| 00:11.22 | CoffeeIV | It seems that my dialplan needed CONFBRIDGE_LEAVE_SOUND and CONFBRIDGE_JOIN_SOUND set, and the dialplan pre-dated those being needed |
| 00:11.33 | CoffeeIV | now I just have to figure out what to set them to |
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| 02:43.55 | *** join/#asterisk infobot (ibot@rikers.org) |
| 02:43.55 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.8.2.3 (2010/01/26), 1.6.2.16.1 (2010/01/18), 1.4.39.1 (2010/01/18), *-Addons 1.6.2.3, 1.4.13 (2010/01/14), dahdi-linux 2.4.0 (2010/09/01), dahdi-tools 2.4.0 (2010/09/01), libpri 1.4.11.5 (2010/11/22) -=- Visit the new official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/ |
| 02:46.47 | *** join/#asterisk IsUp (IsUp@unaffiliated/isup) |
| 02:53.58 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
| 02:55.49 | neurosys | Im using 1.6.2.14 |
| 02:56.14 | p3nguin | I'm a 1.4 type of person. |
| 03:20.08 | *** join/#asterisk Bryanstein (bryan@shellium/admin/bryanstein) |
| 03:23.16 | ChannelZ | going back in time, since Skype is messing with the whole business control panel thing, and Skype For Asterisk can't work with normaly Skype users, it's less interesting to play with for fun |
| 03:23.23 | Kobaz | wow |
| 03:23.26 | Kobaz | this is totally crazy |
| 03:24.51 | *** join/#asterisk Mhaddog (~Mhaddog@z65-50-116-17.ips.direcpath.com) |
| 03:24.57 | *** join/#asterisk youngproguru (~mm@pool-173-64-5-115.bflony.fios.verizon.net) |
| 03:24.59 | Kobaz | SIP/201 rings SIP/202... if SIP/202 picks up and does an attended transfer very very quickly (as in, immediatly completing the transfer after starting the dial), the dialplan will execute on SIP/201 |
| 03:25.33 | Kobaz | if SIP/202 takes it's time, then the dialplan runs on the proper channel, which is the channel making the call: SIP/202 |
| 03:26.52 | *** join/#asterisk timahvo1 (~rogue@41.223.57.77) |
| 03:28.30 | Kobaz | i guess it treats it as a blind transfer if it happens really quickly on these polycom phones |
| 03:37.08 | Kobaz | but the problem is, the event you get still says attended transfer, but asterisk is doing a blind transfer instead |
| 03:37.13 | Kobaz | looks like a bug |
| 03:40.08 | WIMPy | In the phone or in Asterisk? |
| 03:40.12 | Kobaz | in asterisk |
| 03:40.27 | Kobaz | asterisk is telling me it's an attended transfer, but it runs the dialplan on the wrong channel |
| 03:40.33 | Kobaz | but only if the transfer happens really fast |
| 03:40.58 | *** join/#asterisk psilikon (~psilikon@147-112.96-97.tampabay.res.rr.com) |
| 03:41.51 | Kobaz | that's what it seems |
| 03:41.51 | *** join/#asterisk ketas (ketas@ketas6-sixxs.si.pri.ee) |
| 03:42.06 | Kobaz | i haven't tried on a non-polycom, but looking at the events, it looks like an asterisk issue |
| 03:43.39 | WIMPy | But an attended transfer is a new call. How can that become a blind transfer? |
| 03:43.42 | Kobaz | i would think attended transfers should always behave the same way, no matter how fast you do one |
| 03:43.51 | Kobaz | exactly |
| 03:44.20 | WIMPy | So if you place a cll immediately after answering one it becomes a transfer? |
| 03:44.39 | Kobaz | it gets executed as a blind transfer |
| 03:44.46 | Kobaz | but you get an event saying it's attended |
| 03:44.55 | Kobaz | i need to add some more debugging |
| 03:45.39 | Kobaz | this is even crazier |
| 03:45.50 | Kobaz | it's much worse than being treated as a blind transfer |
| 03:45.51 | WIMPy | Why do you get an event about a transfer at all at that point? |
| 03:46.15 | WIMPy | How fast do you have to be? |
| 03:46.18 | Kobaz | if i do a DumpChan on the channel |
| 03:46.21 | Kobaz | i get this: Name= SIP/6208-000000aa |
| 03:46.37 | Kobaz | but the args that AGI is sending me: 'channel' => 'SIP/6288-000000ac', |
| 03:46.50 | *** join/#asterisk csnook (~chris@c-76-19-64-161.hsd1.ma.comcast.net) |
| 03:46.57 | Kobaz | so the agi is executing like a blind transfer, but agi is passing me the wrong channel |
| 03:47.03 | WIMPy | I don't get any odd behavior. |
| 03:47.15 | Kobaz | fast as in like, accept the call, transfer, number transfer |
| 03:47.17 | Kobaz | really fast |
| 03:47.51 | WIMPy | Ok, so it happens when you complete the transfer? |
| 03:48.09 | Kobaz | the oddness of the behavior will depend on how your dialplan is written... if you're not doing anything other than dialing another endpoint, you wouldn't notice any difference |
| 03:48.25 | Kobaz | but i have lots and lots of channel tracking that requires asterisk to be running dialplan on the correct channel |
| 03:48.39 | Kobaz | yeah, completing the transfer very fast |
| 03:48.55 | Kobaz | and, a bigger problem, is if you do it in the 'too fast' timeframe |
| 03:49.03 | Kobaz | you will lose the call on the calling side |
| 03:49.08 | Kobaz | asterisk just dumps it |
| 03:49.22 | Kobaz | but! the destination side is ringing |
| 03:49.26 | Kobaz | the middle phone is hung up |
| 03:49.41 | Kobaz | so now you have a ringing channel, that's totally dead and stuck |
| 03:49.52 | Kobaz | i've seen this many many times, and i finally know how to reproduce it |
| 03:50.10 | Kobaz | the only way to get rid of the ringing channel is to hit reject on the phone |
| 03:50.22 | Kobaz | since according to asterisk it's gone and destroyed, but it never sent a BYE |
| 03:51.06 | WIMPy | I think I read an issue to that effect, yes. |
| 03:52.59 | Kobaz | hmm, i just tried in in 1.8.2.3 at least it doesn't get stuck |
| 03:53.10 | Kobaz | the call just drops if you complete the transfer too fast |
| 03:53.24 | Kobaz | which is still a problem, but at least you dont have people screaming why cant i pick up this call |
| 03:53.49 | WIMPy | I can't get it to drop. |
| 03:54.13 | WIMPy | But then I can't transfer too fast. |
| 03:54.18 | Kobaz | heh |
| 03:54.34 | Kobaz | lemme try a simple setup, i have Originate's and Bridge's involved |
| 03:54.49 | WIMPy | I just hit transfer until it hangs up :-) |
| 03:56.10 | Kobaz | it probably has to do with the Bridge() |
| 03:56.16 | Kobaz | i cant get the same behavior doing a straight dial |
| 03:56.21 | Kobaz | figures |
| 03:56.29 | Kobaz | any time i'm doing complex stuff is when i break asterisk |
| 03:56.30 | WIMPy | I just wonder if the caller id issue has gotten worse. |
| 03:56.30 | Kobaz | heh |
| 03:57.25 | WIMPy | Holy shit |
| 03:57.49 | WIMPy | It depends on how I dial. I didn't notice that before. |
| 03:58.58 | Kobaz | oh |
| 03:58.59 | Kobaz | haha |
| 03:59.13 | WIMPy | I've got overlap dialling enabled for sip. And when I use it, i.e. dial by hand, I get the wrong caller ID. When I rit redial, it's correct. |
| 03:59.21 | Kobaz | just crashed 1.8.2.3 by doing an originate that bridges a call, and then did my fast transfer |
| 03:59.25 | WIMPy | hit |
| 03:59.42 | Kobaz | doesn't crash 1.6.0.26 |
| 04:00.06 | WIMPy | This is getting interesting :-( |
| 04:03.21 | Kobaz | but in 1.6.0.26 you drop the call |
| 04:03.27 | Kobaz | and keeps ringing the other side |
| 04:03.33 | Kobaz | this is good |
| 04:03.55 | Kobaz | well, not good good, but good that i know all the info on how this bug works |
| 04:04.08 | WIMPy | Hmm. Did 'core show channels verbose' always display the dialled extension in the 'CallerID' column of the 2nd call leg? |
| 04:04.14 | Kobaz | dont think so |
| 04:04.18 | Kobaz | okay WIMPy |
| 04:04.18 | Kobaz | try this |
| 04:04.35 | WIMPy | doesn't think so, either. |
| 04:04.42 | WIMPy | Maybe I'm getting nearer as well. |
| 04:05.12 | Kobaz | http://pastebin.com/VtbEUhVx |
| 04:05.30 | Kobaz | it results in a stuck call for me in 1.6.0.26 |
| 04:05.36 | Kobaz | and it results in a crash in 1.8.2.3 |
| 04:05.50 | Kobaz | in 1.8, replace the TrySystem with the commented Originate() |
| 04:07.46 | *** join/#asterisk obnauticus (~obnauticu@about/windows/regular/obnauticus) |
| 04:07.47 | WIMPy | So I dial 1234 and replace 6288 and 6208 with extensions? |
| 04:07.59 | Kobaz | yeah sure |
| 04:08.01 | WIMPy | Or rather the other way round :) |
| 04:08.16 | Kobaz | you need two phones |
| 04:08.27 | Kobaz | one calls 1234 |
| 04:08.42 | Kobaz | the bridge should be using the same device as it's calling from |
| 04:08.48 | Kobaz | so 6208 calls 1234 |
| 04:08.58 | Kobaz | and the originate does a bridge back to 6208 |
| 04:09.23 | Kobaz | so if your phone is 201.... 201 calls 1234.... it should be Bridge,201 |
| 04:09.30 | WIMPy | never used dialplan originate. |
| 04:09.36 | Kobaz | it's new in 1.6.2 |
| 04:09.48 | Kobaz | it's kinda crappy that you cant put in arguments |
| 04:09.56 | WIMPy | I know it exists, but I never used it. |
| 04:09.57 | Kobaz | so i can't use it for real stuff, need to use the AMI |
| 04:11.42 | WIMPy | Oh, it died on 'dialplan reload' :-( |
| 04:13.09 | Kobaz | died as in crash? |
| 04:13.11 | WIMPy | Ok, the call worked. |
| 04:13.14 | Kobaz | yeah |
| 04:13.16 | Kobaz | the call should work |
| 04:13.20 | WIMPy | Yes |
| 04:13.21 | Kobaz | now try transfering |
| 04:13.31 | Kobaz | for me it crashes on your do a 'fast attended transfer' |
| 04:13.32 | WIMPy | So then I transfer that again? |
| 04:13.38 | Kobaz | on 1.8.3.2 anyway |
| 04:13.38 | Kobaz | yeah |
| 04:13.56 | WIMPy | Hmm, need to do it the other way round then. |
| 04:13.57 | Kobaz | what version are you on |
| 04:14.11 | Kobaz | you need three phones actually... i lied before about needing just two |
| 04:14.23 | Kobaz | so |
| 04:14.31 | Kobaz | this is *really* fscked up |
| 04:14.32 | WIMPy | But the intersting thing is that I cannot end that call from the called phone. |
| 04:14.37 | WIMPy | The calling one stays connected. |
| 04:14.39 | Kobaz | back to this blind/attended transfer thing |
| 04:14.43 | Kobaz | yeah |
| 04:14.51 | Kobaz | there's nasty bugs with bridge and transfering |
| 04:14.54 | WIMPy | SVN from yesterday. |
| 04:15.14 | Kobaz | so, get this |
| 04:15.19 | Kobaz | i do my 'fast transfer' |
| 04:15.37 | Kobaz | the new Dial() is running on SIP/6208-00000009 |
| 04:15.46 | Kobaz | $CHANNEL tells me that |
| 04:15.49 | Kobaz | DumpChan tells me that |
| 04:15.59 | Kobaz | but agi sends me SIP/6288-0000000b |
| 04:16.30 | WIMPy | Hmm. 'dialplan reload' seems to always crash. Even without syntax errors. |
| 04:16.31 | Kobaz | so, i have to verify that asterisk sent me the right channel, so... i guess i can't trust that the agi variable is correct |
| 04:16.51 | Kobaz | passes WIMPy the baton |
| 04:17.00 | Kobaz | bug report it is! |
| 04:17.05 | thehar | farts |
| 04:19.00 | *** join/#asterisk MrTelephone (~ropeburn@S01060018e7e1c1b7.ls.shawcable.net) |
| 04:19.20 | WIMPy | Hmm. Does that bridge thing only work with SIP? |
| 04:19.29 | MrTelephone | does anyone have any ideas as how to authenticate an endpoint that support *67 when they share the same IP and souce port? |
| 04:19.49 | Kobaz | WIMPy: should work with any device |
| 04:20.03 | Kobaz | it'll do native bridging |
| 04:20.03 | WIMPy | Or channels that carry the peer name in their name? |
| 04:20.15 | Kobaz | perhaps |
| 04:20.42 | MrTelephone | asterisk is all about bridging |
| 04:20.49 | WIMPy | bridge failed because channel does not exist or can't get lock. |
| 04:21.08 | WIMPy | Ok, fetched another sip phone. |
| 04:21.25 | WIMPy | But that seems to make bridge pretty senseless. |
| 04:21.43 | MrTelephone | what are you bridging |
| 04:22.13 | WIMPy | lcr |
| 04:22.40 | MrTelephone | Asterisk should go through all the register requests when authenticating instead of stopping at the first one with IP and SOURCE port match. Is that a stupid idea? |
| 04:22.44 | MrTelephone | lcr? |
| 04:23.01 | WIMPy | Firs I do an 'svn up' to see if it still crashes in 'dialplan reload'. |
| 04:23.18 | WIMPy | chan_lcr |
| 04:23.37 | WIMPy | I usually only have one sip phone on my desk. |
| 04:24.20 | WIMPy | Nope. Still crashes. :-( |
| 04:25.01 | WIMPy | Damn. In theory I have been asleep for 2.5 hours... |
| 04:25.23 | Kobaz | MrTelephone: bridge() followed by a very fast attended tranafer completion crashes 1.8.2.3 for me |
| 04:26.04 | Kobaz | fast attended transfer completions just wreck havok on my asterisk in general |
| 04:26.24 | Kobaz | asterisk sending me the wrong channel names, executing dialplan on the wrong channel |
| 04:26.27 | Kobaz | you name it |
| 04:26.52 | MrTelephone | im using 1.4.21 in a production environment. I've never used lcr before |
| 04:27.16 | WIMPy | Ok, that worked now. |
| 04:28.22 | MrTelephone | you you see if there is a PRIREVERSECHARGE line in your chan_dahdi.c in your asterisk 1.8 source directory? |
| 04:28.27 | WIMPy | Hmm, not I got a "WARNING: TRANSFER FAILED" from the phone. But it worked anyway. |
| 04:28.30 | MrTelephone | can you |
| 04:29.45 | MrTelephone | interesting. check the sip dialog and see what happened |
| 04:30.50 | MrTelephone | I wonder what gui the programmers use to handle large c files |
| 04:31.06 | MrTelephone | chan_sip.c is retardedly massive |
| 04:31.11 | WIMPy | ~collectdebug |
| 04:31.11 | infobot | extra, extra, read all about it, collectdebug is a method of collecting logs allowing others help troubleshoot an issue. Refer to https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information |
| 04:31.30 | *** join/#asterisk chazzam (~chazz@173-24-239-247.client.mchsi.com) |
| 04:31.45 | Kobaz | i use emacs |
| 04:37.47 | MrTelephone | im looking for where asterisk finds the auth info by registration IP |
| 04:37.58 | MrTelephone | i guess the high command is all in bed right now? |
| 04:38.19 | MrTelephone | Kobaz, can you search for PRIREVERSECHARGE in your chan_dahdi.c? |
| 04:39.39 | Kobaz | dev {/home/markm/asterisk/1.8.2.3-custom} root# ack PRIREVERSECHARGE |
| 04:39.39 | Kobaz | dev {/home/markm/asterisk/1.8.2.3-custom} root# |
| 04:39.40 | Kobaz | nothing |
| 04:41.54 | WIMPy | Seems to be ssl related. |
| 04:41.59 | MrTelephone | hmm |
| 04:42.16 | MrTelephone | wonder why they don't include that in the code |
| 04:42.31 | MrTelephone | supposed to be able to notify you of collect calls |
| 04:43.31 | MrTelephone | kobaz, it's in the subfolder called channels or does that serach for everything? |
| 04:43.59 | WIMPy | MrTelephone: What exactely are you looking for? |
| 04:44.14 | Kobaz | searches everytnig |
| 04:44.51 | MrTelephone | I added some code to detect reverse charges on a pri.. Just curious as to why the code wasn't added to chan_dahdi.c |
| 04:45.10 | Kobaz | issues.asterisk.org |
| 04:45.57 | WIMPy | Never seen that in use. But where do you leave that information, once you find it? |
| 04:47.21 | MrTelephone | yeah someone already had the issue out on the forum. PRIREVERSECHARGE just becomes a dialplan variable |
| 04:47.44 | MrTelephone | if it equals 1 then someone can't afford a proper long distance plan or they are in jail |
| 04:48.40 | WIMPy | Would certainly be interesting to be able to generate such calls. |
| 04:49.16 | WIMPy | I have no idea if anything would actually make use if that information. |
| 04:51.50 | MrTelephone | just to block collect calls so customers can't rack up a 600 dollar bill |
| 04:54.45 | WIMPy | wonders where the link between the dialplan and libcrypto comes from. |
| 04:55.21 | WIMPy | My test system doesn't crash, but I don;t have any IAX or SIPS configured there, |
| 04:59.09 | WIMPy | Ok, it was a syntax error. |
| 05:02.59 | MrTelephone | not sure buddy |
| 05:03.07 | MrTelephone | but who wrote this comment in the source? |
| 05:03.08 | MrTelephone | <PROTECTED> |
| 05:03.17 | MrTelephone | shit it got cut off |
| 05:03.30 | MrTelephone | ...section of SIP can suck my %&#$&* for not putting |
| 05:03.49 | MrTelephone | haha when I write progams I put comments like that when I get pissed off |
| 05:03.55 | MrTelephone | or dont have enough coffee |
| 05:05.49 | MrTelephone | i mean, how do you auth if you *67 and the URI is completely anonymous |
| 05:06.16 | MrTelephone | asterisk doesn't know who to use to make the digest |
| 05:06.30 | WIMPy | What is *67? |
| 05:06.37 | MrTelephone | block caller id |
| 05:06.58 | MrTelephone | outgoing call screening |
| 05:07.21 | MrTelephone | asterisk kicks ass though |
| 05:07.40 | MrTelephone | I phoned ADTRAN last week and found out their atlas 550 can oversubscribe and groom t1 circuits |
| 05:08.03 | MrTelephone | good way to build a redundant system off of 1 pri |
| 05:08.34 | Kobaz | groom? |
| 05:09.02 | MrTelephone | trunk i guess is a better word |
| 05:09.17 | MrTelephone | the said digium is right across the street from adtran |
| 05:09.28 | MrTelephone | the guy |
| 05:10.54 | MrTelephone | have you been to one of those asterisk conventions yet? |
| 05:11.40 | MrTelephone | I want to do a couple hotel installations but noone else knows how to operate asterisk or linux |
| 05:12.03 | *** join/#asterisk nix8n82 (~nate@63.162.28.112) |
| 05:14.07 | MrTelephone | Have you tried oovoo? |
| 05:14.50 | WIMPy | Man, you really got a lot of topics. |
| 05:15.18 | MrTelephone | noones talking, trying to trigger a hot spot |
| 05:15.48 | WIMPy | is going where I should have been a long time ago: To bed. |
| 05:15.58 | MrTelephone | yeah I was thinking of that too |
| 05:16.04 | MrTelephone | midnight |
| 05:16.06 | MrTelephone | :( |
| 05:16.52 | MrTelephone | what is your longest server uptime? |
| 05:17.00 | *** join/#asterisk ketas-av (~ketas@kvlt.eu) |
| 05:18.05 | MrTelephone | 00:17:45 up 580 days, 8:41, 1 user, load average: 0.00, 0.00, 0.00 |
| 05:18.27 | MrTelephone | in another month that thing will have been running for 2 years straight |
| 05:21.53 | ChannelZ | no kernel upgrades for you |
| 05:25.55 | drmessano | What's your favorite color? |
| 05:26.09 | drmessano | How many fingers am I holding up? |
| 05:26.16 | drmessano | Why did Grandma have to go away? |
| 05:26.16 | MrTelephone | no kidding |
| 05:26.28 | MrTelephone | the last kernel update i did was when that exploit vmsplice came out |
| 05:26.35 | MrTelephone | I got hacked severely |
| 05:27.34 | MrTelephone | drmessano: what is your favorite finger? |
| 05:27.59 | drmessano | The one I lost to that hooker in that hotel room in Plano, TX |
| 05:28.08 | drmessano | Almost 4 years ago today |
| 05:28.14 | drmessano | :( |
| 05:28.28 | MrTelephone | yeah that's a messed up state |
| 05:28.38 | MrTelephone | hookers, drugs, death gallore |
| 05:29.05 | MrTelephone | out of all the First 48 episodes I thin 94% of them are in TX |
| 05:29.06 | *** join/#asterisk csnook (~chris@c-76-19-64-161.hsd1.ma.comcast.net) |
| 05:29.46 | drmessano | Good thing it's legal in Texas to punch one in the face when they bite a finger off |
| 05:29.59 | drmessano | Otherwise I wouldn't keep going there for vacation |
| 05:30.15 | MrTelephone | do you rent a room in that building with the green lights? |
| 05:30.49 | drmessano | I normally have my secretary rent a room in the cheapest motel possible.. |
| 05:31.13 | *** join/#asterisk Vrtigo1 (~Vrtigo1@75.112.143.123) |
| 05:31.18 | MrTelephone | i stayed at this place in minneapolis once called the red roof inn. I didn't think I was going to make it back alive |
| 05:31.49 | drmessano | A Red Roof Inn is always good for cheap laughs, cheap women, and a cheap continental breakfast |
| 05:31.53 | MrTelephone | is it normal for them to scan driver licenses at these places? |
| 05:32.09 | MrTelephone | I wasnt sure if the red roof inn was a chain or not |
| 05:32.14 | drmessano | it is |
| 05:32.30 | drmessano | I don't normally hand out my real license, so it really doesn't matter to me |
| 05:32.56 | MrTelephone | I don't really hand out a fake one |
| 05:33.16 | MrTelephone | maybe a copy |
| 05:33.43 | MrTelephone | the police like that when you hand them a black and white photocopy of your original |
| 05:33.44 | Vrtigo1 | Hello. I am trying to price a simple asterisk installation for a small office (only three phones, that would grow to include two phones offsite at employee homes) and am looking for recommendations for a PCI FXO card that can support 3 channels, as well as simple SIP phones that support hold and transfer. I've looked at the wiki, but there is a lot of info there, and not sure the best place |
| 05:33.51 | Vrtigo1 | to start. |
| 05:34.23 | MrTelephone | vrtigo1, three thousand easy |
| 05:34.49 | drmessano | Vrtigo1, Digium TDM410P |
| 05:35.05 | MrTelephone | Adit 600 and adtran make nice channel banks that support fxo and fxs |
| 05:35.08 | drmessano | and Polycom phones |
| 05:35.17 | MrTelephone | don't buy the pci fxo/fxs cards. I had them before and it was a huge mess |
| 05:35.43 | MrTelephone | you need something with automatic gain control on the fxo lines |
| 05:35.45 | Vrtigo1 | are you saying i should handle fxo in a separate, ethernet-connected device? |
| 05:35.53 | MrTelephone | t1 connected device |
| 05:36.03 | Vrtigo1 | gotcha |
| 05:36.21 | drmessano | Vrtigo1, a 4-channel card from Digium works fine |
| 05:36.23 | MrTelephone | then when you expand you can get a PRI/T1 from your telco |
| 05:36.29 | drmessano | People have used them without issue for years |
| 05:36.40 | MrTelephone | i had mega echo problems on mine |
| 05:36.42 | drmessano | 3 channels? Hardly enough to justify a PRI |
| 05:37.07 | drmessano | MrTelephone, easy enough to resolve with Oslec and a little work on the line |
| 05:37.31 | MrTelephone | if you are 20km from the CO they don't work good I guess |
| 05:37.31 | drmessano | or HWEC on the card |
| 05:38.10 | Vrtigo1 | I'm thinking that a channel bank and T1 interface card would probably blow the budget on this particular project - this is for my wife's office. She wants to be able to answer the phone and work at home. |
| 05:38.22 | MrTelephone | faxing doesn't work good from ATA's but they work good when you use a channel bank |
| 05:38.45 | MrTelephone | vrtigo1, then a 4 port card like drmessano is worth a try |
| 05:38.45 | drmessano | I try not to make recommendations based on a specific instance, especially when there are extenuating circumstances. MOST people have NO problem with standard Digium PCI cards |
| 05:39.01 | drmessano | and it would fit this install well |
| 05:39.13 | MrTelephone | what about talkswitch or whatever it is called? |
| 05:39.27 | Vrtigo1 | drmessano, thanks - many years ago i used an x100p card just playing around and it seemed to work fine |
| 05:39.50 | MrTelephone | I wish my card worked but the phone lines were too noisey or something |
| 05:40.21 | drmessano | Well, I would say an x100p card is pretty crap.. you'll find modern cards are much better quality |
| 05:40.39 | MrTelephone | you can sip trunk with voicepulse or something and they might have a local number for you. |
| 05:41.17 | MrTelephone | anyways time for bed. have a good night drmessano and vrtigo1 |
| 05:41.39 | drmessano | Goodnight |
| 05:41.55 | Vrtigo1 | vitelity has local numbers, i use them for my home phone, but the quality isn't where i'd like it to be for business purposes and it'd be a tough sell for me since they're currently under contract on their telephone service |
| 05:42.09 | Vrtigo1 | as far as the tdm410p, do i need to purchase daughter cards to make it work, or does the card itself have all the functionality i would need for fxo? |
| 05:42.19 | *** join/#asterisk timahvo1 (~rogue@41.72.215.94) |
| 05:43.45 | drmessano | Generally you would purchase the card with the daughter cards.. they are sold in various configurations |
| 05:46.07 | Vrtigo1 | i also see that there is an echo cancellation daughter card for it - do you suppose that is something i would need out of the gate, or is that something that is usually purchased on an as-needed basis? |
| 05:54.41 | drmessano | Hard to say.. some people have problems, some don't |
| 05:55.03 | drmessano | I have used Oslec with success and would probably go with it before trying HWEC |
| 06:02.29 | coppice | if you need anything more than OSLEC for a consumer analogue line, something weird is going on |
| 06:04.51 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-uojyuwrxckvcsbdj) |
| 06:19.27 | drmessano | I would think your first start would be getting the line checked.. if there's anything major they would be able to see it and knock it out.. Any other minor mismatches the telco would ignore as within tolerance would be handled with some simple EC |
| 06:21.42 | *** join/#asterisk TTT_Travis (~Travis@cle-bb-cable2-ws-85.dsl.airstreamcomm.net) |
| 06:21.56 | TTT_Travis | I need some recommendations for SIP providers for incoming DID |
| 06:22.00 | coppice | unless the line is broken - cracking or something like that - OSLEC will cancel the mismatches |
| 06:22.36 | coppice | unless the mismatch is so screwed up you're at the point of howling :-) |
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| 06:31.14 | p3nguin | ttt_travis: Which country do you want them to be in? |
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| 07:46.09 | joachim_- | Morning all! |
| 07:47.20 | joachim_- | How can i know for sure that ReInvite behaviour works on my asterisk? Any CLI command ? |
| 07:48.15 | beardy | Sniff the SIP or log verbosely. |
| 07:50.11 | joachim_- | Yeah, ive tried setting sip debugging on and read the sip messages, but i cant really say i can see wheter or not reinvite behaviour, It says locally bridging when making calls between 2 peers on the same lan |
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| 07:53.13 | *** join/#asterisk schmidts (~schmidts@213.235.212.193) |
| 07:53.18 | schmidts | good morning |
| 07:53.26 | beardy | Dump the SIP packets and look what the SDP header says it is sending the media stream to. |
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| 09:30.19 | Tozz_ | We are having an issue that sometimes (once every 2 or 3 months or so) our Asterisk does not function properly anymore. eg. 'core show channels' does not show any output, 'restart now' doesnt do anything |
| 09:30.21 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 09:30.24 | Tozz_ | we have to restart asterisk to get it up again |
| 09:30.27 | Tozz_ | is this a known issue in 1.4? |
| 09:31.37 | plundra | What version of 1.4 are you using? |
| 09:32.05 | Tozz_ | 1.4.18 |
| 09:32.08 | Tozz_ | (yes a bit old, i know ;)) |
| 09:32.41 | plundra | I'd suggest first testing the latest version. Or at least check the (probably HUGE) changelog between 1.4.39.1 and 1.4.18. |
| 09:32.53 | Tozz_ | hehe, will do |
| 09:34.20 | plundra | Other then no output (*no* output or does it say 0 active calls?) of core show channels, does it route calls? Any other show command working? etc. |
| 09:35.56 | Tozz_ | there is really no output, nothing at all |
| 09:36.01 | Tozz_ | it doesn't route any calls anymore |
| 09:37.00 | Tozz_ | I didn't test any other show commands ;( |
| 09:37.05 | Tozz_ | just restarted it, so cant test anymore; ( |
| 09:37.36 | *** join/#asterisk atan (~atan@unaffiliated/atan) |
| 09:38.05 | plundra | It's exactly three years *today* since 1.4.18 was released btw :-) The changelog since is ~15k lines. |
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| 09:39.23 | Tozz_ | hehe yes I'm viewing it now |
| 09:39.41 | schmidts | tozz_ sound like a deadlok |
| 09:39.50 | schmidts | s/deadlok/deadlock/ |
| 09:39.58 | Tozz_ | yes something like that |
| 09:40.14 | Tozz_ | its behaviour is similar to when u run 'restart gracefully' or 'restart when convinient' (not sure which of the 2) |
| 09:40.27 | Tozz_ | if u do that core show channels also often does not output anything anymore |
| 09:40.53 | schmidts | tozz_ this is for sure a deadlock which is allready fixed |
| 09:40.59 | Tozz_ | ah ok, thnx |
| 09:41.23 | Tozz_ | i'll upgrade tonight then ;) |
| 09:41.58 | schmidts | good idea ;) |
| 09:43.37 | *** join/#asterisk tyman (~tyler@173-12-219-178-Fresno.hfc.comcastbusiness.net) |
| 09:55.50 | wdoekes2 | Tozz_: I upgraded friday from 1.4.24, same issue |
| 09:56.04 | Tozz_ | ah |
| 09:56.12 | wdoekes2 | observe that notify replies change from 489 to 481 |
| 09:56.54 | Tozz_ | this always happends during businesshours, so I usually restart as fast as I can ;) |
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| 09:57.48 | wdoekes2 | mm.. we also had cpu usage of 100%.. could be the same issue or a different one |
| 09:58.27 | wdoekes2 | and if you're not doing autoload, app_voicemail requires res_smdi |
| 09:58.30 | Tozz_ | we'll see if the issue comes back in 1.4.39 |
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| 09:59.32 | tyman | I've got a strange new problem with 1.8.2.3 with the Cepstral text-to-speech engine. Fresh install on a new centos 5.5 box. Starting from a fresh boot, asterisk works fine running as user asterisk. If the asterisk is restarted in anyway (init script, or thru console with core restart now, etc) it takes approximately 5 minutes for it to load it's config while apparently waiting for the Cepstral to init. |
| 10:00.27 | tyman | once past the Cepstral loading it loads the rest of the config nearly instantly |
| 10:08.11 | *** join/#asterisk daxt (~daxt@112.135.66.175) |
| 10:08.24 | daxt | Anybody using Visual Dial Plan ? |
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| 10:47.56 | Dovid | tzafrir: Hello there |
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| 10:48.52 | tzafrir | Dovid, hi |
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| 11:04.51 | andylockran | <PROTECTED> |
| 11:04.54 | andylockran | <PROTECTED> |
| 11:04.56 | andylockran | <PROTECTED> |
| 11:04.59 | andylockran | oops |
| 11:05.03 | andylockran | sorry guys - didn't meant to paste that |
| 11:05.15 | andylockran | any idea how to troubleshoot that error though ? |
| 11:07.39 | Tozz_ | your dial string seems weird |
| 11:07.51 | Tozz_ | is one of those numbers an actual IAX peer? |
| 11:08.18 | andylockran | Tozz_: the first one (016166.. is the name of the IAX2 trunk) |
| 11:08.41 | Tozz_ | ah |
| 11:08.42 | Tozz_ | well |
| 11:08.55 | Tozz_ | the error 'everyone is busy' is self explaining |
| 11:09.04 | Tozz_ | your iax peer has refused the call because its busy |
| 11:10.58 | Dovid | tzafrir: Do you mind looking at https://issues.asterisk.org/view.php?id=17954#131100 ? |
| 11:11.00 | andylockran | is that likely if the iax peer is a commercial provider? |
| 11:11.28 | Tozz_ | andylockran: depends if they have a per-client call-limit |
| 11:11.39 | Tozz_ | or maybe they refuse the call because of some other reason |
| 11:11.45 | Tozz_ | (wrong/invalid callerid or something) |
| 11:12.43 | Tozz_ | I would send the provider a mail asking them what y our doing wrong |
| 11:57.55 | creativx | iigh.. call-limit ¤#"¤ |
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| 12:08.37 | creativx | I can't remember how I fixed this.. but call-limit is not working so our queue members gets called on line 2 when they are active on line 1 |
| 12:29.23 | creativx | god damn I am starting to dislike the idea of multiple lines |
| 12:34.44 | *** join/#asterisk Pitel (~pitel@ip-94-113-20-150.net.upcbroadband.cz) |
| 12:35.08 | Pitel | Hi, can someone explain to me the difference between client and component mode in res_jabber? |
| 12:54.52 | leifmadsen | creativx: which version of Asterisk? |
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| 12:59.14 | creativx | leifmadsen: an ancient one.. |
| 12:59.15 | creativx | 1.2 |
| 12:59.27 | creativx | not even ringinuse for queues is available on that version.. sheesh |
| 12:59.32 | leifmadsen | heh odd :) |
| 12:59.47 | creativx | yeah |
| 12:59.49 | leifmadsen | I usually just use Local channels and GROUP() & GROUP_COUNT() for call control |
| 13:00.01 | creativx | so now I'm doing a quick experiment to see if 1.4 enjoys our configs.. |
| 13:00.14 | creativx | yeah I've read a bit about that |
| 13:00.23 | leifmadsen | ~newbook |
| 13:00.24 | infobot | Help review the 3rd edition of the Asterisk book published by O'Reilly at http://ofps.oreilly.com/titles/9780596517342/ Released under a Creative Commons license: http://creativecommons.org/licenses/by-nc-nd/3.0/us/ |
| 13:00.31 | leifmadsen | I've written about that in the ACD chapter |
| 13:01.00 | creativx | I thought about it, but we have an external service that populates queues with members |
| 13:01.06 | creativx | so that would need a rewrite |
| 13:01.46 | creativx | if each agent should be proxied through a local channel that does the group count magic that is |
| 13:04.08 | creativx | leifmadsen: but with ringinuse=no there should be no need for group+groupcount, right? |
| 13:05.57 | *** join/#asterisk zerohalo (~zerohalo@173-13-92-27-NewEngland.hfc.comcastbusiness.net) |
| 13:06.11 | Dovid | leifmadsen: Do you mind looking at https://issues.asterisk.org/view.php?id=17954#131100 ? |
| 13:06.17 | *** join/#asterisk coppice (~chatzilla@m121-202-63-221.smartone-vodafone.com) |
| 13:06.28 | Dovid | leifmadsen: I wonder if it is the same issue or something else |
| 13:06.32 | leifmadsen | Dovid: I will when my work day starts. I'm currently working on writing a book. |
| 13:06.47 | *** join/#asterisk zerohalo (~zerohalo@173-13-92-27-NewEngland.hfc.comcastbusiness.net) |
| 13:06.55 | Dovid | leifmadsen: What time is it out there ? and what book r u writing ? |
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| 13:07.01 | Dovid | TFOT v3 ? |
| 13:07.33 | leifmadsen | Dovid: 8:07am right now. I'm working on the Asterisk Cookbook. We already finished TFOT v3 (which is actually called Asterisk: The Definitive Guide now) |
| 13:07.36 | leifmadsen | ~newbook |
| 13:07.36 | infobot | Help review the 3rd edition of the Asterisk book published by O'Reilly at http://ofps.oreilly.com/titles/9780596517342/ Released under a Creative Commons license: http://creativecommons.org/licenses/by-nc-nd/3.0/us/ |
| 13:08.01 | Dovid | so whats this ? |
| 13:08.11 | leifmadsen | what is what? |
| 13:08.16 | leifmadsen | define: this |
| 13:08.33 | Dovid | the book u r writing now ? |
| 13:08.39 | coppice | a cookbook? like "802.11n: how to microwave your phone calls"? |
| 13:08.44 | leifmadsen | <leifmadsen> Dovid: 8:07am right now. I'm working on the Asterisk Cookbook. |
| 13:08.47 | Dovid | also is it available in pdf ? i will gladly lookit over. |
| 13:08.49 | Dovid | ah cb |
| 13:09.05 | leifmadsen | Dovid: please try clicking on the links I just provided :) |
| 13:09.13 | leifmadsen | we're using OFPS for the review |
| 13:09.28 | leifmadsen | coppice: kind of. They are recipes for how to solve problems :) |
| 13:09.51 | leifmadsen | coppice: We've already got "How To Grow Your Company With PITCH_SHIFT()" in there which I think will be very useful for small companies |
| 13:11.45 | Dovid | leifmadsen: wanted a pdf so i could download to my phone. guess I can use this as well. |
| 13:11.57 | Dovid | so far so good. what do u want people redaing over the book to look for ? |
| 13:12.12 | coppice | MS need a new pitch to shift win phone 7, so it might help big companies too |
| 13:12.28 | leifmadsen | Dovid: http://leifmadsen.wordpress.com/2011/01/06/asterisk-the-definitive-guide-call-for-reviewers/ |
| 13:12.55 | leifmadsen | Dovid: we're just publishing the book via HTML this time around. Ebook versions will be for sale. |
| 13:13.46 | leifmadsen | Dovid: We'll be reviewing this week and next, with all edits by us needing to be given to O'Reilly by the 14th of February. After that, no more changes will be allowed. |
| 13:13.56 | leifmadsen | after that people can file errata :) |
| 13:14.03 | leifmadsen | goes back to the book |
| 13:16.30 | creativx | leifmadsen: before you completely fall into the book.. ringinuse = no exludes the need for group counters? |
| 13:16.34 | coppice | ebooks need a yum like feature |
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| 13:26.09 | creativx | wohoo.. 1.4 liked my 1.2-configs |
| 13:26.22 | Dovid | leifmadsen: Read it. will try to read what i can when i can |
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| 13:26.41 | Dovid | creativx: Careful. Certain things will stop working when you go to 1.6 and then 1.8. |
| 13:26.55 | Dovid | worth looking through the logs and seeing whats been depreceated |
| 13:27.04 | Dovid | and congrats from moving off of 1.2 |
| 13:27.09 | Dovid | i remember the 1.x days...... |
| 13:27.28 | creativx | I am stopping at 1.4 ! |
| 13:27.38 | creativx | all I need is to get the queues working |
| 13:27.41 | Dovid | creativx: why ? |
| 13:27.56 | creativx | agents doesnt like to have all their lines ring when they are in a queue |
| 13:28.03 | Dovid | creativx: I wait b4 upgrading but 1.4.x is king of old school |
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| 13:28.15 | Dovid | boohooo. u pay em. let me get used to it |
| 13:29.34 | creativx | yeah.. paying them and managing them are two different operations |
| 13:29.40 | creativx | anyone can pay an employee |
| 13:29.41 | Dovid | lol |
| 13:29.44 | creativx | not everyone knows how to manage people |
| 13:40.33 | andylockran | hey guys - is there a CLI way of testing asterisk regex |
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| 13:48.48 | plundra | So, am I the only one who wants to specify paths for asterisk without it appending [stuff]/asterisk? :) (The configure-script) |
| 13:49.18 | plundra | Sure it's easy to change manually, but it'd be nice to just do ./configure --mystuff directly in one go. |
| 13:52.34 | andylockran | guys, we've got this REGEX in our asteirks setup. exten => s,1,ExecIf($[${REGEX("^0" ${DIAL_NUMBER})} != "0"],Set,TARGET_FLP_2=44${DIAL_NUMBER:1}) |
| 13:53.14 | andylockran | how do you test the REGEX.. if we don't test the regex, it's set to one.. if I do == 1, then it gets set to 0, and if I do = 1, it also gets set to 0 |
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| 14:01.27 | schmidts | andylockran Function REGEX uses the C builtin regexc function, take a look here: http://manpages.debian.net/cgi-bin/man.cgi?query=regexec |
| 14:01.44 | andylockran | schmidts: thanks |
| 14:02.26 | schmidts | try regex from your bash, it should be the same like its in asterisk |
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| 14:25.59 | andylockran | schmidts: the issue I have though, is that the regex above returns 1 |
| 14:26.06 | andylockran | yet when it's tested, it returns 0 |
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| 14:27.49 | andylockran | I added a line under that one: exten => s,n,Set(TMP_VAR=${REGEX("^0" ${DIAL_NUMBER}) |
| 14:27.54 | andylockran | and TMP_VAR printed as 1 |
| 14:28.17 | andylockran | yet, when I did exten => s,1,ExecIf($[${REGEX("^0" ${DIAL_NUMBER})} != "0"],Set,TARGET_FLP_2=44${DIAL_NUMBER:1}) , it returned as 0 |
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| 14:30.45 | andylockran | schmidts: what is the valid way of testing the regex ? |
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| 14:33.35 | schmidts | sed -e 's/xyz/abc/' IMHO ;) |
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| 14:34.59 | schmidts | andylockran if i understand you right you want to check if DIAL_NUMBER starts with 0 right? |
| 14:35.42 | andylockran | schmidts: yes |
| 14:35.49 | schmidts | maybe it should look like this REGEX("^0[0-9]*" ${DIAL_NUMBER}) |
| 14:35.56 | andylockran | ${REGEX("^00[0-9#\\*\\+]+$" ${DIAL_NUMBER})} |
| 14:36.03 | andylockran | that's what I have at the moment |
| 14:36.09 | andylockran | the example was massively simplified. |
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| 14:36.13 | schmidts | ok :D |
| 14:36.33 | andylockran | schmidts: that ^00 ... does that make sense? |
| 14:37.37 | schmidts | i would read this as, starting with 0 then another 0 and then everything else |
| 14:39.14 | schmidts | what do you want to check? if it starts with 00 or with 0, cause i would use something like this: EXECIF($["${DIAL_NUMBER:0:2]"="00"]..... |
| 14:41.22 | leifmadsen | aye |
| 14:41.26 | leifmadsen | seems easier :) |
| 14:42.13 | schmidts | when it comes to regex allways keeps things as simple as possible ;) |
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| 14:45.23 | schmidts | http://bash.org/?11141 |
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| 14:59.37 | giany | anyone knows what could be the reason of these debug messages: |
| 14:59.39 | giany | <PROTECTED> |
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| 15:04.43 | andylockran | schmidts: http://dpaste.com/371409/ is the problem I'm having.. if you have the time to take a quick look. |
| 15:08.08 | schmidts | andylockrn which is no suprise ;) IMHO your regex looks for something like 00792... not 07921... as you dialed |
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| 15:12.36 | andylockran | ah, even moving the regex down to a single 0 and it fails... |
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| 15:17.57 | andylockran | exten => s,n,Set(TMP_REG=${REGEX("^0([0-9])+$" ${DIAL_NUMBER})}) that fails too |
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| 15:24.24 | Dovid | leifmadsen: @ work yet ? |
| 15:27.41 | *** join/#asterisk tuxxie (~Ryan@rrcs-70-63-90-226.midsouth.biz.rr.com) |
| 15:28.43 | tuxxie | how can i show the concurrent calls using a sip trunk? |
| 15:28.55 | schmidts | andylockran you should really try using it the way i said above, its much easier |
| 15:29.10 | schmidts | tuxxie core show channels |
| 15:30.11 | *** join/#asterisk daxt (~daxt@112.135.81.27) |
| 15:30.28 | tuxxie | schmidts: thanks |
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| 15:38.51 | andylockran | http://dpaste.com/371493/ |
| 15:39.08 | andylockran | right, third line is commented, if I comment out first two lines of exten, and uncomment third line it doesn't work. |
| 15:39.17 | andylockran | however, with it as it is it works. |
| 15:39.23 | andylockran | Can someone explain how? |
| 15:39.24 | tuxxie | schmidts: the total active calls does not seem to match the count on my sip gateway. I am seeing around 75 active calls with core show channel but only 65 on my sip gateway. Am i missing something? |
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| 15:42.04 | Dovid | anyone know's if this isa bug or just how Asterisk is supposed to work ? http://lists.digium.com/pipermail/asterisk-users/2011-January/258628.html |
| 15:43.45 | schmidts | tuxxie are you sure all calls are going out to your gateway? |
| 15:44.15 | schmidts | tuxxie try asterisk -rx"core show channels verbose" | grep "SIP/yourgateway" | wc |
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| 15:44.33 | cusco | lsusb |
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| 16:04.41 | tuxxie | schmidts: i found out why the number differ. Our sip gateway is not showing calls that are in the dailing state. |
| 16:05.24 | tuxxie | so we have around 10-15 calls dailing at anygiven time. |
| 16:08.37 | *** join/#asterisk tokozedg (~toka@94.240.224.234) |
| 16:10.30 | Dovid | anyone here uses ACME packet SBC's ? |
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| 16:22.30 | Dovid | wow the room is dead :( |
| 16:26.14 | *** join/#asterisk daxt (~daxt@112.135.85.103) |
| 16:26.35 | beek | Dovid: It has been since [TK]'s expulsion. |
| 16:26.36 | *** join/#asterisk ClintGoudie-Nice (~clint@smtp.callware.com) |
| 16:27.10 | fenrus | asdf |
| 16:27.10 | fenrus | =) |
| 16:27.22 | ClintGoudie-Nice | Heya all. If I have more than one CA that I need to trust in the tlscafile parameter of the sip config, how do I do it? |
| 16:27.44 | ClintGoudie-Nice | Do I just cat both CA PEM files together? I tried that but it doesn't seem to be working |
| 16:29.05 | malcolmd | try tlscadir instead of tlscafile |
| 16:31.48 | Dovid | beek: sad that he had to go but he was an ass |
| 16:32.16 | beek | Dovid: And one of the best sources of help on the channel. I know he was invaluable to me. |
| 16:32.53 | beek | I see questions go unanswered here daily that were typically taken care of by him. |
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| 16:33.33 | Qwell | beek: You're more than welcome to answer those questions. |
| 16:33.45 | leifmadsen | ideally with less of an attitude |
| 16:33.54 | beek | Qwell: I do when I can. His expertise far exceeds mine. |
| 16:34.04 | Dovid | lol |
| 16:34.27 | Dovid | i dont have time to be here. also i shifted away from asterisk and more to management of systems (servers, routers etc.) |
| 16:34.41 | Dovid | i have not kept up with * excpet for when i play with it |
| 16:34.52 | Dovid | liefmadsen: Were you able to look @ my post ? |
| 16:35.09 | leifmadsen | Dovid: not yet |
| 16:35.11 | leifmadsen | I will when I can |
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| 16:37.42 | Dovid | ok. |
| 16:38.03 | Dovid | goto leave the office. be on here soon. |
| 16:38.04 | Dovid | night |
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| 16:52.26 | ClintGoudie-Nice | Heya guys, can you clue me in on this? tcptls.c: FILE * open failed! |
| 16:52.59 | ClintGoudie-Nice | Is it not reading my certificate? It's in the place specified by my config, and asterisk has full access to it |
| 16:53.29 | *** join/#asterisk telnettech (~telnettec@216.49.139.56) |
| 16:53.36 | ClintGoudie-Nice | next I get SSL Certificate ok and then == Problem setting up ssl connection: error:00000000:lib(0):func(0):reason(0) |
| 16:53.41 | fenrus | lwin 3 |
| 16:53.41 | telnettech | hi guys.....gotta question |
| 16:54.33 | telnettech | can you put this in the sip.conf for a sip trunk and expect the IVR functionality to work?type=peer |
| 16:54.33 | telnettech | insecure=very |
| 16:54.33 | telnettech | allow=ulaw&rfc2833 |
| 16:54.33 | telnettech | dtmfmode=auto |
| 16:54.33 | telnettech | <PROTECTED> |
| 16:54.49 | WIMPy | plonk |
| 16:55.22 | telnettech | i thought that the allow parameter had to be per line for each protocol that you want to use? |
| 16:55.51 | ClintGoudie-Nice | is my side not trusting the other side, vice-versa, or have I busted my cert somehow? |
| 16:57.45 | *** join/#asterisk daxt (~daxt@112.135.94.212) |
| 16:59.38 | Qwell | ClintGoudie-Nice: it's failing to open the file. Either Asterisk doesn't have permission to, or the filename is wrong. |
| 17:05.02 | ClintGoudie-Nice | Qwell: from my sip.conf: tlscertfile=/etc/asterisk/keys/asterisk.pem and everyone can read that file. |
| 17:05.16 | ClintGoudie-Nice | and selinux is off |
| 17:06.33 | ClintGoudie-Nice | tried chmoding the file to 777 and still no luck |
| 17:08.27 | p3nguin | Why in the world would you need to make a pem file 777? |
| 17:08.52 | ClintGoudie-Nice | I wouldn't, but I'm trying to eliminate the variables |
| 17:08.57 | p3nguin | Making it READABLE is enough to read it. |
| 17:09.48 | ClintGoudie-Nice | ok, asterisk is saying it cant read it. Thoughts? |
| 17:10.04 | p3nguin | But you should check the permissions of the entire path to the file. If one of the parent directories prevents it, that'll be the end of the line. |
| 17:10.36 | *** join/#asterisk JustinCampbell (~JustinCam@dsl092-226-147.phl1.dsl.speakeasy.net) |
| 17:10.44 | p3nguin | I use namei to do that at a glance. |
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| 17:13.21 | *** part/#asterisk sekil (~sekil@80.93.247.26) |
| 17:15.44 | ClintGoudie-Nice | namei didn't reveal anything interesting |
| 17:16.39 | p3nguin | pastebin it. |
| 17:17.00 | drift- | has anyone ever gotten google voice to work with asterisk? |
| 17:17.03 | drift- | to make outgoing calls? |
| 17:17.27 | ClintGoudie-Nice | hmm |
| 17:17.51 | ClintGoudie-Nice | So I moved the certificates to / and made sure they were owned and accessible by asterisk, still no go |
| 17:18.38 | p3nguin | namei -l /etc/asterisk/keys/asterisk.pem <--- pastebin the output. |
| 17:19.24 | ClintGoudie-Nice | -l? or -mx? |
| 17:19.31 | ClintGoudie-Nice | at least on my system there's no -l |
| 17:19.56 | p3nguin | If no -l, then use -mov. |
| 17:20.55 | ClintGoudie-Nice | at least on cent, my options are only m and x. |
| 17:21.00 | ClintGoudie-Nice | p3nguin: http://pastebin.com/Q2bNvKsj |
| 17:21.20 | p3nguin | That's a weird namei. |
| 17:22.34 | ClintGoudie-Nice | it's what I've got ;) |
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| 17:23.56 | p3nguin | I'm using namei from util-linux-ng 2.18. Not that you can do anything about yours, but I'm just trying to be complete. |
| 17:24.31 | p3nguin | It's really just a convenience tool. The same information can be found using multiple other utilities. |
| 17:26.38 | p3nguin | Here's an example of what I expect to see from namei -l when I'm looking for what's wrong: http://pastebin.com/DenDSdTe |
| 17:27.52 | p3nguin | This reveals to me that I need to consider chowning some dirs and files after I created them recently. |
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| 17:35.01 | anonymouz666 | nothing is better than vacations |
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| 17:54.27 | *** join/#asterisk Dovid (~Dovid@213.8.121.90) |
| 17:54.48 | Dovid | hello all |
| 17:54.56 | Dovid | leifmadsen: Hi there. back for a bit |
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| 17:59.05 | ClintGoudie-Nice | an strace of the asterisk process seems to show that the file is being opened successfully: http://pastebin.com/c30NC52c |
| 17:59.31 | ClintGoudie-Nice | have I done something wrong with my key? |
| 17:59.57 | miju | hi, i'm playing with asterisk for some days and i must say i am very impressed... but i have one problem. i have a sip account and tried to use ivr solutions but there are no dtmf tones coming through. i played around a bit with dtmfmode options but no success... i was wondering if someone has any suggestions? |
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| 18:11.14 | malcolmd | drift-: https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google |
| 18:12.26 | drift- | yeah that seems to be the magic link |
| 18:12.37 | drift- | has anyone ever got this to work tho? |
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| 18:24.47 | *** join/#asterisk Katty (~Katty@mail.copi-rite.com) |
| 18:25.21 | Dovid | miju: Where are you having issues ? in out ? did you try with a soft phone ? itsp ? |
| 18:25.39 | Dovid | anyone know's if this isa bug or just how Asterisk is supposed to work ? http://lists.digium.com/pipermail/asterisk-users/2011-January/258628.html |
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| 18:32.37 | Katty | AREYOUREADYTO.....monday?! |
| 18:33.00 | Kobaz | it's katty |
| 18:33.30 | Katty | shh, don't tell. it's a secret |
| 18:33.55 | beek | hugs Katty |
| 18:34.41 | Katty | hugs beek |
| 18:40.05 | *** join/#asterisk IsUp (IsUp@unaffiliated/isup) |
| 18:47.43 | diemos | stares at Katty. |
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| 18:55.52 | p3nguin | I want French fries. |
| 19:00.25 | drift- | doin breakfast for lucnh today |
| 19:00.27 | drift- | ;D |
| 19:01.08 | p3nguin | I'm not far from Wendy's... I'm SOOOOO considering grabbing some of those new natural cut fries with sea salt. |
| 19:02.54 | Katty | diemos: umm |
| 19:02.56 | Katty | diemos: hi? |
| 19:03.01 | Katty | diemos: whoarechu |
| 19:03.14 | Katty | diemos: do you need hugs? |
| 19:03.21 | p3nguin | Of course I'm even closer to McDonald's, so it would be more convenient to walk over there and get some of their fast food fries. Without salt, of course. |
| 19:03.26 | Qwell | does |
| 19:03.44 | Katty | :> |
| 19:03.46 | Katty | hugs Qwell |
| 19:03.50 | Qwell | \o/ |
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| 19:04.57 | miju | Dovid: only in when i call from landline. softphone is working and outgoing also. i am using sipgate.de |
| 19:12.52 | *** join/#asterisk bvierra (~bvierra@route1.lax.office.netdna.com) |
| 19:13.00 | bvierra | hey guys |
| 19:13.16 | beardy | Hello |
| 19:13.46 | bvierra | just had an issue where we lost a call, we didnt hang up |
| 19:14.02 | bvierra | looking in logs I see: Macro("SIP/501-00000343", "hangupcall,") |
| 19:14.13 | bvierra | does that mean that our trunk did it or? |
| 19:14.30 | Qwell | bvierra: we'd need to see whatever was before that |
| 19:14.48 | Qwell | also we have no idea what calls that macro, in your dialplan |
| 19:18.32 | bvierra | http://pastebin.com/heN7DHMQ |
| 19:18.37 | bvierra | there ya go :)\ |
| 19:19.32 | p3nguin | There you went. |
| 19:20.43 | Tozz_ | Gone we are! |
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| 19:51.05 | makafre | hey guys, I am using 1.8 for the first time so here is an easy one; how come core set verbose 20 doesn't let regular diaplan messages to be displayed? |
| 19:52.07 | makafre | like " == Everyone is busy/congested at this time (1:0/0/1)" messages |
| 19:52.11 | p3nguin | I guess it's going to be for the same reason core set verbose 5 doesn't, since it doesn't give any additional verbosity past 4. |
| 19:53.06 | makafre | ok, but even dialplan reload just shows me "Dialplan reloaded." |
| 19:53.58 | Qwell | makafre: Do you have verbose messages going to the console? Check logger.conf |
| 19:53.58 | makafre | that's at level 5 |
| 19:54.10 | makafre | ok, let me see |
| 19:54.32 | p3nguin | I just ran core set verbose 4 and then dialplan reload... it spewed the typical stuff about merging and loading things. |
| 19:54.44 | makafre | humm, I have no logger.conf |
| 19:54.59 | p3nguin | Perhaps that is why you aren't getting any messages. |
| 19:55.22 | makafre | maybe, let me try |
| 19:55.55 | p3nguin | I have [logfiles] and then I define full log. Nothing else is uncommented. |
| 19:56.25 | p3nguin | (that includes console being commented out) |
| 19:56.38 | makafre | okido |
| 19:56.51 | makafre | you found the problem ! |
| 19:56.58 | makafre | it was a missing logger.conf |
| 19:57.03 | makafre | yippie |
| 19:57.13 | makafre | thanks a lot |
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| 20:12.30 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 20:13.10 | joako | What is the meaning of the error: Invalid host name in Contact: (can't resolve in DNS) : '' |
| 20:13.22 | joako | ... if the same dialplan was working fine and was not changed! |
| 20:14.44 | Dovid | joako: You have a DNS issue |
| 20:15.07 | Dovid | look at the erorr. it cant resolve a host that you have set. try to ping it from the server and see if you get a dns resoltion. |
| 20:15.24 | Dovid | check /etc/resolv.conf and see what you have ? I use 8.8.8.8 and 4.2.2.2 |
| 20:15.30 | Dovid | they seem to work OK for me |
| 20:17.39 | p3nguin | 4.2.2.1-6 are Verizon's nameservers and rarely fail. |
| 20:18.20 | *** join/#asterisk jsjc (~jsjc@226.Red-80-33-236.staticIP.rima-tde.net) |
| 20:18.35 | jsjc | hi!. is there anyway to check device state in DADHI channels? |
| 20:18.43 | p3nguin | 8.8.8.8 and 8.8.4.4 are Google's, and, for that reason, also probably won't fail. |
| 20:19.48 | jsjc | so when I send a call trough a channel does not give any errors.. |
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| 20:30.08 | joako | Dovid: Host= IP address... very odd for it to fail like that but seems to be a provider issue.... |
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| 21:27.17 | *** join/#asterisk SparFux (~raoul@g231007049.adsl.alicedsl.de) |
| 21:28.06 | SparFux | Hey guys. What do you say, if there was a dahdi driver for HFC-S PCI A single bri cards, what would be the proper name for it? dahdi-zaphfc? dahdi-linux-zaphfc? Or dahdi-hfcsbri or the like? |
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| 21:37.29 | tzafrir | SparFux, zaphfc |
| 21:38.12 | SparFux | tzafrir: hi! Why do you think zaphfc would be best? It's a mixture of "zaptel" and "hfc", but the zaptel is obsolete. |
| 21:38.31 | tzafrir | The name remained |
| 21:38.57 | tzafrir | but it's a dahdi driver |
| 21:39.09 | SparFux | Wouldn't it be a good idea to change the name to something more appropriate? Like dahdi-hfcbri ? |
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| 21:40.52 | SparFux | Advantages of renaming the modules are: new users / admins get less confused, clearer fact that all the "zap" stuff is obsolete (which zaphfc isn't), easy module check with lsmod | grep dahdi. |
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| 21:42.45 | SparFux | tzafrir: There is the whole addition of zaphfc available and it runs fine mostly here. But wouldn't it even be easier to change the xpp bri code to support hfc-s pci a single bri, too? |
| 21:43.03 | SparFux | are hfc-s and hfc-4s so different? |
| 21:43.12 | tzafrir | SparFux, xpp bri is way different |
| 21:43.23 | tzafrir | wcb4xxp would actually be the closest |
| 21:43.57 | tzafrir | and yes, those are different chips. Not exactly sure how different |
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| 21:45.42 | SparFux | Ok then. Perhaps it's a stupid idea to put it into the xpp / wcb4xpp code. |
| 21:47.27 | SparFux | on cologne chip homepage they write using the hfc-s pci a isn't recommended for new projects and I sent email and asked for the reason. |
| 21:51.11 | SparFux | In https://sourceforge.net/projects/dahdi-zaphfc/ I put the zaphfc along with Kbuild file into one git tree for easy building. |
| 21:51.42 | SparFux | but it feels odd to have yet another place where hfc stuff is put. |
| 21:52.52 | ClintGoudie-Nice | is tlscadir supported in 1.8.2.2? |
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| 22:00.04 | malcolmd | ClintGoudie-Nice: works for me |
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| 22:07.14 | ClintGoudie-Nice | malcolmd: I'm stracing the process and I never see it open the directory :S |
| 22:07.24 | ClintGoudie-Nice | at least not on a reload |
| 22:09.02 | malcolmd | is it loaded the first time chan_sip is loaded? |
| 22:09.12 | leifmadsen | malcolmd: o/ |
| 22:09.36 | ClintGoudie-Nice | interesting. Grepping the asterisk source code for 1.8.2.1, it's in configs/sip.conf.sample |
| 22:09.39 | ClintGoudie-Nice | and nowhere else :S |
| 22:10.08 | Katty | heelllllllllloooooo nurse. |
| 22:10.35 | Kobaz | hellloooooo katty |
| 22:11.28 | malcolmd | main/tcptls.c |
| 22:12.28 | Katty | hugs Kobaz |
| 22:12.44 | *** part/#asterisk SparFux (~raoul@g231007049.adsl.alicedsl.de) |
| 22:14.49 | WIMPy | SparFux: Have you created just another version of zaphfc or what? |
| 22:15.02 | ClintGoudie-Nice | malcolmd: I only see tlscafile in tcptls.c but my source is for 1.8.2.1 |
| 22:17.20 | Kobaz | yay |
| 22:19.24 | Qwell | ClintGoudie-Nice: It's tlscapath |
| 22:20.59 | malcolmd | i'm confused. i'm looking at line 351 of tcptls.c and SSL_CTX_load_verify_locations tries both capath and cafile, not in that order. i'm not really inclined to search exhaustively, but i'm also not having an issue right now. maybe you've loaded pem files in the CA dir using "openssl x509 -in myfiles.pem >> CAfile.pem" and then c_rehash and it didn't work for you, but i'm not having a problem...granted, i only loaded one ca into the C |
| 22:20.59 | malcolmd | Afile.pem, so there could be some sort of issue there, but one worked just fine and my client's doing the TLS happy dance |
| 22:21.46 | Qwell | malcolmd: that option is broken. the sample config does not match reality. |
| 22:22.42 | malcolmd | then we'll have to get that fixed :) |
| 22:22.47 | Qwell | yes |
| 22:25.28 | Qwell | malcolmd: done and done |
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| 23:13.04 | *** join/#asterisk Sargun_Screen (~sargun@atarack/Staff/Sargun) |
| 23:13.16 | Sargun_Screen | Is anyone having issues with flowroute |
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| 23:17.23 | paulc | I've build a web front end, allowing team managers to monitor calls of their agents. They choose an active agent from a web browser, click, and I use AGI to redirect a pre-established call between the manager and Asterisk to a context/extension that monitors the agent using ChanSpy. In 1.6.2.9 it works fine, the manager can click between agents, all good. In 1.6.2.16, the first one works, the second click triggers the h extension in that context. Anyone |
| 23:20.12 | joako | Sargun_Screen: I know for a fact I just placed a few calls through them, no issues. |
| 23:23.00 | joako | Sargun_Screen, Inbound on multiple DID (LATA 460) which are even on multiple accoiunts are working fine. |
| 23:25.29 | Sargun_Screen | joako: I'm on the phone with them, and they're a trouble shootin' |
| 23:29.05 | Qwell | Poor Trouble. Always getting shot. |
| 23:31.17 | Kobaz | oh man |
| 23:31.27 | Kobaz | attended transfers are so evil |
| 23:31.40 | Qwell | Kobaz: blonde transfers are even more evil |
| 23:31.53 | Kobaz | blind transfers make programming easy |
| 23:32.02 | Qwell | not blind |
| 23:35.33 | Kobaz | i wouldn't wish this on anyonre |
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| 23:58.44 | paulc | To ask another way.. A monitors B. I use AGI to switch A to monitoring C. But it triggers the h extension in the context A was in.. in 1.6.2.16... but works ok in 1.6.2.9 (h doesn't trigger). Is this expected? bugs.digium.com seems to mention some similar things (involving B getting disconnected, which isn't happening for me) |