
| 20:05.07 | Shinobi | If I purchase a number (DID) and have an internet connection, why do I need a SIP service provider. Can I do everything with a PBX, DID, internet, and some servers? |
| 20:05.12 | putnopvut | dms: ah this time the incoming call is from PJSIP/1000 instead of PJSIP/anonymous. You can see from the incoming SDP offered silk, h.261 and h.263. Endpoint 1000 only allows ulaw, so right there, you've got a problem. |
| 20:05.23 | putnopvut | But let me look further to see if there are other potential issues. |
| 20:05.25 | Shinobi | How would I become a SIP trunk provider? |
| 20:05.38 | myself | who are you purchasing the number from, and how do you connect to whatever you connect to? |
| 20:06.05 | putnopvut | dms: er, wait, I guess you were directing me to the call after you turned logging on. Hold up. |
| 20:06.09 | myself | A DID on the PSTN is meaningless it's reachable from the PSTN |
| 20:07.36 | [TK]D-Fender | [15:04]ShinobiIf I purchase a number (DID) and have an internet connection, why do I need a SIP service provider. Can I do everything with a PBX, DID, internet, and some servers? <_ how do you thing that DID **arrives**at your server in the first place? |
| 20:07.38 | ChannelZ-Wk | The DID has to terminate somewhere. The worldwide phone system is not all VoIP, so you do need *someone* else involved |
| 20:07.39 | [TK]D-Fender | think* |
| 20:08.51 | jmetro | I wonder if large companies dont change their CCM hold music on purpose to let everyone know they use cisco |
| 20:09.22 | ChannelZ-Wk | "I can change it?!" |
| 20:09.24 | drmessano | Probably |
| 20:09.50 | drmessano | "I have CCM.. I have so much money. I R AWESUM" |
| 20:10.09 | ChannelZ-Wk | That'll be $10,000 please. |
| 20:10.33 | eirirs | lol |
| 20:10.34 | Shinobi | [TK]D-Fender: I hear you. I guess I don't understand the architecture of the infrastructure beyond the PBX. Verizon and comcast convert phone lines to voip and somewhere upstream they must just pass data back and forth. Agree? |
| 20:10.46 | drmessano | CCM is hilarious when IT sets up an all-page between 20+ sites and 5000+ users and someone discovers it |
| 20:11.02 | ChannelZ-Wk | The phone system is a spaghetti of equipment |
| 20:11.23 | [TK]D-Fender | Shinobi: Somewhat. They ARE the upstream and they connect directly to one another. |
| 20:11.47 | [TK]D-Fender | Shinobi: A phone call may hop through multiple networks from A to B |
| 20:11.58 | [TK]D-Fender | Shinobi: And forget "IP" for most of that |
| 20:12.24 | [TK]D-Fender | Shinobi: That's where old-school big-league connectivity comes in like OC-384, etc |
| 20:12.34 | Shinobi | [TK]D-Fender: forget IP? not following? |
| 20:12.57 | [TK]D-Fender | Telco's don't use VoIP as a backhaul normally |
| 20:13.10 | [TK]D-Fender | this is for your understanding of the PSNT as a whole |
| 20:13.18 | *** part/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 20:13.34 | [TK]D-Fender | http://en.wikipedia.org/wiki/Optical_Carrier_transmission_rates |
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| 20:14.33 | [TK]D-Fender | Shinobi: And becoming your own mini-telco has a ton of legal requirements. |
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| 20:15.30 | Shinobi | I'm not even at that level, I guess the first thing I need to know is how are SIP addresses and IP addresses mapped? Is there something like DNS for SIP? |
| 20:16.19 | putnopvut | dms: just letting you know I"m still looking. Trying to figure out why PJSIP would say it doesn't have permission to send the INVITE. It could literally be a permission problem on the system, but I'd like to know a bit more before saying that's for sure it. |
| 20:16.34 | Shinobi | [TK]D-Fender: I'm not really trying to be a telco, it was just a way of asking the question so that you know what info I was looking for. |
| 20:16.47 | [TK]D-Fender | Shinobi: There is no such thing as a "SIP address really. |
| 20:17.03 | ChannelZ-Wk | SIP is IP so regular DNS works. But that doesn't get you anywhere with a "real phone number" |
| 20:17.32 | Shinobi | ChannelZ-Wk: I understand. I'm just taking one piece at a time |
| 20:17.44 | ChannelZ-Wk | There is a mechanism for turning a phone number into a voip destination but it's barely supported anywhere AFAIK |
| 20:18.18 | kpettit | Shinobi: You have to tie into the phone network if you want to make calls to normal phone numbers. You tie into the phone network by using a PSTN line or a SIP provider. |
| 20:18.43 | [TK]D-Fender | http://en.wikipedia.org/wiki/Telephone_number_mapping <- enum |
| 20:18.51 | [TK]D-Fender | E.164 |
| 20:18.55 | kpettit | You can't get around not using some sort of telphone provider |
| 20:19.02 | ChannelZ-Wk | that's the one |
| 20:19.49 | Shinobi | but just softphone to softphone, you shouldn't need a thing, correct? |
| 20:19.55 | kpettit | correct |
| 20:20.03 | Shinobi | OK. cool |
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| 20:21.07 | putnopvut | dms: right now permissions issue is the best I've got on it. |
| 20:21.16 | Shinobi | let me read that article... |
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| 20:25.43 | dms | putnopvut: thanks man |
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| 21:12.28 | JamKo | Greetings. Does asterisk version 11 still use the udptl.conf file to set the port range to use? I have it set to use a custom range, yet it still uses the default 4000 range. |
| 21:12.48 | JamKo | Prior to upgrade from 1.8 it was honoring my custom range. |
| 21:14.28 | esaym153 | any tips for using iax2 over a wireless link that has ping times that vary from 80-150 ms? (plus occasional packetloss...) |
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| 21:41.20 | mjordan | JamKo: yes, it does. If it isn't, you may have an error in your udptl.conf file that is causing it to not be processed. You may want to check your ERROR/WARNING messages on startup |
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| 22:05.30 | JamKo | mjordan: thanks. |
| 22:06.52 | JamKo | Anyone up to debug a t.38 passthrough problem I'm having? I have a packet capture ready to go. I've just upgraded to 11.6 from 1.8 to try resolving it. The problem.. |
| 22:07.08 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 22:07.41 | JamKo | is specific to asterisk getting along with sonus session border controllers and t.38 |
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| 22:27.55 | crazed1 | with penalties, i have a problem. we have like 30+ agents in the queue, and we want to have 5 in Tier 0, but the problem is if someone in T0 walks away and leaves their softphone open, (aka logged in the queue) then it only rings that person repeatedly and never moves on to the t1 members |
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| 22:30.04 | danfromuk | Hi, is the queue hold time, an average of all calls queued, or the total of all calls queued? |
| 22:30.39 | danfromuk | Never mind. its an average. |
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| 22:46.04 | drmessano | Hmm |
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| 23:54.11 | bunglex | hi, ive enabled blindxfer in fearues.conf and added an x to the dial command - but the # and * are not being recognised at all ? |
| 23:54.42 | bunglex | im trying to enable a call parking or 'put on hold' feature |
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